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Internet Engineering Task Force (IETF)                    H. SchulzrinneRequest for Comments: 7826                           Columbia UniversityObsoletes:2326                                                   A. RaoCategory: Standards Track                                          CiscoISSN: 2070-1721                                              R. Lanphier                                                           M. Westerlund                                                                Ericsson                                                     M. Stiemerling, Ed.                                University of Applied Sciences Darmstadt                                                           December 2016Real-Time Streaming Protocol Version 2.0Abstract   This memorandum defines the Real-Time Streaming Protocol (RTSP)   version 2.0, which obsoletes RTSP version 1.0 defined inRFC 2326.   RTSP is an application-layer protocol for the setup and control of   the delivery of data with real-time properties.  RTSP provides an   extensible framework to enable controlled, on-demand delivery of   real-time data, such as audio and video.  Sources of data can include   both live data feeds and stored clips.  This protocol is intended to   control multiple data delivery sessions; provide a means for choosing   delivery channels such as UDP, multicast UDP, and TCP; and provide a   means for choosing delivery mechanisms based upon RTP (RFC 3550).Status of This Memo   This is an Internet Standards Track document.   This document is a product of the Internet Engineering Task Force   (IETF).  It represents the consensus of the IETF community.  It has   received public review and has been approved for publication by the   Internet Engineering Steering Group (IESG).  Further information on   Internet Standards is available inSection 2 of RFC 7841.   Information about the current status of this document, any errata,   and how to provide feedback on it may be obtained athttp://www.rfc-editor.org/info/rfc7826.Schulzrinne, et al.          Standards Track                    [Page 1]

RFC 7826                        RTSP 2.0                   December 2016Copyright Notice   Copyright (c) 2016 IETF Trust and the persons identified as the   document authors.  All rights reserved.   This document is subject toBCP 78 and the IETF Trust's Legal   Provisions Relating to IETF Documents   (http://trustee.ietf.org/license-info) in effect on the date of   publication of this document.  Please review these documents   carefully, as they describe your rights and restrictions with respect   to this document.  Code Components extracted from this document must   include Simplified BSD License text as described in Section 4.e of   the Trust Legal Provisions and are provided without warranty as   described in the Simplified BSD License.   This document may contain material from IETF Documents or IETF   Contributions published or made publicly available before November   10, 2008.  The person(s) controlling the copyright in some of this   material may not have granted the IETF Trust the right to allow   modifications of such material outside the IETF Standards Process.   Without obtaining an adequate license from the person(s) controlling   the copyright in such materials, this document may not be modified   outside the IETF Standards Process, and derivative works of it may   not be created outside the IETF Standards Process, except to format   it for publication as an RFC or to translate it into languages other   than English.Table of Contents1. Introduction ...................................................102. Protocol Overview ..............................................112.1. Presentation Description ..................................122.2. Session Establishment .....................................122.3. Media Delivery Control ....................................142.4. Session Parameter Manipulations ...........................152.5. Media Delivery ............................................162.5.1. Media Delivery Manipulations .......................162.6. Session Maintenance and Termination .......................192.7. Extending RTSP ............................................203. Document Conventions ...........................................213.1. Notational Conventions ....................................213.2. Terminology ...............................................214. Protocol Parameters ............................................254.1. RTSP Version ..............................................254.2. RTSP IRI and URI ..........................................254.3. Session Identifiers .......................................28Schulzrinne, et al.          Standards Track                    [Page 2]

RFC 7826                        RTSP 2.0                   December 20164.4. Media-Time Formats ........................................284.4.1. SMPTE-Relative Timestamps ..........................284.4.2. Normal Play Time ...................................294.4.3. Absolute Time ......................................304.5. Feature Tags ..............................................314.6. Message Body Tags .........................................324.7. Media Properties ..........................................324.7.1. Random Access and Seeking ..........................334.7.2. Retention ..........................................344.7.3. Content Modifications ..............................344.7.4. Supported Scale Factors ............................344.7.5. Mapping to the Attributes ..........................355. RTSP Message ...................................................355.1. Message Types .............................................365.2. Message Headers ...........................................365.3. Message Body ..............................................375.4. Message Length ............................................376. General-Header Fields ..........................................377. Request ........................................................397.1. Request Line ..............................................407.2. Request-Header Fields .....................................428. Response .......................................................438.1. Status-Line ...............................................438.1.1. Status Code and Reason Phrase ......................438.2. Response Headers ..........................................479. Message Body ...................................................479.1. Message Body Header Fields ................................489.2. Message Body ..............................................499.3. Message Body Format Negotiation ...........................4910. Connections ...................................................5010.1. Reliability and Acknowledgements .........................5010.2. Using Connections ........................................5110.3. Closing Connections ......................................5410.4. Timing Out Connections and RTSP Messages .................5610.5. Showing Liveness .........................................5710.6. Use of IPv6 ..............................................5810.7. Overload Control .........................................5811. Capability Handling ...........................................6011.1. Feature Tag: play.basic ..................................6212. Pipelining Support ............................................6213. Method Definitions ............................................6313.1. OPTIONS ..................................................6513.2. DESCRIBE .................................................6613.3. SETUP ....................................................6813.3.1. Changing Transport Parameters .....................7113.4. PLAY .....................................................7213.4.1. General Usage .....................................7213.4.2. Aggregated Sessions ...............................77Schulzrinne, et al.          Standards Track                    [Page 3]

RFC 7826                        RTSP 2.0                   December 201613.4.3. Updating Current PLAY Requests ....................7813.4.4. Playing On-Demand Media ...........................8113.4.5. Playing Dynamic On-Demand Media ...................8113.4.6. Playing Live Media ................................8113.4.7. Playing Live with Recording .......................8213.4.8. Playing Live with Time-Shift ......................8313.5. PLAY_NOTIFY ..............................................8313.5.1. End-of-Stream .....................................8413.5.2. Media-Properties-Update ...........................8613.5.3. Scale-Change ......................................8713.6. PAUSE ....................................................8913.7. TEARDOWN .................................................9213.7.1. Client to Server ..................................9213.7.2. Server to Client ..................................9313.8. GET_PARAMETER ............................................9413.9. SET_PARAMETER ............................................9613.10. REDIRECT ................................................9814. Embedded (Interleaved) Binary Data ...........................10115. Proxies ......................................................10315.1. Proxies and Protocol Extensions .........................10415.2. Multiplexing and Demultiplexing of Messages .............10516. Caching ......................................................10616.1. Validation Model ........................................10716.1.1. Last-Modified Dates ..............................10816.1.2. Message Body Tag Cache Validators ................10816.1.3. Weak and Strong Validators .......................108           16.1.4. Rules for When to Use Message Body Tags                   and Last-Modified Dates ..........................11016.1.5. Non-validating Conditionals ......................11216.2. Invalidation after Updates or Deletions .................11217. Status Code Definitions ......................................11317.1. Informational 1xx .......................................11317.1.1. 100 Continue .....................................11317.2. Success 2xx .............................................11317.2.1. 200 OK ...........................................11317.3. Redirection 3xx .........................................11317.3.1. 300 ..............................................11417.3.2. 301 Moved Permanently ............................11417.3.3. 302 Found ........................................11417.3.4. 303 See Other ....................................11517.3.5. 304 Not Modified .................................11517.3.6. 305 Use Proxy ....................................11517.4. Client Error 4xx ........................................11617.4.1. 400 Bad Request ..................................11617.4.2. 401 Unauthorized .................................11617.4.3. 402 Payment Required .............................11617.4.4. 403 Forbidden ....................................11617.4.5. 404 Not Found ....................................116Schulzrinne, et al.          Standards Track                    [Page 4]

RFC 7826                        RTSP 2.0                   December 201617.4.6. 405 Method Not Allowed ...........................11717.4.7. 406 Not Acceptable ...............................11717.4.8. 407 Proxy Authentication Required ................11717.4.9. 408 Request Timeout ..............................11717.4.10. 410 Gone ........................................11817.4.11. 412 Precondition Failed .........................11817.4.12. 413 Request Message Body Too Large ..............11817.4.13. 414 Request-URI Too Long ........................11817.4.14. 415 Unsupported Media Type ......................11917.4.15. 451 Parameter Not Understood ....................11917.4.16. 452 Illegal Conference Identifier ...............11917.4.17. 453 Not Enough Bandwidth ........................11917.4.18. 454 Session Not Found ...........................11917.4.19. 455 Method Not Valid in This State ..............11917.4.20. 456 Header Field Not Valid for Resource .........11917.4.21. 457 Invalid Range ...............................12017.4.22. 458 Parameter Is Read-Only ......................12017.4.23. 459 Aggregate Operation Not Allowed .............12017.4.24. 460 Only Aggregate Operation Allowed ............12017.4.25. 461 Unsupported Transport .......................12017.4.26. 462 Destination Unreachable .....................12017.4.27. 463 Destination Prohibited ......................12017.4.28. 464 Data Transport Not Ready Yet ................12117.4.29. 465 Notification Reason Unknown .................12117.4.30. 466 Key Management Error ........................12117.4.31. 470 Connection Authorization Required ...........12117.4.32. 471 Connection Credentials Not Accepted .........12117.4.33. 472 Failure to Establish Secure Connection ......12117.5. Server Error 5xx ........................................12217.5.1. 500 Internal Server Error ........................12217.5.2. 501 Not Implemented ..............................12217.5.3. 502 Bad Gateway ..................................12217.5.4. 503 Service Unavailable ..........................12217.5.5. 504 Gateway Timeout ..............................12317.5.6. 505 RTSP Version Not Supported ...................12317.5.7. 551 Option Not Supported .........................12317.5.8. 553 Proxy Unavailable ............................12318. Header Field Definitions .....................................12418.1. Accept ..................................................13418.2. Accept-Credentials ......................................13518.3. Accept-Encoding .........................................13518.4. Accept-Language .........................................13618.5. Accept-Ranges ...........................................13718.6. Allow ...................................................13818.7. Authentication-Info .....................................13818.8. Authorization ...........................................13818.9. Bandwidth ...............................................13918.10. Blocksize ..............................................140Schulzrinne, et al.          Standards Track                    [Page 5]

RFC 7826                        RTSP 2.0                   December 201618.11. Cache-Control ..........................................14018.12. Connection .............................................14318.13. Connection-Credentials .................................14318.14. Content-Base ...........................................14418.15. Content-Encoding .......................................14518.16. Content-Language .......................................14518.17. Content-Length .........................................14618.18. Content-Location .......................................14618.19. Content-Type ...........................................14818.20. CSeq ...................................................14818.21. Date ...................................................15018.22. Expires ................................................15118.23. From ...................................................15118.24. If-Match ...............................................15218.25. If-Modified-Since ......................................15218.26. If-None-Match ..........................................15318.27. Last-Modified ..........................................15418.28. Location ...............................................15418.29. Media-Properties .......................................15418.30. Media-Range ............................................15618.31. MTag ...................................................15718.32. Notify-Reason ..........................................15818.33. Pipelined-Requests .....................................15818.34. Proxy-Authenticate .....................................15918.35. Proxy-Authentication-Info ..............................15918.36. Proxy-Authorization ....................................15918.37. Proxy-Require ..........................................16018.38. Proxy-Supported ........................................16018.39. Public .................................................16118.40. Range ..................................................16218.41. Referrer ...............................................16418.42. Request-Status .........................................16418.43. Require ................................................16518.44. Retry-After ............................................16618.45. RTP-Info ...............................................16718.46. Scale ..................................................16918.47. Seek-Style .............................................17018.48. Server .................................................17118.49. Session ................................................17218.50. Speed ..................................................17318.51. Supported ..............................................17418.52. Terminate-Reason .......................................17518.53. Timestamp ..............................................17518.54. Transport ..............................................17618.55. Unsupported ............................................18318.56. User-Agent .............................................18418.57. Via ....................................................18418.58. WWW-Authenticate .......................................185Schulzrinne, et al.          Standards Track                    [Page 6]

RFC 7826                        RTSP 2.0                   December 201619. Security Framework ...........................................18519.1. RTSP and HTTP Authentication ............................18519.1.1. Digest Authentication ............................18619.2. RTSP over TLS ...........................................18719.3. Security and Proxies ....................................18819.3.1. Accept-Credentials ...............................18919.3.2. User-Approved TLS Procedure ......................19020. Syntax .......................................................19220.1. Base Syntax .............................................19320.2. RTSP Protocol Definition ................................19520.2.1. Generic Protocol Elements ........................19520.2.2. Message Syntax ...................................19820.2.3. Header Syntax ....................................20120.3. SDP Extension Syntax ....................................20921. Security Considerations ......................................20921.1. Signaling Protocol Threats ..............................21021.2. Media Stream Delivery Threats ...........................21321.2.1. Remote DoS Attack ................................21521.2.2. RTP Security Analysis ............................21622. IANA Considerations ..........................................21722.1. Feature Tags ............................................21822.1.1. Description ......................................21822.1.2. Registering New Feature Tags with IANA ...........21822.1.3. Registered Entries ...............................21922.2. RTSP Methods ............................................21922.2.1. Description ......................................21922.2.2. Registering New Methods with IANA ................21922.2.3. Registered Entries ...............................22022.3. RTSP Status Codes .......................................22022.3.1. Description ......................................22022.3.2. Registering New Status Codes with IANA ...........22022.3.3. Registered Entries ...............................22122.4. RTSP Headers ............................................22122.4.1. Description ......................................22122.4.2. Registering New Headers with IANA ................22122.4.3. Registered Entries ...............................22222.5. Accept-Credentials ......................................22322.5.1. Accept-Credentials Policies ......................22322.5.2. Accept-Credentials Hash Algorithms ...............22422.6. Cache-Control Cache Directive Extensions ................22422.7. Media Properties ........................................22522.7.1. Description ......................................22522.7.2. Registration Rules ...............................22622.7.3. Registered Values ................................22622.8. Notify-Reason Values ....................................22622.8.1. Description ......................................22622.8.2. Registration Rules ...............................22622.8.3. Registered Values ................................227Schulzrinne, et al.          Standards Track                    [Page 7]

RFC 7826                        RTSP 2.0                   December 201622.9. Range Header Formats ....................................22722.9.1. Description ......................................22722.9.2. Registration Rules ...............................22722.9.3. Registered Values ................................22822.10. Terminate-Reason Header ................................22822.10.1. Redirect Reasons ................................22822.10.2. Terminate-Reason Header Parameters ..............22922.11. RTP-Info Header Parameters .............................22922.11.1. Description .....................................22922.11.2. Registration Rules ..............................22922.11.3. Registered Values ...............................23022.12. Seek-Style Policies ....................................23022.12.1. Description .....................................23022.12.2. Registration Rules ..............................23022.12.3. Registered Values ...............................23022.13. Transport Header Registries ............................23122.13.1. Transport Protocol Identifier ...................23122.13.2. Transport Modes .................................23322.13.3. Transport Parameters ............................23322.14. URI Schemes ............................................23422.14.1. The "rtsp" URI Scheme ...........................23422.14.2. The "rtsps" URI Scheme ..........................23522.14.3. The "rtspu" URI Scheme ..........................23722.15. SDP Attributes .........................................23822.16. Media Type Registration for text/parameters ............23823. References ...................................................24023.1. Normative References ....................................24023.2. Informative References ..................................245Appendix A. Examples .............................................248A.1. Media on Demand (Unicast) ................................248A.2. Media on Demand Using Pipelining .........................251A.3. Secured Media Session for On-Demand Content ..............254A.4. Media on Demand (Unicast) ................................257A.5. Single-Stream Container Files ............................260A.6. Live Media Presentation Using Multicast ..................263A.7. Capability Negotiation ...................................264Appendix B. RTSP Protocol State Machine ..........................265B.1. States ...................................................266B.2. State Variables ..........................................266B.3. Abbreviations ............................................266B.4. State Tables .............................................267Appendix C. Media-Transport Alternatives .........................272C.1. RTP ......................................................272C.1.1. AVP ..................................................272C.1.2. AVP/UDP ..............................................273C.1.3. AVPF/UDP .............................................274C.1.4. SAVP/UDP .............................................275C.1.5. SAVPF/UDP ............................................277Schulzrinne, et al.          Standards Track                    [Page 8]

RFC 7826                        RTSP 2.0                   December 2016C.1.6. RTCP Usage with RTSP .................................278C.2. RTP over TCP .............................................279C.2.1. Interleaved RTP over TCP .............................280C.2.2. RTP over Independent TCP .............................280C.3. Handling Media-Clock Time Jumps in the RTP Media Layer ...284C.4. Handling RTP Timestamps after PAUSE ......................287C.5. RTSP/RTP Integration  ....................................290C.6. Scaling with RTP .........................................290C.7. Maintaining NPT Synchronization with RTP Timestamps ......290C.8. Continuous Audio .........................................290C.9. Multiple Sources in an RTP Session .......................290      C.10. Usage of SSRCs and the RTCP BYE Message during an RTSP            Session .................................................290C.11. Future Additions ........................................291Appendix D. Use of SDP for RTSP Session Descriptions .............292D.1. Definitions  .............................................292D.1.1. Control URI ..........................................292D.1.2. Media Streams ........................................294D.1.3. Payload Type(s) ......................................294D.1.4. Format-Specific Parameters ...........................294D.1.5. Directionality of Media Stream .......................295D.1.6. Range of Presentation ................................295D.1.7. Time of Availability .................................296D.1.8. Connection Information ...............................297D.1.9. Message Body Tag .....................................297D.2. Aggregate Control Not Available ..........................298D.3. Aggregate Control Available ..............................298D.4. Grouping of Media Lines in SDP ...........................299D.5. RTSP External SDP Delivery ...............................300Appendix E. RTSP Use Cases .......................................300E.1. On-Demand Playback of Stored Content .....................300E.2. Unicast Distribution of Live Content .....................302E.3. On-Demand Playback Using Multicast .......................303E.4. Inviting an RTSP Server into a Conference ................303E.5. Live Content Using Multicast .............................304Appendix F. Text Format for Parameters ...........................305Appendix G. Requirements for Unreliable Transport of RTSP ........305Appendix H. Backwards-Compatibility Considerations ...............306H.1. Play Request in Play State ...............................307H.2. Using Persistent Connections .............................307Appendix I. Changes ..............................................307I.1. Brief Overview ...........................................308I.2. Detailed List of Changes .................................309   Acknowledgements .................................................316   Contributors  ....................................................317   Authors' Addresses ...............................................318Schulzrinne, et al.          Standards Track                    [Page 9]

RFC 7826                        RTSP 2.0                   December 20161.  Introduction   This memo defines version 2.0 of the Real-Time Streaming Protocol   (RTSP 2.0).  RTSP 2.0 is an application-layer protocol for the setup   and control over the delivery of data with real-time properties,   typically streaming media.  Streaming media is, for instance, video   on demand or audio live streaming.  Put simply, RTSP acts as a   "network remote control" for multimedia servers.   The protocol operates between RTSP 2.0 clients and servers, but it   also supports the use of proxies placed between clients and servers.   Clients can request information about streaming media from servers by   asking for a description of the media or use media description   provided externally.  The media delivery protocol is used to   establish the media streams described by the media description.   Clients can then request to play out the media, pause it, or stop it   completely.  The requested media can consist of multiple audio and   video streams that are delivered as time-synchronized streams from   servers to clients.   RTSP 2.0 is a replacement of RTSP 1.0 [RFC2326] and this document   obsoletes that specification.  This protocol is based on RTSP 1.0 but   is not backwards compatible other than in the basic version   negotiation mechanism.  The changes between the two documents are   listed inAppendix I.  There are many reasons why RTSP 2.0 can't be   backwards compatible with RTSP 1.0; some of the main ones are as   follows:   o  Most headers that needed to be extensible did not define the      allowed syntax, preventing safe deployment of extensions;   o  the changed behavior of the PLAY method when received in Play      state;   o  the changed behavior of the extensibility model and its mechanism;      and   o  the change of syntax for some headers.   There are so many small updates that changing versions became   necessary to enable clarification and consistent behavior.  Anyone   implementing RTSP for a new use case in which they have not installed   RTSP 1.0 should only implement RTSP 2.0 to avoid having to deal with   RTSP 1.0 inconsistencies.   This document is structured as follows.  It begins with an overview   of the protocol operations and its functions in an informal way.   Then, a set of definitions of terms used and document conventions isSchulzrinne, et al.          Standards Track                   [Page 10]

RFC 7826                        RTSP 2.0                   December 2016   introduced.  These are followed by the actual RTSP 2.0 core protocol   specification.  The appendices describe and define some   functionalities that are not part of the core RTSP specification, but   which are still important to enable some usages.  Among them, the RTP   usage is defined inAppendix C, the Session Description Protocol   (SDP) usage with RTSP is defined inAppendix D, and the "text/   parameters" file formatAppendix F, are three normative specification   appendices.  Other appendices include a number of informational parts   discussing the changes, use cases, different considerations or   motivations.2.  Protocol Overview   This section provides an informative overview of the different   mechanisms in the RTSP 2.0 protocol to give the reader a high-level   understanding before getting into all the specific details.  In case   of conflict with this description and the later sections, the later   sections take precedence.  For more information about use cases   considered for RTSP, seeAppendix E.   RTSP 2.0 is a bidirectional request and response protocol that first   establishes a context including content resources (the media) and   then controls the delivery of these content resources from the   provider to the consumer.  RTSP has three fundamental parts: Session   Establishment, Media Delivery Control, and an extensibility model   described below.  The protocol is based on some assumptions about   existing functionality to provide a complete solution for client-   controlled real-time media delivery.   RTSP uses text-based messages, requests and responses, that may   contain a binary message body.  An RTSP request starts with a method   line that identifies the method, the protocol, and version and the   resource on which to act.  The resource is identified by a URI and   the hostname part of the URI is used by RTSP client to resolve the   IPv4 or IPv6 address of the RTSP server.  Following the method line   are a number of RTSP headers.  These lines are ended by two   consecutive carriage return line feed (CRLF) character pairs.  The   message body, if present, follows the two CRLF character pairs, and   the body's length is described by a message header.  RTSP responses   are similar, but they start with a response line with the protocol   and version followed by a status code and a reason phrase.  RTSP   messages are sent over a reliable transport protocol between the   client and server.  RTSP 2.0 requires clients and servers to   implement TCP and TLS over TCP as mandatory transports for RTSP   messages.Schulzrinne, et al.          Standards Track                   [Page 11]

RFC 7826                        RTSP 2.0                   December 20162.1.  Presentation Description   RTSP exists to provide access to multimedia presentations and content   but tries to be agnostic about the media type or the actual media   delivery protocol that is used.  To enable a client to implement a   complete system, an RTSP-external mechanism for describing the   presentation and the delivery protocol(s) is used.  RTSP assumes that   this description is either delivered completely out of band or as a   data object in the response to a client's request using the DESCRIBE   method (Section 13.2).   Parameters that commonly have to be included in the presentation   description are the following:   o  The number of media streams;   o  the resource identifier for each media stream/resource that is to      be controlled by RTSP;   o  the protocol that will be used to deliver each media stream;   o  the transport protocol parameters that are not negotiated or vary      with each client;   o  the media-encoding information enabling a client to correctly      decode the media upon reception; and   o  an aggregate control resource identifier.   RTSP uses its own URI schemes ("rtsp" and "rtsps") to reference media   resources and aggregates under common control (seeSection 4.2).   This specification describes inAppendix D how one uses SDP [RFC4566]   for describing the presentation.2.2.  Session Establishment   The RTSP client can request the establishment of an RTSP session   after having used the presentation description to determine which   media streams are available, which media delivery protocol is used,   and the resource identifiers of the media streams.  The RTSP session   is a common context between the client and the server that consists   of one or more media resources that are to be under common media   delivery control.   The client creates an RTSP session by sending a request using the   SETUP method (Section 13.3) to the server.  In the Transport header   (Section 18.54) of the SETUP request, the client also includes allSchulzrinne, et al.          Standards Track                   [Page 12]

RFC 7826                        RTSP 2.0                   December 2016   the transport parameters necessary to enable the media delivery   protocol to function.  This includes parameters that are   preestablished by the presentation description but necessary for any   middlebox to correctly handle the media delivery protocols.  The   Transport header in a request may contain multiple alternatives for   media delivery in a prioritized list, which the server can select   from.  These alternatives are typically based on information in the   presentation description.   When receiving a SETUP request, the server determines if the media   resource is available and if one or more of the of the transport   parameter specifications are acceptable.  If that is successful, an   RTSP session context is created and the relevant parameters and state   is stored.  An identifier is created for the RTSP session and   included in the response in the Session header (Section 18.49).  The   SETUP response includes a Transport header that specifies which of   the alternatives has been selected and relevant parameters.   A SETUP request that references an existing RTSP session but   identifies a new media resource is a request to add that media   resource under common control with the already-present media   resources in an aggregated session.  A client can expect this to work   for all media resources under RTSP control within a multimedia   content container.  However, a server will likely refuse to aggregate   resources from different content containers.  Even if an RTSP session   contains only a single media stream, the RTSP session can be   referenced by the aggregate control URI.   To avoid an extra round trip in the session establishment of   aggregated RTSP sessions, RTSP 2.0 supports pipelined requests; i.e.,   the client can send multiple requests back-to-back without waiting   first for the completion of any of them.  The client uses a client-   selected identifier in the Pipelined-Requests header (Section 18.33)   to instruct the server to bind multiple requests together as if they   included the session identifier.   The SETUP response also provides additional information about the   established sessions in a couple of different headers.  The Media-   Properties header (Section 18.29) includes a number of properties   that apply for the aggregate that is valuable when doing media   delivery control and configuring user interface.  The Accept-Ranges   header (Section 18.5) informs the client about range formats that the   server supports for these media resources.  The Media-Range header   (Section 18.30) informs the client about the time range of the media   currently available.Schulzrinne, et al.          Standards Track                   [Page 13]

RFC 7826                        RTSP 2.0                   December 20162.3.  Media Delivery Control   After having established an RTSP session, the client can start   controlling the media delivery.  The basic operations are "begin   playback", using the PLAY method (Section 13.4) and "suspend (pause)   playback" by using the PAUSE method (Section 13.6).  PLAY also allows   for choosing the starting media position from which the server should   deliver the media.  The positioning is done by using the Range header   (Section 18.40) that supports several different time formats: Normal   Play Time (NPT) (Section 4.4.2), Society of Motion Picture and   Television Engineers (SMPTE) Timestamps (Section 4.4.1), and absolute   time (Section 4.4.3).  The Range header also allows the client to   specify a position where delivery should end, thus allowing a   specific interval to be delivered.   The support for positioning/searching within media content depends on   the content's media properties.  Content exists in a number of   different types, such as on-demand, live, and live with simultaneous   recording.  Even within these categories, there are differences in   how the content is generated and distributed, which affect how it can   be accessed for playback.  The properties applicable for the RTSP   session are provided by the server in the SETUP response using the   Media-Properties header (Section 18.29).  These are expressed using   one or several independent attributes.  A first attribute is Random-   Access, which indicates whether positioning is possible, and with   what granularity.  Another aspect is whether the content will change   during the lifetime of the session.  While on-demand content will be   provided in full from the beginning, a live stream being recorded   results in the length of the accessible content growing as the   session goes on.  There also exists content that is dynamically built   by a protocol other than RTSP and, thus, also changes in steps during   the session, but maybe not continuously.  Furthermore, when content   is recorded, there are cases where the complete content is not   maintained, but, for example, only the last hour.  All of these   properties result in the need for mechanisms that will be discussed   below.   When the client accesses on-demand content that allows random access,   the client can issue the PLAY request for any point in the content   between the start and the end.  The server will deliver media from   the closest random access point prior to the requested point and   indicate that in its PLAY response.  If the client issues a PAUSE,   the delivery will be halted and the point at which the server stopped   will be reported back in the response.  The client can later resume   by sending a PLAY request without a Range header.  When the server is   about to complete the PLAY request by delivering the end of the   content or the requested range, the server will send a PLAY_NOTIFY   request (Section 13.5) indicating this.Schulzrinne, et al.          Standards Track                   [Page 14]

RFC 7826                        RTSP 2.0                   December 2016   When playing live content with no extra functions, such as recording,   the client will receive the live media from the server after having   sent a PLAY request.  Seeking in such content is not possible as the   server does not store it, but only forwards it from the source of the   session.  Thus, delivery continues until the client sends a PAUSE   request, tears down the session, or the content ends.   For live sessions that are being recorded, the client will need to   keep track of how the recording progresses.  Upon session   establishment, the client will learn the current duration of the   recording from the Media-Range header.  Because the recording is   ongoing, the content grows in direct relation to the time passed.   Therefore, each server's response to a PLAY request will contain the   current Media-Range header.  The server should also regularly send   (approximately every 5 minutes) the current media range in a   PLAY_NOTIFY request (Section 13.5.2).  If the live transmission ends,   the server must send a PLAY_NOTIFY request with the updated Media-   Properties indicating that the content stopped being a recorded live   session and instead became on-demand content; the request also   contains the final media range.  While the live delivery continues,   the client can request to play the current live point by using the   NPT timescale symbol "now", or it can request a specific point in the   available content by an explicit range request for that point.  If   the requested point is outside of the available interval, the server   will adjust the position to the closest available point, i.e., either   at the beginning or the end.   A special case of recording is that where the recording is not   retained longer than a specific time period; thus, as the live   delivery continues, the client can access any media within a moving   window that covers, for example, "now" to "now" minus 1 hour.  A   client that pauses on a specific point within the content may not be   able to retrieve the content anymore.  If the client waits too long   before resuming the pause point, the content may no longer be   available.  In this case, the pause point will be adjusted to the   closest point in the available media.2.4.  Session Parameter Manipulations   A session may have additional state or functionality that affects how   the server or client treats the session or content, how it functions,   or feedback on how well the session works.  Such extensions are not   defined in this specification, but they may be covered in various   extensions.  RTSP has two methods for retrieving and setting   parameter values on either the client or the server: GET_PARAMETER   (Section 13.8) and SET_PARAMETER (Section 13.9).  These methods carry   the parameters in a message body of the appropriate format.  One can   also use headers to query state with the GET_PARAMETER method.  As anSchulzrinne, et al.          Standards Track                   [Page 15]

RFC 7826                        RTSP 2.0                   December 2016   example, clients needing to know the current media range for a time-   progressing session can use the GET_PARAMETER method and include the   media range.  Furthermore, synchronization information can be   requested by using a combination of RTP-Info (Section 18.45) and   Range (Section 18.40).   RTSP 2.0 does not have a strong mechanism for negotiating the headers   or parameters and their formats.  However, responses will indicate   request-headers or parameters that are not supported.  A priori   determination of what features are available needs to be done through   out-of-band mechanisms, like the session description, or through the   usage of feature tags (Section 4.5).2.5.  Media Delivery   This document specifies how media is delivered with RTP [RFC3550]   over UDP [RFC768], TCP [RFC793], or the RTSP connection.  Additional   protocols may be specified in the future as needed.   The usage of RTP as a media delivery protocol requires some   additional information to function well.  The PLAY response contains   information to enable reliable and timely delivery of how a client   should synchronize different sources in the different RTP sessions.   It also provides a mapping between RTP timestamps and the content-   time scale.  When the server wants to notify the client about the   completion of the media delivery, it sends a PLAY_NOTIFY request to   the client.  The PLAY_NOTIFY request includes information about the   stream end, including the last RTP sequence number for each stream,   thus enabling the client to empty the buffer smoothly.2.5.1.  Media Delivery Manipulations   The basic playback functionality of RTSP enables delivery of a range   of requested content to the client at the pace intended by the   content's creator.  However, RTSP can also manipulate the delivery to   the client in two ways.   Scale:  The ratio of media-content time delivered per unit of      playback time.   Speed:  The ratio of playback time delivered per unit of wallclock      time.   Both affect the media delivery per time unit.  However, they   manipulate two independent timescales and the effects are possible to   combine.Schulzrinne, et al.          Standards Track                   [Page 16]

RFC 7826                        RTSP 2.0                   December 2016   Scale (Section 18.46) is used for fast-forward or slow-motion control   as it changes the amount of content timescale that should be played   back per time unit.  Scale > 1.0, means fast forward, e.g., scale =   2.0 results in that 2 seconds of content being played back every   second of playback.  Scale = 1.0 is the default value that is used if   no scale is specified, i.e., playback at the content's original rate.   Scale values between 0 and 1.0 provide for slow motion.  Scale can be   negative to allow for reverse playback in either regular pace   (scale = -1.0), fast backwards (scale < -1.0), or slow-motion   backwards (-1.0 < scale < 0).  Scale = 0 would be equal to pause and   is not allowed.   In most cases, the realization of scale means server-side   manipulation of the media to ensure that the client can actually play   it back.  The nature of these media manipulations and when they are   needed is highly media-type dependent.  Let's consider two common   media types, audio and video.   It is very difficult to modify the playback rate of audio.   Typically, no more than a factor of two is possible while maintaining   intelligibility by changing the pitch and rate of speech.  Music goes   out of tune if one tries to manipulate the playback rate by   resampling it.  This is a well-known problem, and audio is commonly   muted or played back in short segments with skips to keep up with the   current playback point.   For video, it is possible to manipulate the frame rate, although the   rendering capabilities are often limited to certain frame rates.   Also, the allowed bitrates in decoding, the structure used in the   encoding, and the dependency between frames and other capabilities of   the rendering device limits the possible manipulations.  Therefore,   the basic fast-forward capabilities often are implemented by   selecting certain subsets of frames.   Due to the media restrictions, the possible scale values are commonly   restricted to the set of realizable scale ratios.  To enable the   clients to select from the possible scale values, RTSP can signal the   supported scale ratios for the content.  To support aggregated or   dynamic content, where this may change during the ongoing session and   dependent on the location within the content, a mechanism for   updating the media properties and the scale factor currently in use,   exists.   Speed (Section 18.50) affects how much of the playback timeline is   delivered in a given wallclock period.  The default is Speed = 1   which means to deliver at the same rate the media is consumed.   Speed > 1 means that the receiver will get content faster than it   regularly would consume it.  Speed < 1 means that delivery is slowerSchulzrinne, et al.          Standards Track                   [Page 17]

RFC 7826                        RTSP 2.0                   December 2016   than the regular media rate.  Speed values of 0 or lower have no   meaning and are not allowed.  This mechanism enables two general   functionalities.  One is client-side scale operations, i.e., the   client receives all the frames and makes the adjustment to the   playback locally.  The second is delivery control for the buffering   of media.  By specifying a speed over 1.0, the client can build up   the amount of playback time it has present in its buffers to a level   that is sufficient for its needs.   A naive implementation of Speed would only affect the transmission   schedule of the media and has a clear impact on the needed bandwidth.   This would result in the data rate being proportional to the speed   factor.  Speed = 1.5, i.e., 50% faster than normal delivery, would   result in a 50% increase in the data-transport rate.  Whether or not   that can be supported depends solely on the underlying network path.   Scale may also have some impact on the required bandwidth due to the   manipulation of the content in the new playback schedule.  An example   is fast forward where only the independently decodable intra-frames   are included in the media stream.  This usage of solely intra-frames   increases the data rate significantly compared to a normal sequence   with the same number of frames, where most frames are encoded using   prediction.   This potential increase of the data rate needs to be handled by the   media sender.  The client has requested that the media be delivered   in a specific way, which should be honored.  However, the media   sender cannot ignore if the network path between the sender and the   receiver can't handle the resulting media stream.  In that case, the   media stream needs to be adapted to fit the available resources of   the path.  This can result in a reduced media quality.   The need for bitrate adaptation becomes especially problematic in   connection with the Speed semantics.  If the goal is to fill up the   buffer, the client may not want to do that at the cost of reduced   quality.  If the client wants to make local playout changes, then it   may actually require that the requested speed be honored.  To resolve   this issue, Speed uses a range so that both cases can be supported.   The server is requested to use the highest possible speed value   within the range, which is compatible with the available bandwidth.   As long as the server can maintain a speed value within the range, it   shall not change the media quality, but instead modify the actual   delivery rate in response to available bandwidth and reflect this in   the Speed value in the response.  However, if this is not possible,   the server should instead modify the media quality to respect the   lowest speed value and the available bandwidth.Schulzrinne, et al.          Standards Track                   [Page 18]

RFC 7826                        RTSP 2.0                   December 2016   This functionality enables the local scaling implementation to use a   tight range, or even a range where the lower bound equals the upper   bound, to identify that it requires the server to deliver the   requested amount of media time per delivery time, independent of how   much it needs to adapt the media quality to fit within the available   path bandwidth.  For buffer filling, it is suitable to use a range   with a reasonable span and with a lower bound at the nominal media   rate 1.0, such as 1.0 - 2.5.  If the client wants to reduce the   buffer, it can specify an upper bound that is below 1.0 to force the   server to deliver slower than the nominal media rate.2.6.  Session Maintenance and Termination   The session context that has been established is kept alive by having   the client show liveness.  This is done in two main ways:   o  Media-transport protocol keep-alive.  RTP Control Protocol (RTCP)      may be used when using RTP.   o  Any RTSP request referencing the session context.Section 10.5 discusses the methods for showing liveness in more   depth.  If the client fails to show liveness for more than the   established session timeout value (normally 60 seconds), the server   may terminate the context.  Other values may be selected by the   server through the inclusion of the timeout parameter in the session   header.   The session context is normally terminated by the client sending a   TEARDOWN request (Section 13.7) to the server referencing the   aggregated control URI.  An individual media resource can be removed   from a session context by a TEARDOWN request referencing that   particular media resource.  If all media resources are removed from a   session context, the session context is terminated.   A client may keep the session alive indefinitely if allowed by the   server; however, a client is advised to release the session context   when an extended period of time without media delivery activity has   passed.  The client can re-establish the session context if required   later.  What constitutes an extended period of time is dependent on   the client, server, and their usage.  It is recommended that the   client terminate the session before ten times the session timeout   value has passed.  A server may terminate the session after one   session timeout period without any client activity beyond keep-alive.   When a server terminates the session context, it does so by sending a   TEARDOWN request indicating the reason.Schulzrinne, et al.          Standards Track                   [Page 19]

RFC 7826                        RTSP 2.0                   December 2016   A server can also request that the client tear down the session and   re-establish it at an alternative server, as may be needed for   maintenance.  This is done by using the REDIRECT method   (Section 13.10).  The Terminate-Reason header (Section 18.52) is used   to indicate when and why.  The Location header indicates where it   should connect if there is an alternative server available.  When the   deadline expires, the server simply stops providing the service.  To   achieve a clean closure, the client needs to initiate session   termination prior to the deadline.  In case the server has no other   server to redirect to, and it wants to close the session for   maintenance, it shall use the TEARDOWN method with a Terminate-Reason   header.2.7.  Extending RTSP   RTSP is quite a versatile protocol that supports extensions in many   different directions.  Even this core specification contains several   blocks of functionality that are optional to implement.  The use case   and need for the protocol deployment should determine what parts are   implemented.  Allowing for extensions makes it possible for RTSP to   address additional use cases.  However, extensions will affect the   interoperability of the protocol; therefore, it is important that   they can be added in a structured way.   The client can learn the capability of a server by using the OPTIONS   method (Section 13.1) and the Supported header (Section 18.51).  It   can also try and possibly fail using new methods or require that   particular features be supported using the Require (Section 18.43) or   Proxy-Require (Section 18.37) header.   The RTSP, in itself, can be extended in three ways, listed here in   increasing order of the magnitude of changes supported:   o  Existing methods can be extended with new parameters, for example,      headers, as long as these parameters can be safely ignored by the      recipient.  If the client needs negative acknowledgment when a      method extension is not supported, a tag corresponding to the      extension may be added in the field of the Require or Proxy-      Require headers.   o  New methods can be added.  If the recipient of the message does      not understand the request, it must respond with error code 501      (Not Implemented) so that the sender can avoid using this method      again.  A client may also use the OPTIONS method to inquire about      methods supported by the server.  The server must list the methods      it supports using the Public response-header.Schulzrinne, et al.          Standards Track                   [Page 20]

RFC 7826                        RTSP 2.0                   December 2016   o  A new version of the protocol can be defined, allowing almost all      aspects (except the position of the protocol version number) to      change.  A new version of the protocol must be registered through      a Standards Track document.   The basic capability discovery mechanism can be used to both discover   support for a certain feature and to ensure that a feature is   available when performing a request.  For a detailed explanation of   this, seeSection 11.   New media delivery protocols may be added and negotiated at session   establishment, in addition to extensions to the core protocol.   Certain types of protocol manipulations can be done through parameter   formats using SET_PARAMETER and GET_PARAMETER.3.  Document Conventions3.1.  Notational Conventions   All the mechanisms specified in this document are described in both   prose and the Augmented Backus-Naur form (ABNF) described in detail   in [RFC5234].   Indented paragraphs are used to provide informative background and   motivation.  This is intended to give readers who were not involved   with the formulation of the specification an understanding of why   things are the way they are in RTSP.   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and   "OPTIONAL" in this document are to be interpreted as described in   [RFC2119].   The word, "unspecified" is used to indicate functionality or features   that are not defined in this specification.  Such functionality   cannot be used in a standardized manner without further definition in   an extension specification to RTSP.3.2.  Terminology   Aggregate control:  The concept of controlling multiple streams using      a single timeline, generally one maintained by the server.  A      client, for example, uses aggregate control when it issues a      single play or pause message to simultaneously control both the      audio and video in a movie.  A session that is under aggregate      control is referred to as an "aggregated session".Schulzrinne, et al.          Standards Track                   [Page 21]

RFC 7826                        RTSP 2.0                   December 2016   Aggregate control URI:  The URI used in an RTSP request to refer to      and control an aggregated session.  It normally, but not always,      corresponds to the presentation URI specified in the session      description.  SeeSection 13.3 for more information.   Client:  The client is the requester of media service from the media      server.   Connection:  A transport-layer virtual circuit established between      two programs for the purpose of communication.   Container file:  A file that may contain multiple media streams that      often constitute a presentation when played together.  The concept      of a container file is not embedded in the protocol.  However,      RTSP servers may offer aggregate control on the media streams      within these files.   Continuous media:  Data where there is a timing relationship between      source and sink; that is, the sink needs to reproduce the timing      relationship that existed at the source.  The most common examples      of continuous media are audio and motion video.  Continuous media      can be real time (interactive or conversational), where there is a      "tight" timing relationship between source and sink or it can be      streaming where the relationship is less strict.   Feature tag:  A tag representing a certain set of functionality,      i.e., a feature.   IRI:  An Internationalized Resource Identifier is similar to a URI      but allows characters from the whole Universal Character Set      (Unicode/ISO 10646), rather than the US-ASCII only.  See [RFC3987]      for more information.   Live:  A live presentation or session originates media from an event      taking place at the same time as the media delivery.  Live      sessions often have an unbound or only loosely defined duration      and seek operations may not be possible.   Media initialization:  The datatype- or codec-specific      initialization.  This includes such things as clock rates, color      tables, etc.  Any transport-independent information that is      required by a client for playback of a media stream occurs in the      media initialization phase of stream setup.   Media parameter:  A parameter specific to a media type that may be      changed before or during stream delivery.Schulzrinne, et al.          Standards Track                   [Page 22]

RFC 7826                        RTSP 2.0                   December 2016   Media server:  The server providing media-delivery services for one      or more media streams.  Different media streams within a      presentation may originate from different media servers.  A media      server may reside on the same host or on a different host from      which the presentation is invoked.   (Media) Stream:  A single media instance, e.g., an audio stream or a      video stream as well as a single whiteboard or shared application      group.  When using RTP, a stream consists of all RTP and RTCP      packets created by a media source within an RTP session.   Message:  The basic unit of RTSP communication, consisting of a      structured sequence of octets matching the syntax defined inSection 20 and transmitted over a transport between RTSP agents.      A message is either a request or a response.   Message body:  The information transferred as the payload of a      message (request or response).  A message body consists of meta-      information in the form of message body headers and content in the      form of an arbitrary number of data octets, as described inSection 9.   Non-aggregated control:  Control of a single media stream.   Presentation:  A set of one or more streams presented to the client      as a complete media feed and described by a presentation      description as defined below.  Presentations with more than one      media stream are often handled in RTSP under aggregate control.   Presentation description:  A presentation description contains      information about one or more media streams within a presentation,      such as the set of encodings, network addresses, and information      about the content.  Other IETF protocols, such as SDP ([RFC4566]),      use the term "session" for a presentation.  The presentation      description may take several different formats, including but not      limited to SDP format.   Response:  An RTSP response to a request.  One type of RTSP message.      If an HTTP response is meant, it is indicated explicitly.   Request:  An RTSP request.  One type of RTSP message.  If an HTTP      request is meant, it is indicated explicitly.   Request-URI:  The URI used in a request to indicate the resource on      which the request is to be performed.Schulzrinne, et al.          Standards Track                   [Page 23]

RFC 7826                        RTSP 2.0                   December 2016   RTSP agent:  Either an RTSP client, an RTSP server, or an RTSP proxy.      In this specification, there are many capabilities that are common      to these three entities such as the capability to send requests or      receive responses.  This term will be used when describing      functionality that is applicable to all three of these entities.   RTSP session:  A stateful abstraction upon which the main control      methods of RTSP operate.  An RTSP session is a common context; it      is created and maintained on a client's request and can be      destroyed by either the client or server.  It is established by an      RTSP server upon the completion of a successful SETUP request      (when a 200 OK response is sent) and is labeled with a session      identifier at that time.  The session exists until timed out by      the server or explicitly removed by a TEARDOWN request.  An RTSP      session is a stateful entity; an RTSP server maintains an explicit      session state machine (seeAppendix B) where most state      transitions are triggered by client requests.  The existence of a      session implies the existence of state about the session's media      streams and their respective transport mechanisms.  A given      session can have one or more media streams associated with it.  An      RTSP server uses the session to aggregate control over multiple      media streams.   Origin server:  The server on which a given resource resides.   Seeking:  Requesting playback from a particular point in the content      time line.   Transport initialization:  The negotiation of transport information      (e.g., port numbers, transport protocols) between the client and      the server.   URI:  A Universal Resource Identifier; see [RFC3986].  The URIs used      in RTSP are generally URLs as they give a location for the      resource.  As URLs are a subset of URIs, they will be referred to      as URIs to cover also the cases when an RTSP URI would not be a      URL.   URL:  A Universal Resource Locator is a URI that identifies the      resource through its primary access mechanism rather than      identifying the resource by name or by some other attribute(s) of      that resource.Schulzrinne, et al.          Standards Track                   [Page 24]

RFC 7826                        RTSP 2.0                   December 20164.  Protocol Parameters4.1.  RTSP Version   This specification defines version 2.0 of RTSP.   RTSP uses a "<major>.<minor>" numbering scheme to indicate versions   of the protocol.  The protocol versioning policy is intended to allow   the sender to indicate the format of a message and its capacity for   understanding further RTSP communication rather than the features   obtained via that communication.  No change is made to the version   number for the addition of message components that do not affect   communication behavior or that only add to extensible field values.   The <minor> number is incremented when the changes made to the   protocol add features that do not change the general message parsing   algorithm but that may add to the message semantics and imply   additional capabilities of the sender.  The <major> number is   incremented when the format of a message within the protocol is   changed.  The version of an RTSP message is indicated by an RTSP-   Version field in the first line of the message.  Note that the major   and minor numbers MUST be treated as separate integers and that each   MAY be incremented higher than a single digit.  Thus, RTSP/2.4 is a   lower version than RTSP/2.13, which, in turn, is lower than   RTSP/12.3.  Leading zeros SHALL NOT be sent and MUST be ignored by   recipients.4.2.  RTSP IRI and URI   RTSP 2.0 defines and registers or updates three URI schemes "rtsp",   "rtsps", and "rtspu".  The usage of the last, "rtspu", is unspecified   in RTSP 2.0 and is defined here to register the URI scheme that was   defined in RTSP 1.0.  The "rtspu" scheme indicates unspecified   transport of the RTSP messages over unreliable transport means (UDP   in RTSP 1.0).  An RTSP server MUST respond with an error code   indicating the "rtspu" scheme is not implemented (501) to a request   that carries a "rtspu" URI scheme.   The details of the syntax of "rtsp" and "rtsps" URIs have been   changed from RTSP 1.0.  These changes include the addition of:   o  Support for an IPv6 literal in the host part and future IP      literals through a mechanism defined in [RFC3986].   o  A new relative format to use in the RTSP elements that is not      required to start with "/".Schulzrinne, et al.          Standards Track                   [Page 25]

RFC 7826                        RTSP 2.0                   December 2016   Neither should have any significant impact on interoperability.  If   IPv6 literals are needed in the RTSP URI, then that RTSP server must   be IPv6 capable, and RTSP 1.0 is not a fully IPv6 capable protocol.   If an RTSP 1.0 client attempts to process the URI, the URI will not   match the allowed syntax, it will be considered invalid, and   processing will be stopped.  This is clearly a failure to reach the   resource; however, it is not a signification issue as RTSP 2.0   support was needed anyway in both server and client.  Thus, failure   will only occur in a later step when there is an RTSP version   mismatch between client and server.  The second change will only   occur inside RTSP message headers, as the Request-URI must be an   absolute URI.  Thus, such usages will only occur after an agent has   accepted and started processing RTSP 2.0 messages, and an agent using   RTSP 1.0 only will not be required to parse such types of relative   URIs.   This specification also defines the format of RTSP IRIs [RFC3987]   that can be used as RTSP resource identifiers and locators on web   pages, user interfaces, on paper, etc.  However, the RTSP request   message format only allows usage of the absolute URI format.  The   RTSP IRI format MUST use the rules and transformation for IRIs to   URIs, as defined in [RFC3987].  This allows a URI that matches the   RTSP 2.0 specification, and so is suitable for use in a request, to   be created from an RTSP IRI.   The RTSP IRI and URI are both syntax restricted compared to the   generic syntax defined in [RFC3986] and [RFC3987]:   o  An absolute URI requires the authority part; i.e., a host identity      MUST be provided.   o  Parameters in the path element are prefixed with the reserved      separator ";".   The "scheme" and "host" parts of all URIs [RFC3986] and IRIs   [RFC3987] are case insensitive.  All other parts of RTSP URIs and   IRIs are case sensitive, and they MUST NOT be case mapped.   The fragment identifier is used as defined in Sections3.5 and4.3 of   [RFC3986], i.e., the fragment is to be stripped from the IRI by the   requester and not included in the Request-URI.  The user agent needs   to interpret the value of the fragment based on the media type the   request relates to; i.e., the media type indicated in Content-Type   header in the response to a DESCRIBE request.   The syntax of any URI query string is unspecified and responder   (usually the server) specific.  The query is, from the requester's   perspective, an opaque string and needs to be handled as such.Schulzrinne, et al.          Standards Track                   [Page 26]

RFC 7826                        RTSP 2.0                   December 2016   Please note that relative URIs with queries are difficult to handle   due to the relative URI handling rules ofRFC 3986.  Any change of   the path element using a relative URI results in the stripping of the   query, which means the relative part needs to contain the query.   The URI scheme "rtsp" requires that commands be issued via a reliable   protocol (within the Internet, TCP), while the scheme "rtsps"   identifies a reliable transport using secure transport (TLS   [RFC5246]); seeSection 19.   For the scheme "rtsp", if no port number is provided in the authority   part of the URI, the port number 554 MUST be used.  For the scheme   "rtsps", if no port number is provided in the authority part of the   URI port number, the TCP port 322 MUST be used.   A presentation or a stream is identified by a textual media   identifier, using the character set and escape conventions of URIs   [RFC3986].  URIs may refer to a stream or an aggregate of streams;   i.e., a presentation.  Accordingly, requests described inSection 13   can apply to either the whole presentation or an individual stream   within the presentation.  Note that some request methods can only be   applied to streams, not presentations, and vice versa.   For example, the RTSP URI:      rtsp://media.example.com:554/twister/audiotrack   may identify the audio stream within the presentation "twister",   which can be controlled via RTSP requests issued over a TCP   connection to port 554 of host media.example.com.   Also, the RTSP URI:      rtsp://media.example.com:554/twister   identifies the presentation "twister", which may be composed of audio   and video streams, but could also be something else, such as a random   media redirector.      This does not imply a standard way to reference streams in URIs.      The presentation description defines the hierarchical      relationships in the presentation and the URIs for the individual      streams.  A presentation description may name a stream "a.mov" and      the whole presentation "b.mov".   The path components of the RTSP URI are opaque to the client and do   not imply any particular file system structure for the server.Schulzrinne, et al.          Standards Track                   [Page 27]

RFC 7826                        RTSP 2.0                   December 2016      This decoupling also allows presentation descriptions to be used      with non-RTSP media control protocols simply by replacing the      scheme in the URI.4.3.  Session Identifiers   Session identifiers are strings of a length between 8-128 characters.   A session identifier MUST be generated using methods that make it   cryptographically random (see [RFC4086]).  It is RECOMMENDED that a   session identifier contain 128 bits of entropy, i.e., approximately   22 characters from a high-quality generator (seeSection 21).   However, note that the session identifier does not provide any   security against session hijacking unless it is kept confidential by   the client, server, and trusted proxies.4.4.  Media-Time Formats   RTSP currently supports three different media-time formats defined   below.  Additional time formats may be specified in the future.   These time formats can be used with the Range header (Section 18.40)   to request playback and specify at which media position protocol   requests actually will or have taken place.  They are also used in   description of the media's properties using the Media-Range header   (Section 18.30).  The unqualified format identifier is used on its   own in Accept-Ranges header (Section 18.5) to declare supported time   formats and also in the Range header (Section 18.40) to request the   time format used in the response.4.4.1.  SMPTE-Relative Timestamps   A timestamp may use a format derived from a Society of Motion Picture   and Television Engineers (SMPTE) specification and expresses time   offsets anchored at the start of the media clip.  Relative timestamps   are expressed as SMPTE time codes [SMPTE-TC] for frame-level access   accuracy.  The time code has the format:      hours:minutes:seconds:frames.subframes   with the origin at the start of the clip.  The default SMPTE format   is "SMPTE 30 drop" format, with a frame rate of 29.97 frames per   second.  Other SMPTE codes MAY be supported (such as "SMPTE 25")   through the use of "smpte-type".  For SMPTE 30, the "frames" field in   the time value can assume the values 0 through 29.  The difference   between 30 and 29.97 frames per second is handled by dropping the   first two frame indices (values 00 and 01) of every minute, except   every tenth minute.  If the frame and the subframe values are zero,   they may be omitted.  Subframes are measured in hundredths of a   frame.Schulzrinne, et al.          Standards Track                   [Page 28]

RFC 7826                        RTSP 2.0                   December 2016   Examples:     smpte=10:12:33:20-     smpte=10:07:33-     smpte=10:07:00-10:07:33:05.01     smpte-25=10:07:00-10:07:33:05.014.4.2.  Normal Play Time   Normal Play Time (NPT) indicates the stream-absolute position   relative to the beginning of the presentation.  The timestamp   consists of two parts: The mandatory first part may be expressed in   either seconds only or in hours, minutes, and seconds.  The optional   second part consists of a decimal point and decimal figures and   indicates fractions of a second.   The beginning of a presentation corresponds to 0.0 seconds.  Negative   values are not defined.   The special constant "now" is defined as the current instant of a   live event.  It MAY only be used for live events and MUST NOT be used   for on-demand (i.e., non-live) content.   NPT is defined as in Digital Storage Media Command and Control   (DSMb;CC) [ISO.13818-6.1995]:      Intuitively, NPT is the clock the viewer associates with a      program.  It is often digitally displayed on a DVD player.  NPT      advances normally when in normal play mode (scale = 1), advances      at a faster rate when in fast-scan forward (high positive scale      ratio), decrements when in scan reverse (negative scale ratio) and      is fixed in pause mode.  NPT is (logically) equivalent to SMPTE      time codes.   Examples:     npt=123.45-125     npt=12:05:35.3-     npt=now-Schulzrinne, et al.          Standards Track                   [Page 29]

RFC 7826                        RTSP 2.0                   December 2016   The syntax is based on ISO 8601 [ISO.8601.2000] and expresses the   time elapsed since presentation start, with two different notations   allowed:   o  The npt-hhmmss notation uses an ISO 8601 extended complete      representation of the time of the day format (Section 5.3.1.1 of      [ISO.8601.2000] ) using colons (":") as separators between hours,      minutes, and seconds (hh:mm:ss).  The hour counter is not limited      to 0-24 hours; up to nineteen (19) hour digits are allowed.      *  In accordance with the requirements of the ISO 8601 time         format, the hours, minutes, and seconds MUST all be present,         with two digits used for minutes and for seconds and with at         least two digits for hours.  An NPT of 7 minutes and 0 seconds         is represented as "00:07:00", and an NPT of 392 hours, 0         minutes, and 6 seconds is represented as "392:00:06".      *  RTSP 1.0 allowed NPT in the npt-hhmmss notation without any         leading zeros to ensure that implementations don't fail; for         backward compatibility, all RTSP 2.0 implementations are         REQUIRED to support receiving NPT values, hours, minutes, or         seconds, without leading zeros.   o  The npt-sec notation expresses the time in seconds, using between      one and nineteen (19) digits.   Both notations allow decimal fractions of seconds as specified in   Section 5.3.1.3 of [ISO.8601.2000], using at most nine digits, and   allowing only "." (full stop) as the decimal separator.   The npt-sec notation is optimized for automatic generation; the npt-   hhmmss notation is optimized for consumption by human readers.  The   "now" constant allows clients to request to receive the live feed   rather than the stored or time-delayed version.  This is needed since   neither absolute time nor zero time are appropriate for this case.4.4.3.  Absolute Time   Absolute time is expressed using a timestamp based on ISO 8601   [ISO.8601.2000].  The date is a complete representation of the   calendar date in basic format (YYYYMMDD) without separators (per   Section 5.2.1.1 of [ISO.8601.2000]).  The time of day is provided in   the complete representation basic format (hhmmss) as specified in   Section 5.3.1.1 of [ISO.8601.2000], allowing decimal fractions of   seconds followingSection 5.3.1.3 requiring "." (full stop) as   decimal separator and limiting the number of digits to no more than   nine.  The time expressed MUST use UTC (GMT), i.e., no time zone   offsets are allowed.  The full date and time specification is theSchulzrinne, et al.          Standards Track                   [Page 30]

RFC 7826                        RTSP 2.0                   December 2016   eight-digit date followed by a "T" followed by the six-digit time   value, optionally followed by a full stop followed by one to nine   fractions of a second and ended by "Z", e.g., YYYYMMDDThhmmss.ssZ.      The reasons for this time format rather than using "Date and Time      on the Internet: Timestamps" [RFC3339] are historic.  We continue      to use the format specified in RTSP 1.0.  The motivations raised      inRFC 3339 apply to why a selection from ISO 8601 was made;      however, a different and even more restrictive selection was      applied in this case.   Below are three examples of media time formats, first, a request for   a clock format range request for a starting time of November 8, 1996   at 14 h 37 min and 20 1/4 seconds UTC playing for 10 min and 5   seconds, followed by a Media-Properties header's "Time-Limited" UTC   property for the 24th of December 2014 at 15 hours and 00 minutes,   and finally a Terminate-Reason header "time" property for the 18th of   June 2013 at 16 hours, 12 minutes, and 56 seconds:     clock=19961108T143720.25Z-19961108T144725.25Z     Time-Limited=20141224T1500Z     time=20130618T161256Z4.5.  Feature Tags   Feature tags are unique identifiers used to designate features in   RTSP.  These tags are used in Require (Section 18.43), Proxy-Require   (Section 18.37), Proxy-Supported (Section 18.38), Supported   (Section 18.51), and Unsupported (Section 18.55) header fields.   A feature tag definition MUST indicate which combination of clients,   servers, or proxies to which it applies.   The creator of a new RTSP feature tag should either prefix the   feature tag with a reverse domain name (e.g.,   "com.example.mynewfeature" is an apt name for a feature whose   inventor can be reached at "example.com") or register the new feature   tag with the Internet Assigned Numbers Authority (IANA).  (SeeSection 22, "IANA Considerations".)   The usage of feature tags is further described inSection 11, which   deals with capability handling.Schulzrinne, et al.          Standards Track                   [Page 31]

RFC 7826                        RTSP 2.0                   December 20164.6.  Message Body Tags   Message body tags are opaque strings that are used to compare two   message bodies from the same resource, for example, in caches or to   optimize setup after a redirect.  Message body tags can be carried in   the MTag header (seeSection 18.31) or in SDP (seeAppendix D.1.9).   MTag is similar to ETag in HTTP/1.1 (seeSection 3.11 of [RFC2068]).   A message body tag MUST be unique across all versions of all message   bodies associated with a particular resource.  A given message body   tag value MAY be used for message bodies obtained by requests on   different URIs.  The use of the same message body tag value in   conjunction with message bodies obtained by requests on different   URIs does not imply the equivalence of those message bodies.   Message body tags are used in RTSP to make some methods conditional.   The methods are made conditional through the inclusion of headers;   seeSection 18.24 andSection 18.26 for information on the If-Match   and If-None-Match headers, respectively.  Note that RTSP message body   tags apply to the complete presentation, i.e., both the presentation   description and the individual media streams.  Thus, message body   tags can be used to verify at setup time after a redirect that the   same session description applies to the media at the new location   using the If-Match header.4.7.  Media Properties   When an RTSP server handles media, it is important to consider the   different properties a media instance for delivery and playback can   have.  This specification considers the media properties listed below   in its protocol operations.  They are derived from the differences   between a number of supported usages.   On-demand:  Media that has a fixed (given) duration that doesn't      change during the lifetime of the RTSP session and is known at the      time of the creation of the session.  It is expected that the      content of the media will not change, even if the representation,      such as encoding, or quality, may change.  Generally, one can      seek, i.e., request any range, within the media.   Dynamic On-demand:  This is a variation of the on-demand case where      external methods are used to manipulate the actual content of the      media setup for the RTSP session.  The main example is content      defined by a playlist.Schulzrinne, et al.          Standards Track                   [Page 32]

RFC 7826                        RTSP 2.0                   December 2016   Live:  Live media represents a progressing content stream (such as      broadcast TV) where the duration may or may not be known.  It is      not seekable, only the content presently being delivered can be      accessed.   Live with Recording:  A live stream that is combined with a server-      side capability to store and retain the content of the live      session and allow for random access delivery within the part of      the already-recorded content.  The actual behavior of the media      stream is very much dependent on the retention policy for the      media stream; either the server will be able to capture the      complete media stream or it will have a limitation in how much      will be retained.  The media range will dynamically change as the      session progress.  For servers with a limited amount of storage      available for recording, there will typically be a sliding window      that moves forward while new data is made available and older data      is discarded.   To cover the above usages, the following media properties with   appropriate values are specified.4.7.1.  Random Access and Seeking   Random access is the ability to specify and get media delivered   starting from any time (instant) within the content, an operation   called "seeking".  The Media-Properties header will indicate the   general capability for a media resource to perform random access.   Random-Access:  The media is seekable to any out of a large number of      points within the media.  Due to media-encoding limitations, a      particular point may not be reachable, but seeking to a point      close by is enabled.  A floating-point number of seconds may be      provided to express the worst-case distance between random access      points.   Beginning-Only:  Seeking is only possible to the beginning of the      content.   No-Seeking:  Seeking is not possible at all.   If random access is possible, as indicated by the Media-Properties   header, the actual behavior policy when seeking can be controlled   using the Seek-Style header (Section 18.47).Schulzrinne, et al.          Standards Track                   [Page 33]

RFC 7826                        RTSP 2.0                   December 20164.7.2.  Retention   The following retention policies are used by media to limit possible   protocol operations:   Unlimited:  The media will not be removed as long as the RTSP session      is in existence.   Time-Limited:  The media will not be removed before the given      wallclock time.  After that time, it may or may not be available      anymore.   Time-Duration:  The media (on fragment or unit basis) will be      retained for the specified duration.4.7.3.  Content Modifications   The media content and its timeline can be of different types, e.g.   pre-produced content on demand, a live source that is being generated   as time progresses, or something that is dynamically altered or   recomposed during playback.  Therefore, a media property for content   modifications is needed and the following initial values are defined:   Immutable:  The content of the media will not change, even if the      representation, such as encoding or quality changes.   Dynamic:  The content can change due to external methods or triggers,      such as playlists, but this will be announced by explicit updates.   Time-Progressing:  As time progresses, new content will become      available.  If the content is also retained, it will become longer      as everything between the start point and the point currently      being made available can be accessed.  If the media server uses a      sliding-window policy for retention, the start point will also      change as time progresses.4.7.4.  Supported Scale Factors   A particular media content item often supports only a limited set or   range of scales when delivering the media.  To enable the client to   know what values or ranges of scale operations that the whole content   or the current position supports, a media properties attribute for   this is defined that contains a list with the values or ranges that   are supported.  The attribute is named "Scales".  The "Scales"   attribute may be updated at any point in the content due to content   consisting of spliced pieces or content being dynamically updated by   out-of-band mechanisms.Schulzrinne, et al.          Standards Track                   [Page 34]

RFC 7826                        RTSP 2.0                   December 20164.7.5.  Mapping to the Attributes   This section shows examples of how one would map the above usages to   the properties and their values.   Example of On-Demand:      Random Access: Random-Access=5.0, Content Modifications:      Immutable, Retention: Unlimited or Time-Limited.   Example of Dynamic On-Demand:      Random Access: Random-Access=3.0, Content Modifications: Dynamic,      Retention: Unlimited or Time-Limited.   Example of Live:      Random Access: No-Seeking, Content Modifications: Time-      Progressing, Retention: Time-Duration=0.0   Example of Live with Recording:      Random Access: Random-Access=3.0, Content Modifications: Time-      Progressing, Retention: Time-Duration=7200.05.  RTSP Message   RTSP is a text-based protocol that uses the ISO 10646 character set   in UTF-8 encoding perRFC 3629 [RFC3629].  Lines MUST be terminated   by a CRLF.      Text-based protocols make it easier to add optional parameters in      a self-describing manner.  Since the number of parameters and the      frequency of commands is low, processing efficiency is not a      concern.  Text-based protocols, if used carefully, also allow easy      implementation of research prototypes in scripting languages such      as Python, PHP, Perl and TCL.   The ISO 10646 character set avoids character-set switching, but is   invisible to the application as long as US-ASCII is being used.  This   is also the encoding used for text fields in RTCP [RFC3550].   A request contains a method, the object the method is operating upon,   and parameters to further describe the method.  Methods are   idempotent unless otherwise noted.  Methods are also designed to   require little or no state maintenance at the media server.Schulzrinne, et al.          Standards Track                   [Page 35]

RFC 7826                        RTSP 2.0                   December 20165.1.  Message Types   RTSP messages are either requests from client to server or from   server to client, and responses in the reverse direction.  Request   (Section 7) and response (Section 8) messages use a format based on   the generic message format ofRFC 5322 [RFC5322] for transferring   bodies (the payload of the message).  Both types of messages consist   of a start-line, zero or more header fields (also known as   "headers"), an empty line (i.e., a line with nothing preceding the   CRLF) indicating the end of the headers, and possibly the data of the   message body.  The ABNF [RFC5234] below is for illustration only; the   formal message specification is presented inSection 20.2.2.   generic-message = start-line                   *(rtsp-header CRLF)                     CRLF                   [ message-body-data ]   start-line = Request-Line / Status-Line   In the interest of robustness, agents MUST ignore any empty line(s)   received where a Request-Line or Status-Line is expected.  In other   words, if the agent is reading the protocol stream at the beginning   of a message and receives any number of CRLFs first, it MUST ignore   all of the CRLFs.5.2.  Message Headers   RTSP header fields (seeSection 18) include general-header, request-   header, response-header, and message body header fields.   The order in which header fields with differing field names are   received is not significant.  However, it is "good practice" to send   general-header fields first, followed by a request-header or   response-header field, and ending with the message body header   fields.   Multiple header fields with the same field-name MAY be present in a   message if and only if the entire field-value for that header field   is defined as a comma-separated list.  It MUST be possible to combine   the multiple header fields into one "field-name: field-value" pair,   without changing the semantics of the message, by appending each   subsequent field-value to the first, each separated by a comma.  The   order in which header fields with the same field-name are received is   therefore significant to the interpretation of the combined field   value; thus, a proxy MUST NOT change the order of these field-values   when a message is forwarded.Schulzrinne, et al.          Standards Track                   [Page 36]

RFC 7826                        RTSP 2.0                   December 2016   Unknown message headers MUST be ignored (skipping over the header to   the next protocol element, and not causing an error) by an RTSP   server or client.  An RTSP proxy MUST forward unknown message   headers.  Message headers defined outside of this specification that   are required to be interpreted by the RTSP agent will need to use   feature tags (Section 4.5) and include them in the appropriate   Require (Section 18.43) or Proxy-Require (Section 18.37) header.5.3.  Message Body   The message body (if any) of an RTSP message is used to carry further   information for a particular resource associated with the request or   response.  An example of a message body is an SDP message.   The presence of a message body in either a request or a response MUST   be signaled by the inclusion of a Content-Length header (seeSection 18.17) and Content-Type header (seeSection 18.19).  A   message body MUST NOT be included in a request or response if the   specification of the particular method (see Method Definitions   (Section 13)) does not allow sending a message body.  In case a   message body is received in a message when not expected, the message   body data SHOULD be discarded.  This is to allow future extensions to   define optional use of a message body.5.4.  Message Length   An RTSP message that does not contain any message body is terminated   by the first empty line after the header fields (note: an empty line   is a line with nothing preceding the CRLF.).  In RTSP messages that   contain message bodies, the empty line is followed by the message   body.  The length of that body is determined by the value of the   Content-Length header (Section 18.17).  The value in the header   represents the length of the message body in octets.  If this header   field is not present, a value of zero is assumed, i.e., no message   body present in the message.  Unlike an HTTP message, an RTSP message   MUST contain a Content-Length header whenever it contains a message   body.  Note that RTSP does not support the HTTP/1.1 "chunked"   transfer coding (seeSection 4.1 of [RFC7230]).      Given the moderate length of presentation descriptions returned,      the server should always be able to determine its length, even if      it is generated dynamically, making the chunked transfer encoding      unnecessary.6.  General-Header Fields   General headers are headers that may be used in both requests and   responses.  The general-headers are listed in Table 1:Schulzrinne, et al.          Standards Track                   [Page 37]

RFC 7826                        RTSP 2.0                   December 2016                  +--------------------+----------------+                  | Header Name        | Defined in     |                  +--------------------+----------------+                  | Accept-Ranges      |Section 18.5   |                  |                    |                |                  | Cache-Control      |Section 18.11  |                  |                    |                |                  | Connection         |Section 18.12  |                  |                    |                |                  | CSeq               |Section 18.20  |                  |                    |                |                  | Date               |Section 18.21  |                  |                    |                |                  | Media-Properties   |Section 18.29  |                  |                    |                |                  | Media-Range        |Section 18.30  |                  |                    |                |                  | Pipelined-Requests |Section 18.33  |                  |                    |                |                  | Proxy-Supported    |Section 18.38  |                  |                    |                |                  | Range              |Section 18.40  |                  |                    |                |                  | RTP-Info           |Section 18.45  |                  |                    |                |                  | Scale              |Section 18.46  |                  |                    |                |                  | Seek-Style         |Section 18.47  |                  |                    |                |                  | Server             |Section 18.48  |                  |                    |                |                  | Session            |Section 18.49  |                  |                    |                |                  | Speed              |Section 18.50  |                  |                    |                |                  | Supported          |Section 18.51  |                  |                    |                |                  | Timestamp          |Section 18.53  |                  |                    |                |                  | Transport          |Section 18.54  |                  |                    |                |                  | User-Agent         |Section 18.56  |                  |                    |                |                  | Via                |Section 18.57  |                  +--------------------+----------------+                 Table 1: The General Headers Used in RTSPSchulzrinne, et al.          Standards Track                   [Page 38]

RFC 7826                        RTSP 2.0                   December 20167.  Request   A request message uses the format outlined below regardless of the   direction of a request, whether client to server or server to client:   o  Request line, containing the method to be applied to the resource,      the identifier of the resource, and the protocol version in use;   o  Zero or more Header lines, which can be of the following types:      general-headers (Section 6), request-headers (Section 7.2), or      message body headers (Section 9.1);   o  One empty line (CRLF) to indicate the end of the header section;   o  Optionally, a message body, consisting of one or more lines.  The      length of the message body in octets is indicated by the Content-      Length message header.Schulzrinne, et al.          Standards Track                   [Page 39]

RFC 7826                        RTSP 2.0                   December 20167.1.  Request Line   The request line provides the key information about the request: what   method, on what resources, and using which RTSP version.  The methods   that are defined by this specification are listed in Table 2.                    +---------------+----------------+                    | Method        | Defined in     |                    +---------------+----------------+                    | DESCRIBE      |Section 13.2   |                    |               |                |                    | GET_PARAMETER |Section 13.8   |                    |               |                |                    | OPTIONS       |Section 13.1   |                    |               |                |                    | PAUSE         |Section 13.6   |                    |               |                |                    | PLAY          |Section 13.4   |                    |               |                |                    | PLAY_NOTIFY   |Section 13.5   |                    |               |                |                    | REDIRECT      |Section 13.10  |                    |               |                |                    | SETUP         |Section 13.3   |                    |               |                |                    | SET_PARAMETER |Section 13.9   |                    |               |                |                    | TEARDOWN      |Section 13.7   |                    +---------------+----------------+                         Table 2: The RTSP Methods   The syntax of the RTSP request line has the following:      <Method> SP <Request-URI> SP <RTSP-Version> CRLF   Note: This syntax cannot be freely changed in future versions of   RTSP.  This line needs to remain parsable by older RTSP   implementations since it indicates the RTSP version of the message.   In contrast to HTTP/1.1 [RFC7230], RTSP requests identify the   resource through an absolute RTSP URI (including scheme, host, and   port) (seeSection 4.2) rather than just the absolute path.      HTTP/1.1 requires servers to understand the absolute URI, but      clients are supposed to use the Host request-header.  This is      purely needed for backward compatibility with HTTP/1.0 servers, a      consideration that does not apply to RTSP.Schulzrinne, et al.          Standards Track                   [Page 40]

RFC 7826                        RTSP 2.0                   December 2016   An asterisk "*" can be used instead of an absolute URI in the   Request-URI part to indicate that the request does not apply to a   particular resource but to the server or proxy itself, and is only   allowed when the request method does not necessarily apply to a   resource.   For example:      OPTIONS * RTSP/2.0   An OPTIONS in this form will determine the capabilities of the server   or the proxy that first receives the request.  If the capability of   the specific server needs to be determined, without regard to the   capability of an intervening proxy, the server should be addressed   explicitly with an absolute URI that contains the server's address.   For example:      OPTIONS rtsp://example.com RTSP/2.0Schulzrinne, et al.          Standards Track                   [Page 41]

RFC 7826                        RTSP 2.0                   December 20167.2.  Request-Header Fields   The RTSP headers in Table 3 can be included in a request, as request-   headers, to modify the specifics of the request.                 +---------------------+----------------+                 | Header              | Defined in     |                 +---------------------+----------------+                 | Accept              |Section 18.1   |                 |                     |                |                 | Accept-Credentials  |Section 18.2   |                 |                     |                |                 | Accept-Encoding     |Section 18.3   |                 |                     |                |                 | Accept-Language     |Section 18.4   |                 |                     |                |                 | Authorization       |Section 18.8   |                 |                     |                |                 | Bandwidth           |Section 18.9   |                 |                     |                |                 | Blocksize           |Section 18.10  |                 |                     |                |                 | From                |Section 18.23  |                 |                     |                |                 | If-Match            |Section 18.24  |                 |                     |                |                 | If-Modified-Since   |Section 18.25  |                 |                     |                |                 | If-None-Match       |Section 18.26  |                 |                     |                |                 | Notify-Reason       |Section 18.32  |                 |                     |                |                 | Proxy-Authorization |Section 18.36  |                 |                     |                |                 | Proxy-Require       |Section 18.37  |                 |                     |                |                 | Referrer            |Section 18.41  |                 |                     |                |                 | Request-Status      |Section 18.42  |                 |                     |                |                 | Require             |Section 18.43  |                 |                     |                |                 | Terminate-Reason    |Section 18.52  |                 +---------------------+----------------+                     Table 3: The RTSP Request-Headers   Detailed header definitions are provided inSection 18.Schulzrinne, et al.          Standards Track                   [Page 42]

RFC 7826                        RTSP 2.0                   December 2016   New request-headers may be defined.  If the receiver of the request   is required to understand the request-header, the request MUST   include a corresponding feature tag in a Require or Proxy-Require   header to ensure the processing of the header.8.  Response   After receiving and interpreting a request message, the recipient   responds with an RTSP response message.  Normally, there is only one,   final, response.  Responses using the response code class 1xx is the   only class for which there MAY be sent one or more responses prior to   the final response message.   The valid response codes and the methods they can be used with are   listed in Table 4.8.1.  Status-Line   The first line of a response message is the Status-Line, consisting   of the protocol version followed by a numeric status code and the   textual phrase associated with the status code, with each element   separated by SP characters.  No CR or LF is allowed except in the   final CRLF sequence.   <RTSP-Version> SP <Status-Code> SP <Reason Phrase> CRLF8.1.1.  Status Code and Reason Phrase   The Status-Code element is a 3-digit integer result code of the   attempt to understand and satisfy the request.  These codes are fully   defined inSection 17.  The reason phrase is intended to give a short   textual description of the Status-Code.  The Status-Code is intended   for use by automata and the reason phrase is intended for the human   user.  The client is not required to examine or display the reason   phrase.   The first digit of the Status-Code defines the class of response.   The last two digits do not have any categorization role.  There are   five values for the first digit:   1xx:  Informational - Request received, continuing process   2xx:  Success - The action was successfully received, understood, and         accepted   3rr:  Redirection - Further action needs to be taken in order to         complete the request (3rr rather than 3xx is used as 304 is         excluded; seeSection 17.3)Schulzrinne, et al.          Standards Track                   [Page 43]

RFC 7826                        RTSP 2.0                   December 2016   4xx:  Client Error - The request contains bad syntax or cannot be         fulfilled   5xx:  Server Error - The server failed to fulfill an apparently valid         request   The individual values of the numeric status codes defined for RTSP   2.0, and an example set of corresponding reason phrases, are   presented in Table 4.  The reason phrases listed here are only   recommended; they may be replaced by local equivalents without   affecting the protocol.  Note that RTSP adopted most HTTP/1.1   [RFC2068] status codes and then added RTSP-specific status codes   starting at x50 to avoid conflicts with future HTTP status codes that   are desirable to import into RTSP.  All these codes are RTSP specific   and RTSP has its own registry separate from HTTP for status codes.   RTSP status codes are extensible.  RTSP applications are not required   to understand the meaning of all registered status codes, though such   understanding is obviously desirable.  However, applications MUST   understand the class of any status code, as indicated by the first   digit, and treat any unrecognized response as being equivalent to the   x00 status code of that class, with an exception for unknown 3xx   codes, which MUST be treated as a 302 (Found).  The reason for that   exception is that the status code 300 (Multiple Choices in HTTP) is   not defined for RTSP.  A response with an unrecognized status code   MUST NOT be cached.  For example, if an unrecognized status code of   431 is received by the client, it can safely assume that there was   something wrong with its request and treat the response as if it had   received a 400 status code.  In such cases, user agents SHOULD   present to the user the message body returned with the response,   since that message body is likely to include human-readable   information that will explain the unusual status.   +------+---------------------------------+--------------------------+   | Code | Reason                          | Method                   |   +------+---------------------------------+--------------------------+   | 100  | Continue                        | all                      |   |      |                                 |                          |   | 200  | OK                              | all                      |   |      |                                 |                          |   | 301  | Moved Permanently               | all                      |   |      |                                 |                          |   | 302  | Found                           | all                      |   |      |                                 |                          |   | 303  | See Other                       | n/a                      |   |      |                                 |                          |   | 304  | Not Modified                    | all                      |   |      |                                 |                          |Schulzrinne, et al.          Standards Track                   [Page 44]

RFC 7826                        RTSP 2.0                   December 2016   | 305  | Use Proxy                       | all                      |   |      |                                 |                          |   | 400  | Bad Request                     | all                      |   |      |                                 |                          |   | 401  | Unauthorized                    | all                      |   |      |                                 |                          |   | 402  | Payment Required                | all                      |   |      |                                 |                          |   | 403  | Forbidden                       | all                      |   |      |                                 |                          |   | 404  | Not Found                       | all                      |   |      |                                 |                          |   | 405  | Method Not Allowed              | all                      |   |      |                                 |                          |   | 406  | Not Acceptable                  | all                      |   |      |                                 |                          |   | 407  | Proxy Authentication Required   | all                      |   |      |                                 |                          |   | 408  | Request Timeout                 | all                      |   |      |                                 |                          |   | 410  | Gone                            | all                      |   |      |                                 |                          |   | 412  | Precondition Failed             | DESCRIBE, SETUP          |   |      |                                 |                          |   | 413  | Request Message Body Too Large  | all                      |   |      |                                 |                          |   | 414  | Request-URI Too Long            | all                      |   |      |                                 |                          |   | 415  | Unsupported Media Type          | all                      |   |      |                                 |                          |   | 451  | Parameter Not Understood        | SET_PARAMETER,           |   |      |                                 | GET_PARAMETER            |   |      |                                 |                          |   | 452  | reserved                        | n/a                      |   |      |                                 |                          |   | 453  | Not Enough Bandwidth            | SETUP                    |   |      |                                 |                          |   | 454  | Session Not Found               | all                      |   |      |                                 |                          |   | 455  | Method Not Valid in This State  | all                      |   |      |                                 |                          |   | 456  | Header Field Not Valid for      | all                      |   |      | Resource                        |                          |   |      |                                 |                          |   | 457  | Invalid Range                   | PLAY, PAUSE              |   |      |                                 |                          |   | 458  | Parameter Is Read-Only          | SET_PARAMETER            |   |      |                                 |                          |Schulzrinne, et al.          Standards Track                   [Page 45]

RFC 7826                        RTSP 2.0                   December 2016   | 459  | Aggregate Operation Not Allowed | all                      |   |      |                                 |                          |   | 460  | Only Aggregate Operation        | all                      |   |      | Allowed                         |                          |   |      |                                 |                          |   | 461  | Unsupported Transport           | all                      |   |      |                                 |                          |   | 462  | Destination Unreachable         | all                      |   |      |                                 |                          |   | 463  | Destination Prohibited          | SETUP                    |   |      |                                 |                          |   | 464  | Data Transport Not Ready Yet    | PLAY                     |   |      |                                 |                          |   | 465  | Notification Reason Unknown     | PLAY_NOTIFY              |   |      |                                 |                          |   | 466  | Key Management Error            | all                      |   |      |                                 |                          |   | 470  | Connection Authorization        | all                      |   |      | Required                        |                          |   |      |                                 |                          |   | 471  | Connection Credentials Not      | all                      |   |      | Accepted                        |                          |   |      |                                 |                          |   | 472  | Failure to Establish Secure     | all                      |   |      | Connection                      |                          |   |      |                                 |                          |   | 500  | Internal Server Error           | all                      |   |      |                                 |                          |   | 501  | Not Implemented                 | all                      |   |      |                                 |                          |   | 502  | Bad Gateway                     | all                      |   |      |                                 |                          |   | 503  | Service Unavailable             | all                      |   |      |                                 |                          |   | 504  | Gateway Timeout                 | all                      |   |      |                                 |                          |   | 505  | RTSP Version Not Supported      | all                      |   |      |                                 |                          |   | 551  | Option Not Supported            | all                      |   |      |                                 |                          |   | 553  | Proxy Unavailable               | all                      |   +------+---------------------------------+--------------------------+          Table 4: Status Codes and Their Usage with RTSP MethodsSchulzrinne, et al.          Standards Track                   [Page 46]

RFC 7826                        RTSP 2.0                   December 20168.2.  Response Headers   The response-headers allow the request recipient to pass additional   information about the response that cannot be placed in the Status-   Line.  This header gives information about the server and about   further access to the resource identified by the Request-URI.  All   headers currently classified as response-headers are listed in   Table 5.                +------------------------+----------------+                | Header                 | Defined in     |                +------------------------+----------------+                | Authentication-Info    |Section 18.7   |                |                        |                |                | Connection-Credentials |Section 18.13  |                |                        |                |                | Location               |Section 18.28  |                |                        |                |                | MTag                   |Section 18.31  |                |                        |                |                | Proxy-Authenticate     |Section 18.34  |                |                        |                |                | Public                 |Section 18.39  |                |                        |                |                | Retry-After            |Section 18.44  |                |                        |                |                | Unsupported            |Section 18.55  |                |                        |                |                | WWW-Authenticate       |Section 18.58  |                +------------------------+----------------+                    Table 5: The RTSP Response Headers   Response-header names can be extended reliably only in combination   with a change in the protocol version.  However, the usage of feature   tags in the request allows the responding party to learn the   capability of the receiver of the response.  A new or experimental   header can be given the semantics of response-header if all parties   in the communication recognize them to be a response-header.   Unrecognized headers in responses MUST be ignored.9.  Message Body   Some request and response messages include a message body, if not   otherwise restricted by the request method or response status code.   The message body consists of the content data itself (see alsoSection 5.3).Schulzrinne, et al.          Standards Track                   [Page 47]

RFC 7826                        RTSP 2.0                   December 2016   The SET_PARAMETER and GET_PARAMETER requests and responses, and the   DESCRIBE response as defined by this specification, can have a   message body; the purpose of the message body is defined in each   case.  All 4xx and 5xx responses MAY also have a message body to   carry additional response information.  Generally, a message body MAY   be attached to any RTSP 2.0 request or response, but the content of   the message body MAY be ignored by the receiver.  Extensions to this   specification can specify the purpose and content of message bodies,   including requiring their inclusion.   In this section, both sender and recipient refer to either the client   or the server, depending on who sends and who receives the message   body.9.1.  Message Body Header Fields   Message body header fields define meta-information about the content   data in the message body.  The message body header fields are listed   in Table 6.                   +------------------+----------------+                   | Header           | Defined in     |                   +------------------+----------------+                   | Allow            |Section 18.6   |                   |                  |                |                   | Content-Base     |Section 18.14  |                   |                  |                |                   | Content-Encoding |Section 18.15  |                   |                  |                |                   | Content-Language |Section 18.16  |                   |                  |                |                   | Content-Length   |Section 18.17  |                   |                  |                |                   | Content-Location |Section 18.18  |                   |                  |                |                   | Content-Type     |Section 18.19  |                   |                  |                |                   | Expires          |Section 18.22  |                   |                  |                |                   | Last-Modified    |Section 18.27  |                   +------------------+----------------+                  Table 6: The RTSP Message Body HeadersSchulzrinne, et al.          Standards Track                   [Page 48]

RFC 7826                        RTSP 2.0                   December 2016   The extension-header mechanism allows additional message body header   fields to be defined without changing the protocol, but these fields   cannot be assumed to be recognizable by the recipient.  Unrecognized   header fields MUST be ignored by the recipient and forwarded by   proxies.9.2.  Message Body   An RTSP message with a message body MUST include the Content-Type and   Content-Length headers.  When a message body is included with a   message, the data type of that content data is determined via the   Content-Type and Content-Encoding header fields.   Content-Type specifies the media type of the underlying data.  There   is no default media format and the actual format used in the body is   required to be explicitly stated in the Content-Type header.  By   being explicit and always requiring the inclusion of the Content-Type   header with accurate information, one avoids the many pitfalls in a   heuristic-based interpretation of the body content.  The user   experience of HTTP and email have suffered from relying on such   heuristics.   Content-Encoding may be used to indicate any additional content-   codings applied to the data, usually for the purpose of data   compression, that are a property of the requested resource.  The   default encoding is 'identity', i.e. no transformation of the message   body.   The Content-Length of a message is the length of the content,   measured in octets.9.3.  Message Body Format Negotiation   The content format of the message body is provided using the Content-   Type header (Section 18.19).  To enable the responder of a request to   determine which media type it should use, the requester may include   the Accept header (Section 18.1) in a request to identify supported   media types or media type ranges suitable to the response.  In case   the responder is not supporting any of the specified formats, then   the request response will be a 406 (Not Acceptable) error code.   The media types that may be used on requests with message bodies need   to be determined through the use of feature tags, specification   requirement, or trial and error.  Trial and error works because when   the responder does not support the media type of the message body, it   will respond with a 415 (Unsupported Media Type).Schulzrinne, et al.          Standards Track                   [Page 49]

RFC 7826                        RTSP 2.0                   December 2016   The formats supported and their negotiation is done individually on a   per method and direction (request or response body) direction.   Requirements on supporting particular media types for use as message   bodies in requests and response SHALL also be specified on a per-   method and per-direction basis.10.  Connections   RTSP messages are transferred between RTSP agents and proxies using a   transport connection.  This transport connection uses TCP or TCP/TLS.   This transport connection is referred to as the "connection" or "RTSP   connection" within this document.   RTSP requests can be transmitted using the two different connection   scenarios listed below:   o  persistent - a transport connection is used for several request/      response transactions;   o  transient - a transport connection is used for each single      request/response transaction.RFC 2326 attempted to specify an optional mechanism for transmitting   RTSP messages in connectionless mode over a transport protocol such   as UDP.  However, it was not specified in sufficient detail to allow   for interoperable implementations.  In an attempt to reduce   complexity and scope, and due to lack of interest, RTSP 2.0 does not   attempt to define a mechanism for supporting RTSP over UDP or other   connectionless transport protocols.  A side effect of this is that   RTSP requests MUST NOT be sent to multicast groups since no   connection can be established with a specific receiver in multicast   environments.   Certain RTSP headers, such as the CSeq header (Section 18.20), which   may appear to be relevant only to connectionless transport scenarios,   are still retained and MUST be implemented according to this   specification.  In the case of CSeq, it is quite useful for matching   responses to requests if the requests are pipelined (seeSection 12).   It is also useful in proxies for keeping track of the different   requests when aggregating several client requests on a single TCP   connection.10.1.  Reliability and Acknowledgements   Since RTSP messages are transmitted using reliable transport   protocols, they MUST NOT be retransmitted at the RTSP level.   Instead, the implementation must rely on the underlying transport toSchulzrinne, et al.          Standards Track                   [Page 50]

RFC 7826                        RTSP 2.0                   December 2016   provide reliability.  The RTSP implementation may use any indication   of reception acknowledgment of the message from the underlying   transport protocols to optimize the RTSP behavior.      If both the underlying reliable transport, such as TCP, and the      RTSP application retransmit requests, each packet loss or message      loss may result in two retransmissions.  The receiver typically      cannot take advantage of the application-layer retransmission      since the transport stack will not deliver the application-layer      retransmission before the first attempt has reached the receiver.      If the packet loss is caused by congestion, multiple      retransmissions at different layers will exacerbate the      congestion.   Lack of acknowledgment of an RTSP request should be handled within   the constraints of the connection timeout considerations described   below (Section 10.4).10.2.  Using Connections   A TCP transport can be used for both persistent connections (for   several message exchanges) and transient connections (for a single   message exchange).  Implementations of this specification MUST   support RTSP over TCP.  The scheme of the RTSP URI (Section 4.2)   allows the client to specify the port it will contact the server on,   and defines the default port to use if one is not explicitly given.   In addition to the registered default ports, i.e., 554 (rtsp) and 322   (rtsps), there is an alternative port 8554 registered.  This port may   provide some benefits over non-registered ports if an RTSP server is   unable to use the default ports.  The benefits may include   preconfigured security policies as well as classifiers in network   monitoring tools.   An RTSP client opening a TCP connection to access a particular   resource as identified by a URI uses the IP address and port derived   from the host and port parts of the URI.  The IP address is either   the explicit address provided in the URI or any of the addresses   provided when performing A and AAAA record DNS lookups of the   hostname in the URI.   A server MUST handle both persistent and transient connections.      Transient connections facilitate mechanisms for fault tolerance.      They also allow for application-layer mobility.  A server-and-      client pair that supports transient connections can survive theSchulzrinne, et al.          Standards Track                   [Page 51]

RFC 7826                        RTSP 2.0                   December 2016      loss of a TCP connection; e.g., due to a NAT timeout.  When the      client has discovered that the TCP connection has been lost, it      can set up a new one when there is need to communicate again.   A persistent connection is RECOMMENDED to be used for all   transactions between the server and client, including messages for   multiple RTSP sessions.  However, a persistent connection MAY be   closed after a few message exchanges.  For example, a client may use   a persistent connection for the initial SETUP and PLAY message   exchanges in a session and then close the connection.  Later, when   the client wishes to send a new request, such as a PAUSE for the   session, a new connection would be opened.  This connection may be   either transient or persistent.   An RTSP agent MAY use one connection to handle multiple RTSP sessions   on the same server.  The RTSP agent SHALL NOT use more than one   connection per RTSP session at any given point.      Having only one connection in use at any time avoids confusion      regarding on which connection any server-to-client requests shall      be sent.  Using a single connection for multiple RTSP sessions      also saves complexity by enabling the server to maintain less      state about its connection resources on the server.  Not using      more than one connection at a time for a particular RTSP session      avoids wasting connection resources and allows the server to track      only the most recently used client-to-server connection for each      RTSP session as being the currently valid server-to-client      connection.   RTSP allows a server to send requests to a client.  However, this can   be supported only if a client establishes a persistent connection   with the server.  In cases where a persistent connection does not   exist between a server and its client, due to the lack of a signaling   channel, the server may be forced to silently discard RTSP messages,   and it may even drop an RTSP session without notifying the client.   An example of such a case is when the server desires to send a   REDIRECT request for an RTSP session to the client but is not able to   do so because it cannot reach the client.  A server that attempts to   send a request to a client that has no connection currently to the   server SHALL discard the request.      Without a persistent connection between the client and the server,      the media server has no reliable way of reaching the client.      Because of the likely failure of server-to-client established      connections, the server will not even attempt establishing any      connection.Schulzrinne, et al.          Standards Track                   [Page 52]

RFC 7826                        RTSP 2.0                   December 2016      Queuing of server-to-client requests has been considered.      However, a security issue exists as to how it might be possible to      authorize a client establishing a new connection as being a      legitimate receiver of a request related to a particular RTSP      session, without the client first issuing requests related to the      pending request.  Thus, it would be likely to make any such      requests even more delayed and less useful.   The sending of client and server requests can be asynchronous events.   To avoid deadlock situations, both client and server MUST be able to   send and receive requests simultaneously.  As an RTSP response may be   queued up for transmission, reception or processing behind the peer   RTSP agent's own requests, all RTSP agents are required to have a   certain capability of handling outstanding messages.  A potential   issue is that outstanding requests may time out despite being   processed by the peer; this can be due to the response being caught   in the queue behind a number of requests that the RTSP agent is   processing but that take some time to complete.  To avoid this   problem, an RTSP agent should buffer incoming messages locally so   that any response messages can be processed immediately upon   reception.  If responses are separated from requests and directly   forwarded for processing, not only can the result be used   immediately, the state associated with that outstanding request can   also be released.  However, buffering a number of requests on the   receiving RTSP agent consumes resources and enables a resource   exhaustion attack on the agent.  Therefore, this buffer should be   limited so that an unreasonable number of requests or total message   size is not allowed to consume the receiving agent's resources.  In   most APIs, having the receiving agent stop reading from the TCP   socket will result in TCP's window being clamped, thus forcing the   buffering onto the sending agent when the load is larger than   expected.  However, as both RTSP message sizes and frequency may be   changed in the future by protocol extensions, an agent should be   careful about taking harsher measurements against a potential attack.   When under attack, an RTSP agent can close TCP connections and   release state associated with that TCP connection.   To provide some guidance on what is reasonable, the following   guidelines are given.  It is RECOMMENDED that:   o  an RTSP agent should not have more than 10 outstanding requests      per RTSP session;   o  an RTSP agent should not have more than 10 outstanding requests      that are not related to an RTSP session or that are requesting to      create an RTSP session.Schulzrinne, et al.          Standards Track                   [Page 53]

RFC 7826                        RTSP 2.0                   December 2016   In light of the above, it is RECOMMENDED that clients use persistent   connections whenever possible.  A client that supports persistent   connections MAY "pipeline" its requests (seeSection 12).   RTSP agents can send requests to multiple different destinations,   either server or client contexts over the same connection to a proxy.   Then, the proxy forks the message to the different destinations over   proxy-to-agent connections.  In these cases when multiple requests   are outstanding, the requesting agent MUST be ready to receive the   responses out of order compared to the order they where sent on the   connection.  The order between multiple messages for each destination   will be maintained; however, the order between response from   different destinations can be different.      The reason for this is to avoid a head-of-line blocking situation.      In a sequence of requests, an early outstanding request may take      time to be processed at one destination.  Simultaneously, a      response from any other destination that was later in the sequence      of requests may have arrived at the proxy; thus, allowing out-of-      order responses avoids forcing the proxy to buffer this response      and instead deliver it as soon as possible.  Note, this will not      affect the order in which the messages sent to each separate      destination were processed at the request destination.   This scenario can occur in two cases involving proxies.  The first is   a client issuing requests for sessions on different servers using a   common client-to-proxy connection.  The second is for server-to-   client requests, like REDIRECT being sent by the server over a common   transport connection the proxy created for its different connecting   clients.10.3.  Closing Connections   The client MAY close a connection at any point when no outstanding   request/response transactions exist for any RTSP session being   managed through the connection.  The server, however, SHOULD NOT   close a connection until all RTSP sessions being managed through the   connection have been timed out (Section 18.49).  A server SHOULD NOT   close a connection immediately after responding to a session-level   TEARDOWN request for the last RTSP session being controlled through   the connection.  Instead, the server should wait for a reasonable   amount of time for the client to receive and act upon the TEARDOWNSchulzrinne, et al.          Standards Track                   [Page 54]

RFC 7826                        RTSP 2.0                   December 2016   response and then initiate the connection closing.  The server SHOULD   wait at least 10 seconds after sending the TEARDOWN response before   closing the connection.      This is to ensure that the client has time to issue a SETUP for a      new session on the existing connection after having torn the last      one down.  Ten seconds should give the client ample opportunity to      get its message to the server.   A server SHOULD NOT close the connection directly as a result of   responding to a request with an error code.      Certain error responses such as 460 (Only Aggregate Operation      Allowed) (Section 17.4.24) are used for negotiating capabilities      of a server with respect to content or other factors.  In such      cases, it is inefficient for the server to close a connection on      an error response.  Also, such behavior would prevent      implementation of advanced or special types of requests or result      in extra overhead for the client when testing for new features.      On the other hand, keeping connections open after sending an error      response poses a Denial-of-Service (DoS) security risk      (Section 21).   The server MAY close a connection if it receives an incomplete   message and if the message is not completed within a reasonable   amount of time.  It is RECOMMENDED that the server wait at least 10   seconds for the completion of a message or for the next part of the   message to arrive (which is an indication that the transport and the   client are still alive).  Servers believing they are under attack or   that are otherwise starved for resources during that event MAY   consider using a shorter timeout.   If a server closes a connection while the client is attempting to   send a new request, the client will have to close its current   connection, establish a new connection, and send its request over the   new connection.   An RTSP message SHOULD NOT be terminated by closing the connection.   Such a message MAY be considered to be incomplete by the receiver and   discarded.  An RTSP message is properly terminated as defined inSection 5.Schulzrinne, et al.          Standards Track                   [Page 55]

RFC 7826                        RTSP 2.0                   December 201610.4.  Timing Out Connections and RTSP Messages   Receivers of a request (responders) SHOULD respond to requests in a   timely manner even when a reliable transport such as TCP is used.   Similarly, the sender of a request (requester) SHOULD wait for a   sufficient time for a response before concluding that the responder   will not be acting upon its request.   A responder SHOULD respond to all requests within 5 seconds.  If the   responder recognizes that the processing of a request will take   longer than 5 seconds, it SHOULD send a 100 (Continue) response as   soon as possible.  It SHOULD continue sending a 100 response every 5   seconds thereafter until it is ready to send the final response to   the requester.  After sending a 100 response, the responder MUST send   a final response indicating the success or failure of the request.   A requester SHOULD wait at least 10 seconds for a response before   concluding that the responder will not be responding to its request.   After receiving a 100 response, the requester SHOULD continue waiting   for further responses.  If more than 10 seconds elapse without   receiving any response, the requester MAY assume that the responder   is unresponsive and abort the connection by closing the TCP   connection.   In some cases, multiple RTSP sessions share the same transport   connection; abandoning a request and closing the connection may have   significant impact on those other sessions.  First of all, other RTSP   requests may have become queued up due to the request taking a long   time to process.  Secondly, those sessions also lose the possibility   to receive server-to-client requests.  To mitigate that situation,   the RTSP client or server SHOULD establish a new connection and send   any requests that are queued up or that haven't received a response   on this new connection.  Thirdly, to ensure that the RTSP server   knows which connection is valid for a particular RTSP session, the   RTSP agent SHOULD send a keep-alive request, if no other request will   be sent immediately for that RTSP session, for each RTSP session on   the old connection.  The keep-alive request will normally be a   SET_PARAMETER with a session header to inform the server that this   agent cares about this RTSP session.   A requester SHOULD wait longer than 10 seconds for a response if it   is experiencing significant transport delays on its connection to the   responder.  The requester is capable of determining the Round-Trip   Time (RTT) of the request/response cycle using the Timestamp header   (Section 18.53) in any RTSP request.Schulzrinne, et al.          Standards Track                   [Page 56]

RFC 7826                        RTSP 2.0                   December 2016      The 10-second wait was chosen for the following reasons.  It gives      TCP time to perform a couple of retransmissions, even if operating      on default values.  It is short enough that users may not abandon      the process themselves.  However, it should be noted that 10      seconds can be aggressive on certain types of networks.  The      5-second value for 1xx messages is half the timeout giving a      reasonable chance of successful delivery before timeout happens on      the requester side.10.5.  Showing Liveness   RTSP requires the client to periodically show its liveness to the   server or the server may terminate any session state.  Several   different protocol mechanism include in their usage a liveness proof   from the client.  These mechanisms are RTSP requests with a Session   header to the server; if RTP & RTCP is used for media data transport   and the transport is established, the RTCP message proves liveness;   or through any other used media-transport protocol capable of   indicating liveness of the RTSP client.  It is RECOMMENDED that a   client not wait to the last second of the timeout before trying to   send a liveness message.  The RTSP message may take some time to   arrive safely at the receiver, due to packet loss and TCP   retransmissions.  To show liveness between RTSP requests being issued   to accomplish other things, the following mechanisms can be used, in   descending order of preference:   RTCP: If RTP is used for media transport, RTCP SHOULD be used.  If         RTCP is used to report transport statistics, it will         necessarily also function as a keep-alive.  The server can         determine the client by network address and port together with         the fact that the client is reporting on the server's RTP         sender sources (synchronization source (SSRCs)).  A downside of         using RTCP is that it only gives statistical guarantees of         reaching the server.  However, the probability of a false         client timeout is so low that it can be ignored in most cases.         For example, assume a session with a 60-second timeout and         enough bitrate assigned to RTCP messages to send a message from         client to server on average every 5 seconds.  That client has,         for a network with 5% packet loss, a probability of failing to         confirm liveness within the timeout interval for that session         of 2.4*E-16.  Sessions with shorter timeouts, much higher         packet loss, or small RTCP bandwidths SHOULD also implement one         or more of the mechanisms below.Schulzrinne, et al.          Standards Track                   [Page 57]

RFC 7826                        RTSP 2.0                   December 2016   SET_PARAMETER:  When using SET_PARAMETER for keep-alives, a body         SHOULD NOT be included.  This method is the RECOMMENDED RTSP         method to use for a request intended only to perform keep-         alives.  RTSP servers MUST support the SET_PARAMETER method, so         that clients can always use this mechanism.   GET_PARAMETER:  When using GET_PARAMETER for keep-alives, a body         SHOULD NOT be included, dependent on implementation support in         the server.  Use the OPTIONS method to determine if there is         method support or simply try.   OPTIONS:  This method is also usable, but it causes the server to         perform more unnecessary processing and results in bigger         responses than necessary for the task.  The reason is that the         server needs to determine the capabilities associated with the         media resource to correctly populate the Public and Allow         headers.   The timeout parameter of the Session header (Section 18.49) MAY be   included in a SETUP response and MUST NOT be included in requests.   The server uses it to indicate to the client how long the server is   prepared to wait between RTSP commands or other signs of life before   closing the session due to lack of activity (seeAppendix B).  The   timeout is measured in seconds, with a default of 60 seconds.  The   length of the session timeout MUST NOT be changed in an established   session.10.6.  Use of IPv6   Explicit IPv6 [RFC2460] support was not present in RTSP 1.0.  RTSP   2.0 has been updated for explicit IPv6 support.  Implementations of   RTSP 2.0 MUST understand literal IPv6 addresses in URIs and RTSP   headers.  Although the general URI format envisages potential future   new versions of the literal IP address, usage of any such new version   would require other modifications to the RTSP specification (e.g.,   address fields in the Transport header (Section 18.54)).10.7.  Overload Control   Overload in RTSP can occur when servers and proxies have insufficient   resources to complete the processing of a request.  An improper   handling of such an overload situation at proxies and servers can   impact the operation of the RTSP deployment, and probably worsen the   situation.  RTSP defines the 503 (Service Unavailable) response   (Section 17.5.4) to let servers and proxies notify requesting proxies   and RTSP clients about an overload situation.  In conjunction withSchulzrinne, et al.          Standards Track                   [Page 58]

RFC 7826                        RTSP 2.0                   December 2016   the Retry-After header (Section 18.44), the server or proxy can   indicate the time after which the requesting entity can send another   request to the proxy or server.   There are two scopes of such 503 answers.  The first scope is for an   established RTSP session, where the request resulting in the 503   response as well as the response itself carries a Session header   identifying the session that is suffering overload.  This response   only applies to this particular session.  The other scope is the   general RTSP server as identified by the host in the Request-URI.   Such a 503 answer with any Retry-After header applies to all requests   that are not session specific to that server, including a SETUP   request intended to create a new RTSP session.   Another scope for overload situations exists: the RTSP proxy.  To   enable an RTSP proxy to signal that it is overloaded, or otherwise   unavailable and unable to handle the request, a 553 response code has   been defined with the meaning "Proxy Unavailable".  As with servers,   there is a separation in response scopes between requests associated   with existing RTSP sessions and requests to create new sessions or   general proxy requests.   Simply implementing and using the 503 (Service Unavailable) and 553   (Proxy Unavailable) response codes is not sufficient for properly   handling overload situations.  For instance, a simplistic approach   would be to send the 503 response with a Retry-After header set to a   fixed value.  However, this can cause a situation in which multiple   RTSP clients again send requests to a proxy or server at roughly the   same time, which may again cause an overload situation.  Another   situation would be if the "old" overload situation is not yet   resolved, i.e., the length indicated in the Retry-After header was   too short for the overload situation to subside.   An RTSP server or proxy in an overload situation must select the   value of the Retry-After header carefully, bearing in mind its   current load situation.  It is REQUIRED to increase the timeout   period in proportion to the current load on the server, i.e., an   increasing workload should result in an increased length of the   indicated unavailability.  It is REQUIRED not to send the same value   in the Retry-After header to all requesting proxies and clients, but   to add a variation to the mean value of the Retry-After header.   A more complex case may arise when a load-balancing RTSP proxy is in   use.  This is the case when an RTSP proxy is used to select amongst a   set of RTSP servers to handle the requests or when multiple server   addresses are available for a given server name.  The proxy or client   may receive a 503 (Service Unavailable) or 553 (Proxy Unavailable)   response code from one of its RTSP servers or proxies, or a TCPSchulzrinne, et al.          Standards Track                   [Page 59]

RFC 7826                        RTSP 2.0                   December 2016   timeout (if the server is even unable to handle the request message).   The proxy or client simply retries the other addresses or configured   proxies, but it may also receive a 503 (Service Unavailable) or 553   (Proxy Unavailable) response or TCP timeouts from those addresses.   In such a situation, where none of the RTSP servers/proxies/addresses   can handle the request, the RTSP agent has to wait before it can send   any new requests to the RTSP server.  Any additional request to a   specific address MUST be delayed according to the Retry-After headers   received.  For addresses where no response was received or TCP   timeout occurred, an initial wait timer SHOULD be set to 5 seconds.   That timer MUST be doubled for each additional failure to connect or   receive response until the value exceeds 30 minutes when the timer's   mean value may be set to 30 minutes.  It is REQUIRED not to set the   same value in the timer for each scheduling, but instead to add a   variation to the mean value, resulting in picking a random value   within the range of 0.5 to 1.5 times the mean value.11.  Capability Handling   This section describes the available capability-handling mechanism   that allows RTSP to be extended.  Extensions to this version of the   protocol are basically done in two ways.  Firstly, new headers can be   added.  Secondly, new methods can be added.  The capability-handling   mechanism is designed to handle both cases.   When a method is added, the involved parties can use the OPTIONS   method to discover whether it is supported.  This is done by issuing   an OPTIONS request to the other party.  Depending on the URI, it will   either apply in regard to a certain media resource, the whole server   in general, or simply the next hop.  The OPTIONS response MUST   contain a Public header that declares all methods supported for the   indicated resource.   It is not necessary to use OPTIONS to discover support of a method,   as the client could simply try the method.  If the receiver of the   request does not support the method, it will respond with an error   code indicating the method is either not implemented (501) or does   not apply for the resource (405).  The choice between the two   discovery methods depends on the requirements of the service.   Feature tags are defined to handle functionality additions that are   not new methods.  Each feature tag represents a certain block of   functionality.  The amount of functionality that a feature tag   represents can vary significantly.  For example, a feature tag can   represent the functionality a single RTSP header provides.  Another   feature tag can represent much more functionality, such as the   "play.basic" feature tag (Section 11.1), which represents the minimal   media delivery for playback implementation.Schulzrinne, et al.          Standards Track                   [Page 60]

RFC 7826                        RTSP 2.0                   December 2016   Feature tags are used to determine whether the client, server, or   proxy supports the functionality that is necessary to achieve the   desired service.  To determine support of a feature tag, several   different headers can be used, each explained below:   Supported:  This header is used to determine the complete set of         functionality that both client and server have, in general, and         is not dependent on a specific resource.  The intended usage is         to determine before one needs to use a functionality that it is         supported.  It can be used in any method, but OPTIONS is the         most suitable as it simultaneously determines all methods that         are implemented.  When sending a request, the requester         declares all its capabilities by including all supported         feature tags.  This results in the receiver learning the         requester's feature support.  The receiver then includes its         set of features in the response.   Proxy-Supported:  This header is used in a similar fashion as the         Supported header, but instead of giving the supported         functionality of the client or server, it provides both the         requester and the responder a view of the common functionality         supported in general by all members of the proxy chain between         the client and server; it does not depend on the resource.         Proxies are required to add this header whenever the Supported         header is present, but proxies may also add it independently of         the requester.   Require:  This header can be included in any request where the         endpoint, i.e., the client or server, is required to understand         the feature to correctly perform the request.  This can, for         example, be a SETUP request, where the server is required to         understand a certain parameter to be able to set up the media         delivery correctly.  Ignoring this parameter would not have the         desired effect and is not acceptable.  Therefore, the endpoint         receiving a request containing a Require MUST negatively         acknowledge any feature that it does not understand and not         perform the request.  The response in cases where features are         not supported is 551 (Option Not Supported).  Also, the         features that are not supported are given in the Unsupported         header in the response.   Proxy-Require:  This header has the same purpose and behavior as         Require except that it only applies to proxies and not the         endpoint.  Features that need to be supported by both proxies         and endpoints need to be included in both the Require and         Proxy-Require header.Schulzrinne, et al.          Standards Track                   [Page 61]

RFC 7826                        RTSP 2.0                   December 2016   Unsupported:  This header is used in a 551 (Option Not Supported)         error response, to indicate which features were not supported.         Such a response is only the result of the usage of the Require         or Proxy-Require headers where one or more features were not         supported.  This information allows the requester to make the         best of situations as it knows which features are not         supported.11.1.  Feature Tag: play.basic   An implementation supporting all normative parts of this   specification for the setup and control of playback of media uses the   feature tag "play.basic" to indicate this support.  The appendices   (starting with letters) are not part of the functionality included in   the feature tag unless the appendix is explicitly specified in a main   section as being a required appendix.      Note: This feature tag does not mandate any media delivery      protocol, such as RTP.      In RTSP 1.0, there was a minimal implementation section.  However,      that was not consistent with the rest of the specification.  So,      rather than making an attempt to explicitly enumerate the features      for play.basic, this specification has to be taken as a whole and      the necessary features normatively defined as being required are      included.12.  Pipelining Support   Pipelining is a general method to improve performance of request/   response protocols by allowing the requesting agent to have more than   one request outstanding and to send them over the same persistent   connection.  For RTSP, where the relative order of requests will   matter, it is important to maintain the order of the requests.   Because of this, the responding agent MUST process the incoming   requests in their sending order.  The sending order can be determined   by the CSeq header and its sequence number.  For TCP, the delivery   order will be the same, between two agents, as the sending order.   The processing of the request MUST also have been finished before   processing the next request from the same agent.  The responses MUST   be sent in the order the requests were processed.   RTSP 2.0 has extended support for pipelining beyond the capabilities   in RTSP 1.0.  As a major improvement, all requests involved in   setting up and initiating media delivery can now be pipelined,   indicated by the Pipelined-Request header (seeSection 18.33).  This   header allows a client to request that two or more requests be   processed in the same RTSP session context that the first requestSchulzrinne, et al.          Standards Track                   [Page 62]

RFC 7826                        RTSP 2.0                   December 2016   creates.  In other words, a client can request that two or more media   streams be set up and then played without needing to wait for a   single response.  This speeds up the initial start-up time for an   RTSP session by at least one RTT.   If a pipelined request builds on the successful completion of one or   more prior requests, the requester must verify that all requests were   executed as expected.  A common example will be two SETUP requests   and a PLAY request.  In case one of the SETUP requests fails   unexpectedly, the PLAY request can still be successfully executed.   However, the resulting presentation will not be as expected by the   requesting client, as only a single media instead of two will be   played.  In this case, the client can send a PAUSE request, correct   the failing SETUP request, and then request it be played.13.  Method Definitions   The method indicates what is to be performed on the resource   identified by the Request-URI.  The method name is case sensitive.   New methods may be defined in the future.  Method names MUST NOT   start with a $ character (decimal 36) and MUST be a token as defined   by the ABNF [RFC5234] inSection 20.  The methods are summarized in   Table 7.Schulzrinne, et al.          Standards Track                   [Page 63]

RFC 7826                        RTSP 2.0                   December 2016    +---------------+-----------+--------+-------------+-------------+    | method        | direction | object | Server req. | Client req. |    +---------------+-----------+--------+-------------+-------------+    | DESCRIBE      | C -> S    | P,S    | recommended | recommended |    |               |           |        |             |             |    | GET_PARAMETER | C -> S    | P,S    | optional    | optional    |    |               |           |        |             |             |    |               | S -> C    | P,S    | optional    | optional    |    |               |           |        |             |             |    | OPTIONS       | C -> S    | P,S    | required    | required    |    |               |           |        |             |             |    |               | S -> C    | P,S    | optional    | optional    |    |               |           |        |             |             |    | PAUSE         | C -> S    | P,S    | required    | required    |    |               |           |        |             |             |    | PLAY          | C -> S    | P,S    | required    | required    |    |               |           |        |             |             |    | PLAY_NOTIFY   | S -> C    | P,S    | required    | required    |    |               |           |        |             |             |    | REDIRECT      | S -> C    | P,S    | optional    | required    |    |               |           |        |             |             |    | SETUP         | C -> S    | S      | required    | required    |    |               |           |        |             |             |    | SET_PARAMETER | C -> S    | P,S    | required    | optional    |    |               |           |        |             |             |    |               | S -> C    | P,S    | optional    | optional    |    |               |           |        |             |             |    | TEARDOWN      | C -> S    | P,S    | required    | required    |    |               |           |        |             |             |    |               | S -> C    | P      | required    | required    |    +---------------+-----------+--------+-------------+-------------+                     Table 7: Overview of RTSP Methods      Note on Table 7: This table covers RTSP methods, their direction,      and on what objects (P: presentation, S: stream) they operate.      Further, it indicates whether a server or a client implementation      is required (mandatory), recommended, or optional.      Further note on Table 7: the GET_PARAMETER is optional.  For      example, a fully functional server can be built to deliver media      without any parameters.  However, SET_PARAMETER is required, i.e.,      mandatory to implement for the server; this is due to its usage      for keep-alive.  PAUSE is required because it is the only way of      leaving the Play state without terminating the whole session.Schulzrinne, et al.          Standards Track                   [Page 64]

RFC 7826                        RTSP 2.0                   December 2016   If an RTSP agent does not support a particular method, it MUST return   a 501 (Not Implemented) response code and the requesting RTSP agent,   in turn, SHOULD NOT try this method again for the given agent/   resource combination.  An RTSP proxy whose main function is to log or   audit and not modify transport or media handling in any way MAY   forward RTSP messages with unknown methods.  Note that the proxy   still needs to perform the minimal required processing, like adding   the Via header.13.1.  OPTIONS   The semantics of the RTSP OPTIONS method is similar to that of the   HTTP OPTIONS method described inSection 4.3.7 of [RFC7231].   However, in RTSP, OPTIONS is bidirectional in that a client can send   the request to a server and vice versa.  A client MUST implement the   capability to send an OPTIONS request and a server or a proxy MUST   implement the capability to respond to an OPTIONS request.  In   addition to this "MUST-implement" functionality, clients, servers and   proxies MAY provide support both for sending OPTIONS requests and for   generating responses to the requests.   An OPTIONS request may be issued at any time.  Such a request does   not modify the session state.  However, it may prolong the session   lifespan (see below).  The URI in an OPTIONS request determines the   scope of the request and the corresponding response.  If the Request-   URI refers to a specific media resource on a given host, the scope is   limited to the set of methods supported for that media resource by   the indicated RTSP agent.  A Request-URI with only the host address   limits the scope to the specified RTSP agent's general capabilities   without regard to any specific media.  If the Request-URI is an   asterisk ("*"), the scope is limited to the general capabilities of   the next hop (i.e., the RTSP agent in direct communication with the   request sender).   Regardless of the scope of the request, the Public header MUST always   be included in the OPTIONS response, listing the methods that are   supported by the responding RTSP agent.  In addition, if the scope of   the request is limited to a media resource, the Allow header MUST be   included in the response to enumerate the set of methods that are   allowed for that resource unless the set of methods completely   matches the set in the Public header.  If the given resource is not   available, the RTSP agent SHOULD return an appropriate response code,   such as 3rr or 4xx.  The Supported header MAY be included in the   request to query the set of features that are supported by the   responding RTSP agent.Schulzrinne, et al.          Standards Track                   [Page 65]

RFC 7826                        RTSP 2.0                   December 2016   The OPTIONS method can be used to keep an RTSP session alive.   However, this is not the preferred way of session keep-alive   signaling; seeSection 18.49.  An OPTIONS request intended for   keeping alive an RTSP session MUST include the Session header with   the associated session identifier.  Such a request SHOULD also use   the media or the aggregated control URI as the Request-URI.   Example:     C->S:  OPTIONS rtsp://server.example.com RTSP/2.0            CSeq: 1            User-Agent: PhonyClient/1.2            Proxy-Require: gzipped-messages            Supported: play.basic     S->C:  RTSP/2.0 200 OK            CSeq: 1            Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, OPTIONS            Supported: play.basic, setup.rtp.rtcp.mux, play.scale            Server: PhonyServer/1.1   Note that the "gzipped-messages" feature tag in the Proxy-Require is   a fictitious feature.13.2.  DESCRIBE   The DESCRIBE method is used to retrieve the description of a   presentation or media object from a server.  The Request-URI of the   DESCRIBE request identifies the media resource of interest.  The   client MAY include the Accept header in the request to list the   description formats that it understands.  The server MUST respond   with a description of the requested resource and return the   description in the message body of the response, if the DESCRIBE   method request can be successfully fulfilled.  The DESCRIBE reply-   response pair constitutes the media initialization phase of RTSP.   The DESCRIBE response SHOULD contain all media initialization   information for the resource(s) that it describes.  Servers SHOULD   NOT use the DESCRIBE response as a means of media indirection by   having the description point at another server; instead, using the   3rr responses is RECOMMENDED.      By forcing a DESCRIBE response to contain all media initialization      information for the set of streams that it describes, and      discouraging the use of DESCRIBE for media indirection, any      looping problems can be avoided that might have resulted from      other approaches.Schulzrinne, et al.          Standards Track                   [Page 66]

RFC 7826                        RTSP 2.0                   December 2016   Example:     C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/2.0           CSeq: 312           User-Agent: PhonyClient/1.2           Accept: application/sdp, application/example     S->C: RTSP/2.0 200 OK           CSeq: 312           Date: Thu, 23 Jan 1997 15:35:06 GMT           Server: PhonyServer/1.1           Content-Base: rtsp://server.example.com/fizzle/foo/           Content-Type: application/sdp           Content-Length: 358           v=0           o=MNobody 2890844526 2890842807 IN IP4 192.0.2.46           s=SDP Seminar           i=A Seminar on the session description protocol           u=http://www.example.com/lectures/sdp.ps           e=seminar@example.com (Seminar Management)           c=IN IP4 0.0.0.0           a=control:*           t=2873397496 2873404696           m=audio 3456 RTP/AVP 0           a=control:audio           m=video 2232 RTP/AVP 31           a=control:video   Media initialization is a requirement for any RTSP-based system, but   the RTSP specification does not dictate that this is required to be   done via the DESCRIBE method.  There are three ways that an RTSP   client may receive initialization information:   o  via an RTSP DESCRIBE request   o  via some other protocol (HTTP, email attachment, etc.)   o  via some form of user interface   If a client obtains a valid description from an alternate source, the   client MAY use this description for initialization purposes without   issuing a DESCRIBE request for the same media.  The client should use   any MTag to either validate the presentation description or make the   session establishment conditional on being valid.Schulzrinne, et al.          Standards Track                   [Page 67]

RFC 7826                        RTSP 2.0                   December 2016   It is RECOMMENDED that minimal servers support the DESCRIBE method,   and highly recommended that minimal clients support the ability to   act as "helper applications" that accept a media initialization file   from a user interface, or other means that are appropriate to the   operating environment of the clients.13.3.  SETUP   The description below uses the following states in a protocol state   machine that is related to a specific session when that session has   been created.  The state transitions are driven by protocol   interactions.  For additional information about the state machine,   seeAppendix B.   Init: Initial state.  No session exists.   Ready:  Session is ready to start playing.   Play: Session is playing, i.e., sending media-stream data in the         direction S->C.   The SETUP request for a URI specifies the transport mechanism to be   used for the streamed media.  The SETUP method may be used in two   different cases, namely, creating an RTSP session and changing the   transport parameters of media streams that are already set up.  SETUP   can be used in all three states, Init, Ready, and Play, to change the   transport parameters.  Additionally, Init and Ready can also be used   for the creation of the RTSP session.  The usage of the SETUP method   in the Play state to add a media resource to the session is   unspecified.   The Transport header, seeSection 18.54, specifies the media-   transport parameters acceptable to the client for data transmission;   the response will contain the transport parameters selected by the   server.  This allows the client to enumerate, in descending order of   preference, the transport mechanisms and parameters acceptable to it,   so the server can select the most appropriate.  It is expected that   the session description format used will enable the client to select   a limited number of possible configurations that are offered as   choices to the server.  All transport-related parameters SHALL be   included in the Transport header; the use of other headers for this   purpose is NOT RECOMMENDED due to middleboxes, such as firewalls or   NATs.   For the benefit of any intervening firewalls, a client MUST indicate   the known transport parameters, even if it has no influence over   these parameters, for example, where the server advertises a fixed-   multicast address as destination.Schulzrinne, et al.          Standards Track                   [Page 68]

RFC 7826                        RTSP 2.0                   December 2016      Since SETUP includes all transport initialization information,      firewalls and other intermediate network devices (which need this      information) are spared the more arduous task of parsing the      DESCRIBE response, which has been reserved for media      initialization.   The client MUST include the Accept-Ranges header in the request,   indicating all supported unit formats in the Range header.  This   allows the server to know which formats it may use in future session-   related responses, such as a PLAY response without any range in the   request.  If the client does not support a time format necessary for   the presentation, the server MUST respond using 456 (Header Field Not   Valid for Resource) and include the Accept-Ranges header with the   range unit formats supported for the resource.   In a SETUP response, the server MUST include the Accept-Ranges header   (seeSection 18.5) to indicate which time formats are acceptable to   use for this media resource.   The SETUP 200 OK response MUST include the Media-Properties header   (seeSection 18.29).  The combination of the parameters of the Media-   Properties header indicates the nature of the content present in the   session (see alsoSection 4.7).  For example, a live stream with time   shifting is indicated by   o  Random access set to Random-Access,   o  Content Modifications set to Time-Progressing, and   o  Retention set to Time-Duration (with specific recording window      time value).   The SETUP 200 OK response MUST include the Media-Range header (seeSection 18.30) if the media is Time-Progressing.Schulzrinne, et al.          Standards Track                   [Page 69]

RFC 7826                        RTSP 2.0                   December 2016   A basic example for SETUP:     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0           CSeq: 302           Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589",                      RTP/AVP/TCP;unicast;interleaved=0-1           Accept-Ranges: npt, clock           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 302           Date: Thu, 23 Jan 1997 15:35:06 GMT           Server: PhonyServer/1.1           Session: QKyjN8nt2WqbWw4tIYof52;timeout=60           Transport: RTP/AVP;unicast;dest_addr="192.0.2.53:4588"/                      "192.0.2.53:4589"; src_addr="198.51.100.241:6256"/                      "198.51.100.241:6257"; ssrc=2A3F93ED           Accept-Ranges: npt           Media-Properties: Random-Access=3.2, Time-Progressing,                             Time-Duration=3600.0           Media-Range: npt=0-2893.23   In the above example, the client wants to create an RTSP session   containing the media resource "rtsp://example.com/foo/bar/baz.rm".   The transport parameters acceptable to the client are either RTP/AVP/   UDP (UDP per default) to be received on client port 4588 and 4589 at   the address the RTSP setup connection comes from or RTP/AVP   interleaved on the RTSP control channel.  The server selects the   RTP/AVP/UDP transport and adds the address and ports it will send and   receive RTP and RTCP from, and the RTP SSRC that will be used by the   server.   The server MUST generate a session identifier in response to a   successful SETUP request unless a SETUP request to a server includes   a session identifier or a Pipelined-Requests header referencing an   existing session context.  In that latter case, the server MUST   bundle this SETUP request into the existing session (aggregated   session) or return a 459 (Aggregate Operation Not Allowed) error code   (seeSection 17.4.23).  An aggregate control URI MUST be used to   control an aggregated session.  This URI MUST be different from the   stream control URIs of the individual media streams included in the   aggregate (seeSection 13.4.2 for aggregated sessions and for the   particular URIs seeAppendix D.1.1).  The aggregate control URI is to   be specified by the session description if the server supports   aggregated control and aggregated control is desired for the session.Schulzrinne, et al.          Standards Track                   [Page 70]

RFC 7826                        RTSP 2.0                   December 2016   However, even if aggregated control is offered, the client MAY choose   not to set up the session in aggregated control.  If an aggregate   control URI is not specified in the session description, it is   normally an indication that non-aggregated control should be used.   The SETUP of media streams in an aggregate that has not been given an   aggregated control URI is unspecified.      While the session ID sometimes carries enough information for      aggregate control of a session, the aggregate control URI is still      important for some methods such as SET_PARAMETER where the control      URI enables the resource in question to be easily identified.  The      aggregate control URI is also useful for proxies, enabling them to      route the request to the appropriate server, and for logging,      where it is useful to note the actual resource on which a request      was operating.   A session will exist until it is either removed by a TEARDOWN request   or is timed out by the server.  The server MAY remove a session that   has not demonstrated liveness signs from the client(s) within a   certain timeout period.  The default timeout value is 60 seconds; the   server MAY set this to a different value and indicate so in the   timeout field of the Session header in the SETUP response.  For   further discussion, seeSection 18.49.  Signs of liveness for an RTSP   session include any RTSP requests from a client that contain a   Session header with the ID for that session, as well as RTCP sender   or receiver reports if RTP is used to transport the underlying media   stream.  RTCP sender reports may, for example, be received in session   where the server is invited into a conference session and are thus   valid as a liveness indicator.   If a SETUP request on a session fails for any reason, the session   state, as well as transport and other parameters for associated   streams, MUST remain unchanged from their values as if the SETUP   request had never been received by the server.13.3.1.  Changing Transport Parameters   A client MAY issue a SETUP request for a stream that is already set   up or playing in the session to change transport parameters, which a   server MAY allow.  If it does not allow the changing of parameters,   it MUST respond with error 455 (Method Not Valid in This State).  The   reasons to support changing transport parameters include allowing   application-layer mobility and flexibility to utilize the best   available transport as it becomes available.  If a client receives a   455 error when trying to change transport parameters while the server   is in Play state, it MAY try to put the server in Ready state using   PAUSE before issuing the SETUP request again.  If that also fails,Schulzrinne, et al.          Standards Track                   [Page 71]

RFC 7826                        RTSP 2.0                   December 2016   the changing of transport parameters will require that the client   perform a TEARDOWN of the affected media and then set it up again.   For an aggregated session, not tearing down all the media at the same   time will avoid the creation of a new session.   All transport parameters MAY be changed.  However, the primary usage   expected is to either change the transport protocol completely, like   switching from Interleaved TCP mode to UDP or vice versa, or to   change the delivery address.   In a SETUP response for a request to change the transport parameters   while in Play state, the server MUST include the Range header to   indicate at what point the new transport parameters will be used.   Further, if RTP is used for delivery, the server MUST also include   the RTP-Info header to indicate at what timestamp and RTP sequence   number the change will take place.  If both RTP-Info and Range are   included in the response, the "rtp_time" parameter and start point in   the Range header MUST be for the corresponding time, i.e., be used in   the same way as for PLAY to ensure the correct synchronization   information is available.   If the transport-parameters change that happened while in Play state   results in a change of synchronization-related information, for   example, changing RTP SSRC, the server MUST include the necessary   synchronization information in the SETUP response.  However, the   server SHOULD avoid changing the synchronization information if   possible.13.4.  PLAY   This section describes the usage of the PLAY method in general, for   aggregated sessions, and in different usage scenarios.13.4.1.  General Usage   The PLAY method tells the server to start sending data via the   mechanism specified in SETUP and which part of the media should be   played out.  PLAY requests are valid when the session is in Ready or   Play state.  A PLAY request MUST include a Session header to indicate   to which session the request applies.   Upon receipt of the PLAY request, the server MUST position the normal   play time to the beginning of the range specified in the received   Range header, within the limits of the media resource and in   accordance with the Seek-Style header (Section 18.47).  It MUST   deliver stream data until the end of the range if given, until a new   PLAY request is received, until a PAUSE request (Section 13.5) is   received, or until the end of the media is reached.  If no RangeSchulzrinne, et al.          Standards Track                   [Page 72]

RFC 7826                        RTSP 2.0                   December 2016   header is present in the PLAY request, the server SHALL play from   current pause point until the end of media.  The pause point defaults   at session start to the beginning of the media.  For media that is   time-progressing and has no retention, the pause point will always be   set equal to NPT "now", i.e., the current delivery point.  The pause   point may also be set to a particular point in the media by the PAUSE   method; seeSection 13.6.  The pause point for media that is   currently playing is equal to the current media position.  For time-   progressing media with time-limited retention, if the pause point   represents a position that is older than what is retained by the   server, the pause point will be moved to the oldest retained   position.   What range values are valid depends on the type of content.  For   content that isn't time-progressing, the range value is valid if the   given range is part of any media within the aggregate.  In other   words, the valid media range for the aggregate is the union of all of   the media components in the aggregate.  If a given range value points   outside of the media, the response MUST be the 457 (Invalid Range)   error code and include the Media-Range header (Section 18.30) with   the valid range for the media.  Except for time-progressing content   where the client requests a start point prior to what is retained,   the start point is adjusted to the oldest retained content.  For a   start point that is beyond the media front edge, i.e., beyond the   current value for "now", the server SHALL adjust the start value to   the current front edge.  The Range header's stop point value may   point beyond the current media edge.  In that case, the server SHALL   deliver media from the requested (and possibly adjusted) start point   until the first of either the provided stop point or the end of the   media.  Please note that if one simply wants to play from a   particular start point until the end of media, using a Range header   with an implicit stop point is RECOMMENDED.   If a client requests to start playing at the end of media, either   explicitly with a Range header or implicitly with a pause point that   is at the end of media, a 457 (Invalid Range) error MUST be sent and   include the Media-Range header (Section 18.30).  It is specified   below that the Range header also must be included in the response and   that it will carry the pause point in the media, in the case of the   session being in Ready State.  Note that this also applies if the   pause point or requested start point is at the beginning of the media   and a Scale header (Section 18.46) is included with a negative value   (playing backwards).   For media with random access properties, a client may indicate which   policy for start point selection the server should use.  This is done   by including the Seek-Style header (Section 18.47) in the PLAYSchulzrinne, et al.          Standards Track                   [Page 73]

RFC 7826                        RTSP 2.0                   December 2016   request.  The Seek-Style applied will affect the content of the Range   header as it will be adjusted to indicate from what point the media   actually is delivered.   A client desiring to play the media from the beginning MUST send a   PLAY request with a Range header pointing at the beginning, e.g.,   "npt=0-".  If a PLAY request is received without a Range header and   media delivery has stopped at the end, the server SHOULD respond with   a 457 (Invalid Range) error response.  In that response, the current   pause point MUST be included in a Range header.   All range specifiers in this specification allow for ranges with an   implicit start point (e.g., "npt=-30").  When used in a PLAY request,   the server treats this as a request to start or resume delivery from   the current pause point, ending at the end time specified in the   Range header.  If the pause point is located later than the given end   value, a 457 (Invalid Range) response MUST be returned.   The example below will play seconds 10 through 25.  It also requests   that the server deliver media from the first random access point   prior to the indicated start point.     C->S: PLAY rtsp://audio.example.com/audio RTSP/2.0           CSeq: 835           Session: ULExwZCXh2pd0xuFgkgZJW           Range: npt=10-25           Seek-Style: RAP           User-Agent: PhonyClient/1.2   Servers MUST include a Range header in any PLAY response, even if no   Range header was present in the request.  The response MUST use the   same format as the request's Range header contained.  If no Range   header was in the request, the format used in any previous PLAY   request within the session SHOULD be used.  If no format has been   indicated in a previous request, the server MAY use any time format   supported by the media and indicated in the Accept-Ranges header in   the SETUP request.  It is RECOMMENDED that NPT is used if supported   by the media.   For any error response to a PLAY request, the server's response   depends on the current session state.  If the session is in Ready   state, the current pause point is returned using a Range header with   the pause point as the explicit start point and an implicit stop   point.  For time-progressing content, where the pause-point moves   with real-time due to limited retention, the current pause point is   returned.  For sessions in Play state, the current playout point andSchulzrinne, et al.          Standards Track                   [Page 74]

RFC 7826                        RTSP 2.0                   December 2016   the remaining parts of the range request are returned.  For any media   with retention longer than 0 seconds, the currently valid Media-Range   header SHALL also be included in the response.   A PLAY response MAY include a header carrying synchronization   information.  As the information necessary is dependent on the media-   transport format, further rules specifying the header and its usage   are needed.  For RTP the RTP-Info header is specified, seeSection 18.45, and used in the following example.   Here is a simple example for a single audio stream where the client   requests the media starting from 3.52 seconds and to the end.  The   server sends a 200 OK response with the actual play time, which is 10   ms prior (3.51), and the RTP-Info header that contains the necessary   parameters for the RTP stack.   C->S: PLAY rtsp://example.com/audio RTSP/2.0         CSeq: 836         Session: ULExwZCXh2pd0xuFgkgZJW         Range: npt=3.52-         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 836         Date: Thu, 23 Jan 1997 15:35:06 GMT         Server: PhonyServer/1.0         Range: npt=3.51-324.39         Seek-Style: First-Prior             Session: ULExwZCXh2pd0xuFgkgZJW         RTP-Info:url="rtsp://example.com/audio"            ssrc=0D12F123:seq=14783;rtptime=2345962545   S->C: RTP Packet TS=2345962545 => NPT=3.51         Media duration=0.16 seconds   The server replies with the actual start point that will be   delivered.  This may differ from the requested range if alignment of   the requested range to valid frame boundaries is required for the   media source.  Note that some media streams in an aggregate may need   to be delivered from even earlier points.  Also, some media formats   have a very long duration per individual data unit; therefore, it   might be necessary for the client to parse the data unit, and select   where to start.  The server SHALL also indicate which policy it uses   for selecting the actual start point by including a Seek-Style   header.Schulzrinne, et al.          Standards Track                   [Page 75]

RFC 7826                        RTSP 2.0                   December 2016   In the following example, the client receives the first media packet   that stretches all the way up and past the requested playtime.  Thus,   it is the client's decision whether to render to the user the time   between 3.52 and 7.05 or to skip it.  In most cases, it is probably   most suitable not to render that time period.   C->S: PLAY rtsp://example.com/audio RTSP/2.0         CSeq: 836         Session: ZGGyCJOs8xaLkdNK2dmxQO         Range: npt=7.05-         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 836         Date: Thu, 23 Jan 1997 15:35:06 GMT         Server: PhonyServer/1.0             Session: ZGGyCJOs8xaLkdNK2dmxQO         Range: npt=3.52-         Seek-Style: First-Prior         RTP-Info:url="rtsp://example.com/audio"            ssrc=0D12F123:seq=14783;rtptime=2345962545   S->C: RTP Packet TS=2345962545 => NPT=3.52         Duration=4.15 seconds   After playing the desired range, the presentation does NOT change to   the Ready state, media delivery simply stops.  If it is necessary to   put the stream into the Ready state, a PAUSE request MUST be issued.   A PLAY request while the stream is still in the Play state is legal   and can be issued without an intervening PAUSE request.  Such a   request MUST replace the current PLAY action with the new one   requested, i.e., being handled in the same way as if as the request   was received in Ready state.  In the case that the range in the Range   header has an implicit start time ("-endtime"), the server MUST   continue to play from where it currently was until the specified   endpoint.  This is useful to change the end to at another point than   in the previous request.   The following example plays the whole presentation starting at SMPTE   time code 0:10:20 until the end of the clip.  Note: the RTP-Info   headers have been broken into several lines, where subsequent lines   start with whitespace as allowed by the syntax.Schulzrinne, et al.          Standards Track                   [Page 76]

RFC 7826                        RTSP 2.0                   December 2016   C->S: PLAY rtsp://audio.example.com/twister.en RTSP/2.0         CSeq: 833         Session: N465Wvsv0cjUy6tLqINkcf         Range: smpte=0:10:20-         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 833         Date: Thu, 23 Jan 1997 15:35:06 GMT         Session: N465Wvsv0cjUy6tLqINkcf         Server: PhonyServer/1.0         Range: smpte=0:10:22-0:15:45         Seek-Style: Next         RTP-Info:url="rtsp://example.com/twister.en"            ssrc=0D12F123:seq=14783;rtptime=2345962545   For playing back a recording of a live presentation, it may be   desirable to use clock units:   C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/2.0         CSeq: 835         Session: N465Wvsv0cjUy6tLqINkcf         Range: clock=19961108T142300Z-19961108T143520Z         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 835         Date: Thu, 23 Jan 1997 15:35:06 GMT         Session: N465Wvsv0cjUy6tLqINkcf         Server: PhonyServer/1.0         Range: clock=19961108T142300Z-19961108T143520Z         Seek-Style: Next         RTP-Info:url="rtsp://example.com/meeting.en"            ssrc=0D12F123:seq=53745;rtptime=48458901913.4.2.  Aggregated Sessions   PLAY requests can operate on sessions controlling a single media   stream and on aggregated sessions controlling multiple media streams.   In an aggregated session, the PLAY request MUST contain an aggregated   control URI.  A server MUST respond with a 460 error (Only Aggregate   Operation Allowed) if the client PLAY Request-URI is for a single   media.  The media in an aggregate MUST be played in sync.  If a   client wants individual control of the media, it needs to use   separate RTSP sessions for each media.Schulzrinne, et al.          Standards Track                   [Page 77]

RFC 7826                        RTSP 2.0                   December 2016   For aggregated sessions where the initial SETUP request (creating a   session) is followed by one or more additional SETUP requests, a PLAY   request MAY be pipelined (Section 12) after those additional SETUP   requests without awaiting their responses.  This procedure can reduce   the delay from the start of session establishment until media playout   has started with one RTT.  However, a client needs to be aware that   using this procedure will result in the playout of the server state   established at the time of processing the PLAY, i.e., after the   processing of all the requests prior to the PLAY request in the   pipeline.  This state may not be the intended one due to failure of   any of the prior requests.  A client can easily determine this based   on the responses from those requests.  In case of failure, the client   can halt the media playout using PAUSE and try to establish the   intended state again before issuing another PLAY request.13.4.3.  Updating Current PLAY Requests   Clients can issue PLAY requests while the stream is in Play state and   thus updating their request.   The important difference compared to a PLAY request in Ready state is   the handling of the current play point and how the Range header in   the request is constructed.  The session is actively playing media   and the play point will be moving, making the exact time a request   will take effect hard to predict.  Depending on how the PLAY header   appears, two different cases exist: total replacement or   continuation.  A total replacement is signaled by having the first   range specification have an explicit start value, e.g., "npt=45-" or   "npt=45-60", in which case the server stops playout at the current   playout point and then starts delivering media according to the Range   header.  This is equivalent to having the client first send a PAUSE   and then a new PLAY request that isn't based on the pause point.  In   the case of continuation, the first range specifier has an implicit   start point and an explicit stop value (Z), e.g., "npt=-60", which   indicate that it MUST convert the range specifier being played prior   to this PLAY request (X to Y) into (X to Z) and continue as if this   was the request originally played.  If the current delivery point is   beyond the stop point, the server SHALL immediately pause delivery.   As the request has been completed successfully, it shall be responded   to with a 200 OK response.  A PLAY_NOTIFY with end-of-stream is also   sent to indicate the actual stop point.  The pause point is set to   the requested stop point.   The following is an example of this behavior: The server has received   requests to play ranges 10 to 15.  If the new PLAY request arrives at   the server 4 seconds after the previous one, it will take effectSchulzrinne, et al.          Standards Track                   [Page 78]

RFC 7826                        RTSP 2.0                   December 2016   while the server still plays the first range (10-15).  The server   changes the current play to continue to 25 seconds, i.e., the   equivalent single request would be PLAY with "range: npt=10-25".     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 834           Session: apzA8LnjQ5KWTdw0kUkiRh           Range: npt=10-15           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 834           Date: Thu, 23 Jan 1997 15:35:06 GMT           Session: apzA8LnjQ5KWTdw0kUkiRh           Server: PhonyServer/1.0           Range: npt=10-15           Seek-Style: Next           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"                   ssrc=0D12F123:seq=5712;rtptime=934207921,                   url="rtsp://example.com/fizzle/videotrack"                   ssrc=789DAF12:seq=57654;rtptime=2792482193     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 835           Session: apzA8LnjQ5KWTdw0kUkiRh           Range: npt=-25           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 835           Date: Thu, 23 Jan 1997 15:35:09 GMT           Session: apzA8LnjQ5KWTdw0kUkiRh           Server: PhonyServer/1.0           Range: npt=14-25           Seek-Style: Next           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"                   ssrc=0D12F123:seq=5712;rtptime=934239921,                   url="rtsp://example.com/fizzle/videotrack"                   ssrc=789DAF12:seq=57654;rtptime=2792842193   A common use of a PLAY request while in Play state is changing the   scale of the media, i.e., entering or leaving fast forward or fast   rewind.  The client can issue an updating PLAY request that is either   a continuation or a complete replacement, as discussed above this   section.  Below is an example of a client that is requesting a fast   forward (scale = 2) without giving a stop point and then a change   from fast forward to regular playout (scale = 1).  In the second PLAYSchulzrinne, et al.          Standards Track                   [Page 79]

RFC 7826                        RTSP 2.0                   December 2016   request, the time is set explicitly to be wherever the server   currently plays out (npt=now-) and the server responds with the   actual playback point where the new scale actually takes effect   (npt=02:17:27.144-).     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 2034           Session: apzA8LnjQ5KWTdw0kUkiRh           Range: npt=now-           Scale: 2.0           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 2034           Date: Thu, 23 Jan 1997 15:35:06 GMT           Session: apzA8LnjQ5KWTdw0kUkiRh           Server: PhonyServer/1.0           Range: npt=02:17:21.394-           Seek-Style: Next           Scale: 2.0           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"                   ssrc=0D12F123:seq=5712;rtptime=934207921,                   url="rtsp://example.com/fizzle/videotrack"                   ssrc=789DAF12:seq=57654;rtptime=2792482193   [playing in fast forward and now returning to scale = 1]     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 2035           Session: apzA8LnjQ5KWTdw0kUkiRh           Range: npt=now-           Scale: 1.0           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 2035           Date: Thu, 23 Jan 1997 15:35:09 GMT           Session: apzA8LnjQ5KWTdw0kUkiRh           Server: PhonyServer/1.0           Range: npt=02:17:27.144-           Seek-Style: Next           Scale: 1.0           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"                   ssrc=0D12F123:seq=5712;rtptime=934239921,                   url="rtsp://example.com/fizzle/videotrack"                   ssrc=789DAF12:seq=57654;rtptime=2792842193Schulzrinne, et al.          Standards Track                   [Page 80]

RFC 7826                        RTSP 2.0                   December 201613.4.4.  Playing On-Demand Media   On-demand media is indicated by the content of the Media-Properties   header in the SETUP response when (see alsoSection 18.29):   o  the Random Access property is set to Random-Access;   o  the Content Modifications property is set to Immutable;   o  the Retention property is set to Unlimited or Time-Limited.   Playing on-demand media follows the general usage as described inSection 13.4.1.13.4.5.  Playing Dynamic On-Demand Media   Dynamic on-demand media is indicated by the content of the Media-   Properties header in the SETUP response when (see alsoSection 18.29):   o  the Random Access property is set to Random-Access;   o  the Content Modifications property is set to Dynamic;   o  the Retention property is set to Unlimited or Time-Limited.   Playing on-demand media follows the general usage as described inSection 13.4.1 as long as the media has not been changed.   There are two ways for the client to be informed about changes of   media resources in Play state.  The first being that the client will   receive a PLAY_NOTIFY request with the Notify-Reason header set to   media-properties-update (seeSection 13.5.2).  The client can use the   value of the Media-Range header to decide further actions, if the   Media-Range header is present in the PLAY_NOTIFY request.  The second   way is that the client issues a GET_PARAMETER request without a body   but including a Media-Range header.  The 200 OK response MUST include   the current Media-Range header (seeSection 18.30).13.4.6.  Playing Live Media   Live media is indicated by the content of the Media-Properties header   in the SETUP response when (see alsoSection 18.29):   o  the Random Access property is set to No-Seeking;   o  the Content Modifications property is set to Time-Progressing;Schulzrinne, et al.          Standards Track                   [Page 81]

RFC 7826                        RTSP 2.0                   December 2016   o  the Retention property's Time-Duration is set to 0.0.   For live media, the SETUP 200 OK response MUST include the Media-   Range header (seeSection 18.30).   A client MAY send PLAY requests without the Range header.  If the   request includes the Range header, it MUST use a symbolic value   representing "now".  For NPT, that range specification is "npt=now-".   The server MUST include the Range header in the response, and it MUST   indicate an explicit time value and not a symbolic value.  In other   words, "npt=now-" cannot be used in the response.  Instead, the time   since session start is recommended, expressed as an open interval,   e.g., "npt=96.23-".  An absolute time value (clock) for the   corresponding time MAY be given, i.e., "clock=20030213T143205Z-".   The Absolute Time format can only be used if the client has shown   support for it using the Accept-Ranges header.13.4.7.  Playing Live with Recording   Certain media servers may offer recording services of live sessions   to their clients.  This recording would normally be from the   beginning of the media session.  Clients can randomly access the   media between now and the beginning of the media session.  This live   media with recording is indicated by the content of the Media-   Properties header in the SETUP response when (see alsoSection 18.29):   o  the Random Access property is set to Random-Access;   o  the Content Modifications property is set to Time-Progressing;   o  the Retention property is set to Time-Limited or Unlimited   The SETUP 200 OK response MUST include the Media-Range header (seeSection 18.30) for this type of media.  For live media with   recording, the Range header indicates the current delivery point in   the media and the Media-Range header indicates the currently   available media window around the current time.  This window can   cover recorded content in the past (seen from current time in the   media) or recorded content in the future (seen from current time in   the media).  The server adjusts the delivery point to the requested   border of the window.  If the client requests a delivery point that   is located outside the recording window, e.g., if the requested point   is too far in the past, the server selects the oldest point in the   recording.  The considerations inSection 13.5.3 apply if a client   requests delivery with scale (Section 18.46) values other than 1.0   (normal playback rate) while delivering live media with recording.Schulzrinne, et al.          Standards Track                   [Page 82]

RFC 7826                        RTSP 2.0                   December 201613.4.8.  Playing Live with Time-Shift   Certain media servers may offer time-shift services to their clients.   This time shift records a fixed interval in the past, i.e., a sliding   window recording mechanism, but not past this interval.  Clients can   randomly access the media between now and the interval.  This live   media with recording is indicated by the content of the Media-   Properties header in the SETUP response when (see alsoSection 18.29):   o  the Random Access property is set to Random-Access;   o  the Content Modifications property is set to Time-Progressing;   o  the Retention property is set to Time-Duration and a value      indicating the recording interval (>0).   The SETUP 200 OK response MUST include the Media-Range header (seeSection 18.30) for this type of media.  For live media with   recording, the Range header indicates the current time in the media   and the Media-Range header indicates a window around the current   time.  This window can cover recorded content in the past (seen from   current time in the media) or recorded content in the future (seen   from current time in the media).  The server adjusts the play point   to the requested border of the window, if the client requests a play   point that is located outside the recording windows, e.g., if   requested too far in the past, the server selects the oldest range in   the recording.  The considerations inSection 13.5.3 apply if a   client requests delivery using a scale (Section 18.46) value other   than 1.0 (normal playback rate) while delivering live media with   time-shift.13.5.  PLAY_NOTIFY   The PLAY_NOTIFY method is issued by a server to inform a client about   an asynchronous event for a session in Play state.  The Session   header MUST be presented in a PLAY_NOTIFY request and indicates the   scope of the request.  Sending of PLAY_NOTIFY requests requires a   persistent connection between server and client; otherwise, there is   no way for the server to send this request method to the client.   PLAY_NOTIFY requests have an end-to-end (i.e., server-to-client)   scope, as they carry the Session header, and apply only to the given   session.  The client SHOULD immediately return a response to the   server.Schulzrinne, et al.          Standards Track                   [Page 83]

RFC 7826                        RTSP 2.0                   December 2016   PLAY_NOTIFY requests MAY use both an aggregate control URI and   individual media resource URIs, depending on the scope of the   notification.  This scope may have important distinctions for   aggregated sessions, and each reason for a PLAY_NOTIFY request needs   to specify the interpretation as well as if aggregated control URIs   or individual URIs may be used in requests.   PLAY_NOTIFY requests can be used with a message body, depending on   the value of the Notify-Reason header.  It is described in the   particular section for each Notify-Reason if a message body is used.   However, currently there is no Notify-Reason that allows the use of a   message body.  In this case, there is a need to obey some limitations   when adding new Notify-Reasons that intend to use a message body: the   server can send any type of message body, but it is not ensured that   the client can understand the received message body.  This is related   to DESCRIBE (seeSection 13.2 ); but, in this particular case, the   client can state its acceptable message bodies by using the Accept   header.  In the case of PLAY_NOTIFY, the server does not know which   message bodies are understood by the client.   The Notify-Reason header (seeSection 18.32) specifies the reason why   the server sends the PLAY_NOTIFY request.  This is extensible and new   reasons can be added in the future (seeSection 22.8).  In case the   client does not understand the reason for the notification, it MUST   respond with a 465 (Notification Reason Unknown) (Section 17.4.29)   error code.  This document defines how servers can send PLAY_NOTIFY   with Notify-Reason values of these types:   o  end-of-stream (seeSection 13.5.1);   o  media-properties-update (seeSection 13.5.2);   o  scale-change (seeSection 13.5.3).13.5.1.  End-of-Stream   A PLAY_NOTIFY request with the Notify-Reason header set to end-of-   stream indicates the completion or near completion of the PLAY   request and the ending delivery of the media stream(s).  The request   MUST NOT be issued unless the server is in the Play state.  The end   of the media stream delivery notification may be used either to   indicate a successful completion of the PLAY request currently being   served or to indicate some error resulting in failure to complete the   request.  The Request-Status header (Section 18.42) MUST be included   to indicate which request the notification is for and its completion   status.  The message response status codes (Section 8.1.1) are used   to indicate how the PLAY request concluded.  The sender of a   PLAY_NOTIFY MAY issue an updated PLAY_NOTIFY, in the case of aSchulzrinne, et al.          Standards Track                   [Page 84]

RFC 7826                        RTSP 2.0                   December 2016   PLAY_NOTIFY sent with wrong information.  For instance, a PLAY_NOTIFY   was issued before reaching the end-of-stream, but some error occurred   resulting in that the previously sent PLAY_NOTIFY contained a wrong   time when the stream will end.  In this case, a new PLAY_NOTIFY MUST   be sent including the correct status for the completion and all   additional information.   PLAY_NOTIFY requests with the Notify-Reason header set to end-of-   stream MUST include a Range header and the Scale header if the scale   value is not 1.  The Range header indicates the point in the stream   or streams where delivery is ending with the timescale that was used   by the server in the PLAY response for the request being fulfilled.   The server MUST NOT use the "now" constant in the Range header; it   MUST use the actual numeric end position in the proper timescale.   When end-of-stream notifications are issued prior to having sent the   last media packets, this is made evident because the end time in the   Range header is beyond the current time in the media being received   by the client, e.g., "npt=-15", if npt is currently at 14.2 seconds.   The Scale header is to be included so that it is evident if the media   timescale is moving backwards or has a non-default pace.  The end-of-   stream notification does not prevent the client from sending a new   PLAY request.   If RTP is used as media transport, an RTP-Info header MUST be   included, and the RTP-Info header MUST indicate the last sequence   number in the sequence parameter.   For an RTSP Session where media resources are under aggregated   control, the media resources will normally end at approximately the   same time, although some small differences may exist, on the scale of   a few hundred milliseconds.  In those cases, an RTSP session under   aggregated control SHOULD send only a single PLAY_NOTIFY request.  By   using the aggregate control URI in the PLAY_NOTIFY request, the RTSP   server indicates that this applies to all media resources within the   session.  In cases in which RTP is used for media delivery,   corresponding RTP-Info needs to be included for all media resources.   In cases where one or more media resources have a significantly   shorter duration than some other resources in the aggregated session,   the server MAY send end-of-stream notifications using individual   media resource URIs to indicate to agents that there will be no more   media for this particular media resource related to the current   active PLAY request.  In such cases, when the remaining media   resources come to the end of the stream, they MUST send a PLAY_NOTIFY   request using the aggregate control URI to indicate that no more   resources remain.   A PLAY_NOTIFY request with Notify-Reason header set to end-of-stream   MUST NOT carry a message body.Schulzrinne, et al.          Standards Track                   [Page 85]

RFC 7826                        RTSP 2.0                   December 2016   This example request notifies the client about a future end-of-stream   event:     S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 854           Notify-Reason: end-of-stream           Request-Status: cseq=853 status=200 reason="OK"           Range: npt=-145           RTP-Info:url="rtsp://example.com/fizzle/foo/audio"              ssrc=0D12F123:seq=14783;rtptime=2345962545,              url="rtsp://example.com/fizzle/video"              ssrc=789DAF12:seq=57654;rtptime=2792482193           Session: CDtUJfDQXJWtJ7Iqua2xOi           Date: Mon, 08 Mar 2010 13:37:16 GMT     C->S: RTSP/2.0 200 OK           CSeq: 854           User-Agent: PhonyClient/1.2           Session: CDtUJfDQXJWtJ7Iqua2xOi13.5.2.  Media-Properties-Update   A PLAY_NOTIFY request with a Notify-Reason header set to media-   properties-update indicates an update of the media properties for the   given session (seeSection 18.29) or the available media range that   can be played as indicated by the Media-Range header (Section 18.30).   PLAY_NOTIFY requests with Notify-Reason header set to media-   properties-update MUST include a Media-Properties and Date header and   SHOULD include a Media-Range header.  The Media-Properties header has   session scope; thus, for aggregated sessions, the PLAY_NOTIFY request   MUST use the aggregated control URI.   This notification MUST be sent for media that are time-progressing   every time an event happens that changes the basis for making   estimates on how the available for play-back media range will   progress with wall clock time.  In addition, it is RECOMMENDED that   the server send these notifications approximately every 5 minutes for   time-progressing content to ensure the long-term stability of the   client estimation and allow for clock skew detection by the client.   The time between notifications should be greater than 1 minute and   less than 2 hours.  For the reasons just explained, requests MUST   include a Media-Range header to provide current Media duration and a   Range header to indicate the current playing point and any remaining   parts of the requested range.Schulzrinne, et al.          Standards Track                   [Page 86]

RFC 7826                        RTSP 2.0                   December 2016      The recommendation for sending updates every 5 minutes is due to      any clock skew issues.  In 5 minutes, the clock skew should not      become too significant as this is not used for media playback and      synchronization, it is only for determining which content is      available to the user.   A PLAY_NOTIFY request with Notify-Reason header set to media-   properties-update MUST NOT carry a message body.    S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0           Date: Tue, 14 Apr 2008 15:48:06 GMT           CSeq: 854           Notify-Reason: media-properties-update           Session: CDtUJfDQXJWtJ7Iqua2xOi           Media-Properties: Time-Progressing,                 Time-Limited=20080415T153919.36Z, Random-Access=5.0           Media-Range: npt=00:00:00-01:37:21.394           Range: npt=01:15:49.873-     C->S: RTSP/2.0 200 OK           CSeq: 854           User-Agent: PhonyClient/1.2           Session: CDtUJfDQXJWtJ7Iqua2xOi13.5.3.  Scale-Change   The server may be forced to change the rate of media time per   playback time when a client requests delivery using a scale   (Section 18.46) value other than 1.0 (normal playback rate).  For   time-progressing media with some retention, i.e., the server stores   already-sent content, a client requesting to play with scale values   larger than 1 may catch up with the front end of the media.  The   server will then be unable to continue to provide content at scale   larger than 1 as content is only made available by the server at   scale = 1.  Another case is when scale < 1 and the media retention is   Time-Duration limited.  In this case, the delivery point can reach   the oldest media unit available, and further playback at this scale   becomes impossible as there will be no media available.  To avoid   having the client lose any media, the scale will need to be adjusted   to the same rate at which the media is removed from the storage   buffer, commonly scale = 1.0.   Another case is when the content itself consists of spliced pieces or   is dynamically updated.  In these cases, the server may be required   to change from one supported scale value (different than scale = 1.0)   to another.  In this case, the server will pick the closest value andSchulzrinne, et al.          Standards Track                   [Page 87]

RFC 7826                        RTSP 2.0                   December 2016   inform the client of what it has picked.  In these cases, the media   properties will also be sent, updating the supported scale values.   This enables a client to adjust the scale value used.   To minimize impact on playback in any of the above cases, the server   MUST modify the playback properties, set scale to a supportable   value, and continue delivery of the media.  When doing this   modification, it MUST send a PLAY_NOTIFY message with the Notify-   Reason header set to "scale-change".  The request MUST contain a   Range header with the media time when the change took effect, a Scale   header with the new value in use, a Session header with the   identifier for the session to which it applies, and a Date header   with the server wallclock time of the change.  For time-progressing   content, the Media-Range and the Media-Properties headers at this   point in time also MUST be included.  The Media-Properties header   MUST be included if the scale change was due to the content changing   what scale values ("Scales") are supported.   For media streams delivered using RTP, an RTP-Info header MUST also   be included.  It MUST contain the rtptime parameter with a value   corresponding to the point of change in that media and optionally the   sequence number.   PLAY_NOTIFY requests for aggregated sessions MUST use the aggregated   control URI in the request.  The scale change for any aggregated   session applies to all media streams that are part of the aggregate.   A PLAY_NOTIFY request with Notify-Reason header set to "Scale-Change"   MUST NOT carry a message body.Schulzrinne, et al.          Standards Track                   [Page 88]

RFC 7826                        RTSP 2.0                   December 2016     S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0           Date: Tue, 14 Apr 2008 15:48:06 GMT           CSeq: 854           Notify-Reason: scale-change           Session: CDtUJfDQXJWtJ7Iqua2xOi           Media-Properties: Time-Progressing,                 Time-Limited=20080415T153919.36Z, Random-Access=5.0           Media-Range: npt=00:00:00-01:37:21.394           Range: npt=01:37:21.394-           Scale: 1           RTP-Info: url="rtsp://example.com/fizzle/foo/audio"               ssrc=0D12F123:rtptime=2345962545,               url="rtsp://example.com/fizzle/foo/videotrack"               ssrc=789DAF12:seq=57654;rtptime=2792482193     C->S: RTSP/2.0 200 OK           CSeq: 854           User-Agent: PhonyClient/1.2           Session: CDtUJfDQXJWtJ7Iqua2xOi13.6.  PAUSE   The PAUSE request causes the stream delivery to immediately be   interrupted (halted).  A PAUSE request MUST be made either with the   aggregated control URI for aggregated sessions, resulting in all   media being halted, or with the media URI for non-aggregated   sessions.  Any attempt to mute a single media with a PAUSE request in   an aggregated session MUST be responded to with a 460 (Only Aggregate   Operation Allowed) error.  After resuming playback, synchronization   of the tracks MUST be maintained.  Any server resources are kept,   though servers MAY close the session and free resources after being   paused for the duration specified with the timeout parameter of the   Session header in the SETUP message.   Example:     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 834           Session: OoOUPyUwt0VeY9fFRHuZ6L           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 834           Date: Thu, 23 Jan 1997 15:35:06 GMT                   Session: OoOUPyUwt0VeY9fFRHuZ6L           Range: npt=45.76-75.00Schulzrinne, et al.          Standards Track                   [Page 89]

RFC 7826                        RTSP 2.0                   December 2016   The PAUSE request causes stream delivery to be interrupted   immediately on receipt of the message, and the pause point is set to   the current point in the presentation.  That pause point in the media   stream needs to be maintained.  A subsequent PLAY request without a   Range header resumes from the pause point and plays until media end.   The pause point after any PAUSE request MUST be returned to the   client by adding a Range header with what remains unplayed of the   PLAY request's range.  For media with random access properties, if   one desires to resume playing a ranged request, one simply includes   the Range header from the PAUSE response and includes the Seek-Style   header with the Next policy in the PLAY request.  For media that is   time-progressing and has retention duration=0, the follow-up PLAY   request to start media delivery again MUST use "npt=now-" and not the   answer given in the response to PAUSE.     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 834           Session: OccldOFFq23KwjYpAnBbUr           Range: npt=10-30           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 834           Date: Thu, 23 Jan 1997 15:35:06 GMT           Server: PhonyServer/1.0           Range: npt=10-30           Seek-Style: First-Prior           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"                   ssrc=0D12F123:seq=5712;rtptime=934207921,                   url="rtsp://example.com/fizzle/videotrack"                   ssrc=4FAD8726:seq=57654;rtptime=2792482193           Session: OccldOFFq23KwjYpAnBbUrSchulzrinne, et al.          Standards Track                   [Page 90]

RFC 7826                        RTSP 2.0                   December 2016   After 11 seconds, i.e., at 21 seconds into the presentation:     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 835           Session: OccldOFFq23KwjYpAnBbUr           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 835           Date: 23 Jan 1997 15:35:17 GMT           Server: PhonyServer/1.0           Range: npt=21-30           Session: OccldOFFq23KwjYpAnBbUr   If a client issues a PAUSE request and the server acknowledges and   enters the Ready state, the proper server response, if the player   issues another PAUSE, is still 200 OK.  The 200 OK response MUST   include the Range header with the current pause point.  See examples   below:     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 834           Session: OccldOFFq23KwjYpAnBbUr           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 834           Session: OccldOFFq23KwjYpAnBbUr           Date: Thu, 23 Jan 1997 15:35:06 GMT           Range: npt=45.76-98.36     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 835           Session: OccldOFFq23KwjYpAnBbUr           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 835           Session: OccldOFFq23KwjYpAnBbUr           Date: 23 Jan 1997 15:35:07 GMT           Range: npt=45.76-98.36Schulzrinne, et al.          Standards Track                   [Page 91]

RFC 7826                        RTSP 2.0                   December 201613.7.  TEARDOWN13.7.1.  Client to Server   The TEARDOWN client-to-server request stops the stream delivery for   the given URI, freeing the resources associated with it.  A TEARDOWN   request can be performed on either an aggregated or a media control   URI.  However, some restrictions apply depending on the current   state.  The TEARDOWN request MUST contain a Session header indicating   to what session the request applies.  The TEARDOWN request MUST NOT   include a Terminate-Reason header.   A TEARDOWN using the aggregated control URI or the media URI in a   session under non-aggregated control (single media session) MAY be   done in any state (Ready and Play).  A successful request MUST result   in that media delivery being immediately halted and the session state   being destroyed.  This MUST be indicated through the lack of a   Session header in the response.   A TEARDOWN using a media URI in an aggregated session can only be   done in Ready state.  Such a request only removes the indicated media   stream and associated resources from the session.  This may result in   a session returning to non-aggregated control, because it only   contains a single media after the request's completion.  A session   that will exist after the processing of the TEARDOWN request MUST, in   the response to that TEARDOWN request, contain a Session header.   Thus, the presence of the Session header indicates to the receiver of   the response if the session is still extant or has been removed.Schulzrinne, et al.          Standards Track                   [Page 92]

RFC 7826                        RTSP 2.0                   December 2016   Example:     C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 892           Session: OccldOFFq23KwjYpAnBbUr           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 892           Server: PhonyServer/1.013.7.2.  Server to Client   The server can send TEARDOWN requests in the server-to-client   direction to indicate that the server has been forced to terminate   the ongoing session.  This may happen for several reasons, such as   server maintenance without available backup, or that the session has   been inactive for extended periods of time.  The reason is provided   in the Terminate-Reason header (Section 18.52).   When an RTSP client has maintained an RTSP session that otherwise is   inactive for an extended period of time, the server may reclaim the   resources.  That is done by issuing a TEARDOWN request with the   Terminate-Reason set to "Session-Timeout".  This MAY be done when the   client has been inactive in the RTSP session for more than one   Session Timeout period (Section 18.49).  However, the server is NOT   RECOMMENDED to perform this operation until an extended period of   inactivity of 10 times the Session-Timeout period has passed.  It is   up to the operator of the RTSP server to actually configure how long   this extended period of inactivity is.  An operator should take into   account, when doing this configuration, what the served content is   and what this means for the extended period of inactivity.   In case the server needs to stop providing service to the established   sessions and there is no server to point at in a REDIRECT request,   then TEARDOWN SHALL be used to terminate the session.  This method   can also be used when non-recoverable internal errors have happened   and the server has no other option than to terminate the sessions.   The TEARDOWN request MUST be made only on the session aggregate   control URI (i.e., it is not allowed to terminate individual media   streams, if it is a session aggregate), and it MUST include the   following headers: Session and Terminate-Reason.  The request only   applies to the session identified in the Session header.  The server   may include a message to the client's user with the "user-msg"   parameter.Schulzrinne, et al.          Standards Track                   [Page 93]

RFC 7826                        RTSP 2.0                   December 2016   The TEARDOWN request may alternatively be done on the wildcard URI   "*" and without any session header.  The scope of such a request is   limited to the next-hop (i.e., the RTSP agent in direct communication   with the server) and applies, as well, to the RTSP connection between   the next-hop RTSP agent and the server.  This request indicates that   all sessions and pending requests being managed via the connection   are terminated.  Any intervening proxies SHOULD do all of the   following in the order listed:   1.  respond to the TEARDOWN request   2.  disconnect the control channel from the requesting server   3.  pass the TEARDOWN request to each applicable client (typically       those clients with an active session or an unanswered request)      Note: The proxy is responsible for accepting TEARDOWN responses      from its clients; these responses MUST NOT be passed on to either      the original server or the target server in the redirect.13.8.  GET_PARAMETER   The GET_PARAMETER request retrieves the value of any specified   parameter or parameters for a presentation or stream specified in the   URI.  If the Session header is present in a request, the value of a   parameter MUST be retrieved in the specified session context.  There   are two ways of specifying the parameters to be retrieved.   The first approach includes headers that have been defined to be   usable for this purpose.  Headers for this purpose should allow   empty, or stripped value parts to avoid having to specify bogus data   when indicating the desire to retrieve a value.  The successful   completion of the request should also be evident from any filled out   values in the response.  The headers in this specification that MAY   be used for retrieving their current value using GET_PARAMETER are   listed below; additional headers MAY be specified in the future:   o  Accept-Ranges   o  Media-Range   o  Media-Properties   o  Range   o  RTP-InfoSchulzrinne, et al.          Standards Track                   [Page 94]

RFC 7826                        RTSP 2.0                   December 2016   The other way is to specify a message body that lists the   parameter(s) that are desired to be retrieved.  The Content-Type   header (Section 18.19) is used to specify which format the message   body has.  If the receiver of the request does not support the media   type used for the message body, it SHALL respond using the error code   415 (Unsupported Media Type).  The responder to a GET_PARAMETER   request MUST use the media type of the request for the response.  For   additional considerations regarding message body negotiation, seeSection 9.3.   RTSP agents implementing support for responding to GET_PARAMETER   requests SHALL implement the "text/parameters" format (Appendix F).   This to ensure that at least one known format for parameters is   implemented and, thus, prevent parameter format negotiation failure.   Parameters specified within the body of the message must all be   understood by the request-receiving agent.  If one or more parameters   are not understood a 451 (Parameter Not Understood) MUST be sent   including a body listing the parameters that weren't understood.  If   all parameters are understood, their values are filled in and   returned in the response message body.   The method can also be used without a message body or any header that   requests parameters for keep-alive purposes.  The keep-alive timer   has been updated for any request that is successful, i.e., a 200 OK   response is received.  Any non-required header present in such a   request may or may not have been processed.  Normally, the presence   of filled-out values in the header will be indication that the header   has been processed.  However, for cases when this is difficult to   determine, it is recommended to use a feature tag and the Require   header.  For this reason, it is usually easier if any parameters to   be retrieved are sent in the body, rather than using any header.Schulzrinne, et al.          Standards Track                   [Page 95]

RFC 7826                        RTSP 2.0                   December 2016   Example:     S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 431           User-Agent: PhonyClient/1.2           Session: OccldOFFq23KwjYpAnBbUr           Content-Length: 26           Content-Type: text/parameters           packets_received           jitter     C->S: RTSP/2.0 200 OK           CSeq: 431           Session: OccldOFFq23KwjYpAnBbUr           Server: PhonyServer/1.1           Date: Mon, 08 Mar 2010 13:43:23 GMT           Content-Length: 38           Content-Type: text/parameters           packets_received: 10           jitter: 0.383813.9.  SET_PARAMETER   This method requests the setting of the value of a parameter or a set   of parameters for a presentation or stream specified by the URI.  If   the Session header is present in a request, the value of a parameter   MUST be retrieved in the specified session context.  The method MAY   also be used without a message body.  It is the RECOMMENDED method to   be used in a request sent for the sole purpose of updating the keep-   alive timer.  If this request is successful, i.e., a 200 OK response   is received, then the keep-alive timer has been updated.  Any non-   required header present in such a request may or may not have been   processed.  To allow a client to determine if any such header has   been processed, it is necessary to use a feature tag and the Require   header.  Due to this reason it is RECOMMENDED that any parameters are   sent in the body rather than using any header.   When using a message body to list the parameter(s) desired to be set,   the Content-Type header (Section 18.19) is used to specify which   format the message body has.  If the receiver of the request is not   supporting the media type used for the message body, it SHALL respond   using the error code 415 (Unsupported Media Type).  For additional   considerations regarding message body negotiation, seeSection 9.3.   The responder to a SET_PARAMETER request MUST use the media type of   the request for the response.  For additional considerations   regarding message body negotiation, seeSection 9.3.Schulzrinne, et al.          Standards Track                   [Page 96]

RFC 7826                        RTSP 2.0                   December 2016   RTSP agents implementing support for responding to SET_PARAMETER   requests SHALL implement the text/parameters format (Appendix F).   This is to ensure that at least one known format for parameters is   implemented and, thus, prevent parameter format negotiation failure.   A request is RECOMMENDED to only contain a single parameter to allow   the client to determine why a particular request failed.  If the   request contains several parameters, the server MUST only act on the   request if all of the parameters can be set successfully.  A server   MUST allow a parameter to be set repeatedly to the same value, but it   MAY disallow changing parameter values.  If the receiver of the   request does not understand or cannot locate a parameter, error 451   (Parameter Not Understood) MUST be used.  When a parameter is not   allowed to change, the error code is 458 (Parameter Is Read-Only).   The response body MUST contain only the parameters that have errors.   Otherwise, a body MUST NOT be returned.  The response body MUST use   the media type of the request for the response.   Note: transport parameters for the media stream MUST only be set with   the SETUP command.      Restricting setting transport parameters to SETUP is for the      benefit of firewalls connected to border RTSP proxies.      The parameters are split in a fine-grained fashion so that there      can be more meaningful error indications.  However, it may make      sense to allow the setting of several parameters if an atomic      setting is desirable.  Imagine device control where the client      does not want the camera to pan unless it can also tilt to the      right angle at the same time.Schulzrinne, et al.          Standards Track                   [Page 97]

RFC 7826                        RTSP 2.0                   December 2016   Example:     C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 421           User-Agent: PhonyClient/1.2           Session: iixT43KLc           Date: Mon, 08 Mar 2010 14:45:04 GMT           Content-length: 20           Content-type: text/parameters           barparam: barstuff     S->C: RTSP/2.0 451 Parameter Not Understood           CSeq: 421           Session: iixT43KLc           Server: PhonyServer/1.0           Date: Mon, 08 Mar 2010 14:44:56 GMT           Content-length: 20           Content-type: text/parameters           barparam: barstuff13.10.  REDIRECT   The REDIRECT method is issued by a server to inform a client that the   service provided will be terminated and where a corresponding service   can be provided instead.  This may happen for different reasons.  One   is that the server is being administered such that it must stop   providing service.  Thus, the client is required to connect to   another server location to access the resource indicated by the   Request-URI.   The REDIRECT request SHALL contain a Terminate-Reason header   (Section 18.52) to inform the client of the reason for the request.   Additional parameters related to the reason may also be included.   The intention here is to allow a server administrator to do a   controlled shutdown of the RTSP server.  That requires sufficient   time to inform all entities having associated state with the server   and for them to perform a controlled migration from this server to a   fall-back server.   A REDIRECT request with a Session header has end-to-end (i.e.,   server-to-client) scope and applies only to the given session.  Any   intervening proxies SHOULD NOT disconnect the control channel while   there are other remaining end-to-end sessions.  The REQUIRED Location   header MUST contain a complete absolute URI pointing to the resource   to which the client SHOULD reconnect.  Specifically, the LocationSchulzrinne, et al.          Standards Track                   [Page 98]

RFC 7826                        RTSP 2.0                   December 2016   MUST NOT contain just the host and port.  A client may receive a   REDIRECT request with a Session header, if and only if, an end-to-end   session has been established.   A client may receive a REDIRECT request without a Session header at   any time when it has communication or a connection established with a   server.  The scope of such a request is limited to the next-hop   (i.e., the RTSP agent in direct communication with the server) and   applies to all sessions controlled, as well as the connection between   the next-hop RTSP agent and the server.  A REDIRECT request without a   Session header indicates that all sessions and pending requests being   managed via the connection MUST be redirected.  The Location header,   if included in such a request, SHOULD contain an absolute URI with   only the host address and the OPTIONAL port number of the server to   which the RTSP agent SHOULD reconnect.  Any intervening proxies   SHOULD do all of the following in the order listed:   1.  respond to the REDIRECT request   2.  disconnect the control channel from the requesting server   3.  connect to the server at the given host address   4.  pass the REDIRECT request to each applicable client (typically       those clients with an active session or an unanswered request)      Note: The proxy is responsible for accepting REDIRECT responses      from its clients; these responses MUST NOT be passed on to either      the original server or the redirected server.   A server that needs to terminate a session or all its sessions and   lacks an alternative server to redirect to, SHALL instead use   TEARDOWN requests.   When no Terminate-Reason "time" parameter is included in a REDIRECT   request, the client SHALL perform the redirection immediately and   return a response to the server.  The server shall consider the   session to be terminated and can free any associated state after it   receives the successful (2xx) response.  The server MAY close the   signaling connection upon receiving the response, and the client   SHOULD close the signaling connection after sending the 2xx response.   The exception to this is when the client has several sessions on the   server being managed by the given signaling connection.  In this   case, the client SHOULD close the connection when it has received and   responded to REDIRECT requests for all the sessions managed by the   signaling connection.Schulzrinne, et al.          Standards Track                   [Page 99]

RFC 7826                        RTSP 2.0                   December 2016   The Terminate-Reason header "time" parameter MAY be used to indicate   the wallclock time by which the redirection MUST have taken place.   To allow a client to determine that redirect time without being time   synchronized with the server, the server MUST include a Date header   in the request.  The client should have terminated the session and   closed the connection before the redirection time-line terminated.   The server MAY simply cease to provide service when the deadline time   has been reached, or it can issue a TEARDOWN requests to the   remaining sessions.   If the REDIRECT request times out following the rules inSection 10.4, the server MAY terminate the session or transport   connection that would be redirected by the request.  This is a   safeguard against misbehaving clients that refuse to respond to a   REDIRECT request.  This action removes any incentive of not   acknowledging the reception of a REDIRECT request.   After a REDIRECT request has been processed, a client that wants to   continue to receive media for the resource identified by the Request-   URI will have to establish a new session with the designated host.   If the URI given in the Location header is a valid resource URI, a   client SHOULD issue a DESCRIBE request for the URI.      Note: The media resource indicated by the Location header can be      identical, slightly different, or totally different.  This is the      reason why a new DESCRIBE request SHOULD be issued.   If the Location header contains only a host address, the client may   assume that the media on the new server is identical to the media on   the old server, i.e., all media configuration information from the   old session is still valid except for the host address.  However, the   usage of conditional SETUP using MTag identifiers is RECOMMENDED as a   means to verify the assumption.   This example request redirects traffic for this session to the new   server at the given absolute time:     S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 732           Location: rtsp://s2.example.com:8001/fizzle/foo           Terminate-Reason: Server-Admin ;time=19960213T143205Z           Session: uZ3ci0K+Ld-M           Date: Thu, 13 Feb 1996 14:30:43 GMT     C->S: RTSP/2.0 200 OK           CSeq: 732           User-Agent: PhonyClient/1.2           Session: uZ3ci0K+Ld-MSchulzrinne, et al.          Standards Track                  [Page 100]

RFC 7826                        RTSP 2.0                   December 201614.  Embedded (Interleaved) Binary Data   In order to fulfill certain requirements on the network side, e.g.,   in conjunction with network address translators that block RTP   traffic over UDP, it may be necessary to interleave RTSP messages and   media-stream data.  This interleaving should generally be avoided   unless necessary since it complicates client and server operation and   imposes additional overhead.  Also, head-of-line blocking may cause   problems.  Interleaved binary data SHOULD only be used if RTSP is   carried over TCP.  Interleaved data is not allowed inside RTSP   messages.   Stream data, such as RTP packets, is encapsulated by an ASCII dollar   sign (36 decimal) followed by a one-octet channel identifier and the   length of the encapsulated binary data as a binary, two-octet   unsigned integer in network octet order (Appendix B of [RFC791]).   The stream data follows immediately afterwards, without a CRLF, but   including the upper-layer protocol headers.  Each dollar sign block   MUST contain exactly one upper-layer protocol data unit, e.g., one   RTP packet.      Note that this mechanism does not support PDUs larger than 65535      octets, which matches the maximum payload size of regular, non-      jumbo IPv4 and IPv6 packets.  If the media delivery protocol      intended to be used has larger PDUs than that, a definition of a      PDU fragmentation mechanism will be required to support embedded      binary data.       0                   1                   2                   3       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+      | "$" = 36      | Channel ID    | Length in octets              |      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+      : Binary data (Length according to Length field)                :      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+             Figure 1: Embedded Interleaved Binary Data Format   The channel identifier is defined in the Transport header with the   interleaved parameter (Section 18.54).   When the transport choice is RTP, RTCP messages are also interleaved   by the server over the TCP connection.  The usage of RTCP messages is   indicated by including an interval containing a second channel in the   interleaved parameter of the Transport header (seeSection 18.54).   If RTCP is used, packets MUST be sent on the first available channelSchulzrinne, et al.          Standards Track                  [Page 101]

RFC 7826                        RTSP 2.0                   December 2016   that is higher than the RTP channel.  The channels are bidirectional,   using the same Channel ID in both directions; therefore, RTCP traffic   is sent on the second channel in both directions.      RTCP is sometimes needed for synchronization when two or more      streams are interleaved in such a fashion.  Also, this provides a      convenient way to tunnel RTP/RTCP packets through the RTSP      connection (TCP or TCP/TLS) when required by the network      configuration and to transfer them onto UDP when possible.     C->S: SETUP rtsp://example.com/bar.file RTSP/2.0           CSeq: 2           Transport: RTP/AVP/TCP;unicast;interleaved=0-1           Accept-Ranges: npt, smpte, clock           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 2           Date: Thu, 05 Jun 1997 18:57:18 GMT           Transport: RTP/AVP/TCP;unicast;interleaved=5-6           Session: OccldOFFq23KwjYpAnBbUr           Accept-Ranges: npt           Media-Properties: Random-Access=0.2, Immutable, Unlimited     C->S: PLAY rtsp://example.com/bar.file RTSP/2.0           CSeq: 3           Session: OccldOFFq23KwjYpAnBbUr           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 3           Session: OccldOFFq23KwjYpAnBbUr           Date: Thu, 05 Jun 1997 18:57:19 GMT           RTP-Info: url="rtsp://example.com/bar.file"             ssrc=0D12F123:seq=232433;rtptime=972948234           Range: npt=0-56.8           Seek-Style: RAP     S->C: $005{2 octet length}{"length" octets data, w/RTP header}     S->C: $005{2 octet length}{"length" octets data, w/RTP header}     S->C: $006{2 octet length}{"length" octets  RTCP packet}Schulzrinne, et al.          Standards Track                  [Page 102]

RFC 7826                        RTSP 2.0                   December 201615.  Proxies   RTSP Proxies are RTSP agents that are located in between a client and   a server.  A proxy can take on the roles of both client and server   depending on what it tries to accomplish.  RTSP proxies use two   transport-layer connections: one from the RTSP client to the RTSP   proxy and a second from the RTSP proxy to the RTSP server.  Proxies   are introduced for several different reasons; those listed below are   often combined.   Caching Proxy:  This type of proxy is used to reduce the workload on         servers and connections.  By caching the description and media         streams, i.e., the presentation, the proxy can serve a client         with content, but without requesting it from the server once it         has been cached and has not become stale.  SeeSection 16.         This type of proxy is also expected to understand RTSP endpoint         functionality, i.e., functionality identified in the Require         header in addition to what Proxy-Require demands.   Translator Proxy:  This type of proxy is used to ensure that an RTSP         client gets access to servers and content on an external         network or gets access by using content encodings not supported         by the client.  The proxy performs the necessary translation of         addresses, protocols, or encodings.  This type of proxy is         expected also to understand RTSP endpoint functionality, i.e.,         functionality identified in the Require header in addition to         what Proxy-Require demands.   Access Proxy:  This type of proxy is used to ensure that an RTSP         client gets access to servers on an external network.  Thus,         this proxy is placed on the border between two domains, e.g., a         private address space and the public Internet.  The proxy         performs the necessary translation, usually addresses.  This         type of proxy is required to redirect the media to itself or a         controlled gateway that performs the translation before the         media can reach the client.   Security Proxy:  This type of proxy is used to help facilitate         security functions around RTSP.  For example, in the case of a         firewalled network, the security proxy requests that the         necessary pinholes in the firewall are opened when a client in         the protected network wants to access media streams on the         external side.  This proxy can perform its function without         redirecting the media between the server and client.  However,         in deployments with private address spaces, this proxy is         likely to be combined with the access proxy.  The functionality         of this proxy is usually closely tied into understanding all         aspects of the media transport.Schulzrinne, et al.          Standards Track                  [Page 103]

RFC 7826                        RTSP 2.0                   December 2016   Auditing Proxy:  RTSP proxies can also provide network owners with a         logging and auditing point for RTSP sessions, e.g., for         corporations that track their employees usage of the network.         This type of proxy can perform its function without inserting         itself or any other node in the media transport.  This proxy         type can also accept unknown methods as it doesn't interfere         with the clients' requests.   All types of proxies can also be used when using secured   communication with TLS, as RTSP 2.0 allows the client to approve   certificate chains used for connection establishment from a proxy;   seeSection 19.3.2.  However, that trust model may not be suitable   for all types of deployment.  In those cases, the secured sessions do   bypass the proxies.   Access proxies SHOULD NOT be used in equipment like NATs and   firewalls that aren't expected to be regularly maintained, like home   or small office equipment.  In these cases, it is better to use the   NAT traversal procedures defined for RTSP 2.0 [RFC7825].  The reason   for these recommendations is that any extensions of RTSP resulting in   new media-transport protocols or profiles, new parameters, etc., may   fail in a proxy that isn't maintained.  This would impede RTSP's   future development and usage.15.1.  Proxies and Protocol Extensions   The existence of proxies must always be considered when developing   new RTSP extensions.  Most types of proxies will need to implement   any new method to operate correctly in the presence of that   extension.  New headers can be introduced and will not be blocked by   older proxies.  However, it is important to consider if this header   and its function are required to be understood by the proxy or if it   can be simply forwarded.  If the header needs to be understood, a   feature tag representing the functionality MUST be included in the   Proxy-Require header.  Below are guidelines for analysis whether the   header needs to be understood.  The Transport header and its   parameters are extensible, which requires handling rules for a proxy   in order to ensure a correct interpretation.Schulzrinne, et al.          Standards Track                  [Page 104]

RFC 7826                        RTSP 2.0                   December 2016   Whether or not a proxy needs to understand a header is not easy to   determine as they serve a broad variety of functions.  When   evaluating if a header needs to be understood, one can divide the   functionality into three main categories:   Media modifying:  The caching and translator proxies modify the      actual media and therefore need also to understand the request      directed to the server that affects how the media is rendered.      Thus, this type of proxy also needs to understand the server-side      functionality.   Transport modifying:  The access and the security proxy both need to      understand how the media transport is performed, either for      opening pinholes or translating the outer headers, e.g., IP and      UDP or TCP.   Non-modifying:  The audit proxy is special in that it does not modify      the messages in other ways than to insert the Via header.  That      makes it possible for this type to forward RTSP messages that      contain different types of unknown methods, headers, or header      parameters.   An extension has to be classified as mandatory to be implemented for   a proxy, if an extension has to be understood by a "Transport   modifying" type of proxy.15.2.  Multiplexing and Demultiplexing of Messages   RTSP proxies may have to multiplex several RTSP sessions from their   clients towards RTSP servers.  This requires that RTSP requests from   multiple clients be multiplexed onto a common connection for requests   outgoing to an RTSP server, and, on the way back, the responses be   demultiplexed from the server to per-client responses.  On the   protocol level, this requires that request and response messages be   handled in both directions, requiring that there be a mechanism to   correlate which request/response pair exchanged between proxy and   server is mapped to which client (or client request).   This multiplexing of requests and demultiplexing of responses is done   by using the CSeq header field.  The proxy has to rewrite the CSeq in   requests to the server and responses from the server and remember   which CSeq is mapped to which client.  The proxy also needs to ensure   that the order of the message related to each client is maintained.Section 18.20 defines the handling of how requests and responses are   rewritten.Schulzrinne, et al.          Standards Track                  [Page 105]

RFC 7826                        RTSP 2.0                   December 201616.  Caching   In HTTP, request/response pairs are cached.  RTSP differs   significantly in that respect.  Responses are not cacheable, with the   exception of the presentation description returned by DESCRIBE.   (Since the responses for anything but DESCRIBE and GET_PARAMETER do   not return any data, caching is not really an issue for these   requests.)  However, it is desirable for the continuous media data,   typically delivered out-of-band with respect to RTSP, to be cached,   as well as the session description.   On receiving a SETUP or PLAY request, a proxy ascertains whether it   has an up-to-date copy of the continuous media content and its   description.  It can determine whether the copy is up to date by   issuing a SETUP or DESCRIBE request, respectively, and comparing the   Last-Modified header with that of the cached copy.  If the copy is   not up to date, it modifies the SETUP transport parameters as   appropriate and forwards the request to the origin server.   Subsequent control commands such as PLAY or PAUSE then pass the proxy   unmodified.  The proxy delivers the continuous media data to the   client, while possibly making a local copy for later reuse.  The   exact allowed behavior of the cache is given by the cache-response   directives described inSection 18.11.  A cache MUST answer any   DESCRIBE requests if it is currently serving the stream to the   requester, as it is possible that low-level details of the stream   description may have changed on the origin server.   Note that an RTSP cache is of the "cut-through" variety.  Rather than   retrieving the whole resource from the origin server, the cache   simply copies the streaming data as it passes by on its way to the   client.  Thus, it does not introduce additional latency.   To the client, an RTSP proxy cache appears like a regular media   server.  To the media origin server, an RTSP proxy cache appears like   a client.  Just as an HTTP cache has to store the content type,   content language, and so on for the objects it caches, a media cache   has to store the presentation description.  Typically, a cache   eliminates all transport references (e.g., multicast information)   from the presentation description, since these are independent of the   data delivery from the cache to the client.  Information on the   encodings remains the same.  If the cache is able to translate the   cached media data, it would create a new presentation description   with all the encoding possibilities it can offer.Schulzrinne, et al.          Standards Track                  [Page 106]

RFC 7826                        RTSP 2.0                   December 201616.1.  Validation Model   When a cache has a stale entry that it would like to use as a   response to a client's request, it first has to check with the origin   server (or possibly an intermediate cache with a fresh response) to   see if its cached entry is still usable.  This is called "validating"   the cache entry.  To avoid having to pay the overhead of   retransmitting the full response if the cached entry is good, and at   the same time avoiding having to pay the overhead of an extra round   trip if the cached entry is invalid, RTSP supports the use of   conditional methods.   The key protocol features for supporting conditional methods are   those concerned with "cache validators."  When an origin server   generates a full response, it attaches some sort of validator to it,   which is kept with the cache entry.  When a client (user agent or   proxy cache) makes a conditional request for a resource for which it   has a cache entry, it includes the associated validator in the   request.   The server then checks that validator against the current validator   for the requested resource, and, if they match (seeSection 16.1.3),   it responds with a special status code (usually, 304 (Not Modified))   and no message body.  Otherwise, it returns a full response   (including message body).  Thus, avoiding transmitting the full   response if the validator matches and avoiding an extra round trip if   it does not match.   In RTSP, a conditional request looks exactly the same as a normal   request for the same resource, except that it carries a special   header (which includes the validator) that implicitly turns the   method (usually DESCRIBE or SETUP) into a conditional.   The protocol includes both positive and negative senses of cache-   validating conditions.  That is, it is possible to request that a   method be performed either if and only if a validator matches or if   and only if no validators match.      Note: a response that lacks a validator may still be cached, and      served from cache until it expires, unless this is explicitly      prohibited by a cache directive (seeSection 18.11).  However, a      cache cannot perform a conditional retrieval if it does not have a      validator for the resource, which means it will not be refreshable      after it expires.Schulzrinne, et al.          Standards Track                  [Page 107]

RFC 7826                        RTSP 2.0                   December 2016   Media streams that are being adapted based on the transport capacity   between the server and the cache make caching more difficult.  A   server needs to consider how it views the caching of media streams   that it adapts and potentially instruct any caches not to cache such   streams.16.1.1.  Last-Modified Dates   The Last-Modified header (Section 18.27) value is often used as a   cache validator.  In simple terms, a cache entry is considered to be   valid if the cache entry was created after the Last-Modified time.16.1.2.  Message Body Tag Cache Validators   The MTag response-header field-value, a message body tag, provides   for an "opaque" cache validator.  This might allow more reliable   validation in situations where it is inconvenient to store   modification dates, where the one-second resolution of RTSP-date   values is not sufficient, or where the origin server wishes to avoid   certain paradoxes that might arise from the use of modification   dates.   Message body tags are described inSection 4.616.1.3.  Weak and Strong Validators   Since both origin servers and caches will compare two validators to   decide if they represent the same or different entities, one normally   would expect that if the message body (i.e., the presentation   description) or any associated message body headers changes in any   way, then the associated validator would change as well.  If this is   true, then this validator is a "strong validator".  The Message body   (i.e., the presentation description) or any associated message body   headers is named an entity for a better understanding.   However, there might be cases when a server prefers to change the   validator only on semantically significant changes and not when   insignificant aspects of the entity change.  A validator that does   not always change when the resource changes is a "weak validator".   Message body tags are normally strong validators, but the protocol   provides a mechanism to tag a message body tag as "weak".  One can   think of a strong validator as one that changes whenever the bits of   an entity changes, while a weak value changes whenever the meaning of   an entity changes.  Alternatively, one can think of a strong   validator as part of an identifier for a specific entity, while a   weak validator is part of an identifier for a set of semantically   equivalent entities.Schulzrinne, et al.          Standards Track                  [Page 108]

RFC 7826                        RTSP 2.0                   December 2016      Note: One example of a strong validator is an integer that is      incremented in stable storage every time an entity is changed.      An entity's modification time, if represented with one-second      resolution, could be a weak validator, since it is possible that      the resource might be modified twice during a single second.      Support for weak validators is optional.  However, weak validators      allow for more efficient caching of equivalent objects.   A "use" of a validator is either when a client generates a request   and includes the validator in a validating header field or when a   server compares two validators.   Strong validators are usable in any context.  Weak validators are   only usable in contexts that do not depend on exact equality of an   entity.  For example, either kind is usable for a conditional   DESCRIBE of a full entity.  However, only a strong validator is   usable for a subrange retrieval, since otherwise the client might end   up with an internally inconsistent entity.   Clients MAY issue DESCRIBE requests with either weak or strong   validators.  Clients MUST NOT use weak validators in other forms of   requests.   The only function that RTSP defines on validators is comparison.   There are two validator comparison functions, depending on whether or   not the comparison context allows the use of weak validators:   o  The strong comparison function: in order to be considered equal,      both validators MUST be identical in every way, and both MUST NOT      be weak.   o  The weak comparison function: in order to be considered equal,      both validators MUST be identical in every way, but either or both      of them MAY be tagged as "weak" without affecting the result.   A message body tag is strong unless it is explicitly tagged as weak.   A Last-Modified time, when used as a validator in a request, is   implicitly weak unless it is possible to deduce that it is strong,   using the following rules:   o  The validator is being compared by an origin server to the actual      current validator for the entity and,Schulzrinne, et al.          Standards Track                  [Page 109]

RFC 7826                        RTSP 2.0                   December 2016   o  That origin server reliably knows that the associated entity did      not change more than once during the second covered by the      presented validator.   OR   o  The validator is about to be used by a client in an If-Modified-      Since, because the client has a cache entry for the associated      entity, and   o  That cache entry includes a Date value, which gives the time when      the origin server sent the original response, and   o  The presented Last-Modified time is at least 60 seconds before the      Date value.   OR   o  The validator is being compared by an intermediate cache to the      validator stored in its cache entry for the entity, and   o  That cache entry includes a Date value, which gives the time when      the origin server sent the original response, and   o  The presented Last-Modified time is at least 60 seconds before the      Date value.   This method relies on the fact that if two different responses were   sent by the origin server during the same second, but both had the   same Last-Modified time, then at least one of those responses would   have a Date value equal to its Last-Modified time.  The arbitrary   60-second limit guards against the possibility that the Date and   Last-Modified values are generated from different clocks or at   somewhat different times during the preparation of the response.  An   implementation MAY use a value larger than 60 seconds, if it is   believed that 60 seconds is too short.   If a client wishes to perform a subrange retrieval on a value for   which it has only a Last-Modified time and no opaque validator, it   MAY do this only if the Last-Modified time is strong in the sense   described here.16.1.4.  Rules for When to Use Message Body Tags and Last-Modified Dates   This document adopts a set of rules and recommendations for origin   servers, clients, and caches regarding when various validator types   ought to be used, and for what purposes.Schulzrinne, et al.          Standards Track                  [Page 110]

RFC 7826                        RTSP 2.0                   December 2016   RTSP origin servers:   o  SHOULD send a message body tag validator unless it is not feasible      to generate one.   o  MAY send a weak message body tag instead of a strong message body      tag, if performance considerations support the use of weak message      body tags, or if it is unfeasible to send a strong message body      tag.   o  SHOULD send a Last-Modified value if it is feasible to send one,      unless the risk of a breakdown in semantic transparency that could      result from using this date in an If-Modified-Since header would      lead to serious problems.   In other words, the preferred behavior for an RTSP origin server is   to send both a strong message body tag and a Last-Modified value.   In order to be legal, a strong message body tag MUST change whenever   the associated entity value changes in any way.  A weak message body   tag SHOULD change whenever the associated entity changes in a   semantically significant way.      Note: in order to provide semantically transparent caching, an      origin server MUST avoid reusing a specific strong message body      tag value for two different entities or reusing a specific weak      message body tag value for two semantically different entities.      Cache entries might persist for arbitrarily long periods,      regardless of expiration times, so it might be inappropriate to      expect that a cache will never again attempt to validate an entry      using a validator that it obtained at some point in the past.   RTSP clients:   o  If a message body tag has been provided by the origin server, MUST      use that message body tag in any cache-conditional request (using      If-Match or If-None-Match).   o  If only a Last-Modified value has been provided by the origin      server, SHOULD use that value in non-subrange cache-conditional      requests (using If-Modified-Since).   o  If both a message body tag and a Last-Modified value have been      provided by the origin server, SHOULD use both validators in      cache-conditional requests.   An RTSP origin server, upon receiving a conditional request that   includes both a Last-Modified date (e.g., in an If-Modified-Since   header) and one or more message body tags (e.g., in an If-Match,Schulzrinne, et al.          Standards Track                  [Page 111]

RFC 7826                        RTSP 2.0                   December 2016   If-None-Match, or If-Range header field) as cache validators, MUST   NOT return a response status of 304 (Not Modified) unless doing so is   consistent with all of the conditional header fields in the request.      Note: The general principle behind these rules is that RTSP      servers and clients should transmit as much non-redundant      information as is available in their responses and requests.  RTSP      systems receiving this information will make the most conservative      assumptions about the validators they receive.16.1.5.  Non-validating Conditionals   The principle behind message body tags is that only the service   author knows the semantics of a resource well enough to select an   appropriate cache validation mechanism, and the specification of any   validator comparison function more complex than octet equality would   open up a can of worms.  Thus, comparisons of any other headers are   never used for purposes of validating a cache entry.16.2.  Invalidation after Updates or Deletions   The effect of certain methods performed on a resource at the origin   server might cause one or more existing cache entries to become non-   transparently invalid.  That is, although they might continue to be   "fresh," they do not accurately reflect what the origin server would   return for a new request on that resource.   There is no way for RTSP to guarantee that all such cache entries are   marked invalid.  For example, the request that caused the change at   the origin server might not have gone through the proxy where a cache   entry is stored.  However, several rules help reduce the likelihood   of erroneous behavior.   In this section, the phrase "invalidate an entity" means that the   cache will either remove all instances of that entity from its   storage or mark these as "invalid" and in need of a mandatory   revalidation before they can be returned in response to a subsequent   request.   Some RTSP methods MUST cause a cache to invalidate an entity.  This   is either the entity referred to by the Request-URI or by the   Location or Content-Location headers (if present).  These methods   are:   o  DESCRIBE   o  SETUPSchulzrinne, et al.          Standards Track                  [Page 112]

RFC 7826                        RTSP 2.0                   December 2016   In order to prevent DoS attacks, an invalidation based on the URI in   a Location or Content-Location header MUST only be performed if the   host part is the same as in the Request-URI.   A cache that passes through requests for methods it does not   understand SHOULD invalidate any entities referred to by the Request-   URI.17.  Status Code Definitions   Where applicable, HTTP status codes (seeSection 6 of [RFC7231]) are   reused.  See Table 4 inSection 8.1 for a listing of which status   codes may be returned by which requests.  All error messages, 4xx and   5xx, MAY return a body containing further information about the   error.17.1.  Informational 1xx17.1.1.  100 Continue   The requesting agent SHOULD continue with its request.  This interim   response is used to inform the requesting agent that the initial part   of the request has been received and has not yet been rejected by the   responding agent.  The requesting agent SHOULD continue by sending   the remainder of the request or, if the request has already been   completed, continue to wait for a final response (seeSection 10.4).   The responding agent MUST send a final response after the request has   been completed.17.2.  Success 2xx   This class of status code indicates that the agent's request was   successfully received, understood, and accepted.17.2.1.  200 OK   The request has succeeded.  The information returned with the   response is dependent on the method used in the request.17.3.  Redirection 3xx   The notation "3xx" indicates response codes from 300 to 399 inclusive   that are meant for redirection.  We use the notation "3rr" to   indicate all 3xx codes used for redirection, i.e., excluding 304.   The 304 response code appears here, rather than a 2xx response code,   which would have been appropriate; 304 has also been used in RTSP 1.0   [RFC2326].Schulzrinne, et al.          Standards Track                  [Page 113]

RFC 7826                        RTSP 2.0                   December 2016   Within RTSP, redirection may be used for load-balancing or   redirecting stream requests to a server topologically closer to the   agent.  Mechanisms to determine topological proximity are beyond the   scope of this specification.   A 3rr code MAY be used to respond to any request.  The Location   header MUST be included in any 3rr response.  It is RECOMMENDED that   they are used if necessary before a session is established, i.e., in   response to DESCRIBE or SETUP.  However, in cases where a server is   not able to send a REDIRECT request to the agent, the server MAY need   to resort to using 3rr responses to inform an agent with an   established session about the need for redirecting the session.  If a   3rr response is received for a request in relation to an established   session, the agent SHOULD send a TEARDOWN request for the session and   MAY reestablish the session using the resource indicated by the   Location.   If the Location header is used in a response, it MUST contain an   absolute URI pointing out the media resource the agent is redirected   to; the URI MUST NOT only contain the hostname.   In the event that an unknown 3rr status code is received, the agent   SHOULD behave as if a 302 response code had been received   (Section 17.3.3).17.3.1.  300   The 300 response code is not used in RTSP 2.0.17.3.2.  301 Moved Permanently   The requested resource is moved permanently and resides now at the   URI given by the Location header.  The user agent SHOULD redirect   automatically to the given URI.  This response MUST NOT contain a   message body.  The Location header MUST be included in the response.17.3.3.  302 Found   The requested resource resides temporarily at the URI given by the   Location header.  This response is intended to be used for many types   of temporary redirects, e.g., load balancing.  It is RECOMMENDED that   the server set the reason phrase to something more meaningful than   "Found" in these cases.  The Location header MUST be included in the   response.  The user agent SHOULD redirect automatically to the given   URI.  This response MUST NOT contain a message body.Schulzrinne, et al.          Standards Track                  [Page 114]

RFC 7826                        RTSP 2.0                   December 2016   This example shows a client being redirected to a different server:     C->S: SETUP rtsp://example.com/fizzle/foo RTSP/2.0           CSeq: 2           Transport: RTP/AVP/TCP;unicast;interleaved=0-1           Accept-Ranges: npt, smpte, clock           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 302 Try Other Server           CSeq: 2           Location: rtsp://s2.example.com:8001/fizzle/foo17.3.4.  303 See Other   This status code MUST NOT be used in RTSP 2.0.  However, it was   allowed in RTSP 1.0 [RFC2326].17.3.5.  304 Not Modified   If the agent has performed a conditional DESCRIBE or SETUP (see   Sections18.25 and18.26) and the requested resource has not been   modified, the server SHOULD send a 304 response.  This response MUST   NOT contain a message body.   The response MUST include the following header fields:   o  Date   o  MTag or Content-Location, if the headers would have been sent in a      200 response to the same request.   o  Expires and Cache-Control if the field-value might differ from      that sent in any previous response for the same variant.   This response is independent for the DESCRIBE and SETUP requests.   That is, a 304 response to DESCRIBE does NOT imply that the resource   content is unchanged (only the session description) and a 304   response to SETUP does NOT imply that the resource description is   unchanged.  The MTag and If-Match header (Section 18.24) may be used   to link the DESCRIBE and SETUP in this manner.17.3.6.  305 Use Proxy   The requested resource MUST be accessed through the proxy given by   the Location header that MUST be included.  The Location header   field-value gives the URI of the proxy.  The recipient is expected to   repeat this single request via the proxy. 305 responses MUST only be   generated by origin servers.Schulzrinne, et al.          Standards Track                  [Page 115]

RFC 7826                        RTSP 2.0                   December 201617.4.  Client Error 4xx17.4.1.  400 Bad Request   The request could not be understood by the agent due to malformed   syntax.  The agent SHOULD NOT repeat the request without   modifications.  If the request does not have a CSeq header, the agent   MUST NOT include a CSeq in the response.17.4.2.  401 Unauthorized   The request requires user authentication using the HTTP   authentication mechanism [RFC7235].  The usage of the error code is   defined in [RFC7235] and any applicable HTTP authentication scheme,   such as Digest [RFC7616].  The response is to include a WWW-   Authenticate header (Section 18.58) field containing a challenge   applicable to the requested resource.  The agent can repeat the   request with a suitable Authorization header field.  If the request   already included authorization credentials, then the 401 response   indicates that authorization has been refused for those credentials.   If the 401 response contains the same challenge as the prior   response, and the user agent has already attempted authentication at   least once, then the user SHOULD be presented the message body that   was given in the response, since that message body might include   relevant diagnostic information.17.4.3.  402 Payment Required   This code is reserved for future use.17.4.4.  403 Forbidden   The agent understood the request, but is refusing to fulfill it.   Authorization will not help, and the request SHOULD NOT be repeated.   If the agent wishes to make public why the request has not been   fulfilled, it SHOULD describe the reason for the refusal in the   message body.  If the agent does not wish to make this information   available to the agent, the status code 404 (Not Found) can be used   instead.17.4.5.  404 Not Found   The agent has not found anything matching the Request-URI.  No   indication is given of whether the condition is temporary or   permanent.  The 410 (Gone) status code SHOULD be used if the agent   knows, through some internally configurable mechanism, that an old   resource is permanently unavailable and has no forwarding address.Schulzrinne, et al.          Standards Track                  [Page 116]

RFC 7826                        RTSP 2.0                   December 2016   This status code is commonly used when the agent does not wish to   reveal exactly why the request has been refused, or when no other   response is applicable.17.4.6.  405 Method Not Allowed   The method specified in the request is not allowed for the resource   identified by the Request-URI.  The response MUST include an Allow   header containing a list of valid methods for the requested resource.   This status code is also to be used if a request attempts to use a   method not indicated during SETUP.17.4.7.  406 Not Acceptable   The resource identified by the request is only capable of generating   response message bodies that have content characteristics not   acceptable according to the Accept headers sent in the request.   The response SHOULD include a message body containing a list of   available message body characteristics and location(s) from which the   user or user agent can choose the one most appropriate.  The message   body format is specified by the media type given in the Content-Type   header field.  Depending upon the format and the capabilities of the   user agent, selection of the most appropriate choice MAY be performed   automatically.  However, this specification does not define any   standard for such automatic selection.   If the response could be unacceptable, a user agent SHOULD   temporarily stop receipt of more data and query the user for a   decision on further actions.17.4.8.  407 Proxy Authentication Required   This code is similar to 401 (Unauthorized) (Section 17.4.2), but it   indicates that the client must first authenticate itself with the   proxy.  The usage of this error code is defined in [RFC7235] and any   applicable HTTP authentication scheme, such as Digest [RFC7616].  The   proxy MUST return a Proxy-Authenticate header field (Section 18.34)   containing a challenge applicable to the proxy for the requested   resource.17.4.9.  408 Request Timeout   The agent did not produce a request within the time that the agent   was prepared to wait.  The agent MAY repeat the request without   modifications at any later time.Schulzrinne, et al.          Standards Track                  [Page 117]

RFC 7826                        RTSP 2.0                   December 201617.4.10.  410 Gone   The requested resource is no longer available at the server and the   forwarding address is not known.  This condition is expected to be   considered permanent.  If the server does not know, or has no   facility to determine, whether or not the condition is permanent, the   status code 404 (Not Found) SHOULD be used instead.  This response is   cacheable unless indicated otherwise.   The 410 response is primarily intended to assist the task of   repository maintenance by notifying the recipient that the resource   is intentionally unavailable and that the server owners desire that   remote links to that resource be removed.  Such an event is common   for limited-time, promotional services and for resources belonging to   individuals no longer working at the server's site.  It is not   necessary to mark all permanently unavailable resources as "gone" or   to keep the mark for any length of time -- that is left to the   discretion of the owner of the server.17.4.11.  412 Precondition Failed   The precondition given in one or more of the 'if-' request-header   fields evaluated to false when it was tested on the agent.  See these   sections for the 'if-' headers: If-MatchSection 18.24, If-Modified-   SinceSection 18.25, and If-None-MatchSection 18.26.  This response   code allows the agent to place preconditions on the current resource   meta-information (header field data) and, thus, prevent the requested   method from being applied to a resource other than the one intended.17.4.12.  413 Request Message Body Too Large   The agent is refusing to process a request because the request   message body is larger than the agent is willing or able to process.   The agent MAY close the connection to prevent the requesting agent   from continuing the request.   If the condition is temporary, the agent SHOULD include a Retry-After   header field to indicate that it is temporary and after what time the   requesting agent MAY try again.17.4.13.  414 Request-URI Too Long   The responding agent is refusing to service the request because the   Request-URI is longer than the agent is willing to interpret.  This   rare condition is only likely to occur when an agent has used a   request with long query information, when the agent has descended   into a URI "black hole" of redirection (e.g., a redirected URI prefix   that points to a suffix of itself), or when the agent is under attackSchulzrinne, et al.          Standards Track                  [Page 118]

RFC 7826                        RTSP 2.0                   December 2016   by an agent attempting to exploit security holes present in some   agents using fixed-length buffers for reading or manipulating the   Request-URI.17.4.14.  415 Unsupported Media Type   The server is refusing to service the request because the message   body of the request is in a format not supported by the requested   resource for the requested method.17.4.15.  451 Parameter Not Understood   The recipient of the request does not support one or more parameters   contained in the request.  When returning this error message the   agent SHOULD return a message body containing the offending   parameter(s).17.4.16.  452 Illegal Conference Identifier   This status code MUST NOT be used in RTSP 2.0.  However, it was   allowed in RTSP 1.0 [RFC2326].17.4.17.  453 Not Enough Bandwidth   The request was refused because there was insufficient bandwidth.   This may, for example, be the result of a resource reservation   failure.17.4.18.  454 Session Not Found   The RTSP session identifier in the Session header is missing, is   invalid, or has timed out.17.4.19.  455 Method Not Valid in This State   The agent cannot process this request in its current state.  The   response MUST contain an Allow header to make error recovery   possible.17.4.20.  456 Header Field Not Valid for Resource   The targeted agent could not act on a required request-header.  For   example, if PLAY request contains the Range header field but the   stream does not allow seeking.  This error message may also be used   for specifying when the time format in Range is impossible for the   resource.  In that case, the Accept-Ranges header MUST be returned to   inform the agent of which formats are allowed.Schulzrinne, et al.          Standards Track                  [Page 119]

RFC 7826                        RTSP 2.0                   December 201617.4.21.  457 Invalid Range   The Range value given is out of bounds, e.g., beyond the end of the   presentation.17.4.22.  458 Parameter Is Read-Only   The parameter to be set by SET_PARAMETER can be read but not   modified.  When returning this error message, the sender SHOULD   return a message body containing the offending parameter(s).17.4.23.  459 Aggregate Operation Not Allowed   The requested method may not be applied on the URI in question since   it is an aggregate (presentation) URI.  The method may be applied on   a media URI.17.4.24.  460 Only Aggregate Operation Allowed   The requested method may not be applied on the URI in question since   it is not an aggregate control (presentation) URI.  The method may be   applied on the aggregate control URI.17.4.25.  461 Unsupported Transport   The Transport field did not contain a supported transport   specification.17.4.26.  462 Destination Unreachable   The data transmission channel could not be established because the   agent address could not be reached.  This error will most likely be   the result of an agent attempt to place an invalid dest_addr   parameter in the Transport field.17.4.27.  463 Destination Prohibited   The data transmission channel was not established because the server   prohibited access to the agent address.  This error is most likely   the result of an agent attempt to redirect media traffic to another   destination with a dest_addr parameter in the Transport header.Schulzrinne, et al.          Standards Track                  [Page 120]

RFC 7826                        RTSP 2.0                   December 201617.4.28.  464 Data Transport Not Ready Yet   The data transmission channel to the media destination is not yet   ready for carrying data.  However, the responding agent still expects   that the data transmission channel will be established at some point   in time.  Note, however, that this may result in a permanent failure   like 462 (Destination Unreachable).   An example of when this error may occur is in the case in which a   client sends a PLAY request to a server prior to ensuring that the   TCP connections negotiated for carrying media data were successfully   established (in violation of this specification).  The server would   use this error code to indicate that the requested action could not   be performed due to the failure of completing the connection   establishment.17.4.29.  465 Notification Reason Unknown   This indicates that the client has received a PLAY_NOTIFY   (Section 13.5) with a Notify-Reason header (Section 18.32) unknown to   the client.17.4.30.  466 Key Management Error   This indicates that there has been an error in a Key Management   function used in conjunction with a request.  For example, usage of   Multimedia Internet KEYing (MIKEY) [RFC3830] according toAppendix C.1.4.1 may result in this error.17.4.31.  470 Connection Authorization Required   The secured connection attempt needs user or client authorization   before proceeding.  The next hop's certificate is included in this   response in the Accept-Credentials header.17.4.32.  471 Connection Credentials Not Accepted   When performing a secure connection over multiple connections, an   intermediary has refused to connect to the next hop and carry out the   request due to unacceptable credentials for the used policy.17.4.33.  472 Failure to Establish Secure Connection   A proxy fails to establish a secure connection to the next-hop RTSP   agent.  This is primarily caused by a fatal failure at the TLS   handshake, for example, due to the agent not accepting any cipher   suites.Schulzrinne, et al.          Standards Track                  [Page 121]

RFC 7826                        RTSP 2.0                   December 201617.5.  Server Error 5xx   Response status codes beginning with the digit "5" indicate cases in   which the server is aware that it has erred or is incapable of   performing the request.  The server SHOULD include a message body   containing an explanation of the error situation and whether it is a   temporary or permanent condition.  User agents SHOULD display any   included message body to the user.  These response codes are   applicable to any request method.17.5.1.  500 Internal Server Error   The agent encountered an unexpected condition that prevented it from   fulfilling the request.17.5.2.  501 Not Implemented   The agent does not support the functionality required to fulfill the   request.  This is the appropriate response when the agent does not   recognize the request method and is not capable of supporting it for   any resource.17.5.3.  502 Bad Gateway   The agent, while acting as a gateway or proxy, received an invalid   response from the upstream agent it accessed in attempting to fulfill   the request.17.5.4.  503 Service Unavailable   The server is currently unable to handle the request due to a   temporary overloading or maintenance of the server.  The implication   is that this is a temporary condition that will be alleviated after   some delay.  If known, the length of the delay MAY be indicated in a   Retry-After header.  If no Retry-After is given, the agent SHOULD   handle the response as it would for a 500 response.  The agent MUST   honor the length, if given, in the Retry-After header.         Note: The existence of the 503 status code does not imply that         a server must use it when becoming overloaded.  Some servers         may wish to simply refuse the transport connection.   The response scope is dependent on the request.  If the request was   in relation to an existing RTSP session, the scope of the overload   response is to this individual RTSP session.  If the request was not   session specific or intended to form an RTSP session, it applies to   the RTSP server identified by the hostname in the Request-URI.Schulzrinne, et al.          Standards Track                  [Page 122]

RFC 7826                        RTSP 2.0                   December 201617.5.5.  504 Gateway Timeout   The agent, while acting as a proxy, did not receive a timely response   from the upstream agent specified by the URI or some other auxiliary   server (e.g., DNS) that it needed to access in attempting to complete   the request.17.5.6.  505 RTSP Version Not Supported   The agent does not support, or refuses to support, the RTSP version   that was used in the request message.  The agent is indicating that   it is unable or unwilling to complete the request using the same   major version as the agent other than with this error message.  The   response SHOULD contain a message body describing why that version is   not supported and what other protocols are supported by that agent.17.5.7.  551 Option Not Supported   A feature tag given in the Require or the Proxy-Require fields was   not supported.  The Unsupported header MUST be returned stating the   feature for which there is no support.17.5.8.  553 Proxy Unavailable   The proxy is currently unable to handle the request due to a   temporary overloading or maintenance of the proxy.  The implication   is that this is a temporary condition that will be alleviated after   some delay.  If known, the length of the delay MAY be indicated in a   Retry-After header.  If no Retry-After is given, the agent SHOULD   handle the response as it would for a 500 response.  The agent MUST   honor the length, if given in the Retry-After header.         Note: The existence of the 553 status code does not imply that         a proxy must use it when becoming overloaded.  Some proxies may         wish to simply refuse the connection.   The response scope is dependent on the Request.  If the request was   in relation to an existing RTSP session, the scope of the overload   response is to this individual RTSP session.  If the request was non-   session specific or intended to form an RTSP session, it applies to   all such requests to this proxy.Schulzrinne, et al.          Standards Track                  [Page 123]

RFC 7826                        RTSP 2.0                   December 201618.  Header Field Definitions       +---------------+----------------+--------+---------+------+       | method        | direction      | object | acronym | Body |       +---------------+----------------+--------+---------+------+       | DESCRIBE      | C -> S         | P,S    | DES     | r    |       |               |                |        |         |      |       | GET_PARAMETER | C -> S, S -> C | P,S    | GPR     | R,r  |       |               |                |        |         |      |       | OPTIONS       | C -> S, S -> C | P,S    | OPT     |      |       |               |                |        |         |      |       | PAUSE         | C -> S         | P,S    | PSE     |      |       |               |                |        |         |      |       | PLAY          | C -> S         | P,S    | PLY     |      |       |               |                |        |         |      |       | PLAY_NOTIFY   | S -> C         | P,S    | PNY     | R    |       |               |                |        |         |      |       | REDIRECT      | S -> C         | P,S    | RDR     |      |       |               |                |        |         |      |       | SETUP         | C -> S         | S      | STP     |      |       |               |                |        |         |      |       | SET_PARAMETER | C -> S, S -> C | P,S    | SPR     | R,r  |       |               |                |        |         |      |       | TEARDOWN      | C -> S         | P,S    | TRD     |      |       |               |                |        |         |      |       |               | S -> C         | P      | TRD     |      |       +---------------+----------------+--------+---------+------+   This table is an overview of RTSP methods, their direction, and what   objects (P: presentation, S: stream) they operate on.  "Body" denotes     if a method is allowed to carry body and in which direction; R =    request, r=response.  Note: All error messages for statuses 4xx and                     5xx are allowed to carry a body.                     Table 8: Overview of RTSP Methods   The general syntax for header fields is covered inSection 5.2.  This   section lists the full set of header fields along with notes on   meaning and usage.  The syntax definitions for header fields are   present inSection 20.2.3.  Examples of each header field are given.   Information about header fields in relation to methods and proxy   processing is summarized in Figures 2, 3, 4, and 5.Schulzrinne, et al.          Standards Track                  [Page 124]

RFC 7826                        RTSP 2.0                   December 2016   The "where" column describes the request and response types in which   the header field can be used.  Values in this column are:   R:                header field may only appear in requests;   r:                header field may only appear in responses;   2xx, 4xx, etc.:   numerical value or range indicates response codes                     with which the header field can be used;   c:                header field is copied from the request to the                     response.   G:                header field is a general-header and may be present                     in both requests and responses.   Note: General headers do not always use the "G" value in the "where"   column.  This is due to differences when the header may be applied in   requests compared to responses.  When such differences exist, they   are expressed using two different rows: one with "where" being "R"   and one with it being "r".   The "proxy" column describes the operations a proxy may perform on a   header field.  An empty proxy column indicates that the proxy MUST   NOT make any changes to that header, all allowed operations are   explicitly stated:   a:    A proxy can add or concatenate the header field if not present.   m:    A proxy can modify an existing header field value.   d:    A proxy can delete a header field-value.   r:    A proxy needs to be able to read the header field; thus, this         header field cannot be encrypted.   The rest of the columns relate to the presence of a header field in a   method.  The method names when abbreviated, are according to Table 8:   c:    Conditional; requirements on the header field depend on the         context of the message.   m:    The header field is mandatory.   m*:   The header field SHOULD be sent, but agents need to be prepared         to receive messages without that header field.   o:    The header field is optional.Schulzrinne, et al.          Standards Track                  [Page 125]

RFC 7826                        RTSP 2.0                   December 2016   *:    The header field MUST be present if the message body is not         empty.  See Sections18.17,18.19 and5.3 for details.   -:    The header field is not applicable.   "Optional" means that an agent MAY include the header field in a   request or response.  The agent behavior when receiving such headers   varies; for some, it may ignore the header field.  In other cases, it   is a request to process the header.  This is regulated by the method   and header descriptions.  Examples of headers that require processing   are the Require and Proxy-Require header fields discussed in Sections   18.43 and 18.37.  A "mandatory" header field MUST be present in a   request, and it MUST be understood by the agent receiving the   request.  A mandatory response-header field MUST be present in the   response, and the header field MUST be understood by the processing   the response.  "Not applicable" means that the header field MUST NOT   be present in a request.  If one is placed in a request by mistake,   it MUST be ignored by the agent receiving the request.  Similarly, a   header field labeled "not applicable" for a response means that the   agent MUST NOT place the header field in the response, and the agent   MUST ignore the header field in the response.   An RTSP agent MUST ignore extension headers that are not understood.   The From and Location header fields contain a URI.  If the URI   contains a comma (') or semicolon (;), the URI MUST be enclosed in   double quotes (").  Any URI parameters are contained within these   quotes.  If the URI is not enclosed in double quotes, any semicolon-   delimited parameters are header-parameters, not URI parameters.Schulzrinne, et al.          Standards Track                  [Page 126]

RFC 7826                        RTSP 2.0                   December 2016   +-------------------+------+------+----+----+-----+-----+-----+-----+   | Header            |Where |Proxy |DES | OPT| STP | PLY | PSE | TRD |   +-------------------+------+------+----+----+-----+-----+-----+-----+   | Accept            | R    |      | o  | -  | -   | -   | -   | -   |   | Accept-           | R    | rm   | o  | o  | o   | o   | o   | o   |   | Credentials       |      |      |    |    |     |     |     |     |   | Accept-Encoding   | R    | r    | o  | -  | -   | -   | -   | -   |   | Accept-Language   | R    | r    | o  | -  | -   | -   | -   | -   |   | Accept-Ranges     | G    | r    | -  | -  | m   | -   | -   | -   |   | Accept-Ranges     | 456  | r    | -  | -  | -   | m   | -   | -   |   | Allow             | r    | am   | c  | c  | c   | -   | -   | -   |   | Allow             | 405  | am   | m  | m  | m   | m   | m   | m   |   | Authentication-   | r    |      | o  | o  | o   | o   | o   | o/- |   | Info              |      |      |    |    |     |     |     |     |   | Authorization     | R    |      | o  | o  | o   | o   | o   | o/- |   | Bandwidth         | R    |      | o  | o  | o   | o   | -   | -   |   | Blocksize         | R    |      | o  | -  | o   | o   | -   | -   |   | Cache-Control     | G    | r    | o  | -  | o   | -   | -   | -   |   | Connection        | G    | ad   | o  | o  | o   | o   | o   | o   |   | Connection-       | 470, | ar   | o  | o  | o   | o   | o   | o   |   | Credentials       | 407  |      |    |    |     |     |     |     |   | Content-Base      | r    |      | o  | -  | -   | -   | -   | -   |   | Content-Base      | 4xx, |      | o  | o  | o   | o   | o   | o   |   |                   | 5xx  |      |    |    |     |     |     |     |   | Content-Encoding  | R    | r    | -  | -  | -   | -   | -   | -   |   | Content-Encoding  | r    | r    | o  | -  | -   | -   | -   | -   |   | Content-Encoding  | 4xx, | r    | o  | o  | o   | o   | o   | o   |   |                   | 5xx  |      |    |    |     |     |     |     |   | Content-Language  | R    | r    | -  | -  | -   | -   | -   | -   |   | Content-Language  | r    | r    | o  | -  | -   | -   | -   | -   |   | Content-Language  | 4xx, | r    | o  | o  | o   | o   | o   | o   |   |                   | 5xx  |      |    |    |     |     |     |     |   | Content-Length    | r    | r    | *  | -  | -   | -   | -   | -   |   | Content-Length    | 4xx, | r    | *  | *  | *   | *   | *   | *   |   |                   | 5xx  |      |    |    |     |     |     |     |   | Content-Location  | r    | r    | o  | -  | -   | -   | -   | -   |   | Content-Location  | 4xx, | r    | o  | o  | o   | o   | o   | o   |   |                   | 5xx  |      |    |    |     |     |     |     |   | Content-Type      | r    | r    | *  | -  | -   | -   | -   | -   |   | Content-Type      | 4xx, | ar   | *  | *  | *   | *   | *   | *   |   |                   | 5xx  |      |    |    |     |     |     |     |   | CSeq              | Gc   | rm   | m  | m  | m   | m   | m   | m   |   | Date              | G    | am   | o/*| o/*| o/* | o/* | o/* | o/* |   | Expires           | r    | r    | o  | -  | o   | -   | -   | -   |   | From              | R    | r    | o  | o  | o   | o   | o   | o   |   | If-Match          | R    | r    | -  | -  | o   | -   | -   | -   |   | If-Modified-Since | R    | r    | o  | -  | o   | -   | -   | -   |   | If-None-Match     | R    | r    | o  | -  | o   | -   | -   | -   |Schulzrinne, et al.          Standards Track                  [Page 127]

RFC 7826                        RTSP 2.0                   December 2016   | Last-Modified     | r    | r    | o  | -  | o   | -   | -   | -   |   | Location          | 3rr  |      | m  | m  | m   | m   | m   | m   |   +-------------------+------+------+----+----+-----+-----+-----+-----+   | Header            |Where |Proxy |DES | OPT| STP | PLY | PSE | TRD |   +-------------------+------+------+----+----+-----+-----+-----+-----+     Figure 2: Overview of RTSP Header Fields (A-L) Related to Methods            DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWNSchulzrinne, et al.          Standards Track                  [Page 128]

RFC 7826                        RTSP 2.0                   December 2016   +------------------+---------+-----+----+----+----+-----+-----+-----+   | Header           | Where   |Proxy|DES |OPT |STP | PLY | PSE | TRD |   +------------------+---------+-----+----+----+----+-----+-----+-----+   | Media-Properties | r       |     | -  | -  | m  | o   | o   | -   |   | Media-Range      | r       |     | -  | -  | c  | c   | c   | -   |   | MTag             | r       | r   | o  | -  | o  | -   | -   | -   |   | Pipelined-       | G       | amd | -  | o  | o  | o   | o   | o   |   | Requests         |         | r   |    |    |    |     |     |     |   | Proxy-           | 407     | amr | m  | m  | m  | m   | m   | m   |   | Authenticate     |         |     |    |    |    |     |     |     |   | Proxy-           | r       | amd | o  | o  | o  | o   | o   | o/- |   | Authentication-  |         | r   |    |    |    |     |     |     |   | Info             |         |     |    |    |    |     |     |     |   | Proxy-           | R       | rd  | o  | o  | o  | o   | o   | o   |   | Authorization    |         |     |    |    |    |     |     |     |   | Proxy-Require    | R       | ar  | o  | o  | o  | o   | o   | o   |   | Proxy-Require    | r       | r   | c  | c  | c  | c   | c   | c   |   | Proxy-Supported  | R       | amr | c  | c  | c  | c   | c   | c   |   | Proxy-Supported  | r       |     | c  | c  | c  | c   | c   | c   |   | Public           | r       | amr | -  | m  | -  | -   | -   | -   |   | Public           | 501     | amr | m  | m  | m  | m   | m   | m   |   | Range            | R       |     | -  | -  | -  | o   | -   | -   |   | Range            | r       |     | -  | -  | c  | m   | m   | -   |   | Referrer         | R       |     | o  | o  | o  | o   | o   | o   |   | Request-Status   | R       |     | -  | -  | -  | -   | -   | -   |   | Require          | R       |     | o  | o  | o  | o   | o   | o   |   | Retry-After      | 3rr,503 |     | o  | o  | o  | o   | o   | -   |   |                  | ,553    |     |    |    |    |     |     |     |   | Retry-After      | 413     |     | o  | -  | -  | -   | -   | -   |   | RTP-Info         | r       |     | -  | -  | c  | c   | -   | -   |   | Scale            | R       | r   | -  | -  | -  | o   | -   | -   |   | Scale            | r       | amr | -  | -  | c  | c   | c   | -   |   | Seek-Style       | R       |     | -  | -  | -  | o   | -   | -   |   | Seek-Style       | r       |     | -  | -  | -  | m   | -   | -   |   | Server           | R       | r   | -  | o  | -  | -   | -   | o   |   | Server           | r       | r   | o  | o  | o  | o   | o   | o   |   | Session          | R       | r   | -  | o  | o  | m   | m   | m   |   | Session          | r       | r   | -  | c  | m  | m   | m   | o   |   | Speed            | R       | admr| -  | -  | -  | o   | -   | -   |   | Speed            | r       | admr| -  | -  | -  | c   | -   | -   |   | Supported        | R       | r   | o  | o  | o  | o   | o   | o   |   | Supported        | r       | r   | c  | c  | c  | c   | c   | c   |   | Terminate-Reason | R       | r   | -  | -  | -  | -   | -   | -/o |   | Timestamp        | R       | admr| o  | o  | o  | o   | o   | o   |   | Timestamp        | c       | admr| m  | m  | m  | m   | m   | m   |   | Transport        | G       | mr  | -  | -  | m  | -   | -   | -   |   | Unsupported      | r       |     | c  | c  | c  | c   | c   | c   |   | User-Agent       | R       |     | m* | m* | m* | m*  | m*  | m*  |Schulzrinne, et al.          Standards Track                  [Page 129]

RFC 7826                        RTSP 2.0                   December 2016   | Via              | R       | amr | c  | c  | c  | c   | c   | c   |   | Via              | r       | amr | c  | c  | c  | c   | c   | c   |   | WWW-Authenticate | 401     |     | m  | m  | m  | m   | m   | m   |   +------------------+---------+-----+----+----+----+-----+-----+-----+   | Header           | Where   |Proxy|DES |OPT |STP | PLY | PSE | TRD |   +------------------+---------+-----+----+----+----+-----+-----+-----+     Figure 3: Overview of RTSP Header Fields (M-W) Related to Methods            DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWNSchulzrinne, et al.          Standards Track                  [Page 130]

RFC 7826                        RTSP 2.0                   December 2016   +---------------------------+-------+-------+-----+-----+-----+-----+   | Header                    | Where | Proxy | GPR | SPR | RDR | PNY |   +---------------------------+-------+-------+-----+-----+-----+-----+   | Accept-Credentials        | R     | rm    | o   | o   | o   | -   |   | Accept-Encoding           | R     | r     | o   | o   | o   | -   |   | Accept-Language           | R     | r     | o   | o   | o   | -   |   | Accept-Ranges             | G     | rm    | o   | -   | -   | -   |   | Allow                     | 405   | amr   | m   | m   | m   | m   |   | Authentication-Info       | r     |       | o/- | o/- | -   | -   |   | Authorization             | R     |       | o   | o   | o   | -   |   | Bandwidth                 | R     |       | -   | o   | -   | -   |   | Blocksize                 | R     |       | -   | o   | -   | -   |   | Cache-Control             | G     | r     | o   | o   | -   | -   |   | Connection                | G     |       | o   | o   | o   | o   |   | Connection-Credentials    | 470,  | ar    | o   | o   | o   | -   |   |                           | 407   |       |     |     |     |     |   | Content-Base              | R     |       | o   | o   | -   | o   |   | Content-Base              | r     |       | o   | o   | -   | -   |   | Content-Base              | 4xx,  |       | o   | o   | o   | o   |   |                           | 5xx   |       |     |     |     |     |   | Content-Encoding          | R     | r     | o   | o   | -   | o   |   | Content-Encoding          | r     | r     | o   | o   | -   | -   |   | Content-Encoding          | 4xx,  | r     | o   | o   | o   | o   |   |                           | 5xx   |       |     |     |     |     |   | Content-Language          | R     | r     | o   | o   | -   | o   |   | Content-Language          | r     | r     | o   | o   | -   | -   |   | Content-Language          | 4xx,  | r     | o   | o   | o   | o   |   |                           | 5xx   |       |     |     |     |     |   | Content-Length            | R     | r     | *   | *   | -   | *   |   | Content-Length            | r     | r     | *   | *   | -   | -   |   | Content-Length            | 4xx,  | r     | *   | *   | *   | *   |   |                           | 5xx   |       |     |     |     |     |   | Content-Location          | R     |       | o   | o   | -   | o   |   | Content-Location          | r     |       | o   | o   | -   | -   |   | Content-Location          | 4xx,  |       | o   | o   | o   | o   |   |                           | 5xx   |       |     |     |     |     |   | Content-Type              | R     |       | *   | *   | -   | *   |   | Content-Type              | r     |       | *   | *   | -   | -   |   | Content-Type              | 4xx,  |       | *   | *   | *   | *   |   |                           | 5xx   |       |     |     |     |     |   | CSeq                      | R,c   | mr    | m   | m   | m   | m   |   | Date                      | R     | a     | o/* | o/* | m   | o/* |   | Date                      | r     | am    | o/* | o/* | o/* | o/* |   | Expires                   | r     | r     | -   | -   | -   | -   |   | From                      | R     | r     | o   | o   | o   | -   |   | If-Match                  | R     | r     | -   | -   | -   | -   |   | If-Modified-Since         | R     | am    | o   | -   | -   | -   |   | If-None-Match             | R     | am    | o   | -   | -   | -   |Schulzrinne, et al.          Standards Track                  [Page 131]

RFC 7826                        RTSP 2.0                   December 2016   | Last-Modified             | R     | r     | -   | -   | -   | -   |   | Last-Modified             | r     | r     | o   | -   | -   | -   |   | Location                  | 3rr   |       | m   | m   | m   | -   |   | Location                  | R     |       | -   | -   | m   | -   |   +---------------------------+-------+-------+-----+-----+-----+-----+   | Header                    | Where | Proxy | GPR | SPR | RDR | PNY |   +---------------------------+-------+-------+-----+-----+-----+-----+     Figure 4: Overview of RTSP Header Fields (A-L) Related to Methods          GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFYSchulzrinne, et al.          Standards Track                  [Page 132]

RFC 7826                        RTSP 2.0                   December 2016 +---------------------------+---------+-------+-----+-----+-----+-----+ | Header                    |  Where  | Proxy | GPR | SPR | RDR | PNY | +---------------------------+---------+-------+-----+-----+-----+-----+ | Media-Properties          | R       | amr   | o   | -   | -   | c   | | Media-Properties          | r       | mr    | c   | -   | -   | -   | | Media-Range               | R       |       | o   | -   | -   | c   | | Media-Range               | r       |       | c   | -   | -   | -   | | MTag                      | r       | r     | o   | -   | -   | -   | | Notify-Reason             | R       |       | -   | -   | -   | m   | | Pipelined-Requests        | R       | amdr  | o   | o   | -   | -   | | Proxy-Authenticate        | 407     | amdr  | m   | m   | m   | -   | | Proxy-Authentication-Info | r       | amdr  | o/- | o/- | -   | -   | | Proxy-Authorization       | R       | amdr  | o   | o   | o   | -   | | Proxy-Require             | R       | ar    | o   | o   | o   | -   | | Proxy-Supported           | R       | amr   | c   | c   | c   | -   | | Proxy-Supported           | r       |       | c   | c   | c   | -   | | Public                    | 501     | admr  | m   | m   | m   | -   | | Range                     | R       |       | o   | -   | -   | m   | | Range                     | r       |       | c   | -   | -   | -   | | Referrer                  | R       |       | o   | o   | o   | -   | | Request-Status            | R       | mr    | -   | -   | -   | c   | | Require                   | R       | r     | o   | o   | o   | o   | | Retry-After               | 3rr,503,|       | o   | o   | -   | -   | |                           | 553     |       |     |     |     |     | | Retry-After               | 413     |       | o   | o   | -   | -   | | RTP-Info                  | R       | r     | o   | -   | -   | C   | | RTP-Info                  | r       | r     | c   | -   | -   | -   | | Scale                     | G       |       | c   | -   | c   | c   | | Seek-Style                | G       |       | -   | -   | -   | -   | | Server                    | R       | r     | o   | o   | o   | o   | | Server                    | r       | r     | o   | o   | -   | -   | | Session                   | R       | r     | o   | o   | o   | m   | | Session                   | r       | r     | c   | c   | o   | m   | | Speed                     | G       |       | -   | -   | -   | -   | | Supported                 | R       | r     | o   | o   | o   | -   | | Supported                 | r       | r     | c   | c   | c   | -   | | Terminate-Reason          | R       | r     | -   | -   | m   | -   | | Timestamp                 | R       | adrm  | o   | o   | o   | o   | | Timestamp                 | c       | adrm  | m   | m   | m   | m   | | Transport                 | G       | mr    | -   | -   | -   | -   | | Unsupported               | r       | arm   | c   | c   | c   | c   | | User-Agent                | R       | r     | m*  | m*  | -   | -   | | User-Agent                | r       | r     | m*  | m*  | m*  | m*  | | Via                       | R       | amr   | c   | c   | c   | c   |Schulzrinne, et al.          Standards Track                  [Page 133]

RFC 7826                        RTSP 2.0                   December 2016 | Via                       | r       | amr   | c   | c   | c   | c   | | WWW-Authenticate          | 401     |       | m   | m   | m   | -   | +---------------------------+---------+-------+-----+-----+-----+-----+ | Header                    |  Where  | Proxy | GPR | SPR | RDR | PNY | +---------------------------+---------+-------+-----+-----+-----+-----+     Figure 5: Overview of RTSP Header Fields (M-W) Related to Methods          GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY18.1.  Accept   The Accept request-header field can be used to specify certain   presentation description and parameter media types [RFC6838] that are   acceptable for the response to the DESCRIBE request.   SeeSection 20.2.3 for the syntax.   The asterisk "*" character is used to group media types into ranges,   with "*/*" indicating all media types and "type/*" indicating all   subtypes of that type.  The range MAY include media type parameters   that are generally applicable to that range.   Each media type or range MAY be followed by one or more accept-   params, beginning with the "q" parameter to indicate a relative   quality factor.  The first "q" parameter (if any) separates the media   type or range's parameters from the accept-params.  Quality factors   allow the user or user agent to indicate the relative degree of   preference for that media type, using the qvalue scale from 0 to 1   (Section 5.3.1 of [RFC7231]).  The default value is q=1.   Example of use:     Accept: application/example ;q=0.7, application/sdp   Indicates that the requesting agent prefers the media type   application/sdp through the default 1.0 rating but also accepts the   application/example media type with a 0.7 quality rating.   If no Accept header field is present, then it is assumed that the   client accepts all media types.  If an Accept header field is   present, and if the server cannot send a response that is acceptable   according to the combined Accept field-value, then the server SHOULD   send a 406 (Not Acceptable) response.Schulzrinne, et al.          Standards Track                  [Page 134]

RFC 7826                        RTSP 2.0                   December 201618.2.  Accept-Credentials   The Accept-Credentials header is a request-header used to indicate to   any trusted intermediary how to handle further secured connections to   proxies or servers.  It MUST NOT be included in server-to-client   requests.  SeeSection 19 for the usage of this header   In a request, the header MUST contain the method (User, Proxy, or   Any) for approving credentials selected by the requester.  The method   MUST NOT be changed by any proxy, unless it is "Proxy" when a proxy   MAY change it to "user" to take the role of user approving each   further hop.  If the method is "User", the header contains zero or   more of the credentials that the client accepts.  The header may   contain zero credentials in the first RTSP request to an RTSP server   via a proxy when using the "User" method.  This is because the client   has not yet received any credentials to accept.  Each credential MUST   consist of one URI identifying the proxy or server, the hash   algorithm identifier, and the hash over that agent's ASN.1 DER-   encoded certificate [RFC5280] in Base64, according toSection 4 of   [RFC4648] and where the padding bits are set to zero.  All RTSP   clients and proxies MUST implement the SHA-256 [FIPS180-4] algorithm   for computation of the hash of the DER-encoded certificate.  The   SHA-256 algorithm is identified by the token "sha-256".   The intention of allowing for other hash algorithms is to enable the   future retirement of algorithms that are not implemented somewhere   other than here.  Thus, the definition of future algorithms for this   purpose is intended to be extremely limited.  A feature tag can be   used to ensure that support for the replacement algorithm exists.   Example:   Accept-Credentials:User     "rtsps://proxy2.example.com/";sha-256;exaIl9VMbQMOFGClx5rXnPJKVNI=,     "rtsps://server.example.com/";sha-256;lurbjj5khhB0NhIuOXtt4bBRH1M=18.3.  Accept-Encoding   The Accept-Encoding request-header field is similar to Accept, but it   restricts the content-codings (seeSection 18.15), i.e.,   transformation codings of the message body, such as gzip compression,   that are acceptable in the response.Schulzrinne, et al.          Standards Track                  [Page 135]

RFC 7826                        RTSP 2.0                   December 2016   A server tests whether a content-coding is acceptable, according to   an Accept-Encoding field, using these rules:   1.  If the content-coding is one of the content-codings listed in the       Accept-Encoding field, then it is acceptable, unless it is       accompanied by a qvalue of 0.  (As defined inSection 5.3.1 of       [RFC7231], a qvalue of 0 means "not acceptable.")   2.  The special "*" symbol in an Accept-Encoding field matches any       available content-coding not explicitly listed in the header       field.   3.  If multiple content-codings are acceptable, then the acceptable       content-coding with the highest non-zero qvalue is preferred.   4.  The "identity" content-coding is always acceptable, i.e., no       transformation at all, unless specifically refused because the       Accept-Encoding field includes "identity;q=0" or because the       field includes "*;q=0" and does not explicitly include the       "identity" content-coding.  If the Accept-Encoding field-value is       empty, then only the "identity" encoding is acceptable.   If an Accept-Encoding field is present in a request, and if the   server cannot send a response that is acceptable according to the   Accept-Encoding header, then the server SHOULD send an error response   with the 406 (Not Acceptable) status code.   If no Accept-Encoding field is present in a request, the server MAY   assume that the client will accept any content-coding.  In this case,   if "identity" is one of the available content-codings, then the   server SHOULD use the "identity" content-coding, unless it has   additional information that a different content-coding is meaningful   to the client.18.4.  Accept-Language   The Accept-Language request-header field is similar to Accept, but   restricts the set of natural languages that are preferred as a   response to the request.  Note that the language specified applies to   the presentation description (response message body) and any reason   phrases, but not the media content.   A language tag identifies a natural language spoken, written, or   otherwise conveyed by human beings for communication of information   to other human beings.  Computer languages are explicitly excluded.   The syntax and registry of RTSP 2.0 language tags are the same as   those defined by [RFC5646].Schulzrinne, et al.          Standards Track                  [Page 136]

RFC 7826                        RTSP 2.0                   December 2016   Each language-range MAY be given an associated quality value that   represents an estimate of the user's preference for the languages   specified by that range.  The quality value defaults to "q=1".  For   example:      Accept-Language: da, en-gb;q=0.8, en;q=0.7   would mean: "I prefer Danish, but will accept British English and   other types of English."  A language-range matches a language tag if   it exactly equals the full tag or if it exactly equals a prefix of   the tag, i.e., the primary-tag in the ABNF, such that the character   following primary-tag is "-".  The special range "*", if present in   the Accept-Language field, matches every tag not matched by any other   range present in the Accept-Language field.      Note: This use of a prefix matching rule does not imply that      language tags are assigned to languages in such a way that it is      always true that if a user understands a language with a certain      tag, then this user will also understand all languages with tags      for which this tag is a prefix.  The prefix rule simply allows the      use of prefix tags if this is the case.   In the process of selecting a language, each language tag is assigned   a qualification factor, i.e., if a language being supported by the   client is actually supported by the server and what "preference"   level the language achieves.  The quality value (q-value) of the   longest language-range in the field that matches the language tag is   assigned as the qualification factor for a particular language tag.   If no language-range in the field matches the tag, the language   qualification factor assigned is 0.  If no Accept-Language header is   present in the request, the server SHOULD assume that all languages   are equally acceptable.  If an Accept-Language header is present,   then all languages that are assigned a qualification factor greater   than 0 are acceptable.18.5.  Accept-Ranges   The Accept-Ranges general-header field allows indication of the   format supported in the Range header.  The client MUST include the   header in SETUP requests to indicate which formats are acceptable   when received in PLAY and PAUSE responses and REDIRECT requests.  The   server MUST include the header in SETUP responses and 456 (Header   Field Not Valid for Resource) error responses to indicate the formats   supported for the resource indicated by the Request-URI.  The header   MAY be included in GET_PARAMETER request and response pairs.  The   GET_PARAMETER request MUST contain a Session header to identify theSchulzrinne, et al.          Standards Track                  [Page 137]

RFC 7826                        RTSP 2.0                   December 2016   session context the request is related to.  The requester and   responder will indicate their capabilities regarding Range formats   respectively.      Accept-Ranges: npt, smpte, clock   The syntax is defined inSection 20.2.3.18.6.  Allow   The Allow message body header field lists the methods supported by   the resource identified by the Request-URI.  The purpose of this   field is to inform the recipient of the complete set of valid methods   associated with the resource.  An Allow header field MUST be present   in a 405 (Method Not Allowed) response.  The Allow header MUST also   be present in all OPTIONS responses where the content of the header   will not include exactly the same methods as listed in the Public   header.   The Allow message body header MUST also be included in SETUP and   DESCRIBE responses, if the methods allowed for the resource are   different from the complete set of methods defined in this memo.   Example of use:      Allow: SETUP, PLAY, SET_PARAMETER, DESCRIBE18.7.  Authentication-Info   The Authentication-Info response-header is used by the server to   communicate some information regarding the successful HTTP   authentication [RFC7235] in the response message.  The definition of   the header is in [RFC7615], and any applicable HTTP authentication   schemes appear in other RFCs, such as Digest [RFC7616].  This header   MUST only be used in response messages related to client to server   requests.18.8.  Authorization   An RTSP client that wishes to authenticate itself with a server using   the authentication mechanism from HTTP [RFC7235], usually (but not   necessarily) after receiving a 401 response, does so by including an   Authorization request-header field with the request.  The   Authorization field-value consists of credentials containing the   authentication information of the user agent for the realm of the   resource being requested.  The definition of the header is inSchulzrinne, et al.          Standards Track                  [Page 138]

RFC 7826                        RTSP 2.0                   December 2016   [RFC7235], and any applicable HTTP authentication schemes appear in   other RFCs, such as Digest [RFC7616] and Basic [RFC7617].  This   header MUST only be used in client-to-server requests.   If a request is authenticated and a realm specified, the same   credentials SHOULD be valid for all other requests within this realm   (assuming that the authentication scheme itself does not require   otherwise, such as credentials that vary according to a challenge   value or using synchronized clocks).  Each client-to-server request   MUST be individually authorized by including the Authorization header   with the information.   When a shared cache (seeSection 16) receives a request containing an   Authorization field, it MUST NOT return the corresponding response as   a reply to any other request, unless one of the following specific   exceptions holds:   1.  If the response includes the "max-age" cache directive, the cache       MAY use that response in replying to a subsequent request.  But       (if the specified maximum age has passed) a proxy cache MUST       first revalidate it with the origin server, using the request-       headers from the new request to allow the origin server to       authenticate the new request.  (This is the defined behavior for       max-age.)  If the response includes "max-age=0", the proxy MUST       always revalidate it before reusing it.   2.  If the response includes the "must-revalidate" cache-control       directive, the cache MAY use that response in replying to a       subsequent request.  But if the response is stale, all caches       MUST first revalidate it with the origin server, using the       request-headers from the new request to allow the origin server       to authenticate the new request.   3.  If the response includes the "public" cache directive, it MAY be       returned in reply to any subsequent request.18.9.  Bandwidth   The Bandwidth request-header field describes the estimated bandwidth   available to the client, expressed as a positive integer and measured   in kilobits per second.  The bandwidth available to the client may   change during an RTSP session, e.g., due to mobility, congestion,   etc.   Clients may not be able to accurately determine the available   bandwidth, for example, because the first hop is not a bottleneck.   Such a case is when the local area network (LAN) is not the   bottleneck, instead the LAN's Internet access link is, if the serverSchulzrinne, et al.          Standards Track                  [Page 139]

RFC 7826                        RTSP 2.0                   December 2016   is not in the same LAN.  Thus, link speeds of WLAN or Ethernet   networks are normally not a basis for estimating the available   bandwidth.  Cellular devices or other devices directly connected to a   modem or connection-enabling device may more accurately estimate the   bottleneck bandwidth and what is a reasonable share of it for RTSP-   controlled media.  The client will also need to take into account   other traffic sharing the bottleneck.  For example, by only assigning   a certain fraction to RTSP and its media streams.  It is RECOMMENDED   that only clients that have accurate and explicit information about   bandwidth bottlenecks use this header.   This header is not a substitute for proper congestion control.  It is   only a method providing an initial estimate and coarsely determines   if the selected content can be delivered at all.   Example:     Bandwidth: 6236018.10.  Blocksize   The Blocksize request-header field is sent from the client to the   media server asking the server for a particular media packet size.   This packet size does not include lower-layer headers such as IP,   UDP, or RTP.  The server is free to use a blocksize that is lower   than the one requested.  The server MAY truncate this packet size to   the closest multiple of the minimum, media-specific block size or   override it with the media-specific size, if necessary.  The block   size MUST be a positive decimal number measured in octets.  The   server only returns an error (4xx) if the value is syntactically   invalid.18.11.  Cache-Control   The Cache-Control general-header field is used to specify directives   that MUST be obeyed by all caching mechanisms along the request/   response chain.   Cache directives MUST be passed through by a proxy or gateway   application, regardless of their significance to that application,   since the directives may be applicable to all recipients along the   request/response chain.  It is not possible to specify a cache-   directive for a specific cache.   Cache-Control should only be specified in a DESCRIBE, GET_PARAMETER,   SET_PARAMETER, and SETUP request and its response.  Note: Cache-   Control does not govern only the caching of responses for the RTSP   messages, instead it also applies to the media stream identified bySchulzrinne, et al.          Standards Track                  [Page 140]

RFC 7826                        RTSP 2.0                   December 2016   the SETUP request.  The RTSP requests are generally not cacheable;   for further information, seeSection 16.  Below are the descriptions   of the cache directives that can be included in the Cache-Control   header.   no-cache:  Indicates that the media stream or RTSP response MUST NOT         be cached anywhere.  This allows an origin server to prevent         caching even by caches that have been configured to return         stale responses to client requests.  Note: there is no security         function preventing the caching of content.   public:  Indicates that the media stream or RTSP response is         cacheable by any cache.   private:  Indicates that the media stream or RTSP response is         intended for a single user and MUST NOT be cached by a shared         cache.  A private (non-shared) cache may cache the media         streams.   no-transform:  An intermediate cache (proxy) may find it useful to         convert the media type of a certain stream.  A proxy might, for         example, convert between video formats to save cache space or         to reduce the amount of traffic on a slow link.  Serious         operational problems may occur, however, when these         transformations have been applied to streams intended for         certain kinds of applications.  For example, applications for         medical imaging, scientific data analysis and those using end-         to-end authentication all depend on receiving a stream that is         bit-for-bit identical to the original media stream or RTSP         response.  Therefore, if a response includes the no-transform         directive, an intermediate cache or proxy MUST NOT change the         encoding of the stream or response.  Unlike HTTP, RTSP does not         provide for partial transformation at this point, e.g.,         allowing translation into a different language.   only-if-cached:  In some cases, such as times of extremely poor         network connectivity, a client may want a cache to return only         those media streams or RTSP responses that it currently has         stored and not to receive these from the origin server.  To do         this, the client may include the only-if-cached directive in a         request.  If the cache receives this directive, it SHOULD         either respond using a cached media stream or response that is         consistent with the other constraints of the request or respond         with a 504 (Gateway Timeout) status.  However, if a group of         caches is being operated as a unified system with good internal         connectivity, such a request MAY be forwarded within that group         of caches.Schulzrinne, et al.          Standards Track                  [Page 141]

RFC 7826                        RTSP 2.0                   December 2016   max-stale:  Indicates that the client is willing to accept a media         stream or RTSP response that has exceeded its expiration time.         If max-stale is assigned a value, then the client is willing to         accept a response that has exceeded its expiration time by no         more than the specified number of seconds.  If no value is         assigned to max-stale, then the client is willing to accept a         stale response of any age.   min-fresh:  Indicates that the client is willing to accept a media         stream or RTSP response whose freshness lifetime is no less         than its current age plus the specified time in seconds.  That         is, the client wants a response that will still be fresh for at         least the specified number of seconds.   must-revalidate:  When the must-revalidate directive is present in a         SETUP response received by a cache, that cache MUST NOT use the         cache entry after it becomes stale to respond to a subsequent         request without first revalidating it with the origin server.         That is, the cache is required to do an end-to-end revalidation         every time, if, based solely on the origin server's Expires,         the cached response is stale.   proxy-revalidate:  The proxy-revalidate directive has the same         meaning as the must-revalidate directive, except that it does         not apply to non-shared user agent caches.  It can be used on a         response to an authenticated request to permit the user's cache         to store and later return the response without needing to         revalidate it (since it has already been authenticated once by         that user), while still requiring proxies that service many         users to revalidate each time (in order to make sure that each         user has been authenticated).  Note that such authenticated         responses also need the "public" cache directive in order to         allow them to be cached at all.   max-age:  When an intermediate cache is forced, by means of a max-         age=0 directive, to revalidate its own cache entry, and the         client has supplied its own validator in the request, the         supplied validator might differ from the validator currently         stored with the cache entry.  In this case, the cache MAY use         either validator in making its own request without affecting         semantic transparency.         However, the choice of validator might affect performance.  The         best approach is for the intermediate cache to use its own         validator when making its request.  If the server replies with         304 (Not Modified), then the cache can return its now validated         copy to the client with a 200 (OK) response.  If the server         replies with a new message body and cache validator, however,Schulzrinne, et al.          Standards Track                  [Page 142]

RFC 7826                        RTSP 2.0                   December 2016         the intermediate cache can compare the returned validator with         the one provided in the client's request, using the strong         comparison function.  If the client's validator is equal to the         origin server's, then the intermediate cache simply returns 304         (Not Modified).  Otherwise, it returns the new message body         with a 200 (OK) response.18.12.  Connection   The Connection general-header field allows the sender to specify   options that are desired for that particular connection.  It MUST NOT   be communicated by proxies over further connections.   RTSP 2.0 proxies MUST parse the Connection header field before a   message is forwarded and, for each connection-token in this field,   remove any header field(s) from the message with the same name as the   connection-token.  Connection options are signaled by the presence of   a connection-token in the Connection header field, not by any   corresponding additional header field(s), since the additional header   field may not be sent if there are no parameters associated with that   connection option.   Message headers listed in the Connection header MUST NOT include end-   to-end headers, such as Cache-Control.   RTSP 2.0 defines the "close" connection option for the sender to   signal that the connection will be closed after completion of the   response.  For example, "Connection: close in either the request or   the response-header fields" indicates that the connection SHOULD NOT   be considered "persistent" (Section 10.2) after the current request/   response is complete.   The use of the connection option "close" in RTSP messages SHOULD be   limited to error messages when the server is unable to recover and   therefore sees it necessary to close the connection.  The reason   being that the client has the choice of continuing using a connection   indefinitely, as long as it sends valid messages.18.13.  Connection-Credentials   The Connection-Credentials response-header is used to carry the chain   of credentials for any next hop that needs to be approved by the   requester.  It MUST only be used in server-to-client responses.   The Connection-Credentials header in an RTSP response MUST, if   included, contain the credential information (in the form of a list   of certificates providing the chain of certification) of the next hop   to which an intermediary needs to securely connect.  The header MUSTSchulzrinne, et al.          Standards Track                  [Page 143]

RFC 7826                        RTSP 2.0                   December 2016   include the URI of the next hop (proxy or server) and a   Base64-encoded (according toSection 4 of [RFC4648] and where the   padding bits are set to zero) binary structure containing a sequence   of DER-encoded X.509v3 certificates [RFC5280].   The binary structure starts with the number of certificates   (NR_CERTS) included as a 16-bit unsigned integer.  This is followed   by an NR_CERTS number of 16-bit unsigned integers providing the size,   in octets, of each DER-encoded certificate.  This is followed by an   NR_CERTS number of DER-encoded X.509v3 certificates in a sequence   (chain).  This format is exemplified in Figure 6.  The certificate of   the proxy or server must come first in the structure.  Each following   certificate must directly certify the one preceding it.  Because   certificate validation requires that root keys be distributed   independently, the self-signed certificate that specifies the root   certificate authority may optionally be omitted from the chain, under   the assumption that the remote end must already possess it in order   to validate it in any case.   Example:   Connection-Credentials:"rtsps://proxy2.example.com/";MIIDNTCC...   Where MIIDNTCC... is a Base64 encoding of the following structure:        0                   1                   2                   3        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+       |  Number of certificates       | Size of certificate #1        |       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+       | Size of certificate #2        | Size of certificate #3        |       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+       : DER Encoding of Certificate #1                                :       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+       : DER Encoding of Certificate #2                                :       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+       : DER Encoding of Certificate #3                                :       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   Figure 6: Format Example of Connection-Credentials Header Certificate18.14.  Content-Base   The Content-Base message body header field may be used to specify the   base URI for resolving relative URIs within the message body.   Content-Base: rtsp://media.example.com/movie/twister/Schulzrinne, et al.          Standards Track                  [Page 144]

RFC 7826                        RTSP 2.0                   December 2016   If no Content-Base field is present, the base URI of a message body   is defined by either its Content-Location (if that Content-Location   URI is an absolute URI) or the URI used to initiate the request, in   that order of precedence.  Note, however, that the base URI of the   contents within the message body may be redefined within that message   body.18.15.  Content-Encoding   The Content-Encoding message body header field is used as a modifier   of the media-type.  When present, its value indicates what additional   content-codings have been applied to the message body, and thus what   decoding mechanisms must be applied in order to obtain the media-type   referenced by the Content-Type header field.  Content-Encoding is   primarily used to allow a document to be compressed without losing   the identity of its underlying media type.   The content-coding is a characteristic of the message body identified   by the Request-URI.  Typically, the message body is stored with this   encoding and is only decoded before rendering or analogous usage.   However, an RTSP proxy MAY modify the content-coding if the new   coding is known to be acceptable to the recipient, unless the "no-   transform" cache directive is present in the message.   If the content-coding of a message body is not "identity", then the   message MUST include a Content-Encoding message body header that   lists the non-identity content-coding(s) used.   If the content-coding of a message body in a request message is not   acceptable to the origin server, the server SHOULD respond with a   status code of 415 (Unsupported Media Type).   If multiple encodings have been applied to a message body, the   content-codings MUST be listed in the order in which they were   applied, first to last from left to right.  Additional information   about the encoding parameters MAY be provided by other header fields   not defined by this specification.18.16.  Content-Language   The Content-Language message body header field describes the natural   language(s) of the intended audience for the enclosed message body.   Note that this might not be equivalent to all the languages used   within the message body.Schulzrinne, et al.          Standards Track                  [Page 145]

RFC 7826                        RTSP 2.0                   December 2016   Language tags are mentioned inSection 18.4.  The primary purpose of   Content-Language is to allow a user to identify and differentiate   entities according to the user's own preferred language.  Thus, if   the body content is intended only for a Danish-literate audience, the   appropriate field is      Content-Language: da   If no Content-Language is specified, the default is that the content   is intended for all language audiences.  This might mean that the   sender does not consider it to be specific to any natural language or   that the sender does not know for which language it is intended.   Multiple languages MAY be listed for content that is intended for   multiple audiences.  For example, a rendition of the "Treaty of   Waitangi", presented simultaneously in the original Maori and English   versions, would call for      Content-Language: mi, en   However, just because multiple languages are present within a message   body does not mean that it is intended for multiple linguistic   audiences.  An example would be a beginner's language primer, such as   "A First Lesson in Latin", which is clearly intended to be used by an   English-literate audience.  In this case, the Content-Language would   properly only include "en".   Content-Language MAY be applied to any media type -- it is not   limited to textual documents.18.17.  Content-Length   The Content-Length message body header field contains the length of   the message body of the RTSP message (i.e., after the double CRLF   following the last header) in octets of bits.  Unlike HTTP, it MUST   be included in all messages that carry a message body beyond the   header portion of the RTSP message.  If it is missing, a default   value of zero is assumed.  Any Content-Length greater than or equal   to zero is a valid value.18.18.  Content-Location   The Content-Location message body header field MAY be used to supply   the resource location for the message body enclosed in the message   when that body is accessible from a location separate from the   requested resource's URI.  A server SHOULD provide a Content-Location   for the variant corresponding to the response message body;   especially in the case where a resource has multiple variantsSchulzrinne, et al.          Standards Track                  [Page 146]

RFC 7826                        RTSP 2.0                   December 2016   associated with it, and those entities actually have separate   locations by which they might be individually accessed, the server   SHOULD provide a Content-Location for the particular variant that is   returned.   As an example, if an RTSP client performs a DESCRIBE request on a   given resource, e.g., "rtsp://a.example.com/movie/   Plan9FromOuterSpace", then the server may use additional information,   such as the User-Agent header, to determine the capabilities of the   agent.  The server will then return a media description tailored to   that class of RTSP agents.  To indicate which specific description   the agent receives, the resource identifier   ("rtsp://a.example.com/movie/Plan9FromOuterSpace/FullHD.sdp") is   provided in Content-Location, while the description is still a valid   response for the generic resource identifier, thus enabling both   debugging and cache operation as discussed below.   The Content-Location value is not a replacement for the original   requested URI; it is only a statement of the location of the resource   corresponding to this particular variant at the time of the request.   Future requests MAY specify the Content-Location URI as the Request-   URI if the desire is to identify the source of that particular   variant.  This is useful if the RTSP agent desires to verify if the   resource variant is current through a conditional request.   A cache cannot assume that a message body with a Content-Location   different from the URI used to retrieve it can be used to respond to   later requests on that Content-Location URI.  However, the Content-   Location can be used to differentiate between multiple variants   retrieved from a single requested resource.   If the Content-Location is a relative URI, the relative URI is   interpreted relative to the Request-URI.   Note that Content-Location can be used in some cases to derive the   base-URI for relative URI(s) present in session description formats.   This needs to be taken into account when Content-Location is used.   The easiest way to avoid needing to consider that issue is to include   the Content-Base whenever the Content-Location is included.   Note also, when using Media Tags in conjunction with Content-   Location, it is important that the different versions have different   MTags, even if provided under different Content-Location URIs.  This   is because the different content variants still have been provided in   response to the same request URI.Schulzrinne, et al.          Standards Track                  [Page 147]

RFC 7826                        RTSP 2.0                   December 2016   Note also, as in most cases, the URIs used in the DESCRIBE and the   SETUP requests are different: the URI provided in a DESCRIBE Content-   Location response can't directly be used in a SETUP request.   Instead, the steps of deriving the media resource URIs are necessary.   This commonly involves combing the media description's relative URIs,   e.g., from the SDP's a=control attribute, with the base-URI to create   the absolute URIs needed in the SETUP request.18.19.  Content-Type   The Content-Type message body header indicates the media type of the   message body sent to the recipient.  Note that the content types   suitable for RTSP are likely to be restricted in practice to   presentation descriptions and parameter-value types.18.20.  CSeq   The CSeq general-header field specifies the sequence number (integer)   for an RTSP request/response pair.  This field MUST be present in all   requests and responses.  RTSP agents maintain a sequence number   series for each responder to which they have an open message   transport channel.  For each new RTSP request an agent originates on   a particular RTSP message transport, the CSeq value MUST be   incremented by one.  The initial sequence number can be any number;   however, it is RECOMMENDED to start at 0.  Each sequence number   series is unique between each requester and responder, i.e., the   client has one series for its requests to a server and the server has   another when sending requests to the client.  Each requester and   responder is identified by its socket address (IP address and port   number), i.e., per direction of a TCP connection.  Any retransmitted   request MUST contain the same sequence number as the original, i.e.,   the sequence number is not incremented for retransmissions of the   same request.  The RTSP agent receiving requests MUST process the   requests arriving on a particular transport in the order of the   sequence numbers.  Responses are sent in the order that they are   generated.  The RTSP response MUST have the same sequence number as   was present in the corresponding request.  An RTSP agent receiving a   response MAY receive the responses out of order compared to the order   of the requests it sent.  Thus, the agent MUST use the sequence   number in the response to pair it with the corresponding request.      The main purpose of the sequence number is to map responses to      requests.      The requirement to use a sequence-number increment of one for each      new request is to support any future specification of RTSP message      transport over a protocol that does not provide in-order delivery      or is unreliable.Schulzrinne, et al.          Standards Track                  [Page 148]

RFC 7826                        RTSP 2.0                   December 2016      The above rules relating to the initial sequence number may appear      unnecessarily loose.  The reason for this is to cater to some      common behavior of existing implementations: when using multiple      reliable connections in sequence, it may still be easiest to use a      single sequence-number series for a client connecting with a      particular server.  Thus, the initial sequence number may be      arbitrary depending on the number of previous requests.  For any      unreliable transport, a stricter definition or other solution will      be required to enable detection of any loss of the first request.      When using multiple sequential transport connections, there is no      protocol mechanism to ensure in-order processing as the sequence      number is scoped on the individual transport connection and its      five tuple.  Thus, there are potential issues with opening a new      transport connection to the same host for which there already      exists a transport connection with outstanding requests and      previously dispatched requests related to the same RTSP session.   RTSP Proxies also need to follow the above rules.  This implies that   proxies that aggregate requests from multiple clients onto a single   transport towards a server or a next-hop proxy need to renumber these   requests to form a unified sequence on that transport, fulfilling the   above rules.  A proxy capable of fulfilling some agent's request   without emitting its own request (e.g., a caching proxy that fulfills   a request from its cache) also causes a need to renumber as the   number of received requests with a particular target may not be the   same as the number of emitted requests towards that target agent.  A   proxy that needs to renumber needs to perform the corresponding   renumbering back to the original sequence number for any received   response before forwarding it back to the originator of the request.      A client connected to a proxy, and using that transport to send      requests to multiple servers, creates a situation where it is      quite likely to receive the responses out of order.  This is      because the proxy will establish separate transports from the      proxy to the servers on which to forward the client's requests.      When the responses arrive from the different servers, they will be      forwarded to the client in the order they arrive at the proxy and      can be processed, not the order of the client's original sequence      numbers.  This is intentional to avoid some session's requests      being blocked by another server's slow processing of requests.Schulzrinne, et al.          Standards Track                  [Page 149]

RFC 7826                        RTSP 2.0                   December 201618.21.  Date   The Date general-header field represents the date and time at which   the message was originated.  The inclusion of the Date header in an   RTSP message follows these rules:   o  An RTSP message, sent by either the client or the server,      containing a body MUST include a Date header, if the sending host      has a clock;   o  Clients and servers are RECOMMENDED to include a Date header in      all other RTSP messages, if the sending host has a clock;   o  If the server does not have a clock that can provide a reasonable      approximation of the current time, its responses MUST NOT include      a Date header field.  In this case, this rule MUST be followed:      some origin-server implementations might not have a clock      available.  An origin server without a clock MUST NOT assign      Expires or Last-Modified values to a response, unless these values      were associated with the resource by a system or user with a      reliable clock.  It MAY assign an Expires value that is known, at      or before server-configuration time, to be in the past (this      allows "pre-expiration" of responses without storing separate      Expires values for each resource).   A received message that does not have a Date header field MUST be   assigned one by the recipient if the message will be cached by that   recipient.  An RTSP implementation without a clock MUST NOT cache   responses without revalidating them on every use.  An RTSP cache,   especially a shared cache, SHOULD use a mechanism, such as the   Network Time Protocol (NTP) [RFC5905], to synchronize its clock with   a reliable external standard.   The RTSP-date, a full date as specified bySection 3.3 of [RFC5322],   sent in a Date header SHOULD NOT represent a date and time subsequent   to the generation of the message.  It SHOULD represent the best   available approximation of the date and time of message generation,   unless the implementation has no means of generating a reasonably   accurate date and time.  In theory, the date ought to represent the   moment just before the message body is generated.  In practice, the   date can be generated at any time during the message origination   without affecting its semantic value.      Note: The RTSP 2.0 date format is defined to be the full-date      format inRFC 5322.  This format is more flexible than the date      format inRFC 1123 used by RTSP 1.0.  Thus, implementations should      use single spaces as separators, as recommended byRFC 5322, and      support receiving the obsolete format.Schulzrinne, et al.          Standards Track                  [Page 150]

RFC 7826                        RTSP 2.0                   December 2016      Further, note that the syntax allows for a comment to be added at      the end of the date.18.22.  Expires   The Expires message body header field gives a date and time after   which the description or media-stream should be considered stale.   The interpretation depends on the method:   DESCRIBE response:  The Expires header indicates a date and time         after which the presentation description (body) SHOULD be         considered stale.   SETUP response:  The Expires header indicates a date and time after         which the media stream SHOULD be considered stale.   A stale cache entry should not be returned by a cache (either a proxy   cache or a user agent cache) unless it is first validated with the   origin server (or with an intermediate cache that has a fresh copy of   the message body).  SeeSection 16 for further discussion of the   expiration model.   The presence of an Expires field does not imply that the original   resource will change or cease to exist at, before, or after that   time.   The format is an absolute date and time as defined by RTSP-date.  An   example of its use is     Expires: Wed, 23 Jan 2013 15:36:52 +0000   RTSP 2.0 clients and caches MUST treat other invalid date formats,   especially those including the value "0", as having occurred in the   past (i.e., already expired).   To mark a response as "already expired," an origin server should use   an Expires date that is equal to the Date header value.  To mark a   response as "never expires", an origin server SHOULD use an Expires   date approximately one year from the time the response is sent.  RTSP   2.0 servers SHOULD NOT send Expires dates that are more than one year   in the future.18.23.  From   The From request-header field, if given, SHOULD contain an Internet   email address for the human user who controls the requesting user   agent.  The address SHOULD be machine usable, as defined by "mailbox"   in [RFC1123].Schulzrinne, et al.          Standards Track                  [Page 151]

RFC 7826                        RTSP 2.0                   December 2016   This header field MAY be used for logging purposes and as a means for   identifying the source of invalid or unwanted requests.  It SHOULD   NOT be used as an insecure form of access protection.  The   interpretation of this field is that the request is being performed   on behalf of the person given, who accepts responsibility for the   method performed.  In particular, robot agents SHOULD include this   header so that the person responsible for running the robot can be   contacted if problems occur on the receiving end.   The Internet email address in this field MAY be separate from the   Internet host that issued the request.  For example, when a request   is passed through a proxy, the original issuer's address SHOULD be   used.   The client SHOULD NOT send the From header field without the user's   approval, as it might conflict with the user's privacy interests or   their site's security policy.  It is strongly recommended that the   user be able to disable, enable, and modify the value of this field   at any time prior to a request.18.24.  If-Match   The If-Match request-header field is especially useful for ensuring   the integrity of the presentation description, independent of how the   presentation description was received.  The presentation description   can be fetched via means external to RTSP (such as HTTP) or via the   DESCRIBE message.  In the case of retrieving the presentation   description via RTSP, the server implementation is guaranteeing the   integrity of the description between the time of the DESCRIBE message   and the SETUP message.  By including the MTag given in or with the   session description in an If-Match header part of the SETUP request,   the client ensures that resources set up are matching the   description.  A SETUP request with the If-Match header for which the   MTag validation check fails MUST generate a response using 412   (Precondition Failed).   This validation check is also very useful if a session has been   redirected from one server to another.18.25.  If-Modified-Since   The If-Modified-Since request-header field is used with the DESCRIBE   and SETUP methods to make them conditional.  If the requested variant   has not been modified since the time specified in this field, a   description will not be returned from the server (DESCRIBE) or a   stream will not be set up (SETUP).  Instead, a 304 (Not Modified)   response MUST be returned without any message body.Schulzrinne, et al.          Standards Track                  [Page 152]

RFC 7826                        RTSP 2.0                   December 2016   An example of the field is:     If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT18.26.  If-None-Match   This request-header can be used with one or several message body tags   to make DESCRIBE requests conditional.  A client that has one or more   message bodies previously obtained from the resource can verify that   none of those entities is current by including a list of their   associated message body tags in the If-None-Match header field.  The   purpose of this feature is to allow efficient updates of cached   information with a minimum amount of transaction overhead.  As a   special case, the value "*" matches any current entity of the   resource.   If any of the message body tags match the message body tag of the   message body that would have been returned in the response to a   similar DESCRIBE request (without the If-None-Match header) on that   resource, or if "*" is given and any current entity exists for that   resource, then the server MUST NOT perform the requested method,   unless required to do so because the resource's modification date   fails to match that supplied in an If-Modified-Since header field in   the request.  Instead, if the request method was DESCRIBE, the server   SHOULD respond with a 304 (Not Modified) response, including the   cache-related header fields (particularly MTag) of one of the message   bodies that matched.  For all other request methods, the server MUST   respond with a status of 412 (Precondition Failed).   SeeSection 16.1.3 for rules on how to determine if two message body   tags match.   If none of the message body tags match, then the server MAY perform   the requested method as if the If-None-Match header field did not   exist, but MUST also ignore any If-Modified-Since header field(s) in   the request.  That is, if no message body tags match, then the server   MUST NOT return a 304 (Not Modified) response.   If the request would, without the If-None-Match header field, result   in anything other than a 2xx or 304 status, then the If-None-Match   header MUST be ignored.  (SeeSection 16.1.4 for a discussion of   server behavior when both If-Modified-Since and If-None-Match appear   in the same request.)   The result of a request having both an If-None-Match header field and   an If-Match header field is unspecified and MUST be considered an   illegal request.Schulzrinne, et al.          Standards Track                  [Page 153]

RFC 7826                        RTSP 2.0                   December 201618.27.  Last-Modified   The Last-Modified message body header field indicates the date and   time at which the origin server believes the presentation description   or media stream was last modified.  For the DESCRIBE method, the   header field indicates the last modification date and time of the   description, for the SETUP of the media stream.   An origin server MUST NOT send a Last-Modified date that is later   than the server's time of message origination.  In such cases, where   the resource's last modification would indicate some time in the   future, the server MUST replace that date with the message   origination date.   An origin server SHOULD obtain the Last-Modified value of the message   body as close as possible to the time that it generates the Date   value of its response.  This allows a recipient to make an accurate   assessment of the message body's modification time, especially if the   message body changes near the time that the response is generated.   RTSP servers SHOULD send Last-Modified whenever feasible.18.28.  Location   The Location response-header field is used to redirect the recipient   to a location other than the Request-URI for completion of the   request or identification of a new resource.  For 3rr responses, the   location SHOULD indicate the server's preferred URI for automatic   redirection to the resource.  The field-value consists of a single   absolute URI.   Note: The Content-Location header field (Section 18.18) differs from   Location in that the Content-Location identifies the original   location of the message body enclosed in the request.  Therefore, it   is possible for a response to contain header fields for both Location   and Content-Location.  Also, seeSection 16.2 for cache requirements   of some methods.18.29.  Media-Properties   This general-header is used in SETUP responses or PLAY_NOTIFY   requests to indicate the media's properties that currently are   applicable to the RTSP session.  PLAY_NOTIFY MAY be used to modify   these properties at any point.  However, the client SHOULD have   received the update prior to any action related to the new media   properties taking effect.  For aggregated sessions, the Media-   Properties header will be returned in each SETUP response.  The   header received in the latest response is the one that applies on theSchulzrinne, et al.          Standards Track                  [Page 154]

RFC 7826                        RTSP 2.0                   December 2016   whole session from this point until any future update.  The header   MAY be included without value in GET_PARAMETER requests to the server   with a Session header included to query the current Media-Properties   for the session.  The responder MUST include the current session's   media properties.   The media properties expressed by this header are the ones applicable   to all media in the RTSP session.  For aggregated sessions, the   header expressed the combined media-properties.  As a result,   aggregation of media MAY result in a change of the media properties   and, thus, the content of the Media-Properties header contained in   subsequent SETUP responses.   The header contains a list of property values that are applicable to   the currently setup media or aggregate of media as indicated by the   RTSP URI in the request.  No ordering is enforced within the header.   Property values should be placed into a single group that handles a   particular orthogonal property.  Values or groups that express   multiple properties SHOULD NOT be used.  The list of properties that   can be expressed MAY be extended at any time.  Unknown property   values MUST be ignored.   This specification defines the following four groups and their   property values:   Random Access:      Random-Access:  Indicates that random access is possible.  May         optionally include a floating-point value in seconds indicating         the longest duration between any two random access points in         the media.      Beginning-Only:  Seeking is limited to the beginning only.      No-Seeking:  No seeking is possible.   Content Modifications:      Immutable:  The content will not be changed during the lifetime of         the RTSP session.      Dynamic:  The content may be changed based on external methods or         triggers.      Time-Progressing:  The media accessible progresses as wallclock         time progresses.Schulzrinne, et al.          Standards Track                  [Page 155]

RFC 7826                        RTSP 2.0                   December 2016   Retention:      Unlimited:  Content will be retained for the duration of the         lifetime of the RTSP session.      Time-Limited:  Content will be retained at least until the         specified wallclock time.  The time must be provided in the         absolute time format specified inSection 4.4.3.      Time-Duration:  Each individual media unit is retained for at         least the specified Time-Duration.  This definition allows for         retaining data with a time-based sliding window.  The time         duration is expressed as floating-point number in seconds.  The         value 0.0 is a valid as this indicates that no data is retained         in a time-progressing session.   Supported Scale:      Scales:  A quoted comma-separated list of one or more decimal         values or ranges of scale values supported by the content in         arbitrary order.  A range has a start and stop value separated         by a colon.  A range indicates that the content supports a         fine-grained selection of scale values.  Fine-graining allows         for steps at least as small as one tenth of a scale value.         Content is considered to support fine-grained selection when         the server in response to a given scale value can produce         content with an actual scale that is less than one tenth of         scale unit, i.e., 0.1, from the requested value.  Negative         values are supported.  The value 0 has no meaning and MUST NOT         be used.   Examples of this header for on-demand content and a live stream   without recording are:   On-demand:   Media-Properties: Random-Access=2.5, Unlimited, Immutable,        Scales="-20, -10, -4, 0.5:1.5, 4, 8, 10, 15, 20"   Live stream without recording/timeshifting:   Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0.018.30.  Media-Range   The Media-Range general-header is used to give the range of the media   at the time of sending the RTSP message.  This header MUST be   included in the SETUP response, PLAY and PAUSE responses for media   that are time-progressing, PLAY and PAUSE responses after any change   for media that are Dynamic, and in PLAY_NOTIFY requests that are sentSchulzrinne, et al.          Standards Track                  [Page 156]

RFC 7826                        RTSP 2.0                   December 2016   due to Media-Property-Update.  A Media-Range header without any range   specifications MAY be included in GET_PARAMETER requests to the   server to request the current range.  In this case, the server MUST   include the current range at the time of sending the response.   The header MUST include range specifications for all time formats   supported for the media, as indicated in Accept-Ranges header   (Section 18.5) when setting up the media.  The server MAY include   more than one range specification of any given time format to   indicate media that has non-continuous range.  The range   specifications SHALL be ordered with the range with the lowest value   or earliest start time first, followed by ranges with increasingly   higher values or later start time.   For media that has the time-progressing property, the Media-Range   header values will only be valid for the particular point in time   when it was issued.  As the wallclock progresses, so will the media   range.  However, it shall be assumed that media time progresses in   direct relationship to wallclock time (with the exception of clock   skew) so that a reasonably accurate estimation of the media range can   be calculated.18.31.  MTag   The MTag response-header MAY be included in DESCRIBE, GET_PARAMETER,   or SETUP responses.  The message body tags (Section 4.6) returned in   a DESCRIBE response and the one in SETUP refer to the presentation,   i.e., both the returned session description and the media stream.   This allows for verification that one has the right session   description to a media resource at the time of the SETUP request.   However, it has the disadvantage that a change in any of the parts   results in invalidation of all the parts.   If the MTag is provided both inside the message body, e.g., within   the "a=mtag" attribute in SDP, and in the response message, then both   tags MUST be identical.  It is RECOMMENDED that the MTag be primarily   given in the RTSP response message, to ensure that caches can use the   MTag without requiring content inspection.  However, for session   descriptions that are distributed outside of RTSP, for example, using   HTTP, etc., it will be necessary to include the message body tag in   the session description as specified inAppendix D.1.9.   SETUP and DESCRIBE requests can be made conditional upon the MTag   using the headers If-Match (Section 18.24) and If-None-Match   (Section 18.26).Schulzrinne, et al.          Standards Track                  [Page 157]

RFC 7826                        RTSP 2.0                   December 201618.32.  Notify-Reason   The Notify-Reason response-header is solely used in the PLAY_NOTIFY   method.  It indicates the reason why the server has sent the   asynchronous PLAY_NOTIFY request (seeSection 13.5).18.33.  Pipelined-Requests   The Pipelined-Requests general-header is used to indicate that a   request is to be executed in the context created by a previous   request(s).  The primary usage of this header is to allow pipelining   of SETUP requests so that any additional SETUP request after the   first one does not need to wait for the session ID to be sent back to   the requesting agent.  The header contains a unique identifier that   is scoped by the persistent connection used to send the requests.   Upon receiving a request with the Pipelined-Requests, the responding   agent MUST look up if there exists a binding between this Pipelined-   Requests identifier for the current persistent connection and an RTSP   session ID.  If the binding exists, then the received request is   processed the same way as if it contained the Session header with the   found session ID.  If there does not exist a mapping and no Session   header is included in the request, the responding agent MUST create a   binding upon the successful completion of a session creating request,   i.e., SETUP.  A binding MUST NOT be created, if the request failed to   create an RTSP session.  In case the request contains both a Session   header and the Pipelined-Requests header, the Pipelined-Requests   header MUST be ignored.   Note: Based on the above definition, at least the first request   containing a new unique Pipelined-Requests header will be required to   be a SETUP request (unless the protocol is extended with new methods   of creating a session).  After that first one, additional SETUP   requests or requests of any type using the RTSP session context may   include the Pipelined-Requests header.   When responding to any request that contained the Pipelined-Requests   header, the server MUST also include the Session header when a   binding to a session context exists.  An RTSP agent that knows the   session identifier SHOULD NOT use the Pipelined-Requests header in   any request and only use the Session header.  This as the Session   identifier is persistent across transport contexts, like TCP   connections, which the Pipelined-Requests identifier is not.   The RTSP agent sending the request with a Pipelined-Requests header   has the responsibility for using a unique and previously unused   identifier within the transport context.  Currently, only a TCP   connection is defined as such a transport context.  A server MUSTSchulzrinne, et al.          Standards Track                  [Page 158]

RFC 7826                        RTSP 2.0                   December 2016   delete the Pipelined-Requests identifier and its binding to a session   upon the termination of that session.  Despite the previous mandate,   RTSP agents are RECOMMENDED not to reuse identifiers to allow for   better error handling and logging.   RTSP Proxies may need to translate Pipelined-Requests identifier   values from incoming requests to outgoing to allow for aggregation of   requests onto a persistent connection.18.34.  Proxy-Authenticate   The Proxy-Authenticate response-header field MUST be included as part   of a 407 (Proxy Authentication Required) response.  The field-value   consists of a challenge that indicates the authentication scheme and   parameters applicable to the proxy for this Request-URI.  The   definition of the header is in [RFC7235], and any applicable HTTP   authentication schemes appear in other RFCs, such as Digest [RFC7616]   and Basic [RFC7617].   The HTTP access authentication process is described in [RFC7235].   This header MUST only be used in response messages related to client-   to-server requests.18.35.  Proxy-Authentication-Info   The Proxy-Authentication-Info response-header is used by the proxy to   communicate some information regarding the successful authentication   to the proxy in the message response in some authentication schemes,   such as the Digest scheme [RFC7616].  The definition of the header is   in [RFC7615], and any applicable HTTP authentication schemes appear   in other RFCs.  This header MUST only be used in response messages   related to client-to-server requests.  This header has hop-by-hop   scope.18.36.  Proxy-Authorization   The Proxy-Authorization request-header field allows the client to   identify itself (or its user) to a proxy that requires   authentication.  The Proxy-Authorization field-value consists of   credentials containing the authentication information of the user   agent for the proxy or realm of the resource being requested.  The   definition of the header is in [RFC7235], and any applicable HTTP   authentication schemes appear in other RFCs, such as Digest [RFC7616]   and Basic [RFC7617].Schulzrinne, et al.          Standards Track                  [Page 159]

RFC 7826                        RTSP 2.0                   December 2016   The HTTP access authentication process is described in [RFC7235].   Unlike Authorization, the Proxy-Authorization header field applies   only to the next-hop proxy.  This header MUST only be used in client-   to-server requests.18.37.  Proxy-Require   The Proxy-Require request-header field is used to indicate proxy-   sensitive features that MUST be supported by the proxy.  Any Proxy-   Require header features that are not supported by the proxy MUST be   negatively acknowledged by the proxy to the client using the   Unsupported header.  The proxy MUST use the 551 (Option Not   Supported) status code in the response.  Any feature tag included in   the Proxy-Require does not apply to the endpoint (server or client).   To ensure that a feature is supported by both proxies and servers,   the tag needs to be included in also a Require header.   SeeSection 18.43 for more details on the mechanics of this message   and a usage example.  See discussion in the proxies section   (Section 15.1) about when to consider that a feature requires proxy   support.   Example of use:      Proxy-Require: play.basic18.38.  Proxy-Supported   The Proxy-Supported general-header field enumerates all the   extensions supported by the proxy using feature tags.  The header   carries the intersection of extensions supported by the forwarding   proxies.  The Proxy-Supported header MAY be included in any request   by a proxy.  It MUST be added by any proxy if the Supported header is   present in a request.  When present in a request, the receiver MUST   copy the received Proxy-Supported header in the response.   The Proxy-Supported header field contains a list of feature tags   applicable to proxies, as described inSection 4.5.  The list is the   intersection of all feature tags understood by the proxies.  To   achieve an intersection, the proxy adding the Proxy-Supported header   includes all proxy feature tags it understands.  Any proxy receiving   a request with the header MUST check the list and remove any feature   tag(s) it does not support.  A Proxy-Supported header present in the   response MUST NOT be modified by the proxies.  These feature tags are   the ones the proxy chains support in general and are not specific to   the request resource.Schulzrinne, et al.          Standards Track                  [Page 160]

RFC 7826                        RTSP 2.0                   December 2016   Example:     C->P1: OPTIONS rtsp://example.com/ RTSP/2.0            Supported: foo, bar, blech            User-Agent: PhonyClient/1.2    P1->P2: OPTIONS rtsp://example.com/ RTSP/2.0            Supported: foo, bar, blech            Proxy-Supported: proxy-foo, proxy-bar, proxy-blech            Via: 2.0 pro.example.com    P2->S:  OPTIONS rtsp://example.com/ RTSP/2.0            Supported: foo, bar, blech            Proxy-Supported: proxy-foo, proxy-blech            Via: 2.0 pro.example.com, 2.0 prox2.example.com     S->C:  RTSP/2.0 200 OK            Supported: foo, bar, baz            Proxy-Supported: proxy-foo, proxy-blech            Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN            Via: 2.0 pro.example.com, 2.0 prox2.example.com18.39.  Public   The Public response-header field lists the set of methods supported   by the response sender.  This header applies to the general   capabilities of the sender, and its only purpose is to indicate the   sender's capabilities to the recipient.  The methods listed may or   may not be applicable to the Request-URI; the Allow header field   (Section 18.6) MAY be used to indicate methods allowed for a   particular URI.   Example of use:      Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN   In the event that there are proxies between the sender and the   recipient of a response, each intervening proxy MUST modify the   Public header field to remove any methods that are not supported via   that proxy.  The resulting Public header field will contain an   intersection of the sender's methods and the methods allowed through   by the intervening proxies.      In general, proxies should allow all methods to transparently pass      through from the sending RTSP agent to the receiving RTSP agent,      but there may be cases where this is not desirable for a given      proxy.  Modification of the Public response-header field by theSchulzrinne, et al.          Standards Track                  [Page 161]

RFC 7826                        RTSP 2.0                   December 2016      intervening proxies ensures that the request sender gets an      accurate response indicating the methods that can be used on the      target agent via the proxy chain.18.40.  Range   The Range general-header specifies a time range in PLAY   (Section 13.4), PAUSE (Section 13.6), SETUP (Section 13.3), and   PLAY_NOTIFY (Section 13.5) requests and responses.  It MAY be   included in GET_PARAMETER requests from the client to the server with   only a Range format and no value to request the current media   position, whether the session is in Play or Ready state in the   included format.  The server SHALL, if supporting the range format,   respond with the current playing point or pause point as the start of   the range.  If an explicit stop point was used in the previous PLAY   request, then that value shall be included as stop point.  Note that   if the server is currently under any type of media playback   manipulation affecting the interpretation of the Range header, like   scale value other than 1, that fact is also required to be included   in any GET_PARAMETER response by including the Scale header to   provide complete information.   The range can be specified in a number of units.  This specification   defines smpte (Section 4.4.1), npt (Section 4.4.2), and clock   (Section 4.4.3) range units.  While octet ranges (Byte Ranges) (seeSection 2.1 of [RFC7233]) and other extended units MAY be used, their   behavior is unspecified since they are not normally meaningful in   RTSP.  Servers supporting the Range header MUST understand the NPT   range format and SHOULD understand the SMPTE range format.  If the   Range header is sent in a time format that is not understood, the   recipient SHOULD return 456 (Header Field Not Valid for Resource) and   include an Accept-Ranges header indicating the supported time formats   for the given resource.   Example:     Range: clock=19960213T143205Z-   The Range header contains a range of one single range format.  A   range is a half-open interval with a start and an end point,   including the start point but excluding the end point.  A range may   either be fully specified with explicit values for start point and   end point or have either the start or end point be implicit.  An   implicit start point indicates the session's pause point, and if no   pause point is set, the start of the content.  An implicit end point   indicates the end of the content.  The usage of both implicit startSchulzrinne, et al.          Standards Track                  [Page 162]

RFC 7826                        RTSP 2.0                   December 2016   and end points is not allowed in the same Range header; however, the   omission of the Range header has that meaning, i.e., from pause point   (or start) until end of content.      As noted, Range headers define half-open intervals.  A range of      A-B starts exactly at time A, but ends just before B.  Only the      start time of a media unit such as a video or audio frame is      relevant.  For example, assume that video frames are generated      every 40 ms.  A range of 10.0-10.1 would include a video frame      starting at 10.0 or later time and would include a video frame      starting at 10.08, even though it lasted beyond the interval.  A      range of 10.0-10.08, on the other hand, would exclude the frame at      10.08.      Please note the difference between NPT timescales' "now" and an      implicit start value.  Implicit values reference the current      pause-point, while "now" is the current time.  In a time-      progressing session with recording (retention for some or full      time), the pause point may be 2 min into the session while now      could be 1 hour into the session.   By default, range intervals increase, where the second point is   larger than the first point.   Example:       Range: npt=10-15   However, range intervals can also decrease if the Scale header (seeSection 18.46) indicates a negative scale value.  For example, this   would be the case when a playback in reverse is desired.   Example:       Scale: -1       Range: npt=15-10   Decreasing ranges are still half-open intervals as described above.   Thus, for range A-B, A is closed and B is open.  In the above   example, 15 is closed and 10 is open.  An exception to this rule is   the case when B=0 is in a decreasing range.  In this case, the range   is closed on both ends, as otherwise there would be no way to reach 0   on a reverse playback for formats that have such a notion, like NPT   and SMPTE.Schulzrinne, et al.          Standards Track                  [Page 163]

RFC 7826                        RTSP 2.0                   December 2016   Example:       Scale: -1       Range: npt=15-0   In this range, both 15 and 0 are closed.   A decreasing range interval without a corresponding negative value in   the Scale header is not valid.18.41.  Referrer   The Referrer request-header field allows the client to specify, for   the server's benefit, the address (URI) of the resource from which   the Request-URI was obtained.  The URI refers to that of the   presentation description, typically retrieved via HTTP.  The Referrer   request-header allows a server to generate lists of back-links to   resources for interest, logging, optimized caching, etc.  It also   allows obsolete or mistyped links to be traced for maintenance.  The   Referrer field MUST NOT be sent if the Request-URI was obtained from   a source that does not have its own URI, such as input from the user   keyboard.   If the field-value is a relative URI, it SHOULD be interpreted   relative to the Request-URI.  The URI MUST NOT include a fragment   identifier.   Because the source of a link might be private information or might   reveal an otherwise private information source, it is strongly   recommended that the user be able to select whether or not the   Referrer field is sent.  For example, a streaming client could have a   toggle switch for openly/anonymously, which would respectively   enable/disable the sending of Referrer and From information.   Clients SHOULD NOT include a Referrer header field in an (non-secure)   RTSP request if the referring page was transferred with a secure   protocol.18.42.  Request-Status   This request-header is used to indicate the end result for requests   that take time to complete, such as PLAY (Section 13.4).  It is sent   in PLAY_NOTIFY (Section 13.5) with the end-of-stream reason to report   how the PLAY request concluded, either in success or in failure.  The   header carries a reference to the request it reports on using the   CSeq number and the Session ID used in the request reported on.  This   is not ensured to be unambiguous due to the fact that the CSeq number   is scoped by the transport connection.  Agents originating requestsSchulzrinne, et al.          Standards Track                  [Page 164]

RFC 7826                        RTSP 2.0                   December 2016   can reduce the issue by using a monotonically increasing counter   across all sequential transports used.  The header provides both a   numerical status code (according toSection 8.1.1) and a human-   readable reason phrase.   Example:   Request-Status: cseq=63 status=500 reason="Media data unavailable"   Proxies that renumber the CSeq header need to perform corresponding   remapping of the cseq parameter in this header when forwarding the   request to the next-hop agent.18.43.  Require   The Require request-header field is used by agents to ensure that the   other endpoint supports features that are required in respect to this   request.  It can also be used to query if the other endpoint supports   certain features; however, the use of the Supported general-header   (Section 18.51) is much more effective in this purpose.  In case any   of the feature tags listed by the Require header are not supported by   the server or client receiving the request, it MUST respond to the   request using the error code 551 (Option Not Supported) and include   the Unsupported header listing those feature tags that are NOT   supported.  This header does not apply to proxies; for the same   functionality with respect to proxies, see the Proxy-Require header   (Section 18.37) with the exception of media-modifying proxies.   Media-modifying proxies, due to their nature of handling media in a   way that is very similar to a server, do need to understand also the   server's features to correctly serve the client.      This is to make sure that the client-server interaction will      proceed without delay when all features are understood by both      sides and only slow down if features are not understood (as in the      example below).  For a well-matched client-server pair, the      interaction proceeds quickly, saving a round trip often required      by negotiation mechanisms.  In addition, it also removes state      ambiguity when the client requires features that the server does      not understand.Schulzrinne, et al.          Standards Track                  [Page 165]

RFC 7826                        RTSP 2.0                   December 2016   Example (Not complete):   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/2.0           CSeq: 302           Require: funky-feature           Funky-Parameter: funkystuff   S->C:   RTSP/2.0 551 Option not supported           CSeq: 302           Unsupported: funky-feature   In this example, "funky-feature" is the feature tag that indicates to   the client that the fictional Funky-Parameter field is required.  The   relationship between "funky-feature" and Funky-Parameter is not   communicated via the RTSP exchange, since that relationship is an   immutable property of "funky-feature" and thus should not be   transmitted with every exchange.   Proxies and other intermediary devices MUST ignore this header.  If a   particular extension requires that intermediate devices support it,   the extension should be tagged in the Proxy-Require field instead   (seeSection 18.37).  See discussion in the proxies section   (Section 15.1) about when to consider that a feature requires proxy   support.18.44.  Retry-After   The Retry-After response-header field can be used with a 503 (Service   Unavailable) or 553 (Proxy Unavailable) response to indicate how long   the service is expected to be unavailable to the requesting client.   This field MAY also be used with any 3rr (Redirection) response to   indicate the minimum time the user agent is asked to wait before   issuing the redirected request.  A response using 413 (Request   Message Body Too Large) when the restriction is temporary MAY also   include the Retry-After header.  The value of this field can be   either an RTSP-date or an integer number of seconds (in decimal)   after the time of the response.   Example:   Retry-After: Fri, 31 Dec 1999 23:59:59 GMT   Retry-After: 120   In the latter example, the delay is 2 minutes.Schulzrinne, et al.          Standards Track                  [Page 166]

RFC 7826                        RTSP 2.0                   December 201618.45.  RTP-Info   The RTP-Info general-header field is used to set RTP-specific   parameters in the PLAY and GET_PARAMETER responses or PLAY_NOTIFY and   GET_PARAMETER requests.  For streams using RTP as transport protocol,   the RTP-Info header SHOULD be part of a 200 response to PLAY.      The exclusion of the RTP-Info in a PLAY response for RTP-      transported media will result in a client needing to synchronize      the media streams using RTCP.  This may have negative impact as      the RTCP can be lost and does not need to be particularly timely      in its arrival.  Also, functionality that informs the client from      which packet a seek has occurred is affected.   The RTP-Info MAY be included in SETUP responses to provide   synchronization information when changing transport parameters, seeSection 13.3.  The RTP-Info header and the Range header MAY be   included in a GET_PARAMETER request from client to server without any   values to request the current playback point and corresponding RTP   synchronization information.  When the RTP-Info header is included in   a Request, the Range header MUST also be included.  The server   response SHALL include both the Range header and the RTP-Info header.   If the session is in Play state, then the value of the Range header   SHALL be filled in with the current playback point and with the   corresponding RTP-Info values.  If the server is in another state, no   values are included in the RTP-Info header.  The header is included   in PLAY_NOTIFY requests with the Notify-Reason of the end of stream   to provide RTP information about the end of the stream.   The header can carry the following parameters:   url:  Indicates the stream URI for which the following RTP parameters         correspond; this URI MUST be the same as used in the SETUP         request for this media stream.  Any relative URI MUST use the         Request-URI as base URI.  This parameter MUST be present.   ssrc: The SSRC to which the RTP timestamp and sequence number         provided applies.  This parameter MUST be present.   seq:  Indicates the sequence number of the first packet of the stream         that is direct result of the request.  This allows clients to         gracefully deal with packets when seeking.  The client uses         this value to differentiate packets that originated before the         seek from packets that originated after the seek.  Note that a         client may not receive the packet with the expressed sequence         number and instead may receive packets with a higher sequence         number due to packet loss or reordering.  This parameter is         RECOMMENDED to be present.Schulzrinne, et al.          Standards Track                  [Page 167]

RFC 7826                        RTSP 2.0                   December 2016   rtptime:  MUST indicate the RTP timestamp value corresponding to the         start time value in the Range response-header or, if not         explicitly given, the implied start point.  The client uses         this value to calculate the mapping of RTP time to NPT or other         media timescale.  This parameter SHOULD be present to ensure         inter-media synchronization is achieved.  There exists no         requirement that any received RTP packet will have the same RTP         timestamp value as the one in the parameter used to establish         synchronization.      A mapping from RTP timestamps to NTP format timestamps (wallclock)      is available via RTCP.  However, this information is not      sufficient to generate a mapping from RTP timestamps to media      clock time (NPT, etc.).  Furthermore, in order to ensure that this      information is available at the necessary time (immediately at      startup or after a seek), and that it is delivered reliably, this      mapping is placed in the RTSP control channel.      In order to compensate for drift for long, uninterrupted      presentations, RTSP clients should additionally map NPT to NTP,      using initial RTCP sender reports to do the mapping, and later      reports to check drift against the mapping.   Example:   Range:npt=3.25-15   RTP-Info:url="rtsp://example.com/foo/audio" ssrc=0A13C760:seq=45102;            rtptime=12345678,url="rtsp://example.com/foo/video"            ssrc=9A9DE123:seq=30211;rtptime=29567112   Lets assume that Audio uses a 16 kHz RTP timestamp clock and Video   a 90 kHz RTP timestamp clock.  Then, the media synchronization is   depicted in the following way.   NPT    3.0---3.1---3.2-X-3.3---3.4---3.5---3.6   Audio               PA A   Video                  V    PV   X: NPT time value = 3.25, from Range header.   A: RTP timestamp value for Audio from RTP-Info header (12345678).   V: RTP timestamp value for Video from RTP-Info header (29567112).   PA: RTP audio packet carrying an RTP timestamp of 12344878, which       corresponds to NPT = (12344878 - A) / 16000 + 3.25 = 3.2   PV: RTP video packet carrying an RTP timestamp of 29573412, which       corresponds to NPT = (29573412 - V) / 90000 + 3.25 = 3.32Schulzrinne, et al.          Standards Track                  [Page 168]

RFC 7826                        RTSP 2.0                   December 201618.46.  Scale   The Scale general-header indicates the requested or used view rate   for the media resource being played back.  A scale value of 1   indicates normal play at the normal forward viewing rate.  If not 1,   the value corresponds to the rate with respect to normal viewing   rate.  For example, a value of 2 indicates twice the normal viewing   rate ("fast forward") and a value of 0.5 indicates half the normal   viewing rate.  In other words, a value of 2 has content time increase   at twice the playback time.  For every second of elapsed (wallclock)   time, 2 seconds of content time will be delivered.  A negative value   indicates reverse direction.  For certain media transports, this may   require certain considerations to work consistently; seeAppendix C.1   for description on how RTP handles this.   The transmitted-data rate SHOULD NOT be changed by selection of a   different scale value.  The resulting bitrate should be reasonably   close to the nominal bitrate of the content for scale = 1.  The   server has to actively manipulate the data when needed to meet the   bitrate constraints.  Implementation of scale changes depends on the   server and media type.  For video, a server may, for example, deliver   only key frames or selected frames.  For audio, it may time-scale the   audio while preserving pitch or, less desirably, deliver fragments of   audio, or completely mute the audio.   The server and content may restrict the range of scale values that it   supports.  The supported values are indicated by the Media-Properties   header (Section 18.29).  The client SHOULD only indicate request   values to be supported.  However, as the values may change as the   content progresses, a requested value may no longer be valid when the   request arrives.  Thus, a non-supported value in a request does not   generate an error, it only forces the server to choose the closest   value.  The response MUST always contain the actual scale value   chosen by the server.   If the server does not implement the possibility to scale, it will   not return a Scale header.  A server supporting scale operations for   PLAY MUST indicate this with the use of the "play.scale" feature tag.   When indicating a negative scale for a reverse playback, the Range   header MUST indicate a decreasing range as described inSection 18.40.   Example of playing in reverse at 3.5 times normal rate:     Scale: -3.5     Range: npt=15-10Schulzrinne, et al.          Standards Track                  [Page 169]

RFC 7826                        RTSP 2.0                   December 201618.47.  Seek-Style   When a client sends a PLAY request with a Range header to perform a   random access to the media, the client does not know if the server   will pick the first media samples or the first random access point   prior to the request range.  Depending on the use case, the client   may have a strong preference.  To express this preference and provide   the client with information on how the server actually acted on that   preference, the Seek-Style general-header is defined.   Seek-Style is a general-header that MAY be included in any PLAY   request to indicate the client's preference for any media stream that   has the random access properties.  The server MUST always include the   header in any PLAY response for media with random access properties   to indicate what policy was applied.  A server that receives an   unknown Seek-Style policy MUST ignore it and select the server   default policy.  A client receiving an unknown policy MUST ignore it   and use the Range header and any media synchronization information as   basis to determine what the server did.   This specification defines the following seek policies that may be   requested (see alsoSection 4.7.1):   RAP:  Random Access Point (RAP) is the behavior of requesting the      server to locate the closest previous random access point that      exists in the media aggregate and deliver from that.  By      requesting a RAP, media quality will be the best possible as all      media will be delivered from a point where full media state can be      established in the media decoder.   CoRAP:  Conditional Random Access Point (CoRAP) is a variant of the      above RAP behavior.  This policy is primarily intended for cases      where there is larger distance between the random access points in      the media.  CoRAP uses the RAP policy if the condition that there      is a Random Access Point closer to the requested start point than      to the current pause point is fulfilled.  Otherwise, no seeking is      performed and playback will continue from the current pause point.      This policy assumes that the media state existing prior to the      pause is usable if delivery is continued.  If the client or server      knows that this is not the fact, the RAP policy should be used.      In other words, in most cases when the client requests a start      point prior to the current pause point, a valid decoding      dependency chain from the media delivered prior to the pause and      to the requested media unit will not exist.  If the server      searched to a random access point, the server MUST return the      CoRAP policy in the Seek-Style header and adjust the Range header      to reflect the position of the selected RAP.  In case the random      access point is farther away and the server chooses to continueSchulzrinne, et al.          Standards Track                  [Page 170]

RFC 7826                        RTSP 2.0                   December 2016      from the current pause point, it MUST include the "Next" policy in      the Seek-Style header and adjust the Range header start point to      the current pause point.   First-Prior:  The first-prior policy will start delivery with the      media unit that has a playout time first prior to the requested      time.  For discrete media, that would only include media units      that would still be rendered at the request time.  For continuous      media, that is media that will be rendered during the requested      start time of the range.   Next:  The next media units after the provided start time of the      range: for continuous framed media, that would mean the first next      frame after the provided time and for discrete media, the first      unit that is to be rendered after the provided time.  The main      usage for this case is when the client knows it has all media up      to a certain point and would like to continue delivery so that a      complete uninterrupted media playback can be achieved.  An example      of such a scenario would be switching from a broadcast/multicast      delivery to a unicast-based delivery.  This policy MUST only be      used on the client's explicit request.   Please note that these expressed preferences exist for optimizing the   startup time or the media quality.  The "Next" policy breaks the   normal definition of the Range header to enable a client to request   media with minimal overlap, although some may still occur for   aggregated sessions.  RAP and First-Prior both fulfill the   requirement of providing media from the requested range and forward.   However, unless RAP is used, the media quality for many media codecs   using predictive methods can be severely degraded unless additional   data is available as, for example, already buffered, or through other   side channels.18.48.  Server   The Server general-header field contains information about the   software used by the origin server to create or handle the request.   This field can contain multiple product tokens and comments   identifying the server and any significant subproducts.  The product   tokens are listed in order of their significance for identifying the   application.   Example:   Server: PhonyServer/1.0Schulzrinne, et al.          Standards Track                  [Page 171]

RFC 7826                        RTSP 2.0                   December 2016   If the response is being forwarded through a proxy, the proxy   application MUST NOT modify the Server response-header.  Instead, it   SHOULD include a Via field (Section 18.57).  If the response is   generated by the proxy, the proxy application MUST return the Server   response-header as previously returned by the server.18.49.  Session   The Session general-header field identifies an RTSP session.  An RTSP   session is created by the server as a result of a successful SETUP   request, and in the response, the session identifier is given to the   client.  The RTSP session exists until destroyed by a TEARDOWN or a   REDIRECT or is timed out by the server.   The session identifier is chosen by the server (seeSection 4.3) and   MUST be returned in the SETUP response.  Once a client receives a   session identifier, it MUST be included in any request related to   that session.  This means that the Session header MUST be included in   a request, using the following methods: PLAY, PAUSE, PLAY_NOTIFY and   TEARDOWN.  It MAY be included in SETUP, OPTIONS, SET_PARAMETER,   GET_PARAMETER, and REDIRECT.  It MUST NOT be included in DESCRIBE.   The Session header MUST NOT be included in the following methods, if   these requests are pipelined and if the session identifier is not yet   known: PLAY, PAUSE, TEARDOWN, SETUP, OPTIONS SET_PARAMETER, and   GET_PARAMETER.   In an RTSP response, the session header MUST be included in methods,   SETUP, PLAY, PAUSE, and PLAY_NOTIFY, and it MAY be included in   methods TEARDOWN and REDIRECT.  If included in the request of the   following methods it MUST also be included in the response: OPTIONS,   GET_PARAMETER, and SET_PARAMETER.  It MUST NOT be included in   DESCRIBE responses.   Note that a session identifier identifies an RTSP session across   transport sessions or connections.  RTSP requests for a given session   can use different URIs (Presentation and media URIs).  Note, that   there are restrictions depending on the session as to which URIs are   acceptable for a given method.  However, multiple "user" sessions for   the same URI from the same client will require use of different   session identifiers.      The session identifier is needed to distinguish several delivery      requests for the same URI coming from the same client.   The response 454 (Session Not Found) MUST be returned if the session   identifier is invalid.Schulzrinne, et al.          Standards Track                  [Page 172]

RFC 7826                        RTSP 2.0                   December 2016   The header MAY include a parameter for session timeout period.  If   not explicitly provided, this value is set to 60 seconds.  As this   affects how often session keep-alives are needed, values smaller than   30 seconds are not recommended.  However, larger-than-default values   can be useful in applications of RTSP that have inactive but   established sessions for longer time periods.      The 60-second value was chosen as the session timeout value as it      results in keep-alive messages that are not too frequent and low      sensitivity to variations in request/response timing.  If one      reduces the timeout value to below 30 seconds, the corresponding      request/response timeout becomes a significant part of the session      timeout.  The 60-second value also allows for reasonably rapid      recovery of committed server resources in case of client failure.18.50.  Speed   The Speed general-header field requests the server to deliver   specific amounts of nominal media time per unit of delivery time,   contingent on the server's ability and desire to serve the media   stream at the given speed.  The client requests the delivery speed to   be within a given range with a lower and upper bound.  The server   SHALL deliver at the highest possible speed within the range, but not   faster than the upper bound, for which the underlying network path   can support the resulting transport data rates.  As long as any speed   value within the given range can be provided, the server SHALL NOT   modify the media quality.  Only if the server is unable to deliver   media at the speed value provided by the lower bound shall it reduce   the media quality.   Implementation of the Speed functionality by the server is OPTIONAL.   The server can indicate its support through a feature tag,   play.speed.  The lack of a Speed header in the response is an   indication of lack of support of this functionality.   The speed parameter values are expressed as a positive decimal value,   e.g., a value of 2.0 indicates that data is to be delivered twice as   fast as normal.  A speed value of zero is invalid.  The range is   specified in the form "lower bound - upper bound".  The lower-bound   value may be smaller or equal to the upper bound.  All speeds may not   be possible to support.  Therefore, the server MAY modify the   requested values to the closest supported.  The actual supported   speed MUST be included in the response.  However, note that the use   cases may vary and that Speed value ranges such as 0.7-0.8, 0.3-2.0,   1.0-2.5, and 2.5-2.5 all have their usages.Schulzrinne, et al.          Standards Track                  [Page 173]

RFC 7826                        RTSP 2.0                   December 2016   Example:     Speed: 1.0-2.5   Use of this header changes the bandwidth used for data delivery.  It   is meant for use in specific circumstances where delivery of the   presentation at a higher or lower rate is desired.  The main use   cases are buffer operations or local scale operations.  Implementers   should keep in mind that bandwidth for the session may be negotiated   beforehand (by means other than RTSP) and, therefore, renegotiation   may be necessary.  To perform Speed operations, the server needs to   ensure that the network path can support the resulting bitrate.   Thus, the media transport needs to support feedback so that the   server can react and adapt to the available bitrate.18.51.  Supported   The Supported general-header enumerates all the extensions supported   by the client or server using feature tags.  The header carries the   extensions supported by the message-sending client or server.  The   Supported header MAY be included in any request.  When present in a   request, the receiver MUST respond with its corresponding Supported   header.  Note that the Supported header is also included in 4xx and   5xx responses.   The Supported header contains a list of feature tags, described inSection 4.5, that are understood by the client or server.  These   feature tags are the ones the server or client supports in general   and are not specific to the request resource.   Example:     C->S:  OPTIONS rtsp://example.com/ RTSP/2.0            Supported: foo, bar, blech            User-Agent: PhonyClient/1.2     S->C:  RTSP/2.0 200 OK            Supported: bar, blech, bazSchulzrinne, et al.          Standards Track                  [Page 174]

RFC 7826                        RTSP 2.0                   December 201618.52.  Terminate-Reason   The Terminate-Reason request-header allows the server, when sending a   REDIRECT or TEARDOWN request, to provide a reason for the session   termination and any additional information.  This specification   identifies three reasons for Redirections and may be extended in the   future:   Server-Admin:  The server needs to be shut down for some      administrative reason.   Session-Timeout:  A client's session has been kept alive for extended      periods of time and the server has determined that it needs to      reclaim the resources associated with this session.   Internal-Error  An internal error that is impossible to recover from      has occurred, forcing the server to terminate the session.   The Server may provide additional parameters containing information   around the redirect.  This specification defines the following ones.   time:  Provides a wallclock time when the server will stop providing      any service.   user-msg:  A UTF-8 text string with a message from the server to the      user.  This message SHOULD be displayed to the user.18.53.  Timestamp   The Timestamp general-header describes when the agent sent the   request.  The value of the timestamp is of significance only to the   agent and may use any timescale.  The responding agent MUST echo the   exact same value and MAY, if it has accurate information about this,   add a floating-point number indicating the number of seconds that has   elapsed since it has received the request.  The timestamp can be used   by the agent to compute the round-trip time to the responding agent   so that it can adjust the timeout value for retransmissions when   running over an unreliable protocol.  It also resolves retransmission   ambiguities for unreliable transport of RTSP.   Note that the present specification provides only for reliable   transport of RTSP messages.  The Timestamp general-header is   specified in case the protocol is extended in the future to use   unreliable transport.Schulzrinne, et al.          Standards Track                  [Page 175]

RFC 7826                        RTSP 2.0                   December 201618.54.  Transport   The Transport general-header indicates which transport protocol is to   be used and configures its parameters such as destination address,   compression, multicast time-to-live and destination port for a single   stream.  It sets those values not already determined by a   presentation description.   A Transport request-header MAY contain a list of transport options   acceptable to the client, in the form of multiple transport   specification entries.  Transport specifications are comma separated   and listed in decreasing order of preference.  Each transport   specification consists of a transport protocol identifier, followed   by any number of parameters separated by semicolons.  A Transport   request-header MAY contain multiple transport specifications using   the same transport protocol identifier.  The server MUST return a   Transport response-header in the response to indicate the values   actually chosen, if any.  If no transport specification is supported,   no transport header is returned and the response MUST use the status   code 461 (Unsupported Transport) (Section 17.4.25).  In case more   than one transport specification was present in the request, the   server MUST return the single transport specification (transport-   spec) that was actually chosen, if any.  The number of transport-spec   entries is expected to be limited as the client will receive guidance   on what configurations are possible from the presentation   description.   The Transport header MAY also be used in subsequent SETUP requests to   change transport parameters.  A server MAY refuse to change   parameters of an existing stream.   The transport protocol identifier defines, for each transport   specification, which transport protocol to use and any related rules.   Each transport protocol identifier defines the parameters that are   required to occur; additional optional parameters MAY occur.  This   flexibility is provided as parameters may be different and provide   different options to the RTSP agent.  A transport specification may   only contain one of any given parameter within it.  A parameter   consists of a name and optionally a value string.  Parameters MAY be   given in any order.  Additionally, a transport specification may only   contain either the unicast or the multicast transport type parameter.   The transport protocol identifier, and all parameters, need to be   understood in a transport specification; if not, the transport   specification MUST be ignored.  An RTSP proxy of any type that uses   or modifies the transport specification, e.g., access proxy or   security proxy, MUST remove specifications with unknown parametersSchulzrinne, et al.          Standards Track                  [Page 176]

RFC 7826                        RTSP 2.0                   December 2016   before forwarding the RTSP message.  If that results in no remaining   transport specification, the proxy SHALL send a 461 (Unsupported   Transport) (Section 17.4.25) response without any Transport header.      The Transport header is restricted to describing a single media      stream.  (RTSP can also control multiple streams as a single      entity.)  Making it part of RTSP rather than relying on a      multitude of session description formats greatly simplifies      designs of firewalls.   The general syntax for the transport protocol identifier is a list of   slash-separated tokens:   Value1/Value2/Value3...   Which, for RTP transports, takes the form:   RTP/profile/lower-transport.   The default value for the "lower-transport" parameters is specific to   the profile.  For RTP/AVP, the default is UDP.   There are two different methods for how to specify where the media   should be delivered for unicast transport:   dest_addr:  The presence of this parameter and its values indicates         the destination address or addresses (host address and port         pairs for IP flows) necessary for the media transport.   No dest_addr:  The lack of the dest_addr parameter indicates that the         server MUST send media to the same address from which the RTSP         messages originates.   The choice of method for indicating where the media is to be   delivered depends on the use case.  In some cases, the only allowed   method will be to use no explicit address indication and have the   server deliver media to the source of the RTSP messages.   For multicast, there are several methods for specifying addresses,   but they are different in how they work compared with unicast:   dest_addr with client picked address:  The address and relevant         parameters, like TTL (scope), for the actual multicast group to         deliver the media to.  There are security implications         (Section 21) with this method that need to be addressed because         an RTSP server can be used as a DoS attacker on an existing         multicast group.Schulzrinne, et al.          Standards Track                  [Page 177]

RFC 7826                        RTSP 2.0                   December 2016   dest_addr using Session Description Information:  The information         included in the transport header can all be coming from the         session description, e.g., the SDP "c=" and "m=" lines.  This         mitigates some of the security issues of the previous methods         as it is the session provider that picks the multicast group         and scope.  The client MUST include the information if it is         available in the session description.   No dest_addr:  The behavior when no explicit multicast group is         present in a request is not defined.   An RTSP proxy will need to take care.  If the media is not desired to   be routed through the proxy, the proxy will need to introduce the   destination indication.   Below are the configuration parameters associated with transport:   General parameters:   unicast / multicast:  This parameter is a mutually exclusive         indication of whether unicast or multicast delivery will be         attempted.  One of the two values MUST be specified.  Clients         that are capable of handling both unicast and multicast         transmission need to indicate such capability by including two         full transport-specs with separate parameters for each.   layers:  The number of multicast layers to be used for this media         stream.  The layers are sent to consecutive addresses starting         at the dest_addr address.  If the parameter is not included, it         defaults to a single layer.   dest_addr:  A general destination address parameter that can contain         one or more address specifications.  Each combination of         protocol/profile/lower transport needs to have the format and         interpretation of its address specification defined.  For         RTP/AVP/UDP and RTP/AVP/TCP, the address specification is a         tuple containing a host address and port.  Note, only a single         destination parameter per transport spec is intended.  The         usage of multiple destinations to distribute a single media to         multiple entities is unspecified.         The client originating the RTSP request MAY specify the         destination address of the stream recipient with the host         address as part of the tuple.  When the destination address is         specified, the recipient may be a different party than the         originator of the request.  To avoid becoming the unwitting         perpetrator of a remote-controlled DoS attack, a server MUST         perform security checks (seeSection 21.2.1) and SHOULD logSchulzrinne, et al.          Standards Track                  [Page 178]

RFC 7826                        RTSP 2.0                   December 2016         such attempts before allowing the client to direct a media         stream to a recipient address not chosen by the server.         Implementations cannot rely on TCP as a reliable means of         client identification.  If the server does not allow the host         address part of the tuple to be set, it MUST return 463         (Destination Prohibited).         The host address part of the tuple MAY be empty, for example         ":58044", in cases when it is desired to specify only the         destination port.  Responses to requests including the         Transport header with a dest_addr parameter SHOULD include the         full destination address that is actually used by the server.         The server MUST NOT remove address information that is already         present in the request when responding, unless the protocol         requires it.   src_addr:  A general source address parameter that can contain one or         more address specifications.  Each combination of         protocol/profile/lower transport needs to have the format and         interpretation of its address specification defined.  For         RTP/AVP/UDP and RTP/AVP/TCP, the address specification is a         tuple containing a host address and port.         This parameter MUST be specified by the server if it transmits         media packets from an address other than the one RTSP messages         are sent to.  This will allow the client to verify the source         address and give it a destination address for its RTCP feedback         packets, if RTP is used.  The address or addresses indicated in         the src_addr parameter SHOULD be used both for the sending and         receiving of the media stream's data packets.  The main reasons         are threefold: First, indicating the port and source address(s)         lets the receiver know where from the packets is expected to         originate.  Second, traversal of NATs is greatly simplified         when traffic is flowing symmetrically over a NAT binding.         Third, certain NAT traversal mechanisms need to know to which         address and port to send so-called "binding packets" from the         receiver to the sender, thus creating an address binding in the         NAT that the sender-to-receiver packet flow can use.            This information may also be available through SDP.            However, since this is more a feature of transport than            media initialization, the authoritative source for this            information should be in the SETUP response.Schulzrinne, et al.          Standards Track                  [Page 179]

RFC 7826                        RTSP 2.0                   December 2016   mode: The mode parameter indicates the methods to be supported for         this session.  The currently defined valid value is "PLAY".  If         not provided, the default is "PLAY".  The "RECORD" value was         defined inRFC 2326; in this specification, it is unspecified         but reserved.  RECORD and other values may be specified in the         future.   interleaved:  The interleaved parameter implies mixing the media         stream with the control stream in whatever protocol is being         used by the control stream, using the mechanism defined inSection 14.  The argument provides the channel number to be         used in the $ block (seeSection 14) and MUST be present.  This         parameter MAY be specified as an interval, e.g.,         interleaved=4-5 in cases where the transport choice for the         media stream requires it, e.g., for RTP with RTCP.  The channel         number given in the request is only a guidance from the client         to the server on what channel number(s) to use.  The server MAY         set any valid channel number in the response.  The declared         channels are bidirectional, so both end parties MAY send data         on the given channel.  One example of such usage is the second         channel used for RTCP, where both server and client send RTCP         packets on the same channel.            This allows RTP/RTCP to be handled similarly to the way that            it is done with UDP, i.e., one channel for RTP and the other            for RTCP.   MIKEY:  This parameter is used in conjunction with transport         specifications that can utilize MIKEY [RFC3830] for security         context establishment.  So far, only the SRTP-based RTP         profiles SAVP and SAVPF can utilize MIKEY, and this is defined         inAppendix C.1.4.1.  This parameter can be included both in         request and response messages.  The binary MIKEY message SHALL         be Base64-encoded [RFC4648] before being included in the value         part of the parameter, where the encoding adheres to the         definition inSection 4 of RFC 4648 and where the padding bits         are set to zero.   Multicast-specific:   ttl:  multicast time-to-live for IPv4.  When included in requests,         the value indicates the TTL value that the client requests the         server to use.  In a response, the value actually being used by         the server is returned.  A server will need to consider what         values that are reasonable and also the authority of the user         to set this value.  Corresponding functions are not needed for         IPv6 as the scoping is part of the IPv6 multicast address         [RFC4291].Schulzrinne, et al.          Standards Track                  [Page 180]

RFC 7826                        RTSP 2.0                   December 2016   RTP-specific:   These parameters MAY only be used if the media-transport protocol is   RTP.   ssrc: The ssrc parameter, if included in a SETUP response, indicates         the RTP SSRC [RFC3550] value(s) that will be used by the media         server for RTP packets within the stream.  The values are         expressed as a slash-separated sequence of SSRC values, each         SSRC expressed as an eight-digit hexadecimal value.         The ssrc parameter MUST NOT be specified in requests.  The         functionality of specifying the ssrc parameter in a SETUP         request is deprecated as it is incompatible with the         specification of RTP [RFC3550].  If the parameter is included         in the Transport header of a SETUP request, the server SHOULD         ignore it, and choose appropriate SSRCs for the stream.  The         server SHOULD set the ssrc parameter in the Transport header of         the response.   RTCP-mux:  Used to negotiate the usage of RTP and RTCP multiplexing         [RFC5761] on a single underlying transport stream/flow.  The         presence of this parameter in a SETUP request indicates the         client's support and requires the server to use RTP and RTCP         multiplexing.  The client SHALL only include one transport         stream in the Transport header specification.  To provide the         server with a choice between using RTP/RTCP multiplexing or         not, two different transport header specifications must be         included.   The parameter setup and connection defined below MAY only be used if   the media-transport protocol of the lower-level transport is   connection oriented (such as TCP).  However, these parameters MUST   NOT be used when interleaving data over the RTSP connection.   setup:  Clients use the setup parameter on the Transport line in a         SETUP request to indicate the roles it wishes to play in a TCP         connection.  This parameter is adapted from [RFC4145].  The use         of this parameter in RTP/AVP/TCP non-interleaved transport is         discussed inAppendix C.2.2; the discussion below is limited to         syntactic issues.  Clients may specify the following values for         the setup parameter:         active:  The client will initiate an outgoing connection.         passive:  The client will accept an incoming connection.Schulzrinne, et al.          Standards Track                  [Page 181]

RFC 7826                        RTSP 2.0                   December 2016         actpass:  The client is willing to accept an incoming            connection or to initiate an outgoing connection.         If a client does not specify a setup value, the "active" value         is assumed.         In response to a client SETUP request where the setup parameter         is set to "active", a server's 2xx reply MUST assign the setup         parameter to "passive" on the Transport header line.         In response to a client SETUP request where the setup parameter         is set to "passive", a server's 2xx reply MUST assign the setup         parameter to "active" on the Transport header line.         In response to a client SETUP request where the setup parameter         is set to "actpass", a server's 2xx reply MUST assign the setup         parameter to "active" or "passive" on the Transport header         line.         Note that the "holdconn" value for setup is not defined for         RTSP use, and MUST NOT appear on a Transport line.   connection:  Clients use the connection parameter in a transport         specification part of the Transport header in a SETUP request         to indicate the client's preference for either reusing an         existing connection between client and server (in which case         the client sets the "connection" parameter to "existing") or         requesting the creation of a new connection between client and         server (in which cast the client sets the "connection"         parameter to "new").  Typically, clients use the "new" value         for the first SETUP request for a URL, and "existing" for         subsequent SETUP requests for a URL.         If a client SETUP request assigns the "new" value to         "connection", the server response MUST also assign the "new"         value to "connection" on the Transport line.         If a client SETUP request assigns the "existing" value to         "connection", the server response MUST assign a value of         "existing" or "new" to "connection" on the Transport line, at         its discretion.         The default value of "connection" is "existing", for all SETUP         requests (initial and subsequent).   The combination of transport protocol, profile and lower transport   needs to be defined.  A number of combinations are defined in theAppendix C.Schulzrinne, et al.          Standards Track                  [Page 182]

RFC 7826                        RTSP 2.0                   December 2016   Below is a usage example, showing a client advertising the capability   to handle multicast or unicast, preferring multicast.  Since this is   a unicast-only stream, the server responds with the proper transport   parameters for unicast.     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0           CSeq: 302           Transport: RTP/AVP;multicast;mode="PLAY",               RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/               "192.0.2.5:3457";mode="PLAY"           Accept-Ranges: npt, smpte, clock           User-Agent: PhonyClient/1.2     S->C: RTSP/2.0 200 OK           CSeq: 302           Date: Fri, 20 Dec 2013 10:20:32 +0000           Session: rQi1hBrGlFdiYld241FxUO           Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/              "192.0.2.5:3457";src_addr="192.0.2.224:6256"/              "192.0.2.224:6257";mode="PLAY"           Accept-Ranges: npt           Media-Properties: Random-Access=0.6, Dynamic,                             Time-Limited=20081128T16590018.55.  Unsupported   The Unsupported response-header lists the features not supported by   the responding RTSP agent.  In the case where the feature was   specified via the Proxy-Require field (Section 18.37), if there is a   proxy on the path between the client and the server, the proxy MUST   send a response message with a status code of 551 (Option Not   Supported).  The request MUST NOT be forwarded.   SeeSection 18.43 for a usage example.Schulzrinne, et al.          Standards Track                  [Page 183]

RFC 7826                        RTSP 2.0                   December 201618.56.  User-Agent   The User-Agent general-header field contains information about the   user agent originating the request or producing a response.  This is   for statistical purposes, the tracing of protocol violations, and   automated recognition of user agents for the sake of tailoring   responses to avoid particular user agent limitations.  User agents   SHOULD include this field with requests.  The field can contain   multiple product tokens and comments identifying the agent and any   subproducts which form a significant part of the user agent.  By   convention, the product tokens are listed in order of their   significance for identifying the application.   Example:   User-Agent: PhonyClient/1.218.57.  Via   The Via general-header field MUST be used by proxies to indicate the   intermediate protocols and recipients between the user agent and the   server on requests and between the origin server and the client on   responses.  The field is intended to be used for tracking message   forwards, avoiding request loops, and identifying the protocol   capabilities of all senders along the request/response chain.   Each of multiple values in the Via field represents each proxy that   has forwarded the message.  Each recipient MUST append its   information such that the end result is ordered according to the   sequence of forwarding applications.  So messages originating with   the client or server do not include the Via header.  The first proxy   or other intermediate adds the header and its information into the   field.  Any additional intermediate adds additional field-values.   Resulting in the server seeing the chains of intermediates in a   client-to-server request and the client seeing the full chain in the   response message.   Proxies (e.g., Access Proxy or Translator Proxy) SHOULD NOT, by   default, forward the names and ports of hosts within the private/   protected region.  This information SHOULD only be propagated if   explicitly enabled.  If not enabled, the via-received of any host   behind the firewall/NAT SHOULD be replaced by an appropriate   pseudonym for that host.Schulzrinne, et al.          Standards Track                  [Page 184]

RFC 7826                        RTSP 2.0                   December 2016   For organizations that have strong privacy requirements for hiding   internal structures, a proxy MAY combine an ordered subsequence of   Via header field entries with identical sent-protocol values into a   single such entry.  Applications MUST NOT combine entries that have   different received-protocol values.18.58.  WWW-Authenticate   The WWW-Authenticate header is specified in [RFC7235]; its usage   depends on the used authentication schemes, such as Digest [RFC7616]   and Basic [RFC7617].  The WWW-Authenticate response-header field MUST   be included in 401 (Unauthorized) response messages.  The field-value   consists of at least one challenge that indicates the authentication   scheme(s) and parameters applicable to the Request-URI.  This header   MUST only be used in response messages related to client to server   requests.   The HTTP access authentication process is described in [RFC7235] with   some clarification inSection 19.1.  User agents are advised to take   special care in parsing the WWW-Authenticate field-value as it might   contain more than one challenge, or if more than one WWW-Authenticate   header field is provided, the contents of a challenge itself can   contain a comma-separated list of authentication parameters.19.  Security Framework   The RTSP security framework consists of two high-level components:   the pure authentication mechanisms based on HTTP authentication and   the message transport protection based on TLS, which is independent   of RTSP.  Because of the similarity in syntax and usage between RTSP   servers and HTTP servers, the security for HTTP is reused to a large   extent.19.1.  RTSP and HTTP Authentication   RTSP and HTTP share common authentication schemes; thus, they follow   the same framework as specified in [RFC7235].  RTSP uses the   corresponding RTSP error codes (401 and 407) and headers (WWW-   Authenticate, Authorization, Proxy-Authenticate, Proxy-Authorization)   by importing the definitions from [RFC7235].  Servers SHOULD   implement both the Basic [RFC7617] and the Digest [RFC7616]   authentication schemes.  Clients MUST implement both the Basic and   the Digest authentication schemes so that a server that requires the   client to authenticate can trust that the capability is present.  If   implementing the Digest authentication scheme, then the additional   considerations specified below inSection 19.1.1 MUST be followed.Schulzrinne, et al.          Standards Track                  [Page 185]

RFC 7826                        RTSP 2.0                   December 2016   It should be stressed that using the HTTP authentication alone does   not provide full RTSP message security.  Therefore, TLS SHOULD be   used; seeSection 19.2.  Any RTSP message containing an Authorization   header using the Basic authentication scheme MUST be using a TLS   connection with confidentiality protection enabled, i.e., no NULL   encryption.   In cases where there is a chain of proxies between the client and the   server, each proxy may individually request the client or previous   proxy to authenticate itself.  This is done using the Proxy-   Authenticate (Section 18.34), the Proxy-Authorization   (Section 18.36), and the Proxy-Authentication-Info (Section 18.35)   headers.  These headers are hop-by-hop headers and are only scoped to   the current connection and hop.  Thus, if a proxy chain exists, a   proxy connecting to another proxy will have to act as a client to   authorize itself towards the next proxy.  The WWW-Authenticate   (Section 18.58), Authorization (Section 18.8), and Authentication-   Info (Section 18.7) headers are end-to-end and MUST NOT be modified   by proxies.   This authentication mechanism works only for client-to-server   requests as currently defined.  This leaves server-to-client request   outside of the context of TLS-based communication more vulnerable to   message-injection attacks on the client.  Based on the server-to-   client methods that exist, the potential risks are various: hijacking   (REDIRECT), denial of service (TEARDOWN and PLAY_NOTIFY), or attacks   with uncertain results (SET_PARAMETER).19.1.1.  Digest Authentication   This section describes the modifications and clarifications required   to apply the HTTP Digest authentication scheme to RTSP.  The RTSP   scheme usage is almost completely identical to that for HTTP   [RFC7616].  These modifications are based on the procedures defined   for SIP 2.0 [RFC3261] (inSection 22.4) but updated to useRFC 7235,RFC 7616 andRFC 7615 instead ofRFC 2617.   Digest authentication uses two additional headers, Authentication-   Info and Proxy-Authentication-Info, that are defined as in [RFC7615].   The rules for Digest authentication follow those defined in   [RFC7616], with "HTTP/1.1" replaced by "RTSP/2.0" in addition to the   following differences:   1.  Use the ABNF specified in the referenced documents, with the       difference that the URI parameter uses the request URI format for       RTSP, i.e. the ABNF element: Request-URI (seeSection 20.2.1).       The domain parameter uses the RTSP-URI-Ref element for absolute       and relative URIs.Schulzrinne, et al.          Standards Track                  [Page 186]

RFC 7826                        RTSP 2.0                   December 2016   2.  If MTags are used, then the example procedure for choosing a       nonce based on ETag can work, based on replacing the ETag with       the MTag.   3.  As a clarification to the calculation of the A2 value for message       integrity assurance in the Digest authentication scheme,       implementers should assume, when the entity-body is empty (that       is, when the RTSP messages have no message body) that the hash of       the message body resolves to the hash of an empty string, or:       H(entity-body), example MD5("") =       "d41d8cd98f00b204e9800998ecf8427e".19.2.  RTSP over TLS   RTSP agents MUST implement RTSP over TLS as defined in this section   and the nextSection 19.3.  RTSP MUST follow the same guidelines with   regard to TLS [RFC5246] usage as specified for HTTP; see [RFC2818].   RTSP over TLS is separated from unsecured RTSP both on the URI level   and the port level.  Instead of using the "rtsp" scheme identifier in   the URI, the "rtsps" scheme identifier MUST be used to signal RTSP   over TLS.  If no port is given in a URI with the "rtsps" scheme, port   322 MUST be used for TLS over TCP/IP.   When a client tries to set up an insecure channel to the server   (using the "rtsp" URI), and the policy for the resource requires a   secure channel, the server MUST redirect the client to the secure   service by sending a 301 redirect response code together with the   correct Location URI (using the "rtsps" scheme).  A user or client   MAY upgrade a non secured URI to a secured by changing the scheme   from "rtsp" to "rtsps".  A server implementing support for "rtsps"   MUST allow this.   It should be noted that TLS allows for mutual authentication (when   using both server and client certificates).  Still, one of the more   common ways TLS is used is to provide only server-side authentication   (often to avoid client certificates).  TLS is then used in addition   to HTTP authentication, providing transport security and server   authentication, while HTTP Authentication is used to authenticate the   client.   RTSP includes the possibility to keep a TCP session up between the   client and server, throughout the RTSP session lifetime.  It may be   convenient to keep the TCP session, not only to save the extra setup   time for TCP, but also the extra setup time for TLS (even if TLS uses   the resume function, there will be almost two extra round trips).   Still, when TLS is used, such behavior introduces extra active state   in the server, not only for TCP and RTSP, but also for TLS.  This may   increase the vulnerability to DoS attacks.Schulzrinne, et al.          Standards Track                  [Page 187]

RFC 7826                        RTSP 2.0                   December 2016   There exists a potential security vulnerability when reusing TCP and   TLS state for different resources (URIs).  If two different hostnames   point at the same IP address, it can be desirable to reuse the TCP/   TLS connection to that server.  In that case, the RTSP agent having   the TCP/TLS connection MUST verify that the server certificate   associated with the connection has a SubjectAltName matching the   hostname present in the URI for the resource an RTSP request is to be   issued.   In addition to these recommendations,Section 19.3 gives further   recommendations of TLS usage with proxies.19.3.  Security and Proxies   The nature of a proxy is often to act as a "man in the middle", while   security is often about preventing the existence of one.  This   section provides clients with the possibility to use proxies even   when applying secure transports (TLS) between the RTSP agents.  The   TLS proxy mechanism allows for server and proxy identification using   certificates.  However, the client cannot be identified based on   certificates.  The client needs to select between using the procedure   specified below or using a TLS connection directly (bypassing any   proxies) to the server.  The choice may be dependent on policies.   In general, there are two categories of proxies: the transparent   proxies (of which the client is not aware) and the non-transparent   proxies (of which the client is aware).  This memo specifies only   non-transparent RTSP proxies, i.e., proxies visible to the RTSP   client and RTSP server.  An infrastructure based on proxies requires   that the trust model be such that both client and server can trust   the proxies to handle the RTSP messages correctly.  To be able to   trust a proxy, the client and server also need to be aware of the   proxy.  Hence, transparent proxies cannot generally be seen as   trusted and will not work well with security (unless they work only   at the transport layer).  In the rest of this section, any reference   to "proxy" will be to a non-transparent proxy, which inspects or   manipulates the RTSP messages.   HTTP Authentication is built on the assumption of proxies and can   provide user-proxy authentication and proxy-proxy/server   authentication in addition to the client-server authentication.   When TLS is applied and a proxy is used, the client will connect to   the proxy's address when connecting to any RTSP server.  This implies   that for TLS, the client will authenticate the proxy server and not   the end server.  Note that when the client checks the serverSchulzrinne, et al.          Standards Track                  [Page 188]

RFC 7826                        RTSP 2.0                   December 2016   certificate in TLS, it MUST check the proxy's identity (URI or   possibly other known identity) against the proxy's identity as   presented in the proxy's Certificate message.   The problem is that for a proxy accepted by the client, the proxy   needs to be provided information on which grounds it should accept   the next-hop certificate.  Both the proxy and the user may have rules   for this, and the user should have the possibility to select the   desired behavior.  To handle this case, the Accept-Credentials header   (seeSection 18.2) is used, where the client can request the proxy or   proxies to relay back the chain of certificates used to authenticate   any intermediate proxies as well as the server.  The assumption that   the proxies are viewed as trusted gives the user a possibility to   enforce policies on each trusted proxy of whether it should accept   the next agent in the chain.  However, it should be noted that not   all deployments will return the chain of certificates used to   authenticate any intermediate proxies as well as the server.  An   operator of such a deployment may want to hide its topology from the   client.  It should be noted well that the client does not have any   insight into the proxy's operation.  Even if the proxy is trusted, it   can still return an incomplete chain of certificates.   A proxy MUST use TLS for the next hop if the RTSP request includes an   "rtsps" URI.  TLS MAY be applied on intermediate links (e.g., between   client and proxy or between proxy and proxy) even if the resource and   the end server are not required to use it.  The chain of proxies used   by a client to reach a server and its TLS sessions MUST have   commensurate security.  Therefore, a proxy MUST, when initiating the   next-hop TLS connection, use the incoming TLS connections cipher-   suite list, only modified by removing any cipher suites that the   proxy does not support.  In case a proxy fails to establish a TLS   connection due to cipher-suite mismatch between proxy and next-hop   proxy or server, this is indicated using error code 472 (Failure to   Establish Secure Connection).19.3.1.  Accept-Credentials   The Accept-Credentials header can be used by the client to distribute   simple authorization policies to intermediate proxies.  The client   includes the Accept-Credentials header to dictate how the proxy   treats the server / next proxy certificate.  There are currently   three methods defined:   Any:  With "any", the proxy (or proxies) MUST accept whatever         certificate is presented.  Of course, this is not a recommended         option to use, but it may be useful in certain circumstances         (such as testing).Schulzrinne, et al.          Standards Track                  [Page 189]

RFC 7826                        RTSP 2.0                   December 2016   Proxy:  For the "proxy" method, the proxy (or proxies) MUST use its         own policies to validate the certificate and decide whether or         not to accept it.  This is convenient in cases where the user         has a strong trust relation with the proxy.  Reasons why a         strong trust relation may exist are personal/company proxy or         the proxy has an out-of-band policy configuration mechanism.   User: For the "user" method, the proxy (or proxies) MUST send         credential information about the next hop to the client for         authorization.  The client can then decide whether or not the         proxy should accept the certificate.  SeeSection 19.3.2 for         further details.   If the Accept-Credentials header is not included in the RTSP request   from the client, then the "Proxy" method MUST be used as default.  If   a method other than the "Proxy" is to be used, then the Accept-   Credentials header MUST be included in all of the RTSP requests from   the client.  This is because it cannot be assumed that the proxy   always keeps the TLS state or the user's previous preference between   different RTSP messages (in particular, if the time interval between   the messages is long).   With the "Any" and "Proxy" methods, the proxy will apply the policy   as defined for each method.  If the policy does not accept the   credentials of the next hop, the proxy MUST respond with a message   using status code 471 (Connection Credentials Not Accepted).   An RTSP request in the direction server to client MUST NOT include   the Accept-Credentials header.  As for the non-secured communication,   the possibility for these requests depends on the presence of a   client established connection.  However, if the server-to-client   request is in relation to a session established over a TLS secured   channel, it MUST be sent in a TLS secured connection.  That secured   connection MUST also be the one used by the last client-to-server   request.  If no such transport connection exists at the time when the   server desires to send the request, the server MUST discard the   message.   Further policies MAY be defined and registered, but this should be   done with caution.19.3.2.  User-Approved TLS Procedure   For the "User" method, each proxy MUST perform the following   procedure for each RTSP request:   o  Set up the TLS session to the next hop if not already present      (i.e., run the TLS handshake, but do not send the RTSP request).Schulzrinne, et al.          Standards Track                  [Page 190]

RFC 7826                        RTSP 2.0                   December 2016   o  Extract the peer certificate chain for the TLS session.   o  Check if a matching identity and hash of the peer certificate are      present in the Accept-Credentials header.  If present, send the      message to the next hop and conclude these procedures.  If not, go      to the next step.   o  The proxy responds to the RTSP request with a 470 or 407 response      code.  The 407 response code MAY be used when the proxy requires      both user and connection authorization from user or client.  In      this message the proxy MUST include a Connection-Credentials      header, seeSection 18.13, with the next hop's identity and      certificate.   The client MUST upon receiving a 470 (Connection Authorization   Required) or 407 (Proxy Authentication Required) response with   Connection-Credentials header take the decision on whether or not to   accept the certificate (if it cannot do so, the user SHOULD be   consulted).  Using IP addresses in the next-hop URI and certificates   rather than domain names makes it very difficult for a user to   determine whether or not it should approve the next hop.  Proxies are   RECOMMENDED to use domain names to identify themselves in URIs and in   the certificates.  If the certificate is accepted, the client has to   again send the RTSP request.  In that request, the client has to   include the Accept-Credentials header including the hash over the   DER-encoded certificate for all trusted proxies in the chain.Schulzrinne, et al.          Standards Track                  [Page 191]

RFC 7826                        RTSP 2.0                   December 2016   Example:   C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0         CSeq: 2         Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/                    "192.0.2.5:4589"         Accept-Ranges: npt, smpte, clock         Accept-Credentials: User   P->C: RTSP/2.0 470 Connection Authorization Required         CSeq: 2         Connection-Credentials: "rtsps://test.example.org";         MIIDNTCCAp...   C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0         CSeq: 3         Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/                    "192.0.2.5:4589"         Accept-Credentials: User "rtsps://test.example.org";sha-256;         dPYD7txpoGTbAqZZQJ+vaeOkyH4=         Accept-Ranges: npt, smpte, clock   P->S: SETUP rtsps://test.example.org/secret/audio RTSP/2.0         CSeq: 3         Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/                    "192.0.2.5:4589"         Via: RTSP/2.0 proxy.example.org         Accept-Credentials: User "rtsps://test.example.org";sha-256;         dPYD7txpoGTbAqZZQJ+vaeOkyH4=         Accept-Ranges: npt, smpte, clock   One implication of this process is that the connection for secured   RTSP messages may take significantly more round-trip times for the   first message.  A complete extra message exchange between the proxy   connecting to the next hop and the client results because of the   process for approval for each hop.  However, if each message contains   the chain of proxies that the requester accepts, the remaining   message exchange should not be delayed.  The procedure of including   the credentials in each request rather than building state in each   proxy avoids the need for revocation procedures.20.  Syntax   The RTSP syntax is described in an Augmented Backus-Naur Form (ABNF)   as defined inRFC 5234 [RFC5234].  It uses the basic definitions   present inRFC 5234.Schulzrinne, et al.          Standards Track                  [Page 192]

RFC 7826                        RTSP 2.0                   December 2016   Please note that ABNF strings, e.g., "Accept", are case insensitive   as specified inSection 2.3 of RFC 5234.   The RTSP syntax makes use of the ISO 10646 character set in UTF-8   encoding [RFC3629].20.1.  Base Syntax   RTSP header values can be folded onto multiple lines if the   continuation line begins with a space or horizontal tab.  All linear   whitespace, including folding, has the same semantics as SP.  A   recipient MAY replace any linear whitespace with a single SP before   interpreting the field-value or forwarding the message downstream.   The SWS construct is used when linear whitespace is optional,   generally between tokens and separators.   To separate the header name from the rest of value, a colon is used,   which, by the above rule, allows whitespace before, but no line   break, and whitespace after, including a line break.  The HCOLON   defines this construct.   OCTET           =  %x00-FF ; any 8-bit sequence of data   CHAR            =  %x01-7F ; any US-ASCII character (octets 1 - 127)   UPALPHA         =  %x41-5A ; any US-ASCII uppercase letter "A".."Z"   LOALPHA         =  %x61-7A ; any US-ASCII lowercase letter "a".."z"   ALPHA           =  UPALPHA / LOALPHA   DIGIT           =  %x30-39 ; any US-ASCII digit "0".."9"   CTL             =  %x00-1F / %x7F  ; any US-ASCII control character                      ; (octets 0 - 31) and DEL (127)   CR              =  %x0D ; US-ASCII CR, carriage return (13)   LF              =  %x0A  ; US-ASCII LF, linefeed (10)   SP              =  %x20  ; US-ASCII SP, space (32)   HT              =  %x09  ; US-ASCII HT, horizontal-tab (9)   BACKSLASH       =  %x5C  ; US-ASCII backslash (92)   CRLF            =  CR LF   LWS             =  [CRLF] 1*( SP / HT ) ; Line-breaking whitespace   SWS             =  [LWS] ; Separating whitespace   HCOLON          =  *( SP / HT ) ":" SWS   TEXT            =  %x20-7E / %x80-FF  ; any OCTET except CTLs   tspecials       =  "(" / ")" / "<" / ">" / "@"                   /  "," / ";" / ":" / BACKSLASH / DQUOTE                   /  "/" / "[" / "]" / "?" / "="                   /  "{" / "}" / SP / HT   token           =  1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39                   /  %x41-5A / %x5E-7A / %x7C / %x7E)                      ; 1*<any CHAR except CTLs or tspecials>   quoted-string   =  ( DQUOTE *qdtext DQUOTE )Schulzrinne, et al.          Standards Track                  [Page 193]

RFC 7826                        RTSP 2.0                   December 2016   qdtext          = %x20-21 / %x23-5B / %x5D-7E / quoted-pair                   / UTF8-NONASCII                   ; No DQUOTE and no "\"   quoted-pair     = "\\" / ( "\" DQUOTE )   ctext           =  %x20-27 / %x2A-7E                   /  %x80-FF  ; any OCTET except CTLs, "(" and ")"   generic-param   =  token [ EQUAL gen-value ]   gen-value       =  token / host / quoted-string   safe            =  "$" / "-" / "_" / "." / "+"   extra           =  "!" / "*" / "'" / "(" / ")" / ","   rtsp-extra      =  "!" / "*" / "'" / "(" / ")"   HEX             =  DIGIT / "A" / "B" / "C" / "D" / "E" / "F"                   /  "a" / "b" / "c" / "d" / "e" / "f"   LHEX            =  DIGIT /  "a" / "b" / "c" / "d" / "e" / "f"                      ; lowercase "a-f" Hex   reserved        =  ";" / "/" / "?" / ":" / "@" / "&" / "="   unreserved      =  ALPHA / DIGIT / safe / extra   rtsp-unreserved  =  ALPHA / DIGIT / safe / rtsp-extra   base64          =  *base64-unit [base64-pad]   base64-unit     =  4base64-char   base64-pad      =  (2base64-char "==") / (3base64-char "=")   base64-char     =  ALPHA / DIGIT / "+" / "/"   SLASH    =  SWS "/" SWS ; slash   EQUAL    =  SWS "=" SWS ; equal   LPAREN   =  SWS "(" SWS ; left parenthesis   RPAREN   =  SWS ")" SWS ; right parenthesis   COMMA    =  SWS "," SWS ; comma   SEMI     =  SWS ";" SWS ; semicolon   COLON    =  SWS ":" SWS ; colon   MINUS    =  SWS "-" SWS ; minus/dash   LDQUOT   =  SWS DQUOTE ; open double quotation mark   RDQUOT   =  DQUOTE SWS ; close double quotation mark   RAQUOT   =  ">" SWS ; right angle quote   LAQUOT   =  SWS "<" ; left angle quote   TEXT-UTF8char    =  %x21-7E / UTF8-NONASCII   UTF8-NONASCII    = UTF8-2 / UTF8-3 / UTF8-4   UTF8-1           = <As defined inRFC 3629>   UTF8-2           = <As defined inRFC 3629>   UTF8-3           = <As defined inRFC 3629>   UTF8-4           = <As defined inRFC 3629>   UTF8-tail        = <As defined inRFC 3629>Schulzrinne, et al.          Standards Track                  [Page 194]

RFC 7826                        RTSP 2.0                   December 2016   POS-FLOAT        = 1*12DIGIT ["." 1*9DIGIT]   FLOAT            = ["-"] POS-FLOAT20.2.  RTSP Protocol Definition20.2.1.  Generic Protocol Elements   RTSP-IRI       =  schemes ":" IRI-rest   IRI-rest       =  ihier-part [ "?" iquery ]   ihier-part     =  "//" iauthority ipath-abempty   RTSP-IRI-ref   =  RTSP-IRI / irelative-ref   irelative-ref  =  irelative-part [ "?" iquery ]   irelative-part =  "//" iauthority ipath-abempty                     / ipath-absolute                     / ipath-noscheme                     / ipath-empty   iauthority     =  < As defined inRFC 3987>   ipath          =  ipath-abempty   ; begins with "/" or is empty                     / ipath-absolute  ; begins with "/" but not "//"                     / ipath-noscheme  ; begins with a non-colon segment                     / ipath-rootless  ; begins with a segment                     / ipath-empty     ; zero characters   ipath-abempty   =  *( "/" isegment )   ipath-absolute  =  "/" [ isegment-nz *( "/" isegment ) ]   ipath-noscheme  =  isegment-nz-nc *( "/" isegment )   ipath-rootless  =  isegment-nz *( "/" isegment )   ipath-empty     =  0<ipchar>   isegment        =  *ipchar [";" *ipchar]   isegment-nz     =  1*ipchar [";" *ipchar]                      / ";" *ipchar   isegment-nz-nc  =  (1*ipchar-nc [";" *ipchar-nc])                      / ";" *ipchar-nc                      ; non-zero-length segment without any colon ":"                      ; No parameter (; delimited) inside path.   ipchar         =  iunreserved / pct-encoded / sub-delims / ":" / "@"   ipchar-nc      =  iunreserved / pct-encoded / sub-delims / "@"                     ; sub-delims is different fromRFC 3987                     ; not including ";"   iquery         =  < As defined inRFC 3987>   iunreserved    =  < As defined inRFC 3987>   pct-encoded    =  < As defined inRFC 3987>Schulzrinne, et al.          Standards Track                  [Page 195]

RFC 7826                        RTSP 2.0                   December 2016   RTSP-URI       =  schemes ":" URI-rest   RTSP-REQ-URI   =  schemes ":" URI-req-rest   RTSP-URI-Ref   =  RTSP-URI / RTSP-Relative   RTSP-REQ-Ref   =  RTSP-REQ-URI / RTSP-REQ-Rel   schemes        =  "rtsp" / "rtsps" / scheme   scheme         =  < As defined inRFC 3986>   URI-rest       =  hier-part [ "?" query ]   URI-req-rest   =  hier-part [ "?" query ]                     ; Note fragment part not allowed in requests   hier-part      =  "//" authority path-abempty   RTSP-Relative  =  relative-part [ "?" query ]   RTSP-REQ-Rel   =  relative-part [ "?" query ]   relative-part  =  "//" authority path-abempty                     / path-absolute                     / path-noscheme                     / path-empty   authority      =  < As defined inRFC 3986>   query          =  < As defined inRFC 3986>   path           =  path-abempty    ; begins with "/" or is empty                     / path-absolute ; begins with "/" but not "//"                     / path-noscheme ; begins with a non-colon segment                     / path-rootless ; begins with a segment                     / path-empty    ; zero characters   path-abempty   =  *( "/" segment )   path-absolute  =  "/" [ segment-nz *( "/" segment ) ]   path-noscheme  =  segment-nz-nc *( "/" segment )   path-rootless  =  segment-nz *( "/" segment )   path-empty     =  0<pchar>   segment        =  *pchar [";" *pchar]   segment-nz     =  ( 1*pchar [";" *pchar]) / (";" *pchar)   segment-nz-nc  =  ( 1*pchar-nc [";" *pchar-nc]) / (";" *pchar-nc)                     ; non-zero-length segment without any colon ":"                     ; No parameter (; delimited) inside path.   pchar          =  unreserved / pct-encoded / sub-delims / ":" / "@"   pchar-nc       =  unreserved / pct-encoded / sub-delims / "@"   sub-delims     =  "!" / "$" / "&" / "'" / "(" / ")"                     / "*" / "+" / "," / "="                     ; sub-delims is different fromRFC 3986/3987                     ; not including ";"Schulzrinne, et al.          Standards Track                  [Page 196]

RFC 7826                        RTSP 2.0                   December 2016   smpte-range        =  smpte-type [EQUAL smpte-range-spec]                         ; Seesection 4.4   smpte-range-spec   =  ( smpte-time "-" [ smpte-time ] )                      /  ( "-" smpte-time )   smpte-type         =  "smpte" / "smpte-30-drop"                      /  "smpte-25" / smpte-type-extension                      ; other timecodes may be added   smpte-type-extension  =  "smpte" token   smpte-time         =  1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT                         [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]   npt-range        =  "npt" [EQUAL npt-range-spec]   npt-range-spec   =  ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )   npt-time         =  "now" / npt-sec / npt-hhmmss / npt-hhmmss-comp   npt-sec          =  1*19DIGIT [ "." 1*9DIGIT ]   npt-hhmmss       =  npt-hh ":" npt-mm ":" npt-ss [ "." 1*9DIGIT ]   npt-hh           =  2*19DIGIT   ; any positive number   npt-mm           =  2*2DIGIT  ; 0-59   npt-ss           =  2*2DIGIT  ; 0-59   npt-hhmmss-comp  =  npt-hh-comp ":" npt-mm-comp ":" npt-ss-comp                       [ "." 1*9DIGIT ] ; Compatibility format   npt-hh-comp      =  1*19DIGIT   ; any positive number   npt-mm-comp      =  1*2DIGIT  ; 0-59   npt-ss-comp      =  1*2DIGIT  ; 0-59   utc-range        =  "clock" [EQUAL utc-range-spec]   utc-range-spec   =  ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )   utc-time         =  utc-date "T" utc-clock "Z"   utc-date         =  8DIGIT   utc-clock        =  6DIGIT [ "." 1*9DIGIT ]   feature-tag       =  token   session-id        =  1*256( ALPHA / DIGIT / safe )   extension-header  =  header-name HCOLON header-value   header-name       =  token   header-value      =  *(TEXT-UTF8char / LWS)Schulzrinne, et al.          Standards Track                  [Page 197]

RFC 7826                        RTSP 2.0                   December 201620.2.2.  Message Syntax   RTSP-message  = Request / Response  ; RTSP/2.0 messages   Request       = Request-Line                   *((general-header                   /  request-header                   /  message-body-header) CRLF)                   CRLF                   [ message-body-data ]   Response     = Status-Line                  *((general-header                  /  response-header                  /  message-body-header) CRLF)                  CRLF                  [ message-body-data ]   Request-Line = Method SP Request-URI SP RTSP-Version CRLF   Status-Line  = RTSP-Version SP Status-Code SP Reason-Phrase CRLF   Method  =  "DESCRIBE"           /  "GET_PARAMETER"           /  "OPTIONS"           /  "PAUSE"           /  "PLAY"           /  "PLAY_NOTIFY"           /  "REDIRECT"           /  "SETUP"           /  "SET_PARAMETER"           /  "TEARDOWN"           /  extension-method   extension-method  =  token   Request-URI  =  "*" / RTSP-REQ-URI   RTSP-Version =  "RTSP/" 1*DIGIT "." 1*DIGIT   message-body-data = 1*OCTET   Status-Code  =  "100"  ; Continue                /  "200"  ; OK                /  "301"  ; Moved Permanently                /  "302"  ; Found                /  "303"  ; See Other                /  "304"  ; Not Modified                /  "305"  ; Use ProxySchulzrinne, et al.          Standards Track                  [Page 198]

RFC 7826                        RTSP 2.0                   December 2016                /  "400"  ; Bad Request                /  "401"  ; Unauthorized                /  "402"  ; Payment Required                /  "403"  ; Forbidden                /  "404"  ; Not Found                /  "405"  ; Method Not Allowed                /  "406"  ; Not Acceptable                /  "407"  ; Proxy Authentication Required                /  "408"  ; Request Timeout                /  "410"  ; Gone                /  "412"  ; Precondition Failed                /  "413"  ; Request Message Body Too Large                /  "414"  ; Request-URI Too Long                /  "415"  ; Unsupported Media Type                /  "451"  ; Parameter Not Understood                /  "452"  ; reserved                /  "453"  ; Not Enough Bandwidth                /  "454"  ; Session Not Found                /  "455"  ; Method Not Valid In This State                /  "456"  ; Header Field Not Valid for Resource                /  "457"  ; Invalid Range                /  "458"  ; Parameter Is Read-Only                /  "459"  ; Aggregate Operation Not Allowed                /  "460"  ; Only Aggregate Operation Allowed                /  "461"  ; Unsupported Transport                /  "462"  ; Destination Unreachable                /  "463"  ; Destination Prohibited                /  "464"  ; Data Transport Not Ready Yet                /  "465"  ; Notification Reason Unknown                /  "466"  ; Key Management Error                /  "470"  ; Connection Authorization Required                /  "471"  ; Connection Credentials Not Accepted                /  "472"  ; Failure to Establish Secure Connection                /  "500"  ; Internal Server Error                /  "501"  ; Not Implemented                /  "502"  ; Bad Gateway                /  "503"  ; Service Unavailable                /  "504"  ; Gateway Timeout                /  "505"  ; RTSP Version Not Supported                /  "551"  ; Option Not Supported                /  "553"  ; Proxy Unavailable                /  extension-code   extension-code  =  3DIGIT   Reason-Phrase   =  1*(TEXT-UTF8char / HT / SP)Schulzrinne, et al.          Standards Track                  [Page 199]

RFC 7826                        RTSP 2.0                   December 2016   rtsp-header     = general-header                   / request-header                   / response-header                   / message-body-header   general-header  =  Accept-Ranges                   /  Cache-Control                   /  Connection                   /  CSeq                   /  Date                   /  Media-Properties                   /  Media-Range                   /  Pipelined-Requests                   /  Proxy-Supported                   /  Range                   /  RTP-Info                   /  Scale                   /  Seek-Style                   /  Server                   /  Session                   /  Speed                   /  Supported                   /  Timestamp                   /  Transport                   /  User-Agent                   /  Via                   /  extension-header   request-header  =  Accept                   /  Accept-Credentials                   /  Accept-Encoding                   /  Accept-Language                   /  Authorization                   /  Bandwidth                   /  Blocksize                   /  From                   /  If-Match                   /  If-Modified-Since                   /  If-None-Match                   /  Notify-Reason                   /  Proxy-Authorization                   /  Proxy-Require                   /  Referrer                   /  Request-Status                   /  Require                   /  Terminate-Reason                   /  extension-headerSchulzrinne, et al.          Standards Track                  [Page 200]

RFC 7826                        RTSP 2.0                   December 2016   response-header  =  Authentication-Info                    /  Connection-Credentials                    /  Location                    /  MTag                    /  Proxy-Authenticate                    /  Proxy-Authentication-Info                    /  Public                    /  Retry-After                    /  Unsupported                    /  WWW-Authenticate                    /  extension-header   message-body-header    =  Allow                    /  Content-Base                    /  Content-Encoding                    /  Content-Language                    /  Content-Length                    /  Content-Location                    /  Content-Type                    /  Expires                    /  Last-Modified                    /  extension-header20.2.3.  Header Syntax   Accept            =  "Accept" HCOLON                        [ accept-range *(COMMA accept-range) ]   accept-range      =  media-type-range [SEMI accept-params]   media-type-range  =  ( "*/*"                        / ( m-type SLASH "*" )                        / ( m-type SLASH m-subtype )                       ) *( SEMI m-parameter )   accept-params     =  "q" EQUAL qvalue *(SEMI generic-param )   qvalue            =  ( "0" [ "." *3DIGIT ] )                     /  ( "1" [ "." *3("0") ] )   Accept-Credentials =  "Accept-Credentials" HCOLON cred-decision   cred-decision     =  ("User" [LWS cred-info])                     /  "Proxy"                     /  "Any"                     /  (token [LWS 1*header-value])                                     ; For future extensions   cred-info         =  cred-info-data *(COMMA cred-info-data)   cred-info-data    =  DQUOTE RTSP-REQ-URI DQUOTE SEMI hash-alg                        SEMI base64   hash-alg          =  "sha-256" / extension-alg   extension-alg     =  token   Accept-Encoding   =  "Accept-Encoding" HCOLONSchulzrinne, et al.          Standards Track                  [Page 201]

RFC 7826                        RTSP 2.0                   December 2016                        [ encoding *(COMMA encoding) ]   encoding          =  codings [SEMI accept-params]   codings           =  content-coding / "*"   content-coding    =  "identity" / token   Accept-Language   =  "Accept-Language" HCOLON                        language *(COMMA language)   language          =  language-range [SEMI accept-params]   language-range    =  language-tag / "*"   language-tag      =  primary-tag *( "-" subtag )   primary-tag       =  1*8ALPHA   subtag            =  1*8ALPHA   Accept-Ranges     =  "Accept-Ranges" HCOLON acceptable-ranges   acceptable-ranges =  (range-unit *(COMMA range-unit))   range-unit        =  "npt" / "smpte" / "smpte-30-drop" / "smpte-25"                        / "clock" / extension-format   extension-format  =  token   Allow             =  "Allow" HCOLON Method *(COMMA Method)   Authentication-Info = "Authentication-Info" HCOLON auth-param-list   auth-param-list   =  <As the Authentication-Info element inRFC 7615>   Authorization     =  "Authorization" HCOLON credentials   credentials       =  <As defined byRFC 7235>   Bandwidth         =  "Bandwidth" HCOLON 1*19DIGIT   Blocksize         =  "Blocksize" HCOLON 1*9DIGIT   Cache-Control     =  "Cache-Control" HCOLON cache-directive                        *(COMMA cache-directive)   cache-directive   =  cache-rqst-directive                     /  cache-rspns-directive   cache-rqst-directive =  "no-cache"                        /  "max-stale" [EQUAL delta-seconds]                        /  "min-fresh" EQUAL delta-seconds                        /  "only-if-cached"                        /  cache-extension   cache-rspns-directive =  "public"                            /  "private"                            /  "no-cache"                            /  "no-transform"                            /  "must-revalidate"                            /  "proxy-revalidate"                            /  "max-age" EQUAL delta-seconds                            /  cache-extension   cache-extension   =  token [EQUAL (token / quoted-string)]   delta-seconds     =  1*19DIGITSchulzrinne, et al.          Standards Track                  [Page 202]

RFC 7826                        RTSP 2.0                   December 2016   Connection         =  "Connection" HCOLON connection-token                         *(COMMA connection-token)   connection-token   =  "close" / token   Connection-Credentials = "Connection-Credentials" HCOLON cred-chain   cred-chain         =  DQUOTE RTSP-REQ-URI DQUOTE SEMI base64   Content-Base       =  "Content-Base" HCOLON RTSP-URI   Content-Encoding   =  "Content-Encoding" HCOLON                         content-coding *(COMMA content-coding)   Content-Language   =  "Content-Language" HCOLON                         language-tag *(COMMA language-tag)   Content-Length     =  "Content-Length" HCOLON 1*19DIGIT   Content-Location   =  "Content-Location" HCOLON RTSP-REQ-Ref   Content-Type       =  "Content-Type" HCOLON media-type   media-type         =  m-type SLASH m-subtype *(SEMI m-parameter)   m-type             =  discrete-type / composite-type   discrete-type      =  "text" / "image" / "audio" / "video"                      /  "application" / extension-token   composite-type   =  "message" / "multipart" / extension-token   extension-token  =  ietf-token / x-token   ietf-token       =  token   x-token          =  "x-" token   m-subtype        =  extension-token / iana-token   iana-token       =  token   m-parameter      =  m-attribute EQUAL m-value   m-attribute      =  token   m-value          =  token / quoted-string   CSeq           =  "CSeq" HCOLON cseq-nr   cseq-nr        =  1*9DIGIT   Date           =  "Date" HCOLON RTSP-date   RTSP-date      =  date-time ;   date-time      =  <As defined inRFC 5322>   Expires        =  "Expires" HCOLON RTSP-date   From           =  "From" HCOLON from-spec   from-spec      =  ( name-addr / addr-spec ) *( SEMI from-param )   name-addr      =  [ display-name ] LAQUOT addr-spec RAQUOT   addr-spec      =  RTSP-REQ-URI / absolute-URI   absolute-URI   =  < As defined inRFC 3986>   display-name   =  *(token LWS) / quoted-string   from-param     =  tag-param / generic-param   tag-param      =  "tag" EQUAL token   If-Match       =  "If-Match" HCOLON ("*" / message-tag-list)   message-tag-list =  message-tag *(COMMA message-tag)   message-tag      =  [ weak ] opaque-tag   weak             =  "W/"   opaque-tag       =  quoted-stringSchulzrinne, et al.          Standards Track                  [Page 203]

RFC 7826                        RTSP 2.0                   December 2016   If-Modified-Since  =  "If-Modified-Since" HCOLON RTSP-date   If-None-Match    =  "If-None-Match" HCOLON ("*" / message-tag-list)   Last-Modified    =  "Last-Modified" HCOLON RTSP-date   Location         =  "Location" HCOLON RTSP-REQ-URI   Media-Properties = "Media-Properties" HCOLON [media-prop-list]   media-prop-list  = media-prop-value *(COMMA media-prop-value)   media-prop-value = ("Random-Access" [EQUAL POS-FLOAT])                    / "Beginning-Only"                    / "No-Seeking"                    / "Immutable"                    / "Dynamic"                    / "Time-Progressing"                    / "Unlimited"                    / ("Time-Limited" EQUAL utc-time)                    / ("Time-Duration" EQUAL POS-FLOAT)                    / ("Scales" EQUAL scale-value-list)                    / media-prop-ext   media-prop-ext   = token [EQUAL (1*rtsp-unreserved / quoted-string)]   scale-value-list = DQUOTE scale-entry *(COMMA scale-entry) DQUOTE   scale-entry      = scale-value / (scale-value COLON scale-value)   scale-value      = FLOAT   Media-Range      = "Media-Range" HCOLON [ranges-list]   ranges-list      =  ranges-spec *(COMMA ranges-spec)   MTag             =  "MTag" HCOLON message-tag   Notify-Reason    = "Notify-Reason" HCOLON Notify-Reas-val   Notify-Reas-val  = "end-of-stream"                    / "media-properties-update"                    / "scale-change"                    / Notify-Reason-extension   Notify-Reason-extension  = token   Pipelined-Requests = "Pipelined-Requests" HCOLON startup-id   startup-id  = 1*8DIGIT   Proxy-Authenticate =  "Proxy-Authenticate" HCOLON challenge-list   challenge-list     = <As defined by the WWW-Authenticate inRFC 7235>   Proxy-Authentication-Info = "Proxy-Authentication-Info" HCOLON                         auth-param-list   Proxy-Authorization = "Proxy-Authorization" HCOLON credentials   Proxy-Require      =  "Proxy-Require" HCOLON feature-tag-list   feature-tag-list   =  feature-tag *(COMMA feature-tag)   Proxy-Supported    =  "Proxy-Supported" HCOLON [feature-tag-list]   Public             =  "Public" HCOLON Method *(COMMA Method)   Range              =  "Range" HCOLON ranges-spec   ranges-spec        =  npt-range / utc-range / smpte-range                      /  range-extSchulzrinne, et al.          Standards Track                  [Page 204]

RFC 7826                        RTSP 2.0                   December 2016   range-ext          =  extension-format [EQUAL range-value]   range-value        =  1*(rtsp-unreserved / quoted-string / ":" )   Referrer           =  "Referrer" HCOLON (absolute-URI / RTSP-URI-Ref)   Request-Status     =  "Request-Status" HCOLON req-status-info   req-status-info    =  cseq-info LWS status-info LWS reason-info   cseq-info          =  "cseq" EQUAL cseq-nr   status-info        =  "status" EQUAL Status-Code   reason-info        =  "reason" EQUAL DQUOTE Reason-Phrase DQUOTE   Require            =  "Require" HCOLON feature-tag-listSchulzrinne, et al.          Standards Track                  [Page 205]

RFC 7826                        RTSP 2.0                   December 2016   RTP-Info         =  "RTP-Info" HCOLON [rtsp-info-spec                       *(COMMA rtsp-info-spec)]   rtsp-info-spec   =  stream-url 1*ssrc-parameter   stream-url       =  "url" EQUAL DQUOTE RTSP-REQ-Ref DQUOTE   ssrc-parameter   =  LWS "ssrc" EQUAL ssrc HCOLON                       ri-parameter *(SEMI ri-parameter)   ri-parameter     =  ("seq" EQUAL 1*5DIGIT)                    /  ("rtptime" EQUAL 1*10DIGIT)                    /  generic-param   Retry-After      =  "Retry-After" HCOLON (RTSP-date / delta-seconds)   Scale            =  "Scale" HCOLON scale-value   Seek-Style       =  "Seek-Style" HCOLON Seek-S-values   Seek-S-values    =  "RAP"                    /  "CoRAP"                    /  "First-Prior"                    /  "Next"                    /  Seek-S-value-ext   Seek-S-value-ext =  token   Server           =  "Server" HCOLON ( product / comment )                       *(LWS (product / comment))   product          =  token [SLASH product-version]   product-version  =  token   comment          =  LPAREN *( ctext / quoted-pair) RPAREN   Session          =  "Session" HCOLON session-id                       [ SEMI "timeout" EQUAL delta-seconds ]   Speed            =  "Speed" HCOLON lower-bound MINUS upper-bound   lower-bound      =  POS-FLOAT   upper-bound      =  POS-FLOAT   Supported        =  "Supported" HCOLON [feature-tag-list]Schulzrinne, et al.          Standards Track                  [Page 206]

RFC 7826                        RTSP 2.0                   December 2016   Terminate-Reason      =  "Terminate-Reason" HCOLON TR-Info   TR-Info              =  TR-Reason *(SEMI TR-Parameter)   TR-Reason            =  "Session-Timeout"                        /  "Server-Admin"                        /  "Internal-Error"                        /  token   TR-Parameter         =  TR-time / TR-user-msg / generic-param   TR-time              =  "time" EQUAL utc-time   TR-user-msg          =  "user-msg" EQUAL quoted-string   Timestamp        =  "Timestamp" HCOLON timestamp-value [LWS delay]   timestamp-value  =  *19DIGIT [ "." *9DIGIT ]   delay            =  *9DIGIT [ "." *9DIGIT ]   Transport        =  "Transport" HCOLON transport-spec                       *(COMMA transport-spec)   transport-spec   =  transport-id *trns-parameter   transport-id     =  trans-id-rtp / other-trans   trans-id-rtp     =  "RTP/" profile ["/" lower-transport]                       ; no LWS is allowed inside transport-id   other-trans      =  token *("/" token)Schulzrinne, et al.          Standards Track                  [Page 207]

RFC 7826                        RTSP 2.0                   December 2016   profile           = "AVP" / "SAVP" / "AVPF" / "SAVPF" / token   lower-transport   = "TCP" / "UDP" / token   trns-parameter    = (SEMI ( "unicast" / "multicast" ))                     / (SEMI "interleaved" EQUAL channel ["-" channel])                     / (SEMI "ttl" EQUAL ttl)                     / (SEMI "layers" EQUAL 1*DIGIT)                     / (SEMI "ssrc" EQUAL ssrc *(SLASH ssrc))                     / (SEMI "mode" EQUAL mode-spec)                     / (SEMI "dest_addr" EQUAL addr-list)                     / (SEMI "src_addr" EQUAL addr-list)                     / (SEMI "setup" EQUAL contrans-setup)                     / (SEMI "connection" EQUAL contrans-con)                     / (SEMI "RTCP-mux")                     / (SEMI "MIKEY" EQUAL MIKEY-Value)                     / (SEMI trn-param-ext)   contrans-setup    = "active" / "passive" / "actpass"   contrans-con      = "new" / "existing"   trn-param-ext     = par-name [EQUAL trn-par-value]   par-name          = token   trn-par-value     = *(rtsp-unreserved / quoted-string)   ttl               = 1*3DIGIT ; 0 to 255   ssrc              = 8HEX   channel           = 1*3DIGIT ; 0 to 255   MIKEY-Value       = base64   mode-spec         = ( DQUOTE mode *(COMMA mode) DQUOTE )   mode              = "PLAY" / token   addr-list         = quoted-addr *(SLASH quoted-addr)   quoted-addr       = DQUOTE (host-port / extension-addr) DQUOTE   host-port         = ( host [":" port] )                     / ( ":" port )   extension-addr    = 1*qdtext   host              = < As defined inRFC 3986>   port              = < As defined inRFC 3986>Schulzrinne, et al.          Standards Track                  [Page 208]

RFC 7826                        RTSP 2.0                   December 2016   Unsupported     = "Unsupported" HCOLON feature-tag-list   User-Agent      = "User-Agent" HCOLON ( product / comment )                     *(LWS (product / comment))   Via             = "Via" HCOLON via-parm *(COMMA via-parm)   via-parm        = sent-protocol LWS sent-by *( SEMI via-params )   via-params      = via-ttl / via-maddr                   / via-received / via-extension   via-ttl         = "ttl" EQUAL ttl   via-maddr       = "maddr" EQUAL host   via-received    = "received" EQUAL (IPv4address / IPv6address)   IPv4address     = < As defined inRFC 3986>   IPv6address     = < As defined inRFC 3986>   via-extension   = generic-param   sent-protocol   = protocol-name SLASH protocol-version                     SLASH transport-prot   protocol-name   = "RTSP" / token   protocol-version = token   transport-prot  = "UDP" / "TCP" / "TLS" / other-transport   other-transport = token   sent-by         = host [ COLON port ]   WWW-Authenticate = "WWW-Authenticate" HCOLON challenge-list20.3.  SDP Extension Syntax   This section defines in ABNF the SDP extensions defined for RTSP.   SeeAppendix D for the definition of the extensions in text.   control-attribute   =  "a=control:" *SP RTSP-REQ-Ref CRLF   a-range-def         =  "a=range:" ranges-spec CRLF   a-mtag-def          =  "a=mtag:" message-tag CRLF21.  Security Considerations   The security considerations and threats around RTSP and its usage can   be divided into considerations around the signaling protocol itself   and the issues related to the media-stream delivery.  However, when   it comes to mitigation of security threats, a threat depending on the   media-stream delivery may in fact be mitigated by a mechanism in the   signaling protocol.Schulzrinne, et al.          Standards Track                  [Page 209]

RFC 7826                        RTSP 2.0                   December 2016   There are several chapters and an appendix in this document that   define security solutions for the protocol.  These sections will be   referenced when discussing the threats below.  However, the reader   should take special notice of the Security Framework (Section 19) and   the specification of how to use SRTP and its key-management   (Appendix C.1.4) to achieve certain aspects of the media security.21.1.  Signaling Protocol Threats   This section focuses on issues related to the signaling protocol.   Because of the similarity in syntax and usage between RTSP servers   and HTTP servers, the security considerations outlined in [RFC7230],   [RFC7231], [RFC7232], [RFC7233], [RFC7234], and [RFC7235] apply as   well.   Specifically, please note the following:   Abuse of Server Log Information:  A server is in the position to save         personal data about a user's requests that might identify their         media consumption patterns or subjects of interest.  This         information is clearly confidential in nature, and its handling         can be constrained by law in certain countries.  Log         information needs to be securely stored and appropriate         guidelines followed for its analysis.  SeeSection 9.8 of         [RFC7230] for additional guidelines.   Transfer of Sensitive Information:  There is no reason to believe         that information transferred in RTSP message, such as the URI         and the content of headers, especially the Server, Via,         Referrer, and From headers, may be any less sensitive than when         used in HTTP.  Therefore, all of the precautions regarding the         protection of data privacy and user privacy apply to         implementers of RTSP clients, servers, and proxies.  See         Sections9.3-9.6 of [RFC7231] for further details.         The RTSP methods defined in this document are primarily used to         establish and control the delivery of the media data         represented by the URI; thus, the RTSP message bodies are         generally less sensitive than the ones in HTTP.  Where HTTP         bodies could contain, for example, your medical records, in         RTSP, the sensitive video of your medical operation would be in         the media stream over the media-transport protocol, not in the         RTSP message.  Still, one has to take note of what potential         sensitive information is included in RTSP.  The protection of         the media data is separate, can be applied directly between         client and server, and is dependent on the media-transport         protocol in use.  SeeSection 21.2 for further discussion.         This possibility for separation of security between media-Schulzrinne, et al.          Standards Track                  [Page 210]

RFC 7826                        RTSP 2.0                   December 2016         resource content and the signaling protocol mitigates the risk         of exposing the media content when using hop-by-hop security         for RTSP signaling using proxies (Section 19.3).   Attacks Based On File and Path Names:  Though RTSP URIs are opaque         handles that do not necessarily have file-system semantics, it         is anticipated that many implementations will translate         portions of the Request-URIs directly to file-system calls.  In         such cases, file systems SHOULD follow the precautions outlined         inSection 9.1 of [RFC7231], such as checking for ".." in path         components.   Personal Information:  RTSP clients are often privy to the same         information that HTTP clients are (username, location, etc.)         and thus should be equally sensitive.  SeeSection 9.8 of         [RFC7230], Sections9.3-9.7 of [RFC7231], andSection 8 of         [RFC7234] for further recommendations.   Privacy Issues Connected to Accept Headers:  Since similar usages of         the "Accept" headers exist in RTSP as in HTTP, the same caveats         outlined inSection 9.4 of [RFC7231] with regard to their use         should be followed.   Establishing Authority:  RTSP shares with HTTP the question of how a         client communicates with the authoritative source for media         streams (Section 9.1 of [RFC7230]).  The used DNS servers, the         security of the communication, and any possibility of a man in         the middle, and the trust in any RTSP proxies all affect the         possibility that a client has received a non-authoritative         response to a request.  Ensuring that a client receives an         authoritative response is challenging, although using the         secure communication for RTSP signaling (rtsps) simplifies it         significantly as the server can provide a hostname identity         assertion in the TLS handshake.   Location Headers and Spoofing:  If a single server supports multiple         organizations that do not trust each another, then it MUST         check the values of the Content-Location header fields in         responses that are generated under control of said         organizations to make sure that they do not attempt to         invalidate resources over which they have no authority (seeSection 15.4 of [RFC2616]).   In addition to the recommendations in the current HTTP specifications   ([RFC7230], [RFC7231], [RFC7232], [RFC7233], [RFC7234], and [RFC7235]   as of this writing) and also those of the previous relevant RFCs   [RFC2068] [RFC2616], future HTTP specifications may provide   additional guidance on security issues.Schulzrinne, et al.          Standards Track                  [Page 211]

RFC 7826                        RTSP 2.0                   December 2016   The following are added considerations for RTSP implementations.   Session Hijacking:  Since there is no or little relation between a         transport-layer connection and an RTSP session, it is possible         for a malicious client to issue requests with random session         identifiers that could affect other clients of an unsuspecting         server.  To mitigate this, the server SHALL use a large, random         and non-sequential session identifier to minimize the         possibility of this kind of attack.  However, unless the RTSP         signaling is always confidentiality protected, e.g., using TLS,         an on-path attacker will be able to hijack a session.  Another         choice for preventing session hijacking is to use client         authentication and only allow the authenticated client creating         the session to access that session.   Authentication:  Servers SHOULD implement both basic and Digest         [RFC2617] authentication.  In environments requiring tighter         security for the control messages, the transport-layer         mechanism TLS [RFC5246] SHOULD be used.   Suspicious Behavior:  Upon detecting instances of behavior that is         deemed a security risk, RTSP servers SHOULD return error code         403 (Forbidden).  RTSP servers SHOULD also be aware of attempts         to probe the server for weaknesses and entry points and MAY         arbitrarily disconnect and ignore further requests from clients         that are deemed to be in violation of local security policy.   TLS through Proxies:  If one uses the possibility to connect TLS in         multiple legs (Section 19.3), one really needs to be aware of         the trust model.  This procedure requires trust in all proxies         part of the path to the server.  The proxies one connects         through are identified, assuming the proxies so far connected         through are well behaved and fulfilling the trust.  The         accepted proxies are men in the middle and have access to all         that goes on over the TLS connection.  Thus, it is important to         consider if that trust model is acceptable in the actual         application.  Further discussion of the actual trust model is         inSection 19.3.  It is important to note what difference in         security properties, if any, may exist with the used media-         transport protocol and its security mechanism.  Using SRTP and         the MIKEY-based key-establishment defined inAppendix C.1.4.1         enables media key-establishment to be done end-to-end without         revealing the keys to the proxies.Schulzrinne, et al.          Standards Track                  [Page 212]

RFC 7826                        RTSP 2.0                   December 2016   Resource Exhaustion:  As RTSP is a stateful protocol and establishes         resource usage on the server, there is a clear possibility to         attack the server by trying to overbook these resources to         perform a DoS attack.  This attack can be both against ongoing         sessions and to prevent others from establishing sessions.         RTSP agents will need to have mechanisms to prevent single         peers from consuming extensive amounts of resources.  The         methods for guarding against this are varied and depend on the         agent's role and capabilities and policies.  Each         implementation has to carefully consider its methods and         policies to mitigate this threat.  There are recommendations         regarding the handling of connections inSection 10.7.   The above threats and considerations have resulted in a set of   security functions and mechanisms built into or used by the protocol.   The signaling protocol relies on two security features defined in the   Security Framework (Section 19): namely client authentication using   HTTP authentication and TLS-based transport protection of the   signaling messages.  Both of these mechanisms are required to be   implemented by any RTSP agent.   A number of different security mitigations have been designed into   the protocol and will be instantiated if the specification is   implemented as written, for example, by ensuring sufficient amounts   of entropy in the randomly generated session identifiers when not   using client authentication to minimize the risk of session   hijacking.  When client authentication is used, protection against   hijacking will be greatly improved by scoping the accessible sessions   to the one this client identity has created.  Some of the above   threats are such that the implementation of the RTSP functionality   itself needs to consider which policy and strategy it uses to   mitigate them.21.2.  Media Stream Delivery Threats   The fact that RTSP establishes and controls a media-stream delivery   results in a set of security issues related to the media streams.   This section will attempt to analyze general threats; however, the   choice of media-stream transport protocol, such as RTP, will result   in some differences in threats and what mechanisms exist to mitigate   them.  Thus, it becomes important that each specification of a new   media-stream transport and delivery protocol usable by RTSP requires   its own security analysis.  This section includes one for RTP.Schulzrinne, et al.          Standards Track                  [Page 213]

RFC 7826                        RTSP 2.0                   December 2016   The set of general threats from or by the media-stream delivery   itself are:   Concentrated Denial-of-Service Attack:  The protocol offers the      opportunity for a remote-controlled DoS attack, where the media      stream is the hammer in that DoS attack.  SeeSection 21.2.1.   Media Confidentiality:  The media delivery may contain content of any      type, and it is not possible, in general, to determine how      sensitive this content is from a confidentiality point.  Thus, it      is a strong requirement that any media delivery protocol supply a      method for providing confidentiality of the actual media content.      In addition to the media-level confidentiality, it becomes      critical that no resource identifiers used in the signaling be      exposed to an attacker as they may have human-understandable names      or may be available to the attacker, allowing it to determine the      content the user received.  Thus, the signaling protocol must also      provide confidentiality protection of any information related to      the media resource.   Media Integrity and Authentication:  There are several reasons why an      attacker will be interested in substituting the media stream sent      out from the RTSP server with one of the attacker's creation or      selection, such as discrediting the target and misinformation      about the target.  Therefore, it is important that the media      protocol provide mechanisms to verify the source authentication      and integrity and to prevent replay attacks on the media stream.   Scope of Multicast:  If RTSP is used to control the transmission of      media onto a multicast network, the scope of the delivery must be      considered.  RTSP supports the TTL Transport header parameter to      indicate this scope for IPv4.  IPv6 has a different mechanism for      the scope boundary.  However, such scope control has risks, as it      may be set too large and distribute media beyond the intended      scope.   Below (Section 21.2.2) a protocol-specific analysis of security   considerations for RTP-based media transport is included.  In that   section, the requirements on implementing security functions for RTSP   agents supporting media delivery over RTP are made clear.Schulzrinne, et al.          Standards Track                  [Page 214]

RFC 7826                        RTSP 2.0                   December 201621.2.1.  Remote DoS Attack   The attacker may initiate traffic flows to one or more IP addresses   by specifying them as the destination in SETUP requests.  While the   attacker's IP address may be known in this case, this is not always   useful in the prevention of more attacks or ascertaining the   attacker's identity.  Thus, an RTSP server MUST only allow client-   specified destinations for RTSP-initiated traffic flows if the server   has ensured that the specified destination address accepts receiving   media through different security mechanisms.  Security mechanisms   that are acceptable in order of increasing generality are:   o  Verification of the client's identity against a database of known      users using RTSP authentication mechanisms (preferably Digest      authentication or stronger)   o  A list of addresses that have consented to be media destinations,      especially considering user identity   o  Verification based on media path   The server SHOULD NOT allow the destination field to be set unless a   mechanism exists in the system to authorize the request originator to   direct streams to the recipient.  It is preferred that this   authorization be performed by the media recipient (destination)   itself and the credentials be passed along to the server.  However,   in certain cases, such as when the recipient address is a multicast   group or when the recipient is unable to communicate with the server   in an out-of-band manner, this may not be possible.  In these cases,   the server may choose another method such as a server-resident   authorization list to ensure that the request originator has the   proper credentials to request stream delivery to the recipient.   One solution that performs the necessary verification of acceptance   of media suitable for unicast-based delivery is the NAT traversal   method based on Interactive Connectivity Establishment (ICE)   [RFC5245] described in [RFC7825].  This mechanism uses random   passwords and a username so that the probability of unintended   indication as a valid media destination is very low.  In addition,   the server includes in its Session Traversal Utilities for NAT (STUN)   [RFC5389] requests a cookie (consisting of random material) that the   destination echoes back; thus, the solution also safeguards against   having an off-path attacker being able to spoof the STUN checks.   This leaves this solution vulnerable only to on-path attackers that   can see the STUN requests go to the target of attack and thus forge a   response.Schulzrinne, et al.          Standards Track                  [Page 215]

RFC 7826                        RTSP 2.0                   December 2016   For delivery to multicast addresses, there is a need for another   solution that is not specified in this memo.21.2.2.  RTP Security Analysis   RTP is a commonly used media-transport protocol and has been the most   common choice for RTSP 1.0 implementations.  The core RTP protocol   has been in use for a long time, and it has well-known security   properties and the RTP security consideration (Section 9 of   [RFC3550]) needs to be reviewed.  In perspective of the usage of RTP   in the context of RTSP, the following properties should be noted:   Stream Additions:  RTP has support for multiple simultaneous media      streams in each RTP session.  As some use cases require support      for non-synchronized adding and removal of media streams and their      identifiers, an attacker can easily insert additional media      streams into a session context that, according to protocol design,      is intended to be played out.  Another threat vector is one of DoS      by exhausting the resources of the RTP session receiver, for      example, by using a large number of SSRC identifiers      simultaneously.  The strong mitigation of this is to ensure that      one cryptographically authenticates any incoming packet flow to      the RTP session.  Weak mitigations like blocking additional media      streams in session contexts easily lead to a DoS vulnerability in      addition to preventing certain RTP extensions or use cases that      rely on multiple media streams, such as RTP retransmission      [RFC4588] to function.   Forged Feedback:  The built-in RTCP also offers a large attack      surface for a couple of different types of attacks.  One venue is      to send RTCP feedback to the media sender indicating large amounts      of packet loss and thus trigger a media bitrate adaptation      response from the sender resulting in lowered media quality and      potentially a shutdown of the media stream.  Another attack is to      perform a resource-exhaustion attack on the receiver by using many      SSRC identifiers to create large state tables and increase the      RTCP-related processing demands.   RTP/RTCP Extensions:  RTP and RTCP extensions generally provide      additional and sometimes extremely powerful tools for DoS attacks      or service disruption.  For example, the Code Control Message      [RFC5104] RTCP extensions enables both the lock down of the      bitrate to low values and disruption of video quality by      requesting intra-frames.   Taking into account the above general discussion inSection 21.2 and   the RTP-specific discussion in this section, it is clear that it is   necessary that a strong security mechanism be supported to protectSchulzrinne, et al.          Standards Track                  [Page 216]

RFC 7826                        RTSP 2.0                   December 2016   RTP.  Therefore, this specification has the following requirements on   RTP security functions for all RTSP agents that handle media streams   and where media-stream transport is completed using RTP.   RTSP agents supporting RTP MUST implement Secure RTP (SRTP) [RFC3711]   and, thus, SAVP.  In addition, SAVPF [RFC5124] MUST also be supported   if AVPF is implemented.  This specification requires no additional   cryptographic transforms or configuration values beyond those   specified as mandatory to implement inRFC 3711, i.e., AES-CM and   HMAC-SHA1.  The default key-management mechanism that MUST be   implemented is the one defined in MIKEY Key Establishment   (Appendix C.1.4.1).  The MIKEY implementation MUST implement the   necessary functions for MIKEY-RSA-R mode [RFC4738] and the SRTP   parameter negotiation necessary to negotiate the supported SRTP   transforms and parameters.22.  IANA Considerations   This section describes a number of registries for RTSP 2.0 that have   been established and are maintained by IANA.  These registries are   separate from any registries existing for RTSP 1.0.  For each   registry, there is a description of the required content, the   registration procedures, and the entries that this document   registers.  For more information on extending RTSP, seeSection 2.7.   In addition, this document registers three SDP attributes.   Registries or entries in registries that have been made for RTSP 1.0   are not moved to RTSP 2.0: the registries and entries of RTSP 1.0 and   RTSP 2.0 are independent.  If any registry or entry in a registry is   also required in RTSP 2.0, it MUST follow the procedure defined below   to allocate the registry or entry in a registry.   The sections describing how to register an item use some of the   registration policies described in [RFC5226] -- namely, "First Come   First Served", "Expert Review", "Specification Required", and   "Standards Action".   In case a registry requires a contact person, the authors (with   Magnus Westerlund <magnus.westerlund@ericsson.com> as primary) are   the contact persons for any entries created by this document.   IANA will request the following information for any registration   request:   o  A name of the item to register according to the rules specified by      the intended registrySchulzrinne, et al.          Standards Track                  [Page 217]

RFC 7826                        RTSP 2.0                   December 2016   o  Indication of who has change control over the feature (for      example, the IETF, ISO, ITU-T, other international standardization      bodies, a consortium, a particular company or group of companies,      or an individual)   o  A reference to a further description, if available, for example      (in decreasing order of preference), an RFC, a published standard,      a published paper, a patent filing, a technical report, documented      source code or a computer manual   o  For proprietary features, contact information (postal and email      address)22.1.  Feature Tags22.1.1.  Description   When a client and server try to determine what part and functionality   of the RTSP specification and any future extensions that its   counterpart implements, there is need for a namespace.  This registry   contains named entries representing certain functionality.   The usage of feature tags is explained inSection 11 andSection 13.1.22.1.2.  Registering New Feature Tags with IANA   The registering of feature tags is done on a First Come, First Served   [RFC5226] basis.   The registry entry for a feature tag has the following information:   o  The name of the feature tag      *  If the registrant indicates that the feature is proprietary,         IANA should request a vendor "prefix" portion of the name.  The         name will then be the vendor prefix followed by a "." followed         by the rest of the provided feature name.      *  If the feature is not proprietary, then IANA need not collect a         prefix for the name.   o  A one-paragraph description of what the feature tag represents   o  The applicability (server, client, proxy, or some combination)   o  A reference to a specification, if applicableSchulzrinne, et al.          Standards Track                  [Page 218]

RFC 7826                        RTSP 2.0                   December 2016   Feature tag names (including the vendor prefix) may contain any non-   space and non-control characters.  There is no length limit on   feature tags.   Examples for a vendor tag describing a proprietary feature are:         vendorA.specfeat01         vendorA.specfeat0222.1.3.  Registered Entries   The following feature tags are defined in this specification and   hereby registered.  The change control belongs to the IETF.   play.basic:  The implementation for delivery and playback operations         according to the core RTSP specification, as defined in this         memo.  Applies for clients, servers, and proxies.  SeeSection 11.1.   play.scale:  Support of scale operations for media playback.  Applies         only for servers.  SeeSection 18.46.   play.speed:  Support of the speed functionality for media delivery.         Applies only for servers.  SeeSection 18.50.   setup.rtp.rtcp.mux:  Support of the RTP and RTCP multiplexing as         discussed inAppendix C.1.6.4.  Applies for both client and         servers and any media caching proxy.   The IANA registry is a table with the name, description, and   reference for each feature tag.22.2.  RTSP Methods22.2.1.  Description   Methods are described inSection 13.  Extending the protocol with new   methods allows for totally new functionality.22.2.2.  Registering New Methods with IANA   A new method is registered through a Standards Action [RFC5226]   because new methods may radically change the protocol's behavior and   purpose.Schulzrinne, et al.          Standards Track                  [Page 219]

RFC 7826                        RTSP 2.0                   December 2016   A specification for a new RTSP method consists of the following   items:   o  A method name that follows the ABNF rules for methods.   o  A clear specification of what a request using the method does and      what responses are expected.  In which directions the method is      used: C->S, S->C, or both.  How the use of headers, if any,      modifies the behavior and effect of the method.   o  A list or table specifying which of the IANA-registered headers      that are allowed to be used with the method in the request or/and      response.  The list or table SHOULD follow the format of tables inSection 18.   o  Describe how the method relates to network proxies.22.2.3.  Registered Entries   This specification,RFC 7826, registers 10 methods: DESCRIBE,   GET_PARAMETER, OPTIONS, PAUSE, PLAY, PLAY_NOTIFY, REDIRECT, SETUP,   SET_PARAMETER, and TEARDOWN.  The initial table of the registry is   provided below.   Method         Directionality           Reference   -----------------------------------------------------   DESCRIBE       C->SRFC 7826   GET_PARAMETER  C->S, S->CRFC 7826   OPTIONS        C->S, S->CRFC 7826   PAUSE          C->SRFC 7826   PLAY           C->SRFC 7826   PLAY_NOTIFY    S->CRFC 7826   REDIRECT       S->CRFC 7826   SETUP          C->SRFC 7826   SET_PARAMETER  C->S, S->CRFC 7826   TEARDOWN       C->S, S->CRFC 782622.3.  RTSP Status Codes22.3.1.  Description   A status code is the three-digit number used to convey information in   RTSP response messages; seeSection 8.  The number space is limited,   and care should be taken not to fill the space.Schulzrinne, et al.          Standards Track                  [Page 220]

RFC 7826                        RTSP 2.0                   December 201622.3.2.  Registering New Status Codes with IANA   A new status code registration follows the policy of IETF Review   [RFC5226].  New RTSP functionality requiring Status Codes should   first be registered in the range of x50-x99.  Only when the range is   full should registrations be made in the x00-x49 range, unless it is   to adopt an HTTP extension to RTSP.  This is done to enable any HTTP   extension to be adopted to RTSP without needing to renumber any   related status codes.  A specification for a new status code must   include the following:   o  The registered number.   o  A description of what the status code means and the expected      behavior of the sender and receiver of the code.22.3.3.  Registered EntriesRFC 7826 (this document) registers the numbered status code defined   in the ABNF entry "Status-Code", except "extension-code" (that   defines the syntax allowed for future extensions) inSection 20.2.2.22.4.  RTSP Headers22.4.1.  Description   By specifying new headers, one or more methods can be enhanced in   many different ways.  An unknown header will be ignored by the   receiving agent.  If the new header is vital for certain   functionality, a feature tag for the functionality can be created and   demanded to be used by the counterpart with the inclusion of a   Require header carrying the feature tag.22.4.2.  Registering New Headers with IANA   Registrations can be made following the Expert Review policy   [RFC5226].  A specification is recommended to be provided, preferably   an RFC or other specification from a Standards Developing   Organization.  The minimal information in a registration request is   the header name and the contact information.   The expert reviewer verifies that the registration request contains   the following information:   o  The name of the header.   o  An ABNF specification of the header syntax.Schulzrinne, et al.          Standards Track                  [Page 221]

RFC 7826                        RTSP 2.0                   December 2016   o  A list or table specifying when the header may be used,      encompassing all methods, their request or response, and the      direction (C->S or S->C).   o  How the header is to be handled by proxies.   o  A description of the purpose of the header.22.4.3.  Registered Entries   All headers specified inSection 18 in RFC 7826 have been registered.   The registry includes the header name and reference.   Furthermore, the following legacy RTSP headers defined in other   specifications are registered with header name, and reference   according to below list.  Note: these references may not fulfill all   of the above rules for registrations due to their legacy status.   o  x-wap-profile defined in [TS-26234].  The x-wap-profile request-      header contains one or more absolute URLs to the requesting      agent's device-capability profile.   o  x-wap-profile-diff defined in [TS-26234].  The x-wap-profile-diff      request-header contains a subset of a device-capability profile.   o  x-wap-profile-warning defined in [TS-26234].  The x-wap-profile-      warning is a response-header that contains error codes explaining      to what extent the server has been able to match the terminal      request in regard to device-capability profiles, as described      using x-wap-profile and x-wap-profile-diff headers.   o  x-predecbufsize defined in [TS-26234].  This response-header      provides an RTSP agent with the TS 26.234 Annex G hypothetical      pre-decoder buffer size.   o  x-initpredecbufperiod defined in [TS-26234].  This response-header      provides an RTSP agent with the TS 26.234 Annex G hypothetical      pre-decoder buffering period.   o  x-initpostdecbufperiod defined in [TS-26234].  This response-      header provides an RTSP agent with the TS 26.234 Annex G post-      decoder buffering period.   o  3gpp-videopostdecbufsize defined in [TS-26234].  This response-      header provides an RTSP agent with the TS 26.234 defined post-      decoder buffer size usable for H.264 (AVC) video streams.Schulzrinne, et al.          Standards Track                  [Page 222]

RFC 7826                        RTSP 2.0                   December 2016   o  3GPP-Link-Char defined in [TS-26234].  This request-header      provides the RTSP server with the RTSP client's link      characteristics as determined from the radio interface.  The      information that can be provided are guaranteed bitrate, maximum      bitrate and maximum transfer delay.   o  3GPP-Adaptation defined in [TS-26234].  This general-header is      part of the bitrate adaptation solution specified for the Packet-      switched Streaming Service (PSS).  It provides the RTSP client's      buffer sizes and target buffer levels to the server, and responses      are used to acknowledge the support and values.   o  3GPP-QoE-Metrics defined in [TS-26234].  This general-header is      used by PSS RTSP agents to negotiate the quality of experience      metrics that a client should gather and report to the server.   o  3GPP-QoE-Feedback defined in [TS-26234].  This request-header is      used by RTSP clients supporting PSS to report the actual values of      the metrics gathered in its quality of experience metering.   The use of "x-" is NOT RECOMMENDED, but the above headers in the list   were defined prior to the clarification.22.5.  Accept-Credentials   The security framework's TLS connection mechanism has two   registerable entities.22.5.1.  Accept-Credentials Policies   This registry is for policies for an RTSP proxy's handling and   verification of TLS certificates when establishing an outbound TLS   connection on behalf of a client.  InSection 19.3.1, three policies   for how to handle certificates are specified.  Further policies may   be defined; registration is made through Standards Action [RFC5226].   A registration request is required to contain the following   information:   o  Name of the policy.   o  Text that describes how the policy works for handling the      certificates.   o  A contact person.Schulzrinne, et al.          Standards Track                  [Page 223]

RFC 7826                        RTSP 2.0                   December 2016   This specification registers the following values:   Any:  A policy requiring the proxy to accept any received         certificate.   Proxy:  A policy where the proxy applies its own policies to         determine which certificates are accepted.   User: A policy where the certificate is required to be forwarded down         the proxy chain to the client, thus allowing the user to         decided to accept or refuse a certificate.22.5.2.  Accept-Credentials Hash Algorithms   The Accept-Credentials header (seeSection 18.2) allows for the usage   of other algorithms for hashing the DER records of accepted entities.   The registration of any future algorithm is expected to be extremely   rare and could also cause interoperability problems.  Therefore, the   bar for registering new algorithms is intentionally placed high.   Any registration of a new hash algorithm requires Standards Action   [RFC5226].  The registration needs to fulfill the following   requirement:   o  The algorithms identifier meeting the "token" ABNF requirement.   o  Provide a definition of the algorithm.   The registered value is:   Hash Alg. ID   Reference   ------------------------   sha-256RFC 782622.6.  Cache-Control Cache Directive Extensions   There exist a number of cache directives that can be sent in the   Cache-Control header.  A registry for these cache directives has been   established by IANA.  New registrations in this registry require   Standards Action or IESG Approval [RFC5226].  A registration request   needs to contain the following information.   o  The name of the cache directive.   o  A definition of the parameter value, if any is allowed.   o  The specification if it is a request or response directive.Schulzrinne, et al.          Standards Track                  [Page 224]

RFC 7826                        RTSP 2.0                   December 2016   o  Text that explains how the cache directive is used for RTSP-      controlled media streams.   o  A contact person.   This specification registers the following values:      no-cache:      public:      private:      no-transform:      only-if-cached:      max-stale:      min-fresh:      must-revalidate:      proxy-revalidate:      max-age:   The registry contains the name of the directive and the reference.22.7.  Media Properties22.7.1.  Description   The media streams being controlled by RTSP can have many different   properties.  The media properties required to cover the use cases   that were in mind when writing the specification are defined.   However, it can be expected that further innovation will result in   new use cases or media streams with properties not covered by the   ones specified here.  Thus, new media properties can be specified.   As new media properties may need a substantial amount of new   definitions to correctly specify behavior for this property, the bar   is intended to be high.Schulzrinne, et al.          Standards Track                  [Page 225]

RFC 7826                        RTSP 2.0                   December 201622.7.2.  Registration Rules   Registering a new media property is done following the Specification   Required policy [RFC5226].  The expert reviewer verifies that a   registration request fulfills the following requirements.   o  An ABNF definition of the media property value name that meets      "media-prop-ext" definition is included.   o  A definition of which media property group it belongs to or define      a new group is included.   o  A description of all changes to the behavior of RTSP as result of      these changes is included.   o  A contact person for the registration is listed.22.7.3.  Registered Values   This specification registers the ten values listed inSection 18.29.   The registry contains the property group, the name of the media   property, and the reference.22.8.  Notify-Reason Values22.8.1.  Description   Notify-Reason values are used to indicate the reason the notification   was sent.  Each reason has its associated rules on what headers and   information may or must be included in the notification.  New   notification behaviors need to be specified to enable interoperable   usage; thus, a specification of each new value is required.22.8.2.  Registration Rules   Registrations for new Notify-Reason values follow the Specification   Required policy [RFC5226].  The expert reviewer verifies that the   request fulfills the following requirements:   o  An ABNF definition of the Notify-Reason value name that meets      "Notify-Reason-extension" definition is included.   o  A description of which headers shall be included in the request      and response, when it should be sent, and any effect it has on the      server client state is made clear.   o  A contact person for the registration is listed.Schulzrinne, et al.          Standards Track                  [Page 226]

RFC 7826                        RTSP 2.0                   December 201622.8.3.  Registered Values   This specification registers three values defined in the Notify-Reas-   val ABNF,Section 20.2.3:   end-of-stream:  This Notify-Reason value indicates the end of a media      stream.   media-properties-update:  This Notify-Reason value allows the server      to indicate that the properties of the media have changed during      the playout.   scale-change:  This Notify-Reason value allows the server to notify      the client about a change in the scale of the media.   The registry contains the name, description, and reference.22.9.  Range Header Formats22.9.1.  Description   The Range header (Section 18.40) allows for different range formats.   These range formats also need an identifier to be used in the Accept-   Ranges header (Section 18.5).  New range formats may be registered,   but moderation should be applied as it makes interoperability more   difficult.22.9.2.  Registration Rules   A registration follows the Specification Required policy [RFC5226].   The expert reviewer verifies that the request fulfills the following   requirements:   o  An ABNF definition of the range format that fulfills the "range-      ext" definition is included.   o  The range format identifier used in Accept-Ranges header according      to the "extension-format" definition is defined.   o  Rules for how one handles the range when using a negative Scale      are included.   o  A contact person for the registration is listed.Schulzrinne, et al.          Standards Track                  [Page 227]

RFC 7826                        RTSP 2.0                   December 201622.9.3.  Registered Values   The registry contains the Range header format identifier, the name of   the range format, and the reference.  This specification registers   the following values.   npt:  Normal Play Time   clock:  UTC Absolute Time format   smpte:  SMPTE Timestamps   smpte-30-drop:  SMPTE Timestamps 29.97 Frames/sec (30 Hz with Drop)   smpte-25:  SMPTE Timestamps 25 Frames/sec22.10.  Terminate-Reason Header   The Terminate-Reason header (Section 18.52) has two registries for   extensions.22.10.1.  Redirect Reasons   This registry contains reasons for session termination that can be   included in a Terminate-Reason header (Section 18.52).  Registrations   follow the Expert Review policy [RFC5226].  The expert reviewer   verifies that the registration request contains the following   information:   o  That the value follows the Terminate-Reason ABNF, i.e., be a      token.   o  That the specification provide a definition of what procedures are      to be followed when a client receives this redirect reason.   o  A contact person   This specification registers three values:   o  Session-Timeout   o  Server-Admin   o  Internal-Error   The registry contains the name of the Redirect Reason and the   reference.Schulzrinne, et al.          Standards Track                  [Page 228]

RFC 7826                        RTSP 2.0                   December 201622.10.2.  Terminate-Reason Header Parameters   This registry contains parameters that may be included in the   Terminate-Reason header (Section 18.52) in addition to a reason.   Registrations are made under the Specification Required policy   [RFC5226].  The expert reviewer verifies that the registration   request contains the following:   o  A parameter name.   o  A parameter following the syntax allowed by the RTSP 2.0      specification.   o  A reference to the specification.   o  A contact person.   This specification registers:   o  time   o  user-msg   The registry contains the name of the Terminate Reason and the   reference.22.11.  RTP-Info Header Parameters22.11.1.  Description   The RTP-Info header (Section 18.45) carries one or more parameter   value pairs with information about a particular point in the RTP   stream.  RTP extensions or new usages may need new types of   information.  As RTP information that could be needed is likely to be   generic enough, and to maximize the interoperability, new   registration is made under the Specification Required policy.22.11.2.  Registration Rules   Registrations for new RTP-Info values follow the policy of   Specification Required [RFC5226].  The expert reviewer verifies that   the registration request contains the following information.   o  An ABNF definition that meets the "generic-param" definition.   o  A reference to the specification.   o  A contact person for the registration.Schulzrinne, et al.          Standards Track                  [Page 229]

RFC 7826                        RTSP 2.0                   December 201622.11.3.  Registered Values   This specification registers the following parameter value pairs:   o  url   o  ssrc   o  seq   o  rtptime   The registry contains the name of the parameter and the reference.22.12.  Seek-Style Policies22.12.1.  Description   Seek-Style policy defines how the RTSP agent seeks in media content   when given a position within the media content.  New seek policies   may be registered; however, a large number of these will complicate   implementation substantially.  The impact of unknown policies is that   the server will not honor the unknown and will use the server default   policy instead.22.12.2.  Registration Rules   Registrations of new Seek-Style policies follow the Specification   Required policy [RFC5226].  The expert reviewer verifies that the   registration request fulfills the following requirements:   o  Has an ABNF definition of the Seek-Style policy name that meets      "Seek-S-value-ext" definition.   o  Includes a short description.   o  Lists a contact person for the registration.   o  Includes a description of which headers shall be included in the      request and response, when it should be sent, and any affect it      has on the server-client state.22.12.3.  Registered Values   This specification registers four values (Name - Short Description):   o  RAP - Using the closest Random Access Point prior to or at the      requested start position.Schulzrinne, et al.          Standards Track                  [Page 230]

RFC 7826                        RTSP 2.0                   December 2016   o  CoRAP - Conditional Random Access Point is like RAP, but only if      the RAP is closer prior to the requested start position than      current pause point.   o  First-Prior - The first-prior policy will start delivery with the      media unit that has a playout time first prior to the requested      start position.   o  Next - The next media units after the provided start position.   The registry contains the name of the Seek-Style policy, the   description, and the reference.22.13.  Transport Header Registries   The transport header (Section 18.54) contains a number of parameters   that have possibilities for future extensions.  Therefore, registries   for these are defined below.22.13.1.  Transport Protocol Identifier   A Transport Protocol specification consists of a transport protocol   identifier, representing some combination of transport protocols, and   any number of transport header parameters required or optional to use   with the identified protocol specification.  This registry contains   the identifiers used by registered transport protocol identifiers.   A registration for the parameter transport protocol identifier   follows the Specification Required policy [RFC5226].  The expert   reviewer verifies that the registration request fulfills the   following requirements:   o  A contact person or organization with address and email.   o  A value definition that follows the ABNF syntax definition of      "transport-id"Section 20.2.3.   o  A descriptive text that explains how the registered values are      used in RTSP, which underlying transport protocols are used, and      any required Transport header parameters.   The registry contains the protocol ID string and the reference.Schulzrinne, et al.          Standards Track                  [Page 231]

RFC 7826                        RTSP 2.0                   December 2016   This specification registers the following values:   RTP/AVP:  Use of the RTP [RFC3550] protocol for media transport in         combination with the "RTP Profile for Audio and Video         Conferences with Minimal Control" [RFC3551] over UDP.  The         usage is explained inRFC 7826, Appendix C.1.   RTP/AVP/UDP:  the same as RTP/AVP.   RTP/AVPF:  Use of the RTP [RFC3550] protocol for media transport in         combination with the "Extended RTP Profile for RTCP-based         Feedback (RTP/AVPF)" [RFC4585] over UDP.  The usage is         explained inRFC 7826, Appendix C.1.   RTP/AVPF/UDP:  the same as RTP/AVPF.   RTP/SAVP:  Use of the RTP [RFC3550] protocol for media transport in         combination with the "The Secure Real-time Transport Protocol         (SRTP)" [RFC3711] over UDP.  The usage is explained inRFC7826,Appendix C.1.   RTP/SAVP/UDP:  the same as RTP/SAVP.   RTP/SAVPF:  Use of the RTP [RFC3550] protocol for media transport in         combination with the "Extended Secure RTP Profile for Real-time         Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)"         [RFC5124] over UDP.  The usage is explained inRFC 7826,         Appendix C.1.   RTP/SAVPF/UDP:  the same as RTP/SAVPF.   RTP/AVP/TCP:  Use of the RTP [RFC3550] protocol for media transport         in combination with the "RTP profile for audio and video         conferences with minimal control" [RFC3551] over TCP.  The         usage is explained inRFC 7826, Appendix C.2.2.   RTP/AVPF/TCP:  Use of the RTP [RFC3550] protocol for media transport         in combination with the "Extended RTP Profile for Real-time         Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)"         [RFC4585] over TCP.  The usage is explained inRFC 7826,         Appendix C.2.2.   RTP/SAVP/TCP:  Use of the RTP [RFC3550] protocol for media transport         in combination with the "The Secure Real-time Transport         Protocol (SRTP)" [RFC3711] over TCP.  The usage is explained inRFC 7826, Appendix C.2.2.Schulzrinne, et al.          Standards Track                  [Page 232]

RFC 7826                        RTSP 2.0                   December 2016   RTP/SAVPF/TCP:  Use of the RTP [RFC3550] protocol for media transport         in combination with the "Extended Secure RTP Profile for Real-         time Transport Control Protocol (RTCP)-Based Feedback (RTP/         SAVPF)" [RFC5124] over TCP.  The usage is explained inRFC7826,Appendix C.2.2.22.13.2.  Transport Modes   The Transport Mode is a Transport header (Section 18.54) parameter.   It is used to identify the general mode of media transport.  The PLAY   value registered defines a PLAYBACK mode, where media flows from   server to client.   A registration for the transport parameter mode follows the Standards   Action policy [RFC5226].  The registration request needs to meet the   following requirements:   o  A value definition that follows the ABNF "token" definitionSection 20.2.3.   o  Text that explains how the registered value is used in RTSP.   This specification registers one value:   PLAY: SeeRFC 7826.   The registry contains the transport mode value and the reference.22.13.3.  Transport Parameters   Transport Parameters are different parameters used in a Transport   header's transport specification (Section 18.54) to provide   additional information required beyond the transport protocol   identifier to establish a functioning transport.   A registration for parameters that may be included in the Transport   header follows the Specification Required policy [RFC5226].  The   expert reviewer verifies that the registration request fulfills the   following requirements:   o  A Transport Parameter Name following the "token" ABNF definition.   o  A value definition, if the parameter takes a value, that follows      the ABNF definition of "trn-par-value"Section 20.2.3.   o  Text that explains how the registered value is used in RTSP.Schulzrinne, et al.          Standards Track                  [Page 233]

RFC 7826                        RTSP 2.0                   December 2016   This specification registers all the transport parameters defined inSection 18.54.  This is a copy of that list:   o  unicast   o  multicast   o  interleaved   o  ttl   o  layers   o  ssrc   o  mode   o  dest_addr   o  src_addr   o  setup   o  connection   o  RTCP-mux   o  MIKEY   The registry contains the transport parameter name and the reference.22.14.  URI Schemes   This specification updates two URI schemes: one previously   registered, "rtsp", and one missing in the registry, "rtspu"   (previously only defined in RTSP 1.0 [RFC2326]).  One new URI scheme,   "rtsps", is also registered.  These URI schemes are registered in an   existing registry ("Uniform Resource Identifier (URI) Schemes") not   created by this memo.  Registrations follow [RFC7595].22.14.1.  The "rtsp" URI Scheme   URI scheme name:  rtsp   Status:  Permanent   URI scheme syntax:  SeeSection 20.2.1 of RFC 7826.Schulzrinne, et al.          Standards Track                  [Page 234]

RFC 7826                        RTSP 2.0                   December 2016   URI scheme semantics:  The rtsp scheme is used to indicate resources         accessible through the usage of the Real-Time Streaming         Protocol (RTSP).  RTSP allows different operations on the         resource identified by the URI, but the primary purpose is the         streaming delivery of the resource to a client.  However, the         operations that are currently defined are DESCRIBE,         GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,         SETUP, SET_PARAMETER, and TEARDOWN.   Encoding considerations:  IRIs in this scheme are defined and need to         be encoded as RTSP URIs when used within RTSP.  That encoding         is done according toRFC 3987.   Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC2326), RTSP 2.0 (RFC 7826).   Interoperability considerations:  The extensions in the URI syntax         performed between RTSP 1.0 and 2.0 can create interoperability         issues.  The changes are:            Support for IPv6 literals in the host part and future IP            literals through a mechanism as defined inRFC 3986.            A new relative format to use in RTSP elements that is not            required to start with "/".         The above changes should have no impact on interoperability as         discussed in detail inSection 4.2 of RFC 7826.   Security considerations:  All the security threats identified inSection 7 of RFC 3986 also apply to this scheme.  They need to         be reviewed and considered in any implementation utilizing this         scheme.   Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com   Author/Change controller:  IETF   References:RFC 2326,RFC 3986,RFC 3987, andRFC 782622.14.2.  The "rtsps" URI Scheme   URI scheme name:  rtsps   Status:  Permanent   URI scheme syntax:  SeeSection 20.2.1 of RFC 7826.Schulzrinne, et al.          Standards Track                  [Page 235]

RFC 7826                        RTSP 2.0                   December 2016   URI scheme semantics:  The rtsps scheme is used to indicate resources         accessible through the usage of the Real-Time Streaming         Protocol (RTSP) over TLS.  RTSP allows different operations on         the resource identified by the URI, but the primary purpose is         the streaming delivery of the resource to a client.  However,         the operations that are currently defined are DESCRIBE,         GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,         SETUP, SET_PARAMETER, and TEARDOWN.   Encoding considerations:  IRIs in this scheme are defined and need to         be encoded as RTSP URIs when used within RTSP.  That encoding         is done according toRFC 3987.   Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC2326), RTSP 2.0 (RFC 7826).   Interoperability considerations:  The "rtsps" scheme was never         officially defined for RTSP 1.0; however, it has seen         widespread use in actual deployments of RTSP 1.0.  Therefore,         this section discusses the believed changes between the         unspecified RTSP 1.0 "rtsps" scheme and RTSP 2.0 definition.         The extensions in the URI syntax performed between RTSP 1.0 and         2.0 can create interoperability issues.  The changes are:            Support for IPv6 literals in the host part and future IP            literals through a mechanism as defined byRFC 3986.            A new relative format to use in RTSP elements that is not            required to start with "/".         The above changes should have no impact on interoperability as         discussed in detail inSection 4.2 of RFC 7826.   Security considerations:  All the security threats identified inSection 7 of RFC 3986 also apply to this scheme.  They need to         be reviewed and considered in any implementation utilizing this         scheme.   Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com   Author/Change controller:  IETF   References:RFC 2326,RFC 3986,RFC 3987, andRFC 7826Schulzrinne, et al.          Standards Track                  [Page 236]

RFC 7826                        RTSP 2.0                   December 201622.14.3.  The "rtspu" URI Scheme   URI scheme name:  rtspu   Status:  Permanent   URI scheme syntax:  SeeSection 3.2 of RFC 2326.   URI scheme semantics:  The rtspu scheme is used to indicate resources         accessible through the usage of the Real-Time Streaming         Protocol (RTSP) over unreliable datagram transport.  RTSP         allows different operations on the resource identified by the         URI, but the primary purpose is the streaming delivery of the         resource to a client.  However, the operations that are         currently defined are DESCRIBE, GET_PARAMETER, OPTIONS,         REDIRECT,PLAY, PLAY_NOTIFY, PAUSE, SETUP, SET_PARAMETER, and         TEARDOWN.   Encoding considerations:  This scheme is not intended to be used with         characters outside the US-ASCII repertoire.   Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC2326).   Interoperability considerations:  The definition of the transport         mechanism of RTSP over UDP has interoperability issues.  That         makes the usage of this scheme problematic.   Security considerations:  All the security threats identified inSection 7 of RFC 3986 also apply to this scheme.  They need to         be reviewed and considered in any implementation utilizing this         scheme.   Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com   Author/Change controller:  IETF   References:RFC 2326Schulzrinne, et al.          Standards Track                  [Page 237]

RFC 7826                        RTSP 2.0                   December 201622.15.  SDP Attributes   This specification defines three SDP [RFC4566] attributes that have   been registered by IANA.   SDP Attribute ("att-field"):        Attribute name:     range        Long form:          Media Range Attribute        Type of name:       att-field        Type of attribute:  both session and media level        Subject to charset: No        Purpose:RFC 7826        Reference:RFC 2326,RFC 7826        Values:             See ABNF definition.        Attribute name:     control        Long form:          RTSP control URI        Type of name:       att-field        Type of attribute:  both session and media level        Subject to charset: No        Purpose:RFC 7826        Reference:RFC 2326,RFC 7826        Values:             Absolute or Relative URIs.        Attribute name:     mtag        Long form:          Message Tag        Type of name:       att-field        Type of attribute:  both session and media level        Subject to charset: No        Purpose:RFC 7826        Reference:RFC 7826        Values:             See ABNF definition22.16.  Media Type Registration for text/parameters   Type name:  text   Subtype name:  parameters   Required parameters:   Optional parameters:  charset: The charset parameter is applicable to      the encoding of the parameter values.  The default charset is      UTF-8, if the 'charset' parameter is not present.   Encoding considerations:  8bitSchulzrinne, et al.          Standards Track                  [Page 238]

RFC 7826                        RTSP 2.0                   December 2016   Security considerations:  This format may carry any type of      parameters.  Some can have security requirements, like privacy,      confidentiality, or integrity requirements.  The format has no      built-in security protection.  For the usage, the transport can be      protected between server and client using TLS.  However, care must      be taken to consider if the proxies are also trusted with the      parameters in case hop-by-hop security is used.  If stored as a      file in a file system, the necessary precautions need to be taken      in relation to the parameter requirements including object      security such as S/MIME [RFC5751].   Interoperability considerations:  This media type was mentioned as a      fictional example in [RFC2326], but was not formally specified.      This has resulted in usage of this media type that may not match      its formal definition.   Published specification:RFC 7826, Appendix F.   Applications that use this media type:  Applications that use RTSP      and have additional parameters they like to read and set using the      RTSP GET_PARAMETER and SET_PARAMETER methods.   Additional information:   Magic number(s):  N/A   File extension(s):  N/A   Macintosh file type code(s):  N/A   Person & email address to contact for further information:      Magnus Westerlund (magnus.westerlund@ericsson.com)   Intended usage:   Common   Restrictions on usage:   None   Author:  Magnus Westerlund (magnus.westerlund@ericsson.com)   Change controller:  IETF   Addition Notes:Schulzrinne, et al.          Standards Track                  [Page 239]

RFC 7826                        RTSP 2.0                   December 201623.  References23.1.  Normative References   [FIPS180-4]              National Institute of Standards and Technology (NIST),              "Federal Information Processing Standards Publication:              Secure Hash Standard (SHS)", DOI 10.6028/NIST.FIPS.180-4,              August 2015, <http://nvlpubs.nist.gov/nistpubs/FIPS/NIST.FIPS.180-4.pdf>.   [RFC768]   Postel, J., "User Datagram Protocol", STD 6,RFC 768,              DOI 10.17487/RFC0768, August 1980,              <http://www.rfc-editor.org/info/rfc768>.   [RFC793]   Postel, J., "Transmission Control Protocol", STD 7,RFC 793, DOI 10.17487/RFC0793, September 1981,              <http://www.rfc-editor.org/info/rfc793>.   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate              Requirement Levels",BCP 14,RFC 2119,              DOI 10.17487/RFC2119, March 1997,              <http://www.rfc-editor.org/info/rfc2119>.   [RFC2460]  Deering, S. and R. Hinden, "Internet Protocol, Version 6              (IPv6) Specification",RFC 2460, DOI 10.17487/RFC2460,              December 1998, <http://www.rfc-editor.org/info/rfc2460>.   [RFC2616]  Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,              Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext              Transfer Protocol -- HTTP/1.1",RFC 2616,              DOI 10.17487/RFC2616, June 1999,              <http://www.rfc-editor.org/info/rfc2616>.   [RFC2617]  Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,              Leach, P., Luotonen, A., and L. Stewart, "HTTP              Authentication: Basic and Digest Access Authentication",RFC 2617, DOI 10.17487/RFC2617, June 1999,              <http://www.rfc-editor.org/info/rfc2617>.   [RFC2818]  Rescorla, E., "HTTP Over TLS",RFC 2818,              DOI 10.17487/RFC2818, May 2000,              <http://www.rfc-editor.org/info/rfc2818>.   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.              Jacobson, "RTP: A Transport Protocol for Real-Time              Applications", STD 64,RFC 3550, DOI 10.17487/RFC3550,              July 2003, <http://www.rfc-editor.org/info/rfc3550>.Schulzrinne, et al.          Standards Track                  [Page 240]

RFC 7826                        RTSP 2.0                   December 2016   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and              Video Conferences with Minimal Control", STD 65,RFC 3551,              DOI 10.17487/RFC3551, July 2003,              <http://www.rfc-editor.org/info/rfc3551>.   [RFC3629]  Yergeau, F., "UTF-8, a transformation format of ISO              10646", STD 63,RFC 3629, DOI 10.17487/RFC3629, November              2003, <http://www.rfc-editor.org/info/rfc3629>.   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.              Norrman, "The Secure Real-time Transport Protocol (SRTP)",RFC 3711, DOI 10.17487/RFC3711, March 2004,              <http://www.rfc-editor.org/info/rfc3711>.   [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.              Norrman, "MIKEY: Multimedia Internet KEYing",RFC 3830,              DOI 10.17487/RFC3830, August 2004,              <http://www.rfc-editor.org/info/rfc3830>.   [RFC3986]  Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform              Resource Identifier (URI): Generic Syntax", STD 66,RFC 3986, DOI 10.17487/RFC3986, January 2005,              <http://www.rfc-editor.org/info/rfc3986>.   [RFC3987]  Duerst, M. and M. Suignard, "Internationalized Resource              Identifiers (IRIs)",RFC 3987, DOI 10.17487/RFC3987,              January 2005, <http://www.rfc-editor.org/info/rfc3987>.   [RFC4086]  Eastlake 3rd, D., Schiller, J., and S. Crocker,              "Randomness Requirements for Security",BCP 106,RFC 4086,              DOI 10.17487/RFC4086, June 2005,              <http://www.rfc-editor.org/info/rfc4086>.   [RFC4291]  Hinden, R. and S. Deering, "IP Version 6 Addressing              Architecture",RFC 4291, DOI 10.17487/RFC4291, February              2006, <http://www.rfc-editor.org/info/rfc4291>.   [RFC7595]  Thaler, D., Ed., Hansen, T., and T. Hardie, "Guidelines              and Registration Procedures for URI Schemes",BCP 35,RFC7595, DOI 10.17487/RFC7595, June 2015, <http://www.rfc-editor.org/info/rfc7595>.   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session              Description Protocol",RFC 4566, DOI 10.17487/RFC4566,              July 2006, <http://www.rfc-editor.org/info/rfc4566>.Schulzrinne, et al.          Standards Track                  [Page 241]

RFC 7826                        RTSP 2.0                   December 2016   [RFC4571]  Lazzaro, J., "Framing Real-time Transport Protocol (RTP)              and RTP Control Protocol (RTCP) Packets over Connection-              Oriented Transport",RFC 4571, DOI 10.17487/RFC4571, July              2006, <http://www.rfc-editor.org/info/rfc4571>.   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,              "Extended RTP Profile for Real-time Transport Control              Protocol (RTCP)-Based Feedback (RTP/AVPF)",RFC 4585,              DOI 10.17487/RFC4585, July 2006,              <http://www.rfc-editor.org/info/rfc4585>.   [RFC4648]  Josefsson, S., "The Base16, Base32, and Base64 Data              Encodings",RFC 4648, DOI 10.17487/RFC4648, October 2006,              <http://www.rfc-editor.org/info/rfc4648>.   [RFC4738]  Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY-              RSA-R: An Additional Mode of Key Distribution in              Multimedia Internet KEYing (MIKEY)",RFC 4738,              DOI 10.17487/RFC4738, November 2006,              <http://www.rfc-editor.org/info/rfc4738>.   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for              Real-time Transport Control Protocol (RTCP)-Based Feedback              (RTP/SAVPF)",RFC 5124, DOI 10.17487/RFC5124, February              2008, <http://www.rfc-editor.org/info/rfc5124>.   [RFC5226]  Narten, T. and H. Alvestrand, "Guidelines for Writing an              IANA Considerations Section in RFCs",BCP 26,RFC 5226,              DOI 10.17487/RFC5226, May 2008,              <http://www.rfc-editor.org/info/rfc5226>.   [RFC5234]  Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax              Specifications: ABNF", STD 68,RFC 5234,              DOI 10.17487/RFC5234, January 2008,              <http://www.rfc-editor.org/info/rfc5234>.   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security              (TLS) Protocol Version 1.2",RFC 5246,              DOI 10.17487/RFC5246, August 2008,              <http://www.rfc-editor.org/info/rfc5246>.   [RFC5280]  Cooper, D., Santesson, S., Farrell, S., Boeyen, S.,              Housley, R., and W. Polk, "Internet X.509 Public Key              Infrastructure Certificate and Certificate Revocation List              (CRL) Profile",RFC 5280, DOI 10.17487/RFC5280, May 2008,              <http://www.rfc-editor.org/info/rfc5280>.Schulzrinne, et al.          Standards Track                  [Page 242]

RFC 7826                        RTSP 2.0                   December 2016   [RFC5322]  Resnick, P., Ed., "Internet Message Format",RFC 5322,              DOI 10.17487/RFC5322, October 2008,              <http://www.rfc-editor.org/info/rfc5322>.   [RFC5646]  Phillips, A., Ed. and M. Davis, Ed., "Tags for Identifying              Languages",BCP 47,RFC 5646, DOI 10.17487/RFC5646,              September 2009, <http://www.rfc-editor.org/info/rfc5646>.   [RFC5751]  Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet              Mail Extensions (S/MIME) Version 3.2 Message              Specification",RFC 5751, DOI 10.17487/RFC5751, January              2010, <http://www.rfc-editor.org/info/rfc5751>.   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and              Control Packets on a Single Port",RFC 5761,              DOI 10.17487/RFC5761, April 2010,              <http://www.rfc-editor.org/info/rfc5761>.   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description              Protocol (SDP) Grouping Framework",RFC 5888,              DOI 10.17487/RFC5888, June 2010,              <http://www.rfc-editor.org/info/rfc5888>.   [RFC6838]  Freed, N., Klensin, J., and T. Hansen, "Media Type              Specifications and Registration Procedures",BCP 13,RFC 6838, DOI 10.17487/RFC6838, January 2013,              <http://www.rfc-editor.org/info/rfc6838>.   [RFC7230]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer              Protocol (HTTP/1.1): Message Syntax and Routing",RFC 7230, DOI 10.17487/RFC7230, June 2014,              <http://www.rfc-editor.org/info/rfc7230>.   [RFC7231]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer              Protocol (HTTP/1.1): Semantics and Content",RFC 7231,              DOI 10.17487/RFC7231, June 2014,              <http://www.rfc-editor.org/info/rfc7231>.   [RFC7232]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer              Protocol (HTTP/1.1): Conditional Requests",RFC 7232,              DOI 10.17487/RFC7232, June 2014,              <http://www.rfc-editor.org/info/rfc7232>.   [RFC7233]  Fielding, R., Ed., Lafon, Y., Ed., and J. Reschke, Ed.,              "Hypertext Transfer Protocol (HTTP/1.1): Range Requests",RFC 7233, DOI 10.17487/RFC7233, June 2014,              <http://www.rfc-editor.org/info/rfc7233>.Schulzrinne, et al.          Standards Track                  [Page 243]

RFC 7826                        RTSP 2.0                   December 2016   [RFC7234]  Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke,              Ed., "Hypertext Transfer Protocol (HTTP/1.1): Caching",RFC 7234, DOI 10.17487/RFC7234, June 2014,              <http://www.rfc-editor.org/info/rfc7234>.   [RFC7235]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer              Protocol (HTTP/1.1): Authentication",RFC 7235,              DOI 10.17487/RFC7235, June 2014,              <http://www.rfc-editor.org/info/rfc7235>.   [RFC7615]  Reschke, J., "HTTP Authentication-Info and Proxy-              Authentication-Info Response Header Fields",RFC 7615,              DOI 10.17487/RFC7615, September 2015,              <http://www.rfc-editor.org/info/rfc7615>.   [RFC7616]  Shekh-Yusef, R., Ed., Ahrens, D., and S. Bremer, "HTTP              Digest Access Authentication",RFC 7616,              DOI 10.17487/RFC7616, September 2015,              <http://www.rfc-editor.org/info/rfc7616>.   [RFC7617]  Reschke, J., "The 'Basic' HTTP Authentication Scheme",RFC 7617, DOI 10.17487/RFC7617, September 2015,              <http://www.rfc-editor.org/info/rfc7617>.   [RFC7825]  Goldberg, J., Westerlund, M., and T. Zeng, "A Network              Address Translator (NAT) Traversal Mechanism for Media              Controlled by Real-Time Streaming Protocol (RTSP)",RFC 7825, DOI 10.17487/RFC7825, December 2016,              <http://www.rfc-editor.org/info/rfc7825>.   [RTP-CIRCUIT-BREAKERS]              Perkins, C. and V. Singh, "Multimedia Congestion Control:              Circuit Breakers for Unicast RTP Sessions", Work in              Progress,draft-ietf-avtcore-rtp-circuit-breakers-13,              February 2016.   [SMPTE-TC] Society of Motion Picture and Television Engineers, "ST              12-1:2008 For Television -- Time and Control Code",              DOI 10.5594/SMPTE.ST12-1.2008, February 2008,              <http://ieeexplore.ieee.org/servlet/opac?punumber=7289818>.   [TS-26234] 3rd Generation Partnership Project (3GPP), "Transparent              end-to-end Packet-switched Streaming Service (PSS);              Protocols and codecs", Technical Specification 26.234,              Release 13, September 2015,              <http://www.3gpp.org/DynaReport/26234.htm>.Schulzrinne, et al.          Standards Track                  [Page 244]

RFC 7826                        RTSP 2.0                   December 201623.2.  Informative References   [ISO.13818-6.1995]              International Organization for Standardization,              "Information technology -- Generic coding of moving              pictures and associated audio information - part 6:              Extension for DSM-CC", ISO Draft Standard 13818-6:1998,              October 1998,              <http://www.iso.org/iso/home/store/catalogue_tc/catalogue_detail.htm?csnumber=25039>.   [ISO.8601.2000]              International Organization for Standardization, "Data              elements and interchange formats - Information interchange              - Representation of dates and times", ISO/IEC Standard              8601, December 2000.   [RFC791]   Postel, J., "Internet Protocol", STD 5,RFC 791,              DOI 10.17487/RFC0791, September 1981,              <http://www.rfc-editor.org/info/rfc791>.   [RFC1123]  Braden, R., Ed., "Requirements for Internet Hosts -              Application and Support", STD 3,RFC 1123,              DOI 10.17487/RFC1123, October 1989,              <http://www.rfc-editor.org/info/rfc1123>.   [RFC2068]  Fielding, R., Gettys, J., Mogul, J., Frystyk, H., and T.              Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1",RFC 2068, DOI 10.17487/RFC2068, January 1997,              <http://www.rfc-editor.org/info/rfc2068>.   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time              Streaming Protocol (RTSP)",RFC 2326,              DOI 10.17487/RFC2326, April 1998,              <http://www.rfc-editor.org/info/rfc2326>.   [RFC2663]  Srisuresh, P. and M. Holdrege, "IP Network Address              Translator (NAT) Terminology and Considerations",RFC 2663, DOI 10.17487/RFC2663, August 1999,              <http://www.rfc-editor.org/info/rfc2663>.   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session              Announcement Protocol",RFC 2974, DOI 10.17487/RFC2974,              October 2000, <http://www.rfc-editor.org/info/rfc2974>.Schulzrinne, et al.          Standards Track                  [Page 245]

RFC 7826                        RTSP 2.0                   December 2016   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,              A., Peterson, J., Sparks, R., Handley, M., and E.              Schooler, "SIP: Session Initiation Protocol",RFC 3261,              DOI 10.17487/RFC3261, June 2002,              <http://www.rfc-editor.org/info/rfc3261>.   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model              with Session Description Protocol (SDP)",RFC 3264,              DOI 10.17487/RFC3264, June 2002,              <http://www.rfc-editor.org/info/rfc3264>.   [RFC3339]  Klyne, G. and C. Newman, "Date and Time on the Internet:              Timestamps",RFC 3339, DOI 10.17487/RFC3339, July 2002,              <http://www.rfc-editor.org/info/rfc3339>.   [RFC4145]  Yon, D. and G. Camarillo, "TCP-Based Media Transport in              the Session Description Protocol (SDP)",RFC 4145,              DOI 10.17487/RFC4145, September 2005,              <http://www.rfc-editor.org/info/rfc4145>.   [RFC4567]  Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.              Carrara, "Key Management Extensions for Session              Description Protocol (SDP) and Real Time Streaming              Protocol (RTSP)",RFC 4567, DOI 10.17487/RFC4567, July              2006, <http://www.rfc-editor.org/info/rfc4567>.   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.              Hakenberg, "RTP Retransmission Payload Format",RFC 4588,              DOI 10.17487/RFC4588, July 2006,              <http://www.rfc-editor.org/info/rfc4588>.   [RFC4855]  Casner, S., "Media Type Registration of RTP Payload              Formats",RFC 4855, DOI 10.17487/RFC4855, February 2007,              <http://www.rfc-editor.org/info/rfc4855>.   [RFC4856]  Casner, S., "Media Type Registration of Payload Formats in              the RTP Profile for Audio and Video Conferences",RFC 4856, DOI 10.17487/RFC4856, February 2007,              <http://www.rfc-editor.org/info/rfc4856>.   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,              "Codec Control Messages in the RTP Audio-Visual Profile              with Feedback (AVPF)",RFC 5104, DOI 10.17487/RFC5104,              February 2008, <http://www.rfc-editor.org/info/rfc5104>.Schulzrinne, et al.          Standards Track                  [Page 246]

RFC 7826                        RTSP 2.0                   December 2016   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment              (ICE): A Protocol for Network Address Translator (NAT)              Traversal for Offer/Answer Protocols",RFC 5245,              DOI 10.17487/RFC5245, April 2010,              <http://www.rfc-editor.org/info/rfc5245>.   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,              "Session Traversal Utilities for NAT (STUN)",RFC 5389,              DOI 10.17487/RFC5389, October 2008,              <http://www.rfc-editor.org/info/rfc5389>.   [RFC5583]  Schierl, T. and S. Wenger, "Signaling Media Decoding              Dependency in the Session Description Protocol (SDP)",RFC 5583, DOI 10.17487/RFC5583, July 2009,              <http://www.rfc-editor.org/info/rfc5583>.   [RFC5905]  Mills, D., Martin, J., Ed., Burbank, J., and W. Kasch,              "Network Time Protocol Version 4: Protocol and Algorithms              Specification",RFC 5905, DOI 10.17487/RFC5905, June 2010,              <http://www.rfc-editor.org/info/rfc5905>.   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,              "Computing TCP's Retransmission Timer",RFC 6298,              DOI 10.17487/RFC6298, June 2011,              <http://www.rfc-editor.org/info/rfc6298>.   [Stevens98]              Stevens, W., Fenner, B., and A. Rudoff, "Unix Networking              Programming, Volume 1: The Sockets Networking API (3rd              Edition)", 1998.Schulzrinne, et al.          Standards Track                  [Page 247]

RFC 7826                        RTSP 2.0                   December 2016Appendix A.  Examples   This section contains several different examples trying to illustrate   possible ways of using RTSP.  The examples can also help with the   understanding of how functions of RTSP work.  However, remember that   these are examples and the normative and syntax descriptions in the   other sections take precedence.  Please also note that many of the   examples have been broken into several lines, where following lines   start with whitespace as allowed by the syntax.A.1.  Media on Demand (Unicast)   This is an example of media-on-demand streaming of media stored in a   container file.  For the purposes of this example, a container file   is a storage entity in which multiple continuous media types   pertaining to the same end-user presentation are present.  In effect,   the container file represents an RTSP presentation, with each of its   components being RTSP-controlled media streams.  Container files are   a widely used means to store such presentations.  While the   components are transported as independent streams, it is desirable to   maintain a common context for those streams at the server end.      This enables the server to keep a single storage handle open      easily.  It also allows treating all the streams equally in case      of any prioritization of streams by the server.   It is also possible that the presentation author may wish to prevent   selective retrieval of the streams by the client in order to preserve   the artistic effect of the combined media presentation.  Similarly,   in such a tightly bound presentation, it is desirable to be able to   control all the streams via a single control message using an   aggregate URI.   The following is an example of using a single RTSP session to control   multiple streams.  It also illustrates the use of aggregate URIs.  In   a container file, it is also desirable not to write any URI parts   that are not kept when the container is distributed, like the host   and most of the path element.  Therefore, this example also uses the   "*" and relative URI in the delivered SDP.   Also, this presentation description (SDP) is not cacheable, as the   Expires header is set to an equal value with date indicating   immediate expiration of its validity.   Client C requests a presentation from media server M.  The movie is   stored in a container file.  The client has obtained an RTSP URI to   the container file.Schulzrinne, et al.          Standards Track                  [Page 248]

RFC 7826                        RTSP 2.0                   December 2016   C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0         CSeq: 1         User-Agent: PhonyClient/1.2   M->C: RTSP/2.0 200 OK         CSeq: 1         Server: PhonyServer/1.0         Date: Fri, 20 Dec 2013 10:20:32 +0000         Content-Type: application/sdp         Content-Length: 271         Content-Base: rtsp://example.com/twister.3gp/         Expires: Fri, 20 Dec 2013 12:20:32 +0000         v=0         o=- 2890844256 2890842807 IN IP4 198.51.100.5         s=RTSP Session         i=An Example of RTSP Session Usage         e=adm@example.com         c=IN IP4 0.0.0.0         a=control: *         a=range:npt=00:00:00-00:10:34.10         t=0 0         m=audio 0 RTP/AVP 0         a=control: trackID=1         m=video 0 RTP/AVP 26         a=control: trackID=4   C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0         CSeq: 2         User-Agent: PhonyClient/1.2         Require: play.basic         Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"         Accept-Ranges: npt, smpte, clock   M->C: RTSP/2.0 200 OK         CSeq: 2         Server: PhonyServer/1.0         Transport: RTP/AVP;unicast; ssrc=93CB001E;                    dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";                    src_addr="198.51.100.5:9000"/"198.51.100.5:9001"         Session: OccldOFFq23KwjYpAnBbUr         Expires: Fri, 20 Dec 2013 12:20:33 +0000         Date: Fri, 20 Dec 2013 10:20:33 +0000         Accept-Ranges: npt         Media-Properties: Random-Access=0.02, Immutable, UnlimitedSchulzrinne, et al.          Standards Track                  [Page 249]

RFC 7826                        RTSP 2.0                   December 2016   C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0         CSeq: 3         User-Agent: PhonyClient/1.2         Require: play.basic         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"         Session: OccldOFFq23KwjYpAnBbUr         Accept-Ranges: npt, smpte, clock   M->C: RTSP/2.0 200 OK         CSeq: 3         Server: PhonyServer/1.0         Transport: RTP/AVP;unicast; ssrc=A813FC13;                    dest_addr="192.0.2.53:8002"/"192.0.2.53:8003";                    src_addr="198.51.100.5:9002"/"198.51.100.5:9003";         Session: OccldOFFq23KwjYpAnBbUr         Expires: Fri, 20 Dec 2013 12:20:33 +0000         Date: Fri, 20 Dec 2013 10:20:33 +0000         Accept-Range: NPT         Media-Properties: Random-Access=0.8, Immutable, Unlimited   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0         CSeq: 4         User-Agent: PhonyClient/1.2         Range: npt=30-         Seek-Style: RAP         Session: OccldOFFq23KwjYpAnBbUr   M->C: RTSP/2.0 200 OK         CSeq: 4         Server: PhonyServer/1.0         Date: Fri, 20 Dec 2013 10:20:34 +0000         Session: OccldOFFq23KwjYpAnBbUr         Range: npt=30-634.10         Seek-Style: RAP         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"            ssrc=0D12F123:seq=12345;rtptime=3450012,           url="rtsp://example.com/twister.3gp/trackID=1"            ssrc=4F312DD8:seq=54321;rtptime=2876889   C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/2.0         CSeq: 5         User-Agent: PhonyClient/1.2         Session: OccldOFFq23KwjYpAnBbUr   # Pause happens 0.87 seconds after starting to playSchulzrinne, et al.          Standards Track                  [Page 250]

RFC 7826                        RTSP 2.0                   December 2016   M->C: RTSP/2.0 200 OK         CSeq: 5         Server: PhonyServer/1.0         Date: Fri, 20 Dec 2013 10:20:35 +0000         Session: OccldOFFq23KwjYpAnBbUr         Range: npt=30.87-634.10   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0         CSeq: 6         User-Agent: PhonyClient/1.2         Range: npt=30.87-634.10         Seek-Style: Next         Session: OccldOFFq23KwjYpAnBbUr   M->C: RTSP/2.0 200 OK         CSeq: 6         Server: PhonyServer/1.0         Date: Fri, 20 Dec 2013 10:22:13 +0000         Session: OccldOFFq23KwjYpAnBbUr         Range: npt=30.87-634.10         Seek-Style: Next         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"            ssrc=0D12F123:seq=12555;rtptime=6330012,           url="rtsp://example.com/twister.3gp/trackID=1"            ssrc=4F312DD8:seq=55021;rtptime=3132889   C->M: TEARDOWN rtsp://example.com/twister.3gp/ RTSP/2.0         CSeq: 7         User-Agent: PhonyClient/1.2         Session: OccldOFFq23KwjYpAnBbUr   M->C: RTSP/2.0 200 OK         CSeq: 7         Server: PhonyServer/1.0         Date: Fri, 20 Dec 2013 10:31:53 +0000A.2.  Media on Demand Using Pipelining   This example is basically the example above (Appendix A.1), but now   utilizing pipelining to speed up the setup.  It requires only two   round-trip times until the media starts flowing.  First of all, the   session description is retrieved to determine what media resources   need to be set up.  In the second step, one sends the necessary SETUP   requests and the PLAY request to initiate media delivery.Schulzrinne, et al.          Standards Track                  [Page 251]

RFC 7826                        RTSP 2.0                   December 2016   Client C requests a presentation from media server M.  The movie is   stored in a container file.  The client has obtained an RTSP URI to   the container file.   C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0         CSeq: 1         User-Agent: PhonyClient/1.2   M->C: RTSP/2.0 200 OK         CSeq: 1         Server: PhonyServer/1.0         Date: Fri, 20 Dec 2013 10:20:32 +0000         Content-Type: application/sdp         Content-Length: 271         Content-Base: rtsp://example.com/twister.3gp/         Expires: Fri, 20 Dec 2013 12:20:32 +0000         v=0         o=- 2890844256 2890842807 IN IP4 192.0.2.5         s=RTSP Session         i=An Example of RTSP Session Usage         e=adm@example.com         c=IN IP4 0.0.0.0         a=control: *         a=range:npt=00:00:00-00:10:34.10         t=0 0         m=audio 0 RTP/AVP 0         a=control: trackID=1         m=video 0 RTP/AVP 26         a=control: trackID=4   C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0         CSeq: 2         User-Agent: PhonyClient/1.2         Require: play.basic         Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"         Accept-Ranges: npt, smpte, clock         Pipelined-Requests: 7654Schulzrinne, et al.          Standards Track                  [Page 252]

RFC 7826                        RTSP 2.0                   December 2016   C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0         CSeq: 3         User-Agent: PhonyClient/1.2         Require: play.basic         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"         Accept-Ranges: npt, smpte, clock         Pipelined-Requests: 7654   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0         CSeq: 4         User-Agent: PhonyClient/1.2         Range: npt=0-         Seek-Style: RAP         Pipelined-Requests: 7654   M->C: RTSP/2.0 200 OK         CSeq: 2         Server: PhonyServer/1.0         Transport: RTP/AVP;unicast;                    dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";                    src_addr="198.51.100.5:9000"/"198.51.100.5:9001";                    ssrc=93CB001E         Session: OccldOFFq23KwjYpAnBbUr         Expires: Fri, 20 Dec 2013 12:20:32 +0000         Date: Fri, 20 Dec 2013 10:20:32 +0000         Accept-Ranges: npt         Pipelined-Requests: 7654         Media-Properties: Random-Access=0.2, Immutable, Unlimited   M->C: RTSP/2.0 200 OK         CSeq: 3         Server: PhonyServer/1.0         Transport: RTP/AVP;unicast;                    dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;                    src_addr="198.51.100.5:9002"/"198.51.100.5:9003";                    ssrc=A813FC13         Session: OccldOFFq23KwjYpAnBbUr         Expires: Sat, 21 Dec 2013 10:20:32 +0000         Date: Fri, 20 Dec 2013 10:20:32 +0000         Accept-Range: NPT         Pipelined-Requests: 7654         Media-Properties: Random-Access=0.8, Immutable, UnlimitedSchulzrinne, et al.          Standards Track                  [Page 253]

RFC 7826                        RTSP 2.0                   December 2016   M->C: RTSP/2.0 200 OK         CSeq: 4         Server: PhonyServer/1.0         Date: Fri, 20 Dec 2013 10:20:32 +0000         Session: OccldOFFq23KwjYpAnBbUr         Range: npt=0-623.10         Seek-Style: RAP         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"            ssrc=0D12F123:seq=12345;rtptime=3450012,           url="rtsp://example.com/twister.3gp/trackID=1"            ssrc=4F312DD8:seq=54321;rtptime=2876889         Pipelined-Requests: 7654A.3.  Secured Media Session for On-Demand Content   This example is basically the above example (Appendix A.2), but now   including establishment of SRTP crypto contexts to get a secured   media delivery.  First of all, the client attempts to fetch this   insecurely, but the server redirects to a URI indicating a   requirement on using a secure connection for the RTSP messages.  The   client establishes a TCP/TLS connection, and the session description   is retrieved to determine what media resources need to be set up.  In   the this session description, secure media (SRTP) is indicated.  In   the next step, the client sends the necessary SETUP requests   including MIKEY messages.  This is pipelined with a PLAY request to   initiate media delivery.   Client C requests a presentation from media server M.  The movie is   stored in a container file.  The client has obtained an RTSP URI to   the container file.   Note: The MIKEY messages below are not valid MIKEY messages and are   Base64-encoded random data to represent where the MIKEY messages   would go.   C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0         CSeq: 1         User-Agent: PhonyClient/1.2   M->C: RTSP/2.0 301 Moved Permanently         CSeq: 1         Server: PhonyServer/1.0         Date: Fri, 20 Dec 2013 10:25:32 +0000         Location: rtsps://example.com/twister.3gp   C->M: Establish TCP/TLS connection and verify server's         certificate that represents example.com.         Used for all below RTSP messages.Schulzrinne, et al.          Standards Track                  [Page 254]

RFC 7826                        RTSP 2.0                   December 2016   C->M: DESCRIBE rtsps://example.com/twister.3gp RTSP/2.0         CSeq: 2         User-Agent: PhonyClient/1.2   M->C: RTSP/2.0 200 OK         CSeq: 2         Server: PhonyServer/1.0         Date: Fri, 20 Dec 2013 10:25:33 +0000         Content-Type: application/sdp         Content-Length: 271         Content-Base: rtsps://example.com/twister.3gp/         Expires: Fri, 20 Dec 2013 12:25:33 +0000         v=0         o=- 2890844256 2890842807 IN IP4 192.0.2.5         s=RTSP Session         i=An Example of RTSP Session Usage         e=adm@example.com         c=IN IP4 0.0.0.0         a=control: *         a=range:npt=00:00:00-00:10:34.10         t=0 0         m=audio 0 RTP/SAVP 0         a=control: trackID=1         m=video 0 RTP/SAVP 26         a=control: trackID=4   C->M: SETUP rtsps://example.com/twister.3gp/trackID=1 RTSP/2.0         CSeq: 3         User-Agent: PhonyClient/1.2         Require: play.basic         Transport: RTP/SAVP;unicast;dest_addr=":8000"/":8001";            MIKEY=VGhpcyBpcyB0aGUgZmlyc3Qgc3RyZWFtcyBNSUtFWSBtZXNzYWdl         Accept-Ranges: npt, smpte, clock         Pipelined-Requests: 7654   C->M: SETUP rtsps://example.com/twister.3gp/trackID=4 RTSP/2.0         CSeq: 4         User-Agent: PhonyClient/1.2         Require: play.basic         Transport: RTP/SAVP;unicast;dest_addr=":8002"/":8003";            MIKEY=TUlLRVkgZm9yIHN0cmVhbSB0d2lzdGVyLjNncC90cmFja0lEPTQ=         Accept-Ranges: npt, smpte, clock         Pipelined-Requests: 7654Schulzrinne, et al.          Standards Track                  [Page 255]

RFC 7826                        RTSP 2.0                   December 2016   C->M: PLAY rtsps://example.com/twister.3gp/ RTSP/2.0         CSeq: 5         User-Agent: PhonyClient/1.2         Range: npt=0-         Seek-Style: RAP         Pipelined-Requests: 7654   M->C: RTSP/2.0 200 OK         CSeq: 3         Server: PhonyServer/1.0         Transport: RTP/SAVP;unicast;            dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";            src_addr="198.51.100.5:9000"/"198.51.100.5:9001";            ssrc=93CB001E;            MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD0x         Session: OccldOFFq23KwjYpAnBbUr         Expires: Fri, 20 Dec 2013 12:25:34 +0000         Date: Fri, 20 Dec 2013 10:25:34 +0000         Accept-Ranges: npt         Pipelined-Requests: 7654         Media-Properties: Random-Access=0.2, Immutable, Unlimited   M->C: RTSP/2.0 200 OK         CSeq: 4         Server: PhonyServer/1.0         Transport: RTP/SAVP;unicast;            dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;            src_addr="198.51.100.5:9002"/"198.51.100.5:9003";            ssrc=A813FC13;            MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD00         Session: OccldOFFq23KwjYpAnBbUr         Expires: Fri, 20 Dec 2013 12:25:34 +0000         Date: Fri, 20 Dec 2013 10:25:34 +0000         Accept-Range: NPT         Pipelined-Requests: 7654         Media-Properties: Random-Access=0.8, Immutable, UnlimitedSchulzrinne, et al.          Standards Track                  [Page 256]

RFC 7826                        RTSP 2.0                   December 2016   M->C: RTSP/2.0 200 OK         CSeq: 5         Server: PhonyServer/1.0         Date: Fri, 20 Dec 2013 10:25:34 +0000         Session: OccldOFFq23KwjYpAnBbUr         Range: npt=0-623.10         Seek-Style: RAP         RTP-Info: url="rtsps://example.com/twister.3gp/trackID=4"            ssrc=0D12F123:seq=12345;rtptime=3450012,           url="rtsps://example.com/twister.3gp/trackID=1"            ssrc=4F312DD8:seq=54321;rtptime=2876889;         Pipelined-Requests: 7654A.4.  Media on Demand (Unicast)   An alternative example of media on demand with a few more tweaks is   the following.  Client C requests a movie distributed from two   different media servers A (audio.example.com) and V   (video.example.com).  The media description is stored on a web server   W.  The media description contains descriptions of the presentation   and all its streams, including the codecs that are available and the   protocol stack.   In this example, the client is only interested in the last part of   the movie.   C->W: GET /twister.sdp HTTP/1.1         Host: www.example.com         Accept: application/sdp   W->C: HTTP/1.1 200 OK         Date: Wed, 23 Jan 2013 15:35:06 GMT         Content-Type: application/sdp         Content-Length: 278         Expires: Thu, 24 Jan 2013 15:35:06 GMT         v=0         o=- 2890844526 2890842807 IN IP4 198.51.100.5         s=RTSP Session         e=adm@example.com         c=IN IP4 0.0.0.0         a=range:npt=00:00:00-01:49:34         t=0 0         m=audio 0 RTP/AVP 0         a=control:rtsp://audio.example.com/twister/audio.en         m=video 0 RTP/AVP 31         a=control:rtsp://video.example.com/twister/videoSchulzrinne, et al.          Standards Track                  [Page 257]

RFC 7826                        RTSP 2.0                   December 2016   C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/2.0         CSeq: 1         User-Agent: PhonyClient/1.2         Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057",                    RTP/AVP/TCP;unicast;interleaved=0-1         Accept-Ranges: npt, smpte, clock   A->C: RTSP/2.0 200 OK         CSeq: 1         Session: OccldOFFq23KwjYpAnBbUr         Transport: RTP/AVP/UDP;unicast;                    dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";                    src_addr="198.51.100.5:5000"/"198.51.100.5:5001"         Date: Wed, 23 Jan 2013 15:35:12 +0000         Server: PhonyServer/1.0         Expires: Thu, 24 Jan 2013 15:35:12 +0000         Cache-Control: public         Accept-Ranges: npt, smpte         Media-Properties: Random-Access=0.02, Immutable, Unlimited   C->V: SETUP rtsp://video.example.com/twister/video RTSP/2.0         CSeq: 1         User-Agent: PhonyClient/1.2         Transport: RTP/AVP/UDP;unicast;                    dest_addr="192.0.2.53:3058"/"192.0.2.53:3059",                    RTP/AVP/TCP;unicast;interleaved=0-1         Accept-Ranges: npt, smpte, clockSchulzrinne, et al.          Standards Track                  [Page 258]

RFC 7826                        RTSP 2.0                   December 2016   V->C: RTSP/2.0 200 OK         CSeq: 1         Session: P5it3pMo6xHkjUcDrNkBjf         Transport: RTP/AVP/UDP;unicast;            dest_addr="192.0.2.53:3058"/"192.0.2.53:3059";            src_addr="198.51.100.5:5002"/"198.51.100.5:5003"         Date: Wed, 23 Jan 2013 15:35:12 +0000         Server: PhonyServer/1.0         Cache-Control: public         Expires: Thu, 24 Jan 2013 15:35:12 +0000         Accept-Ranges: npt, smpte         Media-Properties: Random-Access=1.2, Immutable, Unlimited   C->V: PLAY rtsp://video.example.com/twister/video RTSP/2.0         CSeq: 2         User-Agent: PhonyClient/1.2         Session: P5it3pMo6xHkjUcDrNkBjf         Range: smpte=0:10:00-   V->C: RTSP/2.0 200 OK         CSeq: 2         Session: P5it3pMo6xHkjUcDrNkBjf         Range: smpte=0:10:00-1:49:23         Seek-Style: First-Prior         RTP-Info: url="rtsp://video.example.com/twister/video"                   ssrc=A17E189D:seq=12312232;rtptime=78712811         Server: PhonyServer/2.0         Date: Wed, 23 Jan 2013 15:35:13 +0000   C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/2.0         CSeq: 2         User-Agent: PhonyClient/1.2         Session: OccldOFFq23KwjYpAnBbUr         Range: smpte=0:10:00-   A->C: RTSP/2.0 200 OK         CSeq: 2         Session: OccldOFFq23KwjYpAnBbUr         Range: smpte=0:10:00-1:49:23         Seek-Style: First-Prior         RTP-Info: url="rtsp://audio.example.com/twister/audio.en"                   ssrc=3D124F01:seq=876655;rtptime=1032181         Server: PhonyServer/1.0         Date: Wed, 23 Jan 2013 15:35:13 +0000Schulzrinne, et al.          Standards Track                  [Page 259]

RFC 7826                        RTSP 2.0                   December 2016   C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/2.0         CSeq: 3         User-Agent: PhonyClient/1.2         Session: OccldOFFq23KwjYpAnBbUr   A->C: RTSP/2.0 200 OK         CSeq: 3         Server: PhonyServer/1.0         Date: Wed, 23 Jan 2013 15:36:52 +0000   C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/2.0         CSeq: 3         User-Agent: PhonyClient/1.2         Session: P5it3pMo6xHkjUcDrNkBjf   V->C: RTSP/2.0 200 OK         CSeq: 3         Server: PhonyServer/2.0         Date: Wed, 23 Jan 2013 15:36:52 +0000   Even though the audio and video track are on two different servers   that may start at slightly different times and may drift with respect   to each other over time, the client can perform initial   synchronization of the two media using RTP-Info and Range received in   the PLAY responses.  If the two servers are time synchronized, the   RTCP packets can also be used to maintain synchronization.A.5.  Single-Stream Container Files   Some RTSP servers may treat all files as though they are "container   files", yet other servers may not support such a concept.  Because of   this, clients needs to use the rules set forth in the session   description for Request-URIs rather than assuming that a consistent   URI may always be used throughout.  Below is an example of how a   multi-stream server might expect a single-stream file to be served:Schulzrinne, et al.          Standards Track                  [Page 260]

RFC 7826                        RTSP 2.0                   December 2016   C->S: DESCRIBE rtsp://foo.example.com/test.wav RTSP/2.0         Accept: application/x-rtsp-mh, application/sdp         CSeq: 1         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 1         Content-base: rtsp://foo.example.com/test.wav/         Content-type: application/sdp         Content-length: 163         Server: PhonyServer/1.0         Date: Wed, 23 Jan 2013 15:36:52 +0000         Expires: Thu, 24 Jan 2013 15:36:52 +0000         v=0         o=- 872653257 872653257 IN IP4 192.0.2.5         s=mu-law wave file         i=audio test         c=IN IP4 0.0.0.0         t=0 0         a=control: *         m=audio 0 RTP/AVP 0         a=control:streamid=0   C->S: SETUP rtsp://foo.example.com/test.wav/streamid=0 RTSP/2.0         Transport: RTP/AVP/UDP;unicast;            dest_addr=":6970"/":6971";mode="PLAY"         CSeq: 2         User-Agent: PhonyClient/1.2         Accept-Ranges: npt, smpte, clock   S->C: RTSP/2.0 200 OK         Transport: RTP/AVP/UDP;unicast;             dest_addr="192.0.2.53:6970"/"192.0.2.53:6971";             src_addr="198.51.100.5:6970"/"198.51.100.5:6971";             mode="PLAY";ssrc=EAB98712         CSeq: 2         Session: NYkqQYKk0bb12BY3goyoyO         Expires: Thu, 24 Jan 2013 15:36:52 +0000         Server: PhonyServer/1.0         Date: Wed, 23 Jan 2013 15:36:52 +0000         Accept-Ranges: npt         Media-Properties: Random-Access=0.5, Immutable, UnlimitedSchulzrinne, et al.          Standards Track                  [Page 261]

RFC 7826                        RTSP 2.0                   December 2016   C->S: PLAY rtsp://foo.example.com/test.wav/ RTSP/2.0         CSeq: 3         User-Agent: PhonyClient/1.2         Session: NYkqQYKk0bb12BY3goyoyO   S->C: RTSP/2.0 200 OK         CSeq: 3         Server: PhonyServer/1.0         Date: Wed, 23 Jan 2013 15:36:52 +0000         Session: NYkqQYKk0bb12BY3goyoyO         Range: npt=0-600         Seek-Style: RAP         RTP-Info: url="rtsp://foo.example.com/test.wav/streamid=0"            ssrc=0D12F123:seq=981888;rtptime=3781123   Note the different URI in the SETUP command and then the switch back   to the aggregate URI in the PLAY command.  This makes complete sense   when there are multiple streams with aggregate control, but it is   less than intuitive in the special case where the number of streams   is one.  However, the server has declared the aggregated control URI   in the SDP; therefore, this is legal.   In this case, it is also required that servers accept implementations   that use the non-aggregated interpretation and use the individual   media URI, like this:   C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0         CSeq: 3         User-Agent: PhonyClient/1.2         Session: NYkqQYKk0bb12BY3goyoyOSchulzrinne, et al.          Standards Track                  [Page 262]

RFC 7826                        RTSP 2.0                   December 2016A.6.  Live Media Presentation Using Multicast   The media server M chooses the multicast address and port.  Here, it   is assumed that the web server only contains a pointer to the full   description, while the media server M maintains the full description.   C->W: GET /sessions.html HTTP/1.1         Host: www.example.com   W->C: HTTP/1.1 200 OK         Content-Type: text/html         <html>           ...           <a href "rtsp://live.example.com/concert/audio">              Streamed Live Music performance </a>           ...         </html>   C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/2.0         CSeq: 1         Supported: play.basic, play.scale         User-Agent: PhonyClient/1.2   M->C: RTSP/2.0 200 OK         CSeq: 1         Content-Type: application/sdp         Content-Length: 183         Server: PhonyServer/1.0         Date: Wed, 23 Jan 2013 15:36:52 +0000         Supported: play.basic         v=0         o=- 2890844526 2890842807 IN IP4 192.0.2.5         s=RTSP Session         t=0 0         m=audio 3456 RTP/AVP 0         c=IN IP4 233.252.0.54/16         a=control: rtsp://live.example.com/concert/audio         a=range:npt=0-Schulzrinne, et al.          Standards Track                  [Page 263]

RFC 7826                        RTSP 2.0                   December 2016   C->M: SETUP rtsp://live.example.com/concert/audio RTSP/2.0         CSeq: 2         Transport: RTP/AVP;multicast;              dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16         Accept-Ranges: npt, smpte, clock         User-Agent: PhonyClient/1.2   M->C: RTSP/2.0 200 OK         CSeq: 2         Server: PhonyServer/1.0         Date: Wed, 23 Jan 2013 15:36:52 +0000         Transport: RTP/AVP;multicast;              dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16              ;ssrc=4D12AB92/0DF876A3         Session: qHj4jidpmF6zy9v9tNbtxr         Accept-Ranges: npt, clock         Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0   C->M: PLAY rtsp://live.example.com/concert/audio RTSP/2.0         CSeq: 3         Session: qHj4jidpmF6zy9v9tNbtxr         User-Agent: PhonyClient/1.2   M->C: RTSP/2.0 200 OK         CSeq: 3         Server: PhonyServer/1.0         Date: Wed, 23 Jan 2013 15:36:52 +0000         Session: qHj4jidpmF6zy9v9tNbtxr         Seek-Style: Next         Range:npt=1256-         RTP-Info: url="rtsp://live.example.com/concert/audio"                   ssrc=0D12F123:seq=1473; rtptime=80000A.7.  Capability Negotiation   This example illustrates how the client and server determine their   capability to support a special feature, in this case, "play.scale".   The server, through the client request and the included Supported   header, learns that the client supports RTSP 2.0 and also supports   the playback time scaling feature of RTSP.  The server's response   contains the following feature-related information to the client; it   supports the basic media delivery functions (play.basic), the   extended functionality of time scaling of content (play.scale), and   one "example.com" proprietary feature (com.example.flight).  The   client also learns the methods supported (Public header) by the   server for the indicated resource.Schulzrinne, et al.          Standards Track                  [Page 264]

RFC 7826                        RTSP 2.0                   December 2016   C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/2.0         CSeq: 1         Supported: play.basic, play.scale         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 1         Public:OPTIONS,SETUP,PLAY,PAUSE,TEARDOWN,DESCRIBE,GET_PARAMETER         Allow: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN, DESCRIBE         Server: PhonyServer/2.0         Supported: play.basic, play.scale, com.example.flight   When the client sends its SETUP request, it tells the server that it   requires support of the play.scale feature for this session by   including the Require header.   C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/2.0         CSeq: 3         User-Agent: PhonyClient/1.2         Transport: RTP/AVP/UDP;unicast;                    dest_addr="192.0.2.53:3056"/"192.0.2.53:3057",                    RTP/AVP/TCP;unicast;interleaved=0-1         Require: play.scale         Accept-Ranges: npt, smpte, clock         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 3         Session: OccldOFFq23KwjYpAnBbUr         Transport: RTP/AVP/UDP;unicast;            dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";            src_addr="198.51.100.5:5000"/"198.51.100.5:5001"         Server: PhonyServer/2.0         Accept-Ranges: npt, smpte         Media-Properties: Random-Access=0.8, Immutable, UnlimitedAppendix B.  RTSP Protocol State Machine   The RTSP session state machine describes the behavior of the protocol   from RTSP session initialization through RTSP session termination.   It is probably easiest to think of this as the server's state and   then view the client as needing to track what it believes the   server's state will be based on sent or received RTSP messages.   Thus, in most cases, the state tables below can be read as: if the   client does X, and assuming it fulfills any prerequisite(s), the   (server) state will move to the new state and the indicated response   will returned.  However, there are also server-to-client   notifications or requests, where the action describes whatSchulzrinne, et al.          Standards Track                  [Page 265]

RFC 7826                        RTSP 2.0                   December 2016   notification or request will occur, its requisites, what new state   will result after the server has received the response, as well as   describing the client's response to the action.   The State machine is defined on a per-session basis, which is   uniquely identified by the RTSP session identifier.  The session may   contain one or more media streams depending on state.  If a single   media stream is part of the session, it is in non-aggregated control.   If two or more are part of the session, it is in aggregated control.   The below state machine is an informative description of the   protocol's behavior.  In case of ambiguity with the earlier parts of   this specification, the description in the earlier parts take   precedence.B.1.  States   The state machine contains three states, described below.  For each   state, there exists a table that shows which requests and events are   allowed and whether they will result in a state change.   Init: Initial state, no session exists.   Ready:  Session is ready to start playing.   Play: Session is playing, i.e., sending media-stream data in the         direction S->C.B.2.  State Variables   This representation of the state machine needs more than its state to   work.  A small number of variables are also needed, and they are   explained below.   NRM:  The number of media streams that are part of this session.   RP:   Resume point, the point in the presentation time line at which         a request to continue playing will resume from.  A time format         for the variable is not mandated.B.3.  Abbreviations   To make the state tables more compact, a number of abbreviations are   used, which are explained below.   IFI:  IF Implemented.   md:   MediaSchulzrinne, et al.          Standards Track                  [Page 266]

RFC 7826                        RTSP 2.0                   December 2016   PP:   Pause Point, the point in the presentation timeline at which         the presentation was paused.   Prs:  Presentation, the complete multimedia presentation.   RedP: Redirect Point, the point in the presentation timeline at which         a REDIRECT was specified to occur.   SES:  Session.B.4.  State Tables   This section contains a table for each state.  The table contains all   the requests and events on which this state is allowed to act.  The   events that are method names are, unless noted, requests with the   given method in the direction client to server (C->S).  In some   cases, there exists one or more requisites.  The response column   tells what type of response actions should be performed.  Possible   actions that are requested for an event include: response codes,   e.g., 200, headers that need to be included in the response, setting   of state variables, or settings of other session-related parameters.   The new state column tells which state the state machine changes to.   The response to a valid request meeting the requisites is normally a   2xx (SUCCESS) unless otherwise noted in the response column.  The   exceptions need to be given a response according to the response   column.  If the request does not meet the requisite, is erroneous, or   some other type of error occurs, the appropriate response code is to   be sent.  If the response code is a 4xx, the session state is   unchanged.  A response code of 3rr will result in that the session   being ended and its state changed to Init.  A response code of 304   results in no state change.  However, there are restrictions to when   a 3rr response may be used.  A 5xx response does not result in any   change of the session state, except if the error is not possible to   recover from.  An unrecoverable error results in the ending of the   session.  In the general case, if it can't be determined whether or   not it was an unrecoverable error, the client will be required to   test.  In the case that the next request after a 5xx is responded to   with a 454 (Session Not Found), the client knows that the session has   ended.  For any request message that cannot be responded to within   the time defined inSection 10.4, a 100 response must be sent.   The server will time out the session after the period of time   specified in the SETUP response, if no activity from the client is   detected.  Therefore, there exists a timeout event for all states   except Init.Schulzrinne, et al.          Standards Track                  [Page 267]

RFC 7826                        RTSP 2.0                   December 2016   In the case that NRM = 1, the presentation URI is equal to the media   URI or a specified presentation URI.  For NRM > 1, the presentation   URI needs to be other than any of the media that are part of the   session.  This applies to all states.   +---------------+-----------------+---------------------------------+   | Event         | Prerequisite    | Response                        |   +---------------+-----------------+---------------------------------+   | DESCRIBE      | Needs REDIRECT  | 3rr, Redirect                   |   |               |                 |                                 |   | DESCRIBE      |                 | 200, Session description        |   |               |                 |                                 |   | OPTIONS       | Session ID      | 200, Reset session timeout      |   |               |                 | timer                           |   |               |                 |                                 |   | OPTIONS       |                 | 200                             |   |               |                 |                                 |   | SET_PARAMETER | Valid parameter | 200, change value of parameter  |   |               |                 |                                 |   | GET_PARAMETER | Valid parameter | 200, return value of parameter  |   +---------------+-----------------+---------------------------------+                Table 9: Non-State-Machine Changing Events   The methods in Table 9 do not have any effect on the state machine or   the state variables.  However, some methods do change other session-   related parameters, for example, SET_PARAMETER, which will set the   parameter(s) specified in its body.  Also, all of these methods that   allow the Session header will also update the keep-alive timer for   the session.   +------------------+----------------+-----------+-------------------+   | Action           | Requisite      | New State | Response          |   +------------------+----------------+-----------+-------------------+   | SETUP            |                | Ready     | NRM=1, RP=0.0     |   |                  |                |           |                   |   | SETUP            | Needs Redirect | Init      | 3rr Redirect      |   |                  |                |           |                   |   | S -> C: REDIRECT | No Session hdr | Init      | Terminate all SES |   +------------------+----------------+-----------+-------------------+                           Table 10: State: Init   The initial state of the state machine (Table 10) can only be left by   processing a correct SETUP request.  As seen in the table, the two   state variables are also set by a correct request.  This table also   shows that a correct SETUP can in some cases be redirected to another   URI or server by a 3rr response.Schulzrinne, et al.          Standards Track                  [Page 268]

RFC 7826                        RTSP 2.0                   December 2016   +-------------+------------------------+---------+------------------+   | Action      | Requisite              | New     | Response         |   |             |                        | State   |                  |   +-------------+------------------------+---------+------------------+   | SETUP       | New URI                | Ready   | NRM +=1          |   |             |                        |         |                  |   | SETUP       | URI Setup prior        | Ready   | Change transport |   |             |                        |         | param            |   |             |                        |         |                  |   | TEARDOWN    | Prs URI,               | Init    | No session hdr,  |   |             |                        |         | NRM = 0          |   |             |                        |         |                  |   | TEARDOWN    | md URI,NRM=1           | Init    | No Session hdr,  |   |             |                        |         | NRM = 0          |   |             |                        |         |                  |   | TEARDOWN    | md URI,NRM>1           | Ready   | Session hdr, NRM |   |             |                        |         | -= 1             |   |             |                        |         |                  |   | PLAY        | Prs URI, No range      | Play    | Play from RP     |   |             |                        |         |                  |   | PLAY        | Prs URI, Range         | Play    | According to     |   |             |                        |         | range            |   |             |                        |         |                  |   | PLAY        | md URI, NRM=1, Range   | Play    | According to     |   |             |                        |         | range            |   |             |                        |         |                  |   | PLAY        | md URI, NRM=1          | Play    | Play from RP     |   |             |                        |         |                  |   | PAUSE       | Prs URI                | Ready   | Return PP        |   |             |                        |         |                  |   | SC:REDIRECT | Terminate-Reason       | Ready   | Set RedP         |   |             |                        |         |                  |   | SC:REDIRECT | No Terminate-Reason    | Init    | Session is       |   |             | time parameter         |         | removed          |   |             |                        |         |                  |   | Timeout     |                        | Init    |                  |   |             |                        |         |                  |   | RedP        |                        | Init    | TEARDOWN of      |   | reached     |                        |         | session          |   +-------------+------------------------+---------+------------------+                          Table 11: State: Ready   In the Ready state (Table 11), some of the actions depend on the   number of media streams (NRM) in the session, i.e., aggregated or   non-aggregated control.  A SETUP request in the Ready state can   either add one more media stream to the session or, if the media   stream (same URI) already is part of the session, change theSchulzrinne, et al.          Standards Track                  [Page 269]

RFC 7826                        RTSP 2.0                   December 2016   transport parameters.  TEARDOWN depends on both the Request-URI and   the number of media streams within the session.  If the Request-URI   is the presentation URI, the whole session is torn down.  If a media   URI is used in the TEARDOWN request and more than one media exists in   the session, the session will remain and a session header is returned   in the response.  If only a single media stream remains in the   session when performing a TEARDOWN with a media URI, the session is   removed.  The number of media streams remaining after tearing down a   media stream determines the new state.Schulzrinne, et al.          Standards Track                  [Page 270]

RFC 7826                        RTSP 2.0                   December 2016   +----------------+-----------------------+--------+-----------------+   | Action         | Requisite             | New    | Response        |   |                |                       | State  |                 |   +----------------+-----------------------+--------+-----------------+   | PAUSE          | Prs URI               | Ready  | Set RP to       |   |                |                       |        | present point   |   |                |                       |        |                 |   | End of media   | All media             | Play   | Set RP = End of |   |                |                       |        | media           |   |                |                       |        |                 |   | End of range   |                       | Play   | Set RP = End of |   |                |                       |        | range           |   |                |                       |        |                 |   | PLAY           | Prs URI, No range     | Play   | Play from       |   |                |                       |        | present point   |   |                |                       |        |                 |   | PLAY           | Prs URI, Range        | Play   | According to    |   |                |                       |        | range           |   |                |                       |        |                 |   | SC:PLAY_NOTIFY |                       | Play   | 200             |   |                |                       |        |                 |   | SETUP          | New URI               | Play   | 455             |   |                |                       |        |                 |   | SETUP          | md URI                | Play   | 455             |   |                |                       |        |                 |   | SETUP          | md URI, IFI           | Play   | Change          |   |                |                       |        | transport param.|   |                |                       |        |                 |   | TEARDOWN       | Prs URI               | Init   | No session hdr  |   |                |                       |        |                 |   | TEARDOWN       | md URI,NRM=1          | Init   | No Session hdr, |   |                |                       |        | NRM=0           |   |                |                       |        |                 |   | TEARDOWN       | md URI                | Play   | 455             |   |                |                       |        |                 |   | SC:REDIRECT    | Terminate Reason with | Play   | Set RedP        |   |                | Time parameter        |        |                 |   |                |                       |        |                 |   | SC:REDIRECT    |                       | Init   | Session is      |   |                |                       |        | removed         |   |                |                       |        |                 |   | RedP reached   |                       | Init   | TEARDOWN of     |   |                |                       |        | session         |   |                |                       |        |                 |   | Timeout        |                       | Init   | Stop Media      |   |                |                       |        | playout         |   +----------------+-----------------------+--------+-----------------+                           Table 12: State: PlaySchulzrinne, et al.          Standards Track                  [Page 271]

RFC 7826                        RTSP 2.0                   December 2016   The Play state table (Table 12) contains a number of requests that   need a presentation URI (labeled as Prs URI) to work on (i.e., the   presentation URI has to be used as the Request-URI).  This is due to   the exclusion of non-aggregated stream control in sessions with more   than one media stream.   To avoid inconsistencies between the client and server, automatic   state transitions are avoided.  This can be seen at, for example, an   "End of media" event when all media has finished playing but the   session still remains in Play state.  An explicit PAUSE request needs   to be sent to change the state to Ready.  It may appear that there   exist automatic transitions in "RedP reached" and "PP reached".   However, they are requested and acknowledged before they take place.   The time at which the transition will happen is known by looking at   the Terminate-Reason header's time parameter and Range header,   respectively.  If the client sends a request close in time to these   transitions, it needs to be prepared for receiving error messages, as   the state may or may not have changed.Appendix C.  Media-Transport Alternatives   This section defines how certain combinations of protocols, profiles,   and lower transports are used.  This includes the usage of the   Transport header's source and destination address parameters:   "src_addr" and "dest_addr".C.1.  RTP   This section defines the interaction of RTSP with respect to the RTP   protocol [RFC3550].  It also defines any necessary media-transport   signaling with regard to RTP.   The available RTP profiles and lower-layer transports are described   below along with rules on signaling the available combinations.C.1.1.  AVP   The usage of the "RTP Profile for Audio and Video Conferences with   Minimal Control" [RFC3551] when using RTP for media transport over   different lower-layer transport protocols is defined below in regard   to RTSP.   One such case is defined within this document: the use of embedded   (interleaved) binary data as defined inSection 14.  The usage of   this method is indicated by including the "interleaved" parameter.Schulzrinne, et al.          Standards Track                  [Page 272]

RFC 7826                        RTSP 2.0                   December 2016   When using embedded binary data, "src_addr" and "dest_addr" MUST NOT   be used.  This addressing and multiplexing is used as defined with   use of channel numbers and the interleaved parameter.C.1.2.  AVP/UDP   This part describes the sending of RTP [RFC3550] over lower-   transport-layer UDP [RFC768] according to the profile "RTP Profile   for Audio and Video Conferences with Minimal Control" defined in   [RFC3551].  Implementations of RTP/AVP/UDP MUST implement RTCP   (Appendix C.1.6).  This profile requires one or two unidirectional or   bidirectional UDP flows per media stream.  The first UDP flow is for   RTP and the second is for RTCP.  Multiplexing of RTP and RTCP   (Appendix C.1.6.4) MAY be used, in which case, a single UDP flow is   used for both parts.  Embedding of RTP data with the RTSP messages,   in accordance withSection 14, SHOULD NOT be performed when RTSP   messages are transported over unreliable transport protocols, like   UDP [RFC768].   The RTP/UDP and RTCP/UDP flows can be established using the Transport   header's "src_addr" and "dest_addr" parameters.   In RTSP PLAY mode, the transmission of RTP packets from client to   server is unspecified.  The behavior in regard to such RTP packets   MAY be defined in future.   The "src_addr" and "dest_addr" parameters are used in the following   way for media delivery and playback mode, i.e., Mode=PLAY:   o  The "src_addr" and "dest_addr" parameters MUST contain either 1 or      2 address specifications.  Note that two address specifications      MAY be provided even if RTP and RTCP multiplexing is negotiated.   o  Each address specification for RTP/AVP/UDP or RTP/AVP/TCP MUST      contain either:      *  both an address and a port number, or      *  a port number without an address.   o  The first address specification given in either of the parameters      applies to the RTP stream.  The second specification, if present,      applies to the RTCP stream, unless in the case RTP and RTCP      multiplexing is negotiated where both RTP and RTCP will use the      first specification.Schulzrinne, et al.          Standards Track                  [Page 273]

RFC 7826                        RTSP 2.0                   December 2016   o  The RTP/UDP packets from the server to the client MUST be sent to      the address and port given by the first address specification of      the "dest_addr" parameter.   o  The RTCP/UDP packets from the server to the client MUST be sent to      the address and port given by the second address specification of      the "dest_addr" parameter, unless RTP and RTCP multiplexing has      been negotiated, in which case RTCP MUST be sent to the first      address specification.  If no second pair is specified and RTP and      RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent.   o  The RTCP/UDP packets from the client to the server MUST be sent to      the address and port given by the second address specification of      the "src_addr" parameter, unless RTP and RTCP multiplexing has      been negotiated, in which case RTCP MUST be sent to the first      address specification.  If no second pair is specified and RTP and      RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent.   o  The RTP/UDP packets from the client to the server MUST be sent to      the address and port given by the first address specification of      the "src_addr" parameter.   o  RTP and RTCP packets SHOULD be sent from the corresponding      receiver port, i.e., RTCP packets from the server should be sent      from the "src_addr" parameters second address port pair, unless      RTP and RTCP multiplexing has been negotiated in which case the      first address port pair is used.C.1.3.  AVPF/UDP   The RTP profile "Extended RTP Profile for RTCP-based Feedback (RTP/   AVPF)" [RFC4585] MAY be used as RTP profiles in sessions using RTP.   All that is defined for AVP MUST also apply for AVPF.   The usage of AVPF is indicated by the media initialization protocol   used.  In the case of SDP, it is indicated by media lines ("m=")   containing the profile RTP/AVPF.  That SDP MAY also contain further   AVPF-related SDP attributes configuring the AVPF session regarding   reporting interval and feedback messages to be used [RFC4585].  This   configuration MUST be followed.Schulzrinne, et al.          Standards Track                  [Page 274]

RFC 7826                        RTSP 2.0                   December 2016C.1.4.  SAVP/UDP   The RTP profile "The Secure Real-time Transport Protocol (SRTP)"   [RFC3711] is an RTP profile (SAVP) that MAY be used in RTSP sessions   using RTP.  All that is defined for AVP MUST also apply for SAVP.   The usage of SRTP requires that a security context be established.   The default key-management unless otherwise signaled SHALL be MIKEY   in RSA-R mode as defined inAppendix C.1.4.1 and not according to the   procedure defined in "Key Management Extensions for Session   Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)"   [RFC4567].  The reason is thatRFC 4567 sends the initial MIKEY   message in SDP, thus, both requiring the usage of the DESCRIBE method   and forcing the server to keep state for clients performing DESCRIBE   in anticipation that they might require key management.   MIKEY is selected as the default method for establishing SRTP   cryptographic context within an RTSP session as it can be embedded in   the RTSP messages while still ensuring confidentiality of content of   the keying material, even when using hop-by-hop TLS security for the   RTSP messages.  This method also supports pipelining of the RTSP   messages.C.1.4.1.  MIKEY Key Establishment   This method for using MIKEY [RFC3830] to establish the SRTP   cryptographic context is initiated in the client's SETUP request, and   the server's response to the SETUP carries the MIKEY response.  This   ensures that the crypto context establishment happens simultaneously   with the establishment of the media stream being protected.  By using   MIKEY's RSA-R mode [RFC4738] the client can be the initiator and   still allow the server to set the parameters in accordance with the   actual media stream.   The SRTP cryptographic context establishment is done according to the   following process:   1.   The client determines that SAVP or SAVPF shall be used from the        media-description format, e.g., SDP.  If no other key-management        method is explicitly signaled, then MIKEY SHALL be used as        defined herein.  The use of SRTP with RTSP is only defined with        MIKEY with keys established as defined in this section.  Future        documents may define how an RTSP implementation treats SDP that        indicates some other key mechanism to be used.  The need for        such specification includes [RFC4567], which is not defined for        use in RTSP 2.0 within this document.Schulzrinne, et al.          Standards Track                  [Page 275]

RFC 7826                        RTSP 2.0                   December 2016   2.   The client SHALL establish a TLS connection for RTSP messages,        directly or hop-by-hop with the server.  If hop-by-hop TLS        security is used, the User method SHALL be indicated in the        Accept-Credentials header.  Note that using hop-by-hop does        allow the proxy to insert itself as a man in the middle.  This        can also occur in the MIKEY exchange by the proxy providing one        of its certificates rather than the server's in the Connection-        Credentials header.  Therefore, the client SHALL validate the        server certificate.   3.   The client retrieves the server's certificate from a direct TLS        connection or hop-by-hop from a Connection-Credentials header.        The client then checks that the server certificate is valid and        belongs to the server.   4.   The client forms the MIKEY Initiator message using RSA-R mode in        unicast mode as specified in [RFC4738].  The client SHOULD use        the same certificate for TLS and MIKEY to enable the server to        bind the two together.  The client's certificate SHALL be        included in the MIKEY message.  The client SHALL indicate its        SRTP capabilities in the message.   5.   The MIKEY message from the previous step is base64-encoded        [RFC4648] and becomes the value of the MIKEY parameter that is        included in the transport specification(s) that specifies an        SRTP-based profile (SAVP, SAVPF) in the SETUP request.   6.   Any proxy encountering the MIKEY parameter SHALL forward it        without modification.  A proxy that is required to understand        the Transport specifications will need to understand SAVP/SAVPF        with MIKEY to enable the default keying for SRTP-protected media        streams.  If such a proxy does not support SAVP/SAVPF with        MIKEY, it will discard the whole transport specification.  Most        types of proxies can easily support SAVP and SAVPF with MIKEY.        If a client encounters a proxy not supporting SAVP/SAVPF with        MIKEY, the client should attempt bypassing that proxy.   7.   The server, upon receiving the SETUP request, will need to        decide upon the transport specification to use, if multiple are        included by the client.  In the determination of which transport        specifications are supported and preferred, the server SHOULD        decode the MIKEY message to take the embedded SRTP parameters        into account.  If all transport spec require SRTP but no MIKEY        parameter or other supported keying method is included, the        server SHALL respond with 403 (Forbidden).Schulzrinne, et al.          Standards Track                  [Page 276]

RFC 7826                        RTSP 2.0                   December 2016   8.   Upon generating a response, the following outcomes can occur:        *  A transport spec not using SRTP and MIKEY is selected.  Thus,           the response will not contain any MIKEY parameters.        *  A transport spec using SRTP and MIKEY is selected but an           error is encountered in the MIKEY processing.  In this case,           an RTSP error response code of 466 (Key Management Error)           SHALL be used.  A MIKEY message describing the error MAY be           included.        *  A transport spec using SRTP and MIKEY is selected and a MIKEY           response message can be created.  The server SHOULD use the           same certificate for TLS and in MIKEY to enable the client to           bind the two together.  If a different certificate is used,           it SHALL be included in the MIKEY message.  It is RECOMMENDED           that the envelope key-cache type be set to 'Cache' and that a           single envelope key is reused for all MIKEY messages to the           client.  That message is included in the MIKEY parameter part           of the single selected transport specification in the SETUP           response.  The server will set the SRTP parameters as           preferred for this media stream within the supported range by           the client.   9.   The server transmits the SETUP response back to the client.   10.  The client receives the SETUP response and, if the response code        indicates a successful request, it decodes the MIKEY message and        establishes the SRTP cryptographic context from the parameters        in the MIKEY response.   In the above method, the client's certificate may be self signed in   cases where the client's identity is not necessary to authenticate   and the security goal is only to ensure that the RTSP signaling   client is the same as the one receiving the SRTP security context.C.1.5.  SAVPF/UDP   The RTP profile "Extended Secure RTP Profile for Real-time Transport   Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124] is an   RTP profile (SAVPF) that MAY be used in RTSP sessions using RTP.  All   that is defined for AVPF MUST also apply for SAVPF.   The usage of SRTP requires that a cryptographic context be   established.  The default mechanism for establishing that security   association is to use MIKEY[RFC3830] with RTSP as defined inAppendix C.1.4.1.Schulzrinne, et al.          Standards Track                  [Page 277]

RFC 7826                        RTSP 2.0                   December 2016C.1.6.  RTCP Usage with RTSP   RTCP has several usages when RTP is implemented for media transport   as explained below.  Thus, RTCP MUST be supported if an RTSP agent   handles RTP.C.1.6.1.  Media Synchronization   RTCP provides media synchronization and clock-drift compensation.   The initial media synchronization is available from RTP-Info header.   However, to be able to handle any clock drift between the media   streams, RTCP is needed.C.1.6.2.  RTSP Session Keep-Alive   RTCP traffic from the RTSP client to the RTSP server MUST function as   keep-alive.  This requires an RTSP server supporting RTP to use the   received RTCP packets as indications that the client desires the   related RTSP session to be kept alive.C.1.6.3.  Bitrate Adaption   RTCP Receiver reports and any additional feedback from the client   MUST be used to adapt the bitrate used over the transport for all   cases when RTP is sent over UDP.  An RTP sender without reserved   resources MUST NOT use more than its fair share of the available   resources.  This can be determined by comparing on short-to-medium   terms (some seconds) the used bitrate and adapting it so that the RTP   sender sends at a bitrate comparable to what a TCP sender would   achieve on average over the same path.   To ensure that the implementation's adaptation mechanism has a well-   defined outer envelope, all implementations using a non-congestion-   controlled unicast transport protocol, like UDP, MUST implement   "Multimedia Congestion Control: Circuit Breakers for Unicast RTP   Sessions" [RTP-CIRCUIT-BREAKERS].C.1.6.4.  RTP and RTCP Multiplexing   RTSP can be used to negotiate the usage of RTP and RTCP multiplexing   as described in [RFC5761].  This allows servers and client to reduce   the amount of resources required for the session by only requiring   one underlying transport stream per media stream instead of two when   using RTP and RTCP.  This lessens the server-port consumption and   also the necessary state and keep-alive work when operating across   NATs [RFC2663].Schulzrinne, et al.          Standards Track                  [Page 278]

RFC 7826                        RTSP 2.0                   December 2016   Content must be prepared with some consideration for RTP and RTCP   multiplexing, mainly ensuring that the RTP payload types used do not   collide with the ones used for RTCP packet types.  This option likely   needs explicit support from the content unless the RTP payload types   can be remapped by the server and that is correctly reflected in the   session description.  Beyond that, support of this feature should   come at little cost and much gain.   It is recommended that, if the content and server support RTP and   RTCP multiplexing, this is indicated in the session description, for   example, using the SDP attribute "a=rtcp-mux".  If the SDP message   contains the "a=rtcp-mux" attribute for a media stream, the server   MUST support RTP and RTCP multiplexing.  If indicated or otherwise   desired by the client, it can include the Transport parameter "RTCP-   mux" in any transport specification where it desires to use "RTCP-   mux".  The server will indicate if it supports "RTCP-mux".  Servers   and Clients SHOULD support RTP and RTCP multiplexing.   For capability exchange, an RTSP feature tag for RTP and RTCP   multiplexing is defined: "setup.rtp.rtcp.mux".   To minimize the risk of negotiation failure while using RTP and RTCP   multiplexing, some recommendations are here provided.  If the session   description includes explicit indication of support ("a=rtcp-mux" in   SDP), then an RTSP agent can safely create a SETUP request with a   transport specification with only a single "dest_addr" parameter   address specification.  If no such explicit indication is provided,   then even if the feature tag "setup.rtp.rtcp.mux" is provided in a   Supported header by the RTSP server or the feature tag included in   the Required header in the SETUP request, the media resource may not   support RTP and RTCP multiplexing.  Thus, to maximize the probability   of successful negotiation, the RTSP agent is recommended to include   two "dest_addr" parameter address specifications in the first or   first set (if pipelining is used) of SETUP request(s) for any media   resource aggregate.  That way, the RTSP server can accept RTP and   RTCP multiplexing and only use the first address specification or, if   not, use both specifications.  The RTSP agent, after having received   the response for a successful negotiation of the usage of RTP and   RTCP multiplexing, can then release the resources associated with the   second address specification.C.2.  RTP over TCP   Transport of RTP over TCP can be done in two ways: over independent   TCP connections using [RFC4571] or interleaved in the RTSP   connection.  In both cases, the protocol MUST be "rtp" and the lower-   layer MUST be TCP.  The profile may be any of the above specified   ones: AVP, AVPF, SAVP, or SAVPF.Schulzrinne, et al.          Standards Track                  [Page 279]

RFC 7826                        RTSP 2.0                   December 2016C.2.1.  Interleaved RTP over TCP   The use of embedded (interleaved) binary data transported on the RTSP   connection is possible as specified inSection 14.  When using this   declared combination of interleaved binary data, the RTSP messages   MUST be transported over TCP.  TLS may or may not be used.  If TLS is   used, both RTSP messages and the binary data will be protected by   TLS.   One should, however, consider that this will result in all media   streams going through any proxy.  Using independent TCP connections   can avoid that issue.C.2.2.  RTP over Independent TCP   In this section, the sending of RTP [RFC3550] over lower-layer   transport TCP [RFC793] according to "Framing Real-time Transport   Protocol (RTP) and RTP Control Protocol (RTCP) Packets over   Connection-Oriented Transport" [RFC4571] is described.  This section   adapts the guidelines for using RTP over TCP within SIP/SDP [RFC4145]   to work with RTSP.   A client codes the support of RTP over independent TCP by specifying   an RTP/AVP/TCP transport option without an interleaved parameter in   the Transport line of a SETUP request.  This transport option MUST   include the "unicast" parameter.   If the client wishes to use RTP with RTCP, two address specifications   need to be included in the "dest_addr" parameter.  If the client   wishes to use RTP without RTCP, one address specification is included   in the "dest_addr" parameter.  If the client wishes to multiplex RTP   and RTCP on a single transport flow (seeAppendix C.1.6.4), one or   two address specifications are included in the "dest_addr" parameter   in addition to the "RTCP-mux" transport parameter.  Two address   specifications are allowed to facilitate successful negotiation when   the server or content can't support RTP and RTCP multiplexing.   Ordering rules of dest_addr ports follow the rules for RTP/AVP/UDP.   If the client wishes to play the active role in initiating the TCP   connection, it MAY set the setup parameter (seeSection 18.54) on the   Transport line to be "active", or it MAY omit the setup parameter, as   active is the default.  If the client signals the active role, the   ports in the address specifications in the "dest_addr" parameter MUST   be set to 9 (the discard port).   If the client wishes to play the passive role in TCP connection   initiation, it MUST set the setup parameter on the Transport line to   be "passive".  If the client is able to assume the active or theSchulzrinne, et al.          Standards Track                  [Page 280]

RFC 7826                        RTSP 2.0                   December 2016   passive role, it MUST set the setup parameter on the Transport line   to be "actpass".  In either case, the "dest_addr" parameter's address   specification port value for RTP MUST be set to the TCP port number   on which the client is expecting to receive the TCP connection for   RTP, and the "dest_addr" address specification port value for RTCP   MUST be set to the TCP port number on which the client is expecting   to receive the TCP connection for RTCP.  In the case that the client   wishes to multiplex RTP and RTCP on a single transport flow, the   "RTCP-mux" parameter is included and one or two "dest_addr" parameter   address specifications are included, as mentioned earlier in this   section.   Upon receipt of a non-interleaved RTP/AVP/TCP SETUP request, if a   server decides to accept this requested option, the 2xx reply MUST   contain a Transport option that specifies RTP/AVP/TCP (without using   the interleaved parameter and using the unicast parameter).  The   "dest_addr" parameter value MUST be echoed from the parameter value   in the client request unless the destination address (only port) was   not provided; in which case, the server MAY include the source   address of the RTSP TCP connection with the port number unchanged.   In addition, the server reply MUST set the setup parameter on the   Transport line, to indicate the role the server will play in the   connection setup.  Permissible values are "active" (if a client set   setup to "passive" or "actpass") and "passive" (if a client set setup   to "active" or "actpass").   If a server sets setup to "passive", the "src_addr" in the reply MUST   indicate the ports on which the server is willing to receive a TCP   connection for RTP and (if the client requested a TCP connection for   RTCP by specifying two "dest_addr" address specifications) a TCP/   RTCP connection.  If a server sets setup to "active", the ports   specified in "src_addr" address specifications MUST be set to 9.  The   server MAY use the "ssrc" parameter, following the guidance inSection 18.54.  The server sets only one address specification in the   case that the client has indicated only a single address   specification or in case RTP and RTCP multiplexing was requested and   accepted by the server.  Port ordering for "src_addr" follows the   rules for RTP/AVP/UDP.   Servers MUST support taking the passive role and MAY support taking   the active role.  Servers with a public IP address take the passive   role, thus enabling clients behind NATs and firewalls a better chance   of successful connect to the server by actively connecting outwards.   Therefore, the clients are RECOMMENDED to take the active role.Schulzrinne, et al.          Standards Track                  [Page 281]

RFC 7826                        RTSP 2.0                   December 2016   After sending (receiving) a 2xx reply for a SETUP method for a non-   interleaved RTP/AVP/TCP media stream, the active party SHOULD   initiate the TCP connection as soon as possible.  The client MUST NOT   send a PLAY request prior to the establishment of all the TCP   connections negotiated using SETUP for the session.  In case the   server receives a PLAY request in a session that has not yet   established all the TCP connections, it MUST respond using the 464   (Data Transport Not Ready Yet) (Section 17.4.28) error code.   Once the PLAY request for a media resource transported over non-   interleaved RTP/AVP/TCP occurs, media begins to flow from server to   client over the RTP TCP connection, and RTCP packets flow   bidirectionally over the RTCP TCP connection.  Unless RTP and RTCP   multiplexing has been negotiated; in which case, RTP and RTCP will   flow over a common TCP connection.  As in the RTP/UDP case, client-   to-server traffic on an RTP-only TCP session is unspecified by this   memo.  The packets that travel on these connections MUST be framed   using the protocol defined in [RFC4571], not by the framing defined   for interleaving RTP over the RTSP connection defined inSection 14.   A successful PAUSE request for media being transported over RTP/AVP/   TCP pauses the flow of packets over the connections, without closing   the connections.  A successful TEARDOWN request signals that the TCP   connections for RTP and RTCP are to be closed by the RTSP client as   soon as possible.   Subsequent SETUP requests using a URI already set up in an RTSP   session using an RTP/AVP/TCP transport specification may be ambiguous   in the following way: does the client wish to open up a new TCP   connection for RTP or RTCP for the URI, or does the client wish to   continue using the existing TCP connections?  The client SHOULD use   the "connection" parameter (defined inSection 18.54) on the   Transport line to make its intention clear (by setting "connection"   to "new" if new connections are needed, and by setting "connection"   to "existing" if the existing connections are to be used).  After a   2xx reply for a SETUP request for a new connection, parties should   close the preexisting connections, after waiting a suitable period   for any stray RTP or RTCP packets to arrive.   The usage of SRTP, i.e., either SAVP or SAVPF profiles, requires that   a security association be established.  The default mechanism for   establishing that security association is to use MIKEY[RFC3830] with   RTSP as definedAppendix C.1.4.1.Schulzrinne, et al.          Standards Track                  [Page 282]

RFC 7826                        RTSP 2.0                   December 2016   Below, a rewritten version of the example "Media on Demand"   (Appendix A.1) shows the use of RTP/AVP/TCP non-interleaved:      C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0            CSeq: 1            User-Agent: PhonyClient/1.2      M->C: RTSP/2.0 200 OK            CSeq: 1            Server: PhonyServer/1.0            Date: Wed, 23 Jan 2013 15:36:52 +0000            Content-Type: application/sdp            Content-Length: 227            Content-Base: rtsp://example.com/twister.3gp/            Expires: Thu, 24 Jan 2013 15:36:52 +0000            v=0            o=- 2890844256 2890842807 IN IP4 198.51.100.34            s=RTSP Session            i=An Example of RTSP Session Usage            e=adm@example.com            c=IN IP4 0.0.0.0            a=control: *            a=range:npt=00:00:00-00:10:34.10            t=0 0            m=audio 0 RTP/AVP 0            a=control: trackID=1      C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0            CSeq: 2            User-Agent: PhonyClient/1.2            Require: play.basic            Transport: RTP/AVP/TCP;unicast;dest_addr=":9"/":9";                       setup=active;connection=new            Accept-Ranges: npt, smpte, clock      M->C: RTSP/2.0 200 OK            CSeq: 2            Server: PhonyServer/1.0            Transport: RTP/AVP/TCP;unicast;                       dest_addr=":9"/":9";                       src_addr="198.51.100.5:53478"/"198.51.100:54091";                       setup=passive;connection=new;ssrc=93CB001E            Session: OccldOFFq23KwjYpAnBbUr            Expires: Thu, 24 Jan 2013 15:36:52 +0000            Date: Wed, 23 Jan 2013 15:36:52 +0000            Accept-Ranges: npt            Media-Properties: Random-Access=0.8, Immutable, UnlimitedSchulzrinne, et al.          Standards Track                  [Page 283]

RFC 7826                        RTSP 2.0                   December 2016      C->M: TCP Connection Establishment x2      C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0            CSeq: 4            User-Agent: PhonyClient/1.2            Range: npt=30-            Session: OccldOFFq23KwjYpAnBbUr      M->C: RTSP/2.0 200 OK            CSeq: 4            Server: PhonyServer/1.0            Date: Wed, 23 Jan 2013 15:36:54 +0000            Session: OccldOFFq23KwjYpAnBbUr            Range: npt=30-623.10            Seek-Style: First-Prior            RTP-Info:  url="rtsp://example.com/twister.3gp/trackID=1"               ssrc=4F312DD8:seq=54321;rtptime=2876889C.3.  Handling Media-Clock Time Jumps in the RTP Media Layer   RTSP allows media clients to control selected, non-contiguous   sections of media presentations, rendering those streams with an RTP   media layer [RFC3550].  Two cases occur, the first is when a new PLAY   request replaces an old ongoing request and the new request results   in a jump in the media.  This should produce continuous media stream   at the RTP layer.  A client may also immediately follow a completed   PLAY request with a new PLAY request.  This will result in some gap   in the media layer.  The below text will look into both cases.   A PLAY request that replaces an ongoing PLAY request allows the media   layer rendering the RTP stream to do so continuously without being   affected by jumps in media-clock time.  The RTP timestamps for the   new media range are set so that they become continuous with the   previous media range in the previous request.  The RTP sequence   number for the first packet in the new range will be the next   following the last packet in the previous range, i.e., monotonically   increasing.  The goal is to allow the media-rendering layer to work   without interruption or reconfiguration across the jumps in media   clock.  This should be possible in all cases of replaced PLAY   requests for media that has random access properties.  In this case,   care is needed to align frames or similar media-dependent structures.   In cases where jumps in media-clock time are a result of RTSP   signaling operations arriving after a completed PLAY operation, the   request timing will result in that media becoming non-continuous.   The server becomes unable to send the media so that it arrives timely   and still carries timestamps to make the media stream continuous.  In   these situations, the server will produce RTP streams where there areSchulzrinne, et al.          Standards Track                  [Page 284]

RFC 7826                        RTSP 2.0                   December 2016   gaps in the RTP timeline for the media.  If the media has frame   structure, aligning the timestamp for the next frame with the   previous structure reduces the burden to render this media.  The gap   should represent the time the server hasn't been serving media, e.g.,   the time between the end of the media stream or a PAUSE request and   the new PLAY request.  In these cases, the RTP sequence number would   normally be monotonically increasing across the gap.   For RTSP sessions with media that lacks random access properties,   such as live streams, any media-clock jump is commonly the result of   a correspondingly long pause of delivery.  The RTP timestamp will   have increased in direct proportion to the duration of the paused   delivery.  Note also that in this case the RTP sequence number should   be the next packet number.  If not, the RTCP packet loss reporting   will indicate as loss all packets not received between the point of   pausing and later resuming.  This may trigger congestion avoidance   mechanisms.  An allowed exception from the above recommendation on   monotonically increasing RTP sequence number is live media streams,   likely being relayed.  In this case, when the client resumes   delivery, it will get the media that is currently being delivered to   the server itself.  For this type of basic delivery of live streams   to multiple users over unicast, individual rewriting of RTP sequence   numbers becomes quite a burden.  For solutions that already cache   media or perform time shifting, the rewriting should impose only a   minor burden.   The goal when handling jumps in media-clock time is that the provided   stream is continuous without gaps in RTP timestamp or sequence   number.  However, when delivery has been halted for some reason, the   RTP timestamp, when resuming, MUST represent the duration that the   delivery was halted.  An RTP sequence number MUST generally be the   next number, i.e., monotonically increasing modulo 65536.  For media   resources with the properties Time-Progressing and Time-Duration=0.0,   the server MAY create RTP media streams with RTP sequence number   jumps in them due to the client first halting delivery and later   resuming it (PAUSE and then later PLAY).  However, servers utilizing   this exception must take into consideration the resulting RTCP   receiver reports that likely contain loss reports for all the packets   that were a part of the discontinuity.  A client cannot rely on the   fact that a server will align when resuming play, even if it is   RECOMMENDED.  The RTP-Info header will provide information on how the   server acts in each case.      One cannot assume that the RTSP client can communicate with the      RTP media agent, as the two may be independent processes.  If the      RTP timestamp shows the same gap as the NPT, the media agent will      assume that there is a pause in the presentation.  If the jump in      NPT is large enough, the RTP timestamp may roll over and the mediaSchulzrinne, et al.          Standards Track                  [Page 285]

RFC 7826                        RTSP 2.0                   December 2016      agent may believe later packets to be duplicates of packets just      played out.  Having the RTP timestamp jump will also affect the      RTCP measurements based on this.   As an example, assume an RTP timestamp frequency of 8000 Hz, a   packetization interval of 100 ms, and an initial sequence number and   timestamp of zero.      C->S: PLAY rtsp://example.com/fizzle RTSP/2.0        CSeq: 4        Session: ymIqLXufHkMHGdtENdblWK        Range: npt=10-15        User-Agent: PhonyClient/1.2      S->C: RTSP/2.0 200 OK        CSeq: 4        Session: ymIqLXufHkMHGdtENdblWK        Range: npt=10-15        RTP-Info: url="rtsp://example.com/fizzle/audiotrack"                  ssrc=0D12F123:seq=0;rtptime=0   The ensuing RTP data stream is depicted below:      S -> C: RTP packet - seq = 0,  rtptime = 0,     NPT time = 10s      S -> C: RTP packet - seq = 1,  rtptime = 800,   NPT time = 10.1s       . . .      S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s   Upon the completion of the requested delivery, the server sends a   PLAY_NOTIFY.        S->C: PLAY_NOTIFY rtsp://example.com/fizzle RTSP/2.0              CSeq: 5              Notify-Reason: end-of-stream              Request-Status: cseq=4 status=200 reason="OK"              Range: npt=-15              RTP-Info:url="rtsp://example.com/fizzle/audiotrack"                 ssrc=0D12F123:seq=49;rtptime=39200              Session: ymIqLXufHkMHGdtENdblWK        C->S: RTSP/2.0 200 OK              CSeq: 5              User-Agent: PhonyClient/1.2   Upon the completion of the play range, the client follows up with a   request to PLAY from a new NPT.Schulzrinne, et al.          Standards Track                  [Page 286]

RFC 7826                        RTSP 2.0                   December 2016   C->S: PLAY rtsp://example.com/fizzle RTSP/2.0         CSeq: 6         Session: ymIqLXufHkMHGdtENdblWK         Range: npt=18-20         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 6         Session: ymIqLXufHkMHGdtENdblWK         Range: npt=18-20         RTP-Info: url="rtsp://example.com/fizzle/audiotrack"                   ssrc=0D12F123:seq=50;rtptime=40100   The ensuing RTP data stream is depicted below:      S->C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s      S->C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s       . . .      S->C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s   In this example, first, NPT 10 through 15 are played, then the client   requests the server to skip ahead and play NPT 18 through 20.  The   first segment is presented as RTP packets with sequence numbers 0   through 49 and timestamps 0 through 39,200.  The second segment   consists of RTP packets with sequence numbers 50 through 69, with   timestamps 40,100 through 55,200.  While there is a gap in the NPT,   there is no gap in the sequence-number space of the RTP data stream.   The RTP timestamp gap is present in the above example due to the time   it takes to perform the second play request, in this case, 12.5 ms   (100/8000).C.4.  Handling RTP Timestamps after PAUSE   During a PAUSE/PLAY interaction in an RTSP session, the duration of   time for which the RTP transmission was halted MUST be reflected in   the RTP timestamp of each RTP stream.  The duration can be calculated   for each RTP stream as the time elapsed from when the last RTP packet   was sent before the PAUSE request was received and when the first RTP   packet was sent after the subsequent PLAY request was received.  The   duration includes all latency incurred and processing time required   to complete the request.RFC 3550 [RFC3550] states that: "the RTP timestamp for each unit      [packet] would be related to the wallclock time at which the unit      becomes current on the virtual presentation timeline".Schulzrinne, et al.          Standards Track                  [Page 287]

RFC 7826                        RTSP 2.0                   December 2016      In order to satisfy the requirements of [RFC3550], the RTP      timestamp space needs to increase continuously with real time.      While this is not optimal for stored media, it is required for RTP      and RTCP to function as intended.  Using a continuous RTP      timestamp space allows the same timestamp model for both stored      and live media and allows better opportunity to integrate both      types of media under a single control.   As an example, assume a clock frequency of 8000 Hz, a packetization   interval of 100 ms, and an initial sequence number and timestamp of   zero.   C->S: PLAY rtsp://example.com/fizzle RTSP/2.0         CSeq: 4         Session: ymIqLXufHkMHGdtENdblWK         Range: npt=10-15         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 4         Session: ymIqLXufHkMHGdtENdblWK         Range: npt=10-15         RTP-Info: url="rtsp://example.com/fizzle/audiotrack"                   ssrc=0D12F123:seq=0;rtptime=0   The ensuing RTP data stream is depicted below:      S -> C: RTP packet - seq = 0, rtptime = 0,    NPT time = 10s      S -> C: RTP packet - seq = 1, rtptime = 800,  NPT time = 10.1s      S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s      S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3sSchulzrinne, et al.          Standards Track                  [Page 288]

RFC 7826                        RTSP 2.0                   December 2016   The client then sends a PAUSE request:   C->S: PAUSE rtsp://example.com/fizzle RTSP/2.0         CSeq: 5         Session: ymIqLXufHkMHGdtENdblWK         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 5         Session: ymIqLXufHkMHGdtENdblWK         Range: npt=10.4-15   20 seconds elapse and then the client sends a PLAY request.  In   addition, the server requires 15 ms to process the request:   C->S: PLAY rtsp://example.com/fizzle RTSP/2.0         CSeq: 6         Session: ymIqLXufHkMHGdtENdblWK         User-Agent: PhonyClient/1.2   S->C: RTSP/2.0 200 OK         CSeq: 6         Session: ymIqLXufHkMHGdtENdblWK         Range: npt=10.4-15         RTP-Info: url="rtsp://example.com/fizzle/audiotrack"                   ssrc=0D12F123:seq=4;rtptime=164400   The ensuing RTP data stream is depicted below:      S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s      S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s      S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s   First, NPT 10 through 10.3 is played, then a PAUSE is received by the   server.  After 20 seconds, a PLAY is received by the server that   takes 15 ms to process.  The duration of time for which the session   was paused is reflected in the RTP timestamp of the RTP packets sent   after this PLAY request.   A client can use the RTSP Range header and RTP-Info header to map NPT   time of a presentation with the RTP timestamp.   Note: inRFC 2326 [RFC2326], this matter was not clearly defined and   was misunderstood commonly.  However, for RTSP 2.0, it is expected   that this will be handled correctly and no exception handling will be   required.Schulzrinne, et al.          Standards Track                  [Page 289]

RFC 7826                        RTSP 2.0                   December 2016   Note further: it may be required to reset some of the state to ensure   the correct media decoding and the usual jitter-buffer handling when   issuing a PLAY request.C.5.  RTSP/RTP Integration   For certain data types, tight integration between the RTSP layer and   the RTP layer will be necessary.  This by no means precludes the   above restrictions.  Combined RTSP/RTP media clients should use the   RTP-Info field to determine whether incoming RTP packets were sent   before or after a seek or before or after a PAUSE.C.6.  Scaling with RTP   For scaling (seeSection 18.46), RTP timestamps should correspond to   the rendering timing.  For example, when playing video recorded at 30   frames per second at a scale of two and speed (Section 18.50) of one,   the server would drop every second frame to maintain and deliver   video packets with the normal timestamp spacing of 3,000 per frame,   but NPT would increase by 1/15 second for each video frame.      Note: the above scaling puts requirements on the media codec or a      media stream to support it.  For example, motion JPEG or other      non-predictive video coding can easier handle the above example.C.7.  Maintaining NPT Synchronization with RTP Timestamps   The client can maintain a correct display of NPT by noting the RTP   timestamp value of the first packet arriving after repositioning.   The sequence parameter of the RTP-Info (Section 18.45) header   provides the first sequence number of the next segment.C.8.  Continuous Audio   For continuous audio, the server SHOULD set the RTP marker bit at the   beginning of serving a new PLAY request or at jumps in timeline.   This allows the client to perform playout delay adaptation.C.9.  Multiple Sources in an RTP Session   Note that more than one SSRC MAY be sent in the media stream.  If it   happens, all sources are expected to be rendered simultaneously.C.10.  Usage of SSRCs and the RTCP BYE Message during an RTSP Session   The RTCP BYE message indicates the end of use of a given SSRC.  If   all sources leave an RTP session, it can, in most cases, be assumed   to have ended.  Therefore, a client or server MUST NOT send an RTCPSchulzrinne, et al.          Standards Track                  [Page 290]

RFC 7826                        RTSP 2.0                   December 2016   BYE message until it has finished using a SSRC.  A server SHOULD keep   using an SSRC until the RTP session is terminated.  Prolonging the   use of a SSRC allows the established synchronization context   associated with that SSRC to be used to synchronize subsequent PLAY   requests even if the PLAY response is late.   An SSRC collision with the SSRC that transmits media does also have   consequences, as it will normally force the media sender to change   its SSRC in accordance with the RTP specification [RFC3550].   However, an RTSP server may wait and see if the client changes and   thus resolve the conflict to minimize the impact.  As media sender,   SSRC change will result in a loss of synchronization context and   require any receiver to wait for RTCP sender reports for all media   requiring synchronization before being able to play out synchronized.   Due to these reasons, a client joining a session should take care not   to select the same SSRC(s) as the server indicates in the ssrc   Transport header parameter.  Any SSRC signaled in the Transport   header MUST be avoided.  A client detecting a collision prior to   sending any RTP or RTCP messages SHALL also select a new SSRC.C.11.  Future Additions   It is the intention that any future protocol or profile regarding   media delivery and lower transport should be easy to add to RTSP.   This section provides the necessary steps that need to be met.   The following things need to be considered when adding a new protocol   or profile for use with RTSP:   o  The protocol or profile needs to define a name tag representing      it.  This tag is required to be an ABNF "token" to be possible to      use in the Transport header specification.   o  The useful combinations of protocol, profiles, and lower-layer      transport for this extension need to be defined.  For each      combination, declare the necessary parameters to use in the      Transport header.   o  For new media protocols, the interaction with RTSP needs to be      addressed.  One important factor will be the media      synchronization.  It may be necessary to have new headers similar      to RTP info to carry this information.   o  Discussion needs to occur regarding congestion control for media,      especially if transport without built-in congestion control is      used.Schulzrinne, et al.          Standards Track                  [Page 291]

RFC 7826                        RTSP 2.0                   December 2016   See the IANA Considerations section (Section 22) for information on   how to register new attributes.Appendix D.  Use of SDP for RTSP Session Descriptions   The Session Description Protocol (SDP, [RFC4566]) may be used to   describe streams or presentations in RTSP.  This description is   typically returned in reply to a DESCRIBE request on a URI from a   server to a client or received via HTTP from a server to a client.   This appendix describes how an SDP file determines the operation of   an RTSP session.  Thus, it is worth pointing out that the   interpretation of the SDP is done in the context of the SDP receiver,   which is the one being configured.  This is the same as in SAP   [RFC2974]; this differs from SDP Offer/Answer [RFC3264] where each   SDP is interpreted in the context of the agent providing it.   SDP as is provides no mechanism by which a client can distinguish,   without human guidance, between several media streams to be rendered   simultaneously and a set of alternatives (e.g., two audio streams   spoken in different languages).  The SDP extension found in "The   Session Description Protocol (SDP) Grouping Framework" [RFC5888]   provides such functionality to some degree.Appendix D.4 describes   the usage of SDP media line grouping for RTSP.D.1.  Definitions   The terms "session-level", "media-level", and other key/attribute   names and values used in this appendix are to be used as defined in   SDP [RFC4566]:D.1.1.  Control URI   The "a=control" attribute is used to convey the control URI.  This   attribute is used both for the session and media descriptions.  If   used for individual media, it indicates the URI to be used for   controlling that particular media stream.  If found at the session   level, the attribute indicates the URI for aggregate control   (presentation URI).  The session-level URI MUST be different from any   media-level URI.  The presence of a session-level control attribute   MUST be interpreted as support for aggregated control.  The control   attribute MUST be present on the media level unless the presentation   only contains a single media stream; in which case, the attribute MAY   be present on the session level only and then also apply to that   single media stream.   ABNF for the attribute is defined inSection 20.3.Schulzrinne, et al.          Standards Track                  [Page 292]

RFC 7826                        RTSP 2.0                   December 2016   Example:     a=control:rtsp://example.com/foo   This attribute MAY contain either relative or absolute URIs,   following the rules and conventions set out inRFC 3986 [RFC3986].   Implementations MUST look for a base URI in the following order:   1.  the RTSP Content-Base field;   2.  the RTSP Content-Location field;   3.  the RTSP Request-URI.   If this attribute contains only an asterisk (*), then the URI MUST be   treated as if it were an empty embedded URI; thus, it will inherit   the entire base URI.      Note:RFC 2326 was very unclear on the processing of relative URIs      and several RTSP 1.0 implementations at the point of publishing      this document did not performRFC 3986 processing to determine the      resulting URI; instead, simple concatenation is common.  To avoid      this issue completely, it is recommended to use absolute URIs in      the SDP.   The URI handling for SDPs from container files needs special   consideration.  For example, let's assume that a container file has   the URI: "rtsp://example.com/container.mp4".  Let's further assume   this URI is the base URI and that there is an absolute media-level   URI: "rtsp://example.com/container.mp4/trackID=2".  A relative media-   level URI that resolves in accordance withRFC 3986 [RFC3986] to the   above given media URI is "container.mp4/trackID=2".  It is usually   not desirable to need to include or modify the SDP stored within the   container file with the server local name of the container file.  To   avoid this, one can modify the base URI used to include a trailing   slash, e.g., "rtsp://example.com/container.mp4/".  In this case, the   relative URI for the media will only need to be "trackID=2".   However, this will also mean that using "*" in the SDP will result in   the control URI including the trailing slash, i.e.,   "rtsp://example.com/container.mp4/".      Note: the usage of TrackID in the above is not a standardized      form, but one example out of several similar strings such as      TrackID, Track_ID, StreamID that is used by different server      vendors to indicate a particular piece of media inside a container      file.Schulzrinne, et al.          Standards Track                  [Page 293]

RFC 7826                        RTSP 2.0                   December 2016D.1.2.  Media Streams   The "m=" field is used to enumerate the streams.  It is expected that   all the specified streams will be rendered with appropriate   synchronization.  If the session is over multicast, the port number   indicated SHOULD be used for reception.  The client MAY try to   override the destination port, through the Transport header.  The   servers MAY allow this: the response will indicate whether or not   this is allowed.  If the session is unicast, the port numbers are the   ones RECOMMENDED by the server to the client, about which receiver   ports to use; the client MUST still include its receiver ports in its   SETUP request.  The client MAY ignore this recommendation.  If the   server has no preference, it SHOULD set the port number value to   zero.   The "m=" lines contain information about which transport protocol,   profile, and possibly lower-layer are to be used for the media   stream.  The combination of transport, profile, and lower layer, like   RTP/AVP/UDP, needs to be defined for how to be used with RTSP.  The   currently defined combinations are discussed inAppendix C; further   combinations MAY be specified.   Example:     m=audio 0 RTP/AVP 31D.1.3.  Payload Type(s)   The payload type or types are specified in the "m=" line.  In case   the payload type is a static payload type fromRFC 3551 [RFC3551], no   other information may be required.  In case it is a dynamic payload   type, the media attribute "rtpmap" is used to specify what the media   is.  The "encoding name" within the "rtpmap" attribute may be one of   those specified in [RFC4856], a media type registered with IANA   according to [RFC4855], or an experimental encoding as specified in   SDP [RFC4566]).  Codec-specific parameters are not specified in this   field, but rather in the "fmtp" attribute described below.   The selection of the RTP payload type numbers used may be required to   consider RTP and RTCP Multiplexing [RFC5761], if that is to be   supported by the server.D.1.4.  Format-Specific Parameters   Format-specific parameters are conveyed using the "fmtp" media   attribute.  The syntax of the "fmtp" attribute is specific to the   encoding(s) to which the attribute refers.  Note that some of theSchulzrinne, et al.          Standards Track                  [Page 294]

RFC 7826                        RTSP 2.0                   December 2016   format-specific parameters may be specified outside of the "fmtp"   parameters, for example, like the "ptime" attribute for most audio   encodings.D.1.5.  Directionality of Media Stream   The SDP attributes "a=sendrecv", "a=recvonly", and "a=sendonly"   provide instructions about the direction the media streams flow   within a session.  When using RTSP, the SDP can be delivered to a   client using either RTSP DESCRIBE or a number of RTSP external   methods, like HTTP, FTP, and email.  Based on this, the SDP applies   to how the RTSP client will see the complete session.  Thus, media   streams delivered from the RTSP server to the client would be given   the "a=recvonly" attribute.   "a=recvonly" in an SDP provided to the RTSP client indicates that   media delivery will only occur in the direction from the RTSP server   to the client.  SDP provided to the RTSP client that lacks any of the   directionality attributes ("a=recvonly", "a=sendonly", "a=sendrecv")   would be interpreted as having "a=sendrecv".  At the time of writing,   there exists no RTSP mode suitable for media traffic in the direction   from the RTSP client to the server.  Thus, all RTSP SDP SHOULD have   an "a=recvonly" attribute when using the PLAY mode defined in this   document.  If future modes are defined for media in the client-to-   server direction, then usage of "a=sendonly" or "a=sendrecv" may   become suitable to indicate intended media directions.D.1.6.  Range of Presentation   The "a=range" attribute defines the total time range of the stored   session or an individual media.  Live sessions that are not seekable   can be indicated as specified below; whereas the length of live   sessions can be deduced from the "t=" and "r=" SDP parameters.   The attribute is both a session- and a media-level attribute.  For   presentations that contain media streams of the same duration, the   range attribute SHOULD only be used at the session level.  In case of   different lengths, the range attribute MUST be given at media level   for all media and SHOULD NOT be given at the session level.  If the   attribute is present at both media level and session level, the   media-level values MUST be used.   Note: usually one will specify the same length for all media, even if   there isn't media available for the full duration on all media.   However, that requires that the server accept PLAY requests within   that range.Schulzrinne, et al.          Standards Track                  [Page 295]

RFC 7826                        RTSP 2.0                   December 2016   Servers MUST take care to provide RTSP Range (seeSection 18.40)   values that are consistent with what is presented in the SDP for the   content.  There is no reason for non dynamic content, like media   clips provided on demand to have inconsistent values.  Inconsistent   values between the SDP and the actual values for the content handled   by the server is likely to generate some failure, like 457 "Invalid   Range", in case the client uses PLAY requests with a Range header.   In case the content is dynamic in length and it is infeasible to   provide a correct value in the SDP, the server is recommended to   describe this as content that is not seekable (see below).  The   server MAY override that property in the response to a PLAY request   using the correct values in the Range header.   The unit is specified first, followed by the value range.  The units   and their values are as defined inSection 4.4.1,Section 4.4.2, andSection 4.4.3 and MAY be extended with further formats.  Any open-   ended range (start-), i.e., without stop range, is of unspecified   duration and MUST be considered as content that is not seekable   unless this property is overridden.  Multiple instances carrying   different clock formats MAY be included at either session or media   level.   ABNF for the attribute is defined inSection 20.3.   Examples:     a=range:npt=0-34.4368     a=range:clock=19971113T211503Z-19971113T220300Z     Non-seekable stream of unknown duration:     a=range:npt=0-D.1.7.  Time of Availability   The "t=" field defines when the SDP is valid.  For on-demand content,   the server SHOULD indicate a stop time value for which it guarantees   the description to be valid and a start time that is equal to or   before the time at which the DESCRIBE request was received.  It MAY   also indicate start and stop times of 0, meaning that the session is   always available.   For sessions that are of live type, i.e., specific start time,   unknown stop time, likely not seekable, the "t=" and "r=" field   SHOULD be used to indicate the start time of the event.  The stop   time SHOULD be given so that the live event will have ended at that   time, while still not being unnecessary far into the future.Schulzrinne, et al.          Standards Track                  [Page 296]

RFC 7826                        RTSP 2.0                   December 2016D.1.8.  Connection Information   In SDP used with RTSP, the "c=" field contains the destination   address for the media stream.  If a multicast address is specified,   the client SHOULD use this address in any SETUP request as   destination address, including any additional parameters, such as   TTL.  For on-demand unicast streams and some multicast streams, the   destination address MAY be specified by the client via the SETUP   request, thus overriding any specified address.  To identify streams   without a fixed destination address, where the client is required to   specify a destination address, the "c=" field SHOULD be set to a null   value.  For addresses of type "IP4", this value MUST be "0.0.0.0";   and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0" (can also be   written as "::"), i.e., the unspecified address according toRFC 4291   [RFC4291].D.1.9.  Message Body Tag   The optional "a=mtag" attribute identifies a version of the session   description.  It is opaque to the client.  SETUP requests may include   this identifier in the If-Match field (seeSection 18.24) to allow   session establishment only if this attribute value still corresponds   to that of the current description.  The attribute value is opaque   and may contain any character allowed within SDP attribute values.   ABNF for the attribute is defined inSection 20.3.   Example:     a=mtag:"158bb3e7c7fd62ce67f12b533f06b83a"      One could argue that the "o=" field provides identical      functionality.  However, it does so in a manner that would put      constraints on servers that need to support multiple session      description types other than SDP for the same piece of media      content.Schulzrinne, et al.          Standards Track                  [Page 297]

RFC 7826                        RTSP 2.0                   December 2016D.2.  Aggregate Control Not Available   If a presentation does not support aggregate control, no session-   level "a=control" attribute is specified.  For an SDP with multiple   media sections specified, each section will have its own control URI   specified via the "a=control" attribute.   Example:   v=0   o=- 2890844256 2890842807 IN IP4 192.0.2.56   s=I came from a web page   e=adm@example.com   c=IN IP4 0.0.0.0   t=0 0   m=video 8002 RTP/AVP 31   a=control:rtsp://audio.example.com/movie.aud   m=audio 8004 RTP/AVP 3   a=control:rtsp://video.example.com/movie.vid   Note that the position of the control URI in the description implies   that the client establishes separate RTSP control sessions to the   servers audio.example.com and video.example.com.   It is recommended that an SDP file contain the complete media-   initialization information even if it is delivered to the media   client through non-RTSP means.  This is necessary as there is no   mechanism to indicate that the client should request more detailed   media stream information via DESCRIBE.D.3.  Aggregate Control Available   In this scenario, the server has multiple streams that can be   controlled as a whole.  In this case, there are both a media-level   "a=control" attribute, which is used to specify the stream URIs, and   a session-level "a=control" attribute, which is used as the Request-   URI for aggregate control.  If the media-level URI is relative, it is   resolved to absolute URIs according toAppendix D.1.1 above.Schulzrinne, et al.          Standards Track                  [Page 298]

RFC 7826                        RTSP 2.0                   December 2016   Example:   C->M: DESCRIBE rtsp://example.com/movie RTSP/2.0         CSeq: 1         User-Agent: PhonyClient/1.2   M->C: RTSP/2.0 200 OK         CSeq: 1         Date: Wed, 23 Jan 2013 15:36:52 +0000         Expires: Wed, 23 Jan 2013 16:36:52 +0000         Content-Type: application/sdp         Content-Base: rtsp://example.com/movie/         Content-Length: 227         v=0         o=- 2890844256 2890842807 IN IP4 192.0.2.211         s=I contain         i=<more info>         e=adm@example.com         c=IN IP4 0.0.0.0         a=control:*         t=0 0         m=video 8002 RTP/AVP 31         a=control:trackID=1         m=audio 8004 RTP/AVP 3         a=control:trackID=2   In this example, the client is recommended to establish a single RTSP   session to the server, and it uses the URIs rtsp://example.com/movie/   trackID=1 and rtsp://example.com/movie/trackID=2 to set up the video   and audio streams, respectively.  The URI rtsp://example.com/movie/,   which is resolved from the "*", controls the whole presentation   (movie).   A client is not required to issue SETUP requests for all streams   within an aggregate object.  Servers should allow the client to ask   for only a subset of the streams.D.4.  Grouping of Media Lines in SDP   For some types of media, it is desirable to express a relationship   between various media components, for instance, for lip   synchronization or Scalable Video Codec (SVC) [RFC5583].  This   relationship is expressed on the SDP level by grouping of media   lines, as described in [RFC5888], and can be exposed to RTSP.Schulzrinne, et al.          Standards Track                  [Page 299]

RFC 7826                        RTSP 2.0                   December 2016   For RTSP, it is mainly important to know how to handle grouped media   received by means of SDP, i.e., if the media are under aggregate   control (seeAppendix D.3) or if aggregate control is not available   (seeAppendix D.2).   It is RECOMMENDED that grouped media are handled by aggregate   control, to give the client the ability to control either the whole   presentation or single media.D.5.  RTSP External SDP Delivery   There are some considerations that need to be made when the session   description is delivered to the client outside of RTSP, for example   via HTTP or email.   First of all, the SDP needs to contain absolute URIs, since relative   will, in most cases, not work as the delivery will not correctly   forward the base URI.   The writing of the SDP session availability information, i.e., "t="   and "r=", needs to be carefully considered.  When the SDP is fetched   by the DESCRIBE method, the probability that it is valid is very   high.  However, the same is much less certain for SDPs distributed   using other methods.  Therefore, the publisher of the SDP should take   care to follow the recommendations about availability in the SDP   specification [RFC4566] inSection 4.2.Appendix E.  RTSP Use Cases   This appendix describes the most important and considered use cases   for RTSP.  They are listed in descending order of importance in   regard to ensuring that all necessary functionality is present.  This   specification only fully supports usage of the two first.  Also, in   these first two cases, there are special cases or exceptions that are   not supported without extensions, e.g., the redirection of media   delivery to an address other than the controlling agent's (client's).E.1.  On-Demand Playback of Stored Content   An RTSP-capable server stores content suitable for being streamed to   a client.  A client desiring playback of any of the stored content   uses RTSP to set up the media transport required to deliver the   desired content.  RTSP is then used to initiate, halt, and manipulate   the actual transmission (playout) of the content.  RTSP is also   required to provide the necessary description and synchronization   information for the content.Schulzrinne, et al.          Standards Track                  [Page 300]

RFC 7826                        RTSP 2.0                   December 2016   The above high-level description can be broken down into a number of   functions of which RTSP needs to be capable.   Presentation Description:  Provide initialization information about         the presentation (content); for example, which media codecs are         needed for the content.  Other information that is important         includes the number of media streams the presentation contains,         the transport protocols used for the media streams, and         identifiers for these media streams.  This information is         required before setup of the content is possible and to         determine if the client is even capable of using the content.         This information need not be sent using RTSP; other external         protocols can be used to transmit the transport presentation         descriptions.  Two good examples are the use of HTTP [RFC7230]         or email to fetch or receive presentation descriptions like SDP         [RFC4566]   Setup:  Set up some or all of the media streams in a presentation.         The setup itself consists of selecting the protocol for media         transport and the necessary parameters for the protocol, like         addresses and ports.   Control of Transmission:  After the necessary media streams have been         established, the client can request the server to start         transmitting the content.  The client must be allowed to start         or stop the transmission of the content at arbitrary times.         The client must also be able to start the transmission at any         point in the timeline of the presentation.   Synchronization:  For media-transport protocols like RTP [RFC3550],         it might be beneficial to carry synchronization information         within RTSP.  This may be due to either the lack of inter-media         synchronization within the protocol itself or the potential         delay before the synchronization is established (which is the         case for RTP when using RTCP).   Termination:  Terminate the established contexts.   For this use case, there are a number of assumptions about how it   works.  These are:   On-Demand content:  The content is stored at the server and can be         accessed at any time during a time period when it is intended         to be available.Schulzrinne, et al.          Standards Track                  [Page 301]

RFC 7826                        RTSP 2.0                   December 2016   Independent sessions:  A server is capable of serving a number of         clients simultaneously, including from the same piece of         content at different points in that presentations timeline.   Unicast Transport:  Content for each individual client is transmitted         to them using unicast traffic.   It is also possible to redirect the media traffic to a different   destination than that of the agent controlling the traffic.  However,   allowing this without appropriate mechanisms for checking that the   destination approves of this allows for Distributed DoS (DDoS).E.2.  Unicast Distribution of Live Content   This use case is similar to the above on-demand content case (seeAppendix E.1), the difference is the nature of the content itself.   Live content is continuously distributed as it becomes available from   a source; i.e., the main difference from on-demand is that one starts   distributing content before the end of it has become available to the   server.   In many cases, the consumer of live content is only interested in   consuming what actually happens "now"; i.e., very similar to   broadcast TV.  However, in this case, it is assumed that there exists   no broadcast or multicast channel to the users, and instead the   server functions as a distribution node, sending the same content to   multiple receivers, using unicast traffic between server and client.   This unicast traffic and the transport parameters are individually   negotiated for each receiving client.   Another aspect of live content is that it often has a very limited   time of availability, as it is only available for the duration of the   event the content covers.  An example of such live content could be a   music concert that lasts two hours and starts at a predetermined   time.  Thus, there is a need to announce when and for how long the   live content is available.   In some cases, the server providing live content may be saving some   or all of the content to allow clients to pause the stream and resume   it from the paused point, or to "rewind" and play continuously from a   point earlier than the live point.  Hence, this use case does not   necessarily exclude playing from other than the live point of the   stream, playing with scales other than 1.0, etc.Schulzrinne, et al.          Standards Track                  [Page 302]

RFC 7826                        RTSP 2.0                   December 2016E.3.  On-Demand Playback Using Multicast   It is possible to use RTSP to request that media be delivered to a   multicast group.  The entity setting up the session (the controller)   will then control when and what media is delivered to the group.   This use case has some potential for DoS attacks by flooding a   multicast group.  Therefore, a mechanism is needed to indicate that   the group actually accepts the traffic from the RTSP server.   An open issue in this use case is how one ensures that all receivers   listening to the multicast or broadcast receives the session   presentation configuring the receivers.  This specification has to   rely on an external solution to solve this issue.E.4.  Inviting an RTSP Server into a Conference   If one has an established conference or group session, it is possible   to have an RTSP server distribute media to the whole group.   Transmission to the group is simplest when controlled by a single   participant or leader of the conference.  Shared control might be   possible, but would require further investigation and possibly   extensions.   This use case assumes that there exists either a multicast or a   conference focus that redistributes media to all participants.   This use case is intended to be able to handle the following   scenario: a conference leader or participant (hereafter called the   "controller") has some pre-stored content on an RTSP server that he   wants to share with the group.  The controller sets up an RTSP   session at the streaming server for this content and retrieves the   session description for the content.  The destination for the media   content is set to the shared multicast group or conference focus.   When desired by the controller, he/she can start and stop the   transmission of the media to the conference group.   There are several issues with this use case that are not solved by   this core specification for RTSP:   DoS:  To avoid an RTSP server from being an unknowing participant in         a DoS attack, the server needs to be able to verify the         destination's acceptance of the media.  Such a mechanism to         verify the approval of received media does not yet exist;         instead, only policies can be used, which can be made to work         in controlled environments.Schulzrinne, et al.          Standards Track                  [Page 303]

RFC 7826                        RTSP 2.0                   December 2016   Distributing the presentation description to all participants in the   group:            To enable a media receiver to correctly decode the content,            the media configuration information needs to be distributed            reliably to all participants.  This will most likely require            support from an external protocol.      Passing control of the session:  If it is desired to pass control            of the RTSP session between the participants, some support            will be required by an external protocol to exchange state            information and possibly floor control of who is controlling            the RTSP session.E.5.  Live Content Using Multicast   This use case in its simplest form does not require any use of RTSP   at all; this is what multicast conferences being announced with SAP   [RFC2974] and SDP are intended to handle.  However, in use cases   where more advanced features like access control to the multicast   session are desired, RTSP could be used for session establishment.   A client desiring to join a live multicasted media session with   cryptographic (encryption) access control could use RTSP in the   following way.  The source of the session announces the session and   gives all interested an RTSP URI.  The client connects to the server   and requests the presentation description, allowing configuration for   reception of the media.  In this step, it is possible for the client   to use secured transport and any desired level of authentication; for   example, for billing or access control.  An RTSP link also allows for   load balancing between multiple servers.   If these were the only goals, they could be achieved by simply using   HTTP.  However, for cases where the sender likes to keep track of   each individual receiver of a session, and possibly use the session   as a side channel for distributing key-updates or other information   on a per-receiver basis, and the full set of receivers is not known   prior to the session start, the state establishment that RTSP   provides can be beneficial.  In this case, a client would establish   an RTSP session for this multicast group with the RTSP server.  The   RTSP server will not transmit any media, but instead will point to   the multicast group.  The client and server will be able to keep the   session alive for as long as the receiver participates in the session   thus enabling, for example, the server to push updates to the client.   This use case will most likely not be able to be implemented without   some extensions to the server-to-client push mechanism.  Here the   PLAY_NOTIFY method (seeSection 13.5) with a suitable extension could   provide clear benefits.Schulzrinne, et al.          Standards Track                  [Page 304]

RFC 7826                        RTSP 2.0                   December 2016Appendix F.  Text Format for Parameters   A resource of type "text/parameters" consists of either 1) a list of   parameters (for a query) or 2) a list of parameters and associated   values (for a response or setting of the parameter).  Each entry of   the list is a single line of text.  Parameters are separated from   values by a colon.  The parameter name MUST only use US-ASCII visible   characters while the values are UTF-8 text strings.  The media type   registration form is inSection 22.16.   There is a potential interoperability issue for this format.  It was   named inRFC 2326 but never defined, even if used in examples that   hint at the syntax.  This format matches the purpose and its syntax   supports the examples provided.  However, it goes further by allowing   UTF-8 in the value part; thus, usage of UTF-8 strings may not be   supported.  However, as individual parameters are not defined, the   implementing application needs to have out-of-band agreement or using   feature tag anyway to determine if the endpoint supports the   parameters.   The ABNF [RFC5234] grammar for "text/parameters" content is:   file             = *((parameter / parameter-value) CRLF)   parameter        = 1*visible-except-colon   parameter-value  = parameter *WSP ":" value   visible-except-colon = %x21-39 / %x3B-7E    ; VCHAR - ":"   value            = *(TEXT-UTF8char / WSP)   TEXT-UTF8char    = <as defined inSection 20.1>   WSP              = <SeeRFC 5234> ; Space or HTAB   VCHAR            = <SeeRFC 5234>   CRLF             = <SeeRFC 5234>Appendix G.  Requirements for Unreliable Transport of RTSP   This appendix provides guidance for those who want to implement RTSP   messages over unreliable transports as has been defined in RTSP 1.0   [RFC2326].RFC 2326 defined the "rtspu" URI scheme and provided some   basic information for the transport of RTSP messages over UDP.  The   information is being provided here as there has been at least one   commercial implementation and compatibility with that should be   maintained.Schulzrinne, et al.          Standards Track                  [Page 305]

RFC 7826                        RTSP 2.0                   December 2016   The following points should be considered for an interoperable   implementation:   o  Requests shall be acknowledged by the receiver.  If there is no      acknowledgement, the sender may resend the same message after a      timeout of one round-trip time (RTT).  Any retransmissions due to      lack of acknowledgement must carry the same sequence number as the      original request.   o  The RTT can be estimated as in TCP (RFC 6298) [RFC6298], with an      initial round-trip value of 500 ms.  An implementation may cache      the last RTT measurement as the initial value for future      connections.   o  The Timestamp header (Section 18.53) is used to avoid the      retransmission ambiguity problem [Stevens98].   o  The registered default port for RTSP over UDP for the server is      554.   o  RTSP messages can be carried over any lower-layer transport      protocol that is 8-bit clean.   o  RTSP messages are vulnerable to bit errors and should not be      subjected to them.   o  Source authentication, or at least validation that RTSP messages      comes from the same entity becomes extremely important, as session      hijacking may be substantially easier for RTSP message transport      using an unreliable protocol like UDP than for TCP.   There are two RTSP headers that are primarily intended for being used   by the unreliable handling of RTSP messages and which will be   maintained:   o  CSeq: SeeSection 18.20.  It should be noted that the CSeq header      is also required to match requests and responses independent      whether a reliable or unreliable transport is used.   o  Timestamp: SeeSection 18.53Appendix H.  Backwards-Compatibility Considerations   This section contains notes on issues about backwards compatibility   with clients or servers being implemented according toRFC 2326   [RFC2326].  Note that there exists no requirement to implement RTSP   1.0; in fact, this document recommends against it as it is difficult   to do in an interoperable way.Schulzrinne, et al.          Standards Track                  [Page 306]

RFC 7826                        RTSP 2.0                   December 2016   A server implementing RTSP 2.0 MUST include an RTSP-Version of   "RTSP/2.0" in all responses to requests containing RTSP-Version value   of "RTSP/2.0".  If a server receives an RTSP 1.0 request, it MAY   respond with an RTSP 1.0 response if it chooses to supportRFC 2326.   If the server chooses not to supportRFC 2326, it MUST respond with a   505 (RTSP Version Not Supported) status code.  A server MUST NOT   respond to an RTSP 1.0 request with an RTSP 2.0 response.   Clients implementing RTSP 2.0 MAY use an OPTIONS request with an   RTSP-Version of "RTSP/2.0" to determine whether a server supports   RTSP 2.0.  If the server responds with either an RTSP-Version of   "RTSP/1.0" or a status code of 505 (RTSP Version Not Supported), the   client will have to use RTSP 1.0 requests if it chooses to supportRFC 2326.H.1.  Play Request in Play State   The behavior in the server when a Play is received in Play state has   changed (Section 13.4).  InRFC 2326, the new PLAY request would be   queued until the current Play completed.  Any new PLAY request now   takes effect immediately replacing the previous request.H.2.  Using Persistent Connections   Some server implementations ofRFC 2326 maintain a one-to-one   relationship between a connection and an RTSP session.  Such   implementations require clients to use a persistent connection to   communicate with the server and when a client closes its connection,   the server may remove the RTSP session.  This is worth noting if an   RTSP 2.0 client also supporting 1.0 connects to a 1.0 server.Appendix I.  Changes   This appendix briefly lists the differences between RTSP 1.0   [RFC2326] and RTSP 2.0 for an informational purpose.  For   implementers of RTSP 2.0, it is recommended to read carefully through   this memo and not to rely on the list of changes below to adapt from   RTSP 1.0 to RTSP 2.0, as RTSP 2.0 is not intended to be backwards   compatible with RTSP 1.0 [RFC2326] other than the version negotiation   mechanism.Schulzrinne, et al.          Standards Track                  [Page 307]

RFC 7826                        RTSP 2.0                   December 2016I.1.  Brief Overview   The following protocol elements were removed in RTSP 2.0 compared to   RTSP 1.0:   o  the RECORD and ANNOUNCE methods and all related functionality      (including 201 (Created) and 250 (Low On Storage Space) status      codes);   o  the use of UDP for RTSP message transport (due to missing interest      and to broken specification);   o  the use of PLAY method for keep-alive in Play state.   The following protocol elements were added or changed in RTSP 2.0   compared to RTSP 1.0:   o  RTSP session TEARDOWN from the server to the client;   o  IPv6 support;   o  extended IANA registries (e.g., transport headers parameters,      transport-protocol, profile, lower-transport, and mode);   o  request pipelining for quick session start-up;   o  fully reworked state machine;   o  RTSP messages now use URIs rather than URLs;   o  incorporated much of related HTTP text ([RFC2616]) in this memo,      compared to just referencing the sections in HTTP, to avoid      ambiguities;   o  the REDIRECT method was expanded and diversified for different      situations;   o  Includes a new section about how to set up different media-      transport alternatives and their profiles in addition to lower-      layer protocols.  This caused the appendix on RTP interaction to      be moved to the new section instead of being in the part that      describes RTP.  The section also includes guidelines what to      consider when writing usage guidelines for new protocols and      profiles;Schulzrinne, et al.          Standards Track                  [Page 308]

RFC 7826                        RTSP 2.0                   December 2016   o  Added an asynchronous notification method PLAY_NOTIFY.  This      method is used by the RTSP server to asynchronously notify clients      about session changes while in Play state.  To a limited extent,      this is comparable with some implementations of ANNOUNCE in RTSP      1.0 not intended for Recording.I.2.  Detailed List of Changes   The below changes have been made to RTSP 1.0 (RFC 2326) when defining   RTSP 2.0.  Note that this list does not reflect minor changes in   wording or correction of typographical errors.   o  The section on minimal implementation was deleted.  Instead, the      main part of the specification defines the core of RTSP 2.0.   o  The Transport header has been changed in the following ways:      *  The ABNF has been changed to define that extensions are         possible and that unknown parameters result in servers ignoring         the transport specification.      *  To prevent backwards compatibility issues, any extension or new         parameter requires the usage of a feature tag combined with the         Require header.      *  Syntax ambiguities with the Mode parameter have been resolved.      *  Syntax error with ";" for multicast and unicast has been         resolved.      *  Two new addressing parameters have been defined: src_addr and         dest_addr.  These replace the parameters "port", "client_port",         "server_port", "destination", and "source".      *  Support for IPv6 explicit addresses in all address fields has         been included.      *  To handle URI definitions that contain ";" or ",", a quoted-URI         format has been introduced and is required.      *  IANA registries for the transport header parameters, transport-         protocol, profile, lower-transport, and mode have been defined.      *  The Transport header's interleaved parameter's text was made         more strict and uses formal requirements levels.  It was also         clarified that the interleaved channels are symmetric and that         it is the server that sets the channel numbers.Schulzrinne, et al.          Standards Track                  [Page 309]

RFC 7826                        RTSP 2.0                   December 2016      *  It has been clarified that the client can't request of the         server to use a certain RTP SSRC, using a request with the         transport parameter SSRC.      *  Syntax definition for SSRC has been clarified to require 8HEX.         It has also been extended to allow multiple values for clients         supporting this version.      *  Clarified the text on the Transport header's "dest_addr"         parameters regarding what security precautions the server is         required to perform.   o  The Range formats have been changed in the following way:      *  The NPT format has been given an initial NPT identifier that         must now be used.      *  All formats now support initial open-ended formats of type         "npt=-10" and also format only "Range: smpte" ranges for usage         with GET_PARAMETER requests.      *  The npt-hhmmss notation now follows ISO 8601 more strictly.   o  RTSP message handling has been changed in the following ways:      *  RTSP messages now use URIs rather than URLs.      *  It has been clarified that a 4xx message due to a missing CSeq         header shall be returned without a CSeq header.      *  The 300 (Multiple Choices) response code has been removed.      *  Rules for how to handle the timing out RTSP messages have been         added.      *  Extended Pipelining rules allowing for quick session startup.      *  Sequence numbering and proxy handling of sequence numbers have         been defined, including cases when responses arrive out of         order.   o  The HTTP references have been updated to first RFCs 2616 and 2617      and then toRFC 7230-7235.  Most of the text has been copied and      then altered to fit RTSP into this specification.  The Public and      the Content-Base headers have also been imported fromRFC 2068 so      that they are defined in the RTSP specification.  Known effects on      RTSP due to HTTP clarifications:Schulzrinne, et al.          Standards Track                  [Page 310]

RFC 7826                        RTSP 2.0                   December 2016      *  Content-Encoding header can include encoding of type         "identity".   o  The state machine section has been completely rewritten.  It now      includes more details and is also more clear about the model used.   o  An IANA section has been included that contains a number of      registries and their rules.  This will allow us to use IANA to      keep track of RTSP extensions.   o  The transport of RTSP messages has seen the following changes:      *  The use of UDP for RTSP message transport has been deprecated         due to missing interest and to broken specification.      *  The rules for how TCP connections are to be handled have been         clarified.  Now it is made clear that servers should not close         the TCP connection unless they have been unused for significant         time.      *  Strong recommendations why servers and clients should use         persistent connections have also been added.      *  There is now a requirement on the servers to handle non-         persistent connections as this provides fault tolerance.      *  Added wording on the usage of Connection:Close for RTSP.      *  Specified usage of TLS for RTSP messages, including a scheme to         approve a proxy's TLS connection to the next hop.   o  The following header-related changes have been made:      *  Accept-Ranges response-header has been added.  This header         clarifies which range formats can be used for a resource.      *  Fixed the missing definitions for the Cache-Control header.         Also added to the syntax definition the missing delta-seconds         for max-stale and min-fresh parameters.      *  Put requirement on CSeq header that the value is increased by         one for each new RTSP request.  A recommendation to start at 0         has also been added.      *  Added a requirement that the Date header must be used for all         messages with a message body and the Server should always         include it.Schulzrinne, et al.          Standards Track                  [Page 311]

RFC 7826                        RTSP 2.0                   December 2016      *  Removed the possibility of using Range header with Scale header         to indicate when it is to be activated, since it can't work as         defined.  Also, added a rule that lack of Scale header in a         response indicates lack of support for the header.  feature         tags for scaled playback have been defined.      *  The Speed header must now be responded to in order to indicate         support and the actual speed going to be used.  A feature tag         is defined.  Notes on congestion control were also added.      *  The Supported header was borrowed from SIP [RFC3261] to help         with the feature negotiation in RTSP.      *  Clarified that the Timestamp header can be used to resolve         retransmission ambiguities.      *  The Session header text has been expanded with an explanation         on keep-alive and which methods to use.  SET_PARAMETER is now         recommended to use if only keep-alive within RTSP is desired.      *  It has been clarified how the Range header formats are used to         indicate pause points in the PAUSE response.      *  Clarified that RTP-Info URIs that are relative use the Request-         URI as base URI.  Also clarified that the used URI must be the         one that was used in the SETUP request.  The URIs are now also         required to be quoted.  The header also expresses the SSRC for         the provided RTP timestamp and sequence number values.      *  Added text that requires the Range to always be present in PLAY         responses.  Clarified what should be sent in case of live         streams.      *  The headers table has been updated using a structure borrowed         from SIP.  Those tables convey much more information and should         provide a good overview of the available headers.      *  It has been clarified that any message with a message body is         required to have a Content-Length header.  This was the case inRFC 2326, but could be misinterpreted.      *  ETag has changed its name to MTag.      *  To resolve functionality around MTag, the MTag and If-None-         Match header have been added from HTTP with necessary         clarification in regard to RTSP operation.Schulzrinne, et al.          Standards Track                  [Page 312]

RFC 7826                        RTSP 2.0                   December 2016      *  Imported the Public header from HTTP (RFC 2068 [RFC2068]) since         it has been removed from HTTP due to lack of use.  Public is         used quite frequently in RTSP.      *  Clarified rules for populating the Public header so that it is         an intersection of the capabilities of all the RTSP agents in a         chain.      *  Added the Media-Range header for listing the current         availability of the media range.      *  Added the Notify-Reason header for giving the reason when         sending PLAY_NOTIFY requests.      *  A new header Seek-Style has been defined to direct and inform         how any seek operation should/have been performed.   o  The Protocol Syntax has been changed in the following way:      *  All ABNF definitions are updated according to the rules defined         inRFC 5234 [RFC5234] and have been gathered in a separate         section (Section 20).      *  The ABNF for the User-Agent and Server headers have been         corrected.      *  Some definitions in the introduction regarding the RTSP session         have been changed.      *  The protocol has been made fully IPv6 capable.      *  The CHAR rule has been changed to exclude NULL.   o  The Status codes have been changed in the following ways:      *  The use of status code 303 (See Other) has been deprecated as         it does not make sense to use in RTSP.      *  The never-defined status code 411 "Length Required" has been         completely removed.      *  When sending response 451 (Parameter Not Understood) and 458         (Parameter Is Read-Only), the response body should contain the         offending parameters.Schulzrinne, et al.          Standards Track                  [Page 313]

RFC 7826                        RTSP 2.0                   December 2016      *  Clarification on when a 3rr redirect status code can be         received has been added.  This includes receiving 3rr as a         result of a request within an established session.  This         provides clarification to a previous unspecified behavior.      *  Removed the 201 (Created) and 250 (Low On Storage Space) status         codes as they are only relevant to recording, which is         deprecated.      *  Several new status codes have been defined: 464 (Data Transport         Not Ready Yet), 465 (Notification Reason Unknown), 470         (Connection Authorization Required), 471 (Connection         Credentials Not Accepted), and 472 (Failure to Establish Secure         Connection).   o  The following functionality has been deprecated from the protocol:      *  The use of Queued Play.      *  The use of PLAY method for keep-alive in Play state.      *  The RECORD and ANNOUNCE methods and all related functionality.         Some of the syntax has been removed.      *  The possibility to use timed execution of methods with the time         parameter in the Range header.      *  The description on how rtspu works is not part of the core         specification and will require external description.  Only that         it exists is mentioned here and some requirements for the         transport are provided.   o  The following changes have been made in relation to methods:      *  The OPTIONS method has been clarified with regard to the use of         the Public and Allow headers.      *  Added text clarifying the usage of SET_PARAMETER for keep-alive         and usage without a body.      *  PLAY method is now allowed to be pipelined with the pipelining         of one or more SETUP requests following the initial that         generates the session for aggregated control.      *  REDIRECT has been expanded and diversified for different         situations.Schulzrinne, et al.          Standards Track                  [Page 314]

RFC 7826                        RTSP 2.0                   December 2016      *  Added a new method PLAY_NOTIFY.  This method is used by the         RTSP server to asynchronously notify clients about session         changes.   o  Wrote a new section about how to set up different media-transport      alternatives and their profiles as well as lower-layer protocols.      This caused the appendix on RTP interaction to be moved to the new      section instead of being in the part that describes RTP.  The new      section also includes guidelines what to consider when writing      usage guidelines for new protocols and profiles.   o  Setup and usage of independent TCP connections for transport of      RTP has been specified.   o  Added a new section describing the available mechanisms to      determine if functionality is supported, called "Capability      Handling".  Renamed option-tags to feature tags.   o  Added a Contributors section with people who have contributed      actual text to the specification.   o  Added a section "Use Cases" that describes the major use cases for      RTSP.   o  Clarified the usage of a=range and how to indicate live content      that are not seekable with this header.   o  Text specifying the special behavior of PLAY for live content.   o  Security features of RTSP have been clarified:      *  HTTP-based authorization has been clarified requiring both         Basic and Digest support      *  TLS support has been mandated      *  If one implements RTP, then SRTP and defined MIKEY-based key-         exchange must be supported      *  Various minor mitigations discussed or resulted in protocol         changes.Schulzrinne, et al.          Standards Track                  [Page 315]

RFC 7826                        RTSP 2.0                   December 2016Acknowledgements   This memorandum defines RTSP version 2.0, which is a revision of the   Proposed Standard RTSP version 1.0 defined in [RFC2326].  The authors   ofRFC 2326 are Henning Schulzrinne, Anup Rao, and Robert Lanphier.   Both RTSP version 1.0 and RTSP version 2.0 borrow format and   descriptions from HTTP/1.1.   Robert Sparks and especially Elwyn Davies provided very valuable and   detailed reviews in the IETF Last Call that greatly improved the   document and resolved many issues, especially regarding consistency.   This document has benefited greatly from the comments of all those   participating in the MMUSIC WG.  In addition to those already   mentioned, the following individuals have contributed to this   specification:   Rahul Agarwal, Claudio Allocchio, Jeff Ayars, Milko Boic, Torsten   Braun, Brent Browning, Bruce Butterfield, Steve Casner, Maureen   Chesire, Jinhang Choi, Francisco Cortes, Elwyn Davies, Spencer   Dawkins, Kelly Djahandari, Martin Dunsmuir, Adrian Farrel, Stephen   Farrell, Ross Finlayson, Eric Fleischman, Jay Geagan, Andy Grignon,   Christian Groves, V.  Guruprasad, Peter Haight, Mark Handley, Brad   Hefta-Gaub, Volker Hilt, John K.  Ho, Patrick Hoffman, Go Hori,   Philipp Hoschka, Anne Jones, Ingemar Johansson, Jae-Hwan Kim, Anders   Klemets, Ruth Lang, Barry Leiba, Stephanie Leif, Jonathan Lennox,   Eduardo F.  Llach, Chris Lonvick, Xavier Marjou, Thomas Marshall, Rob   McCool, Martti Mela, David Oran, Joerg Ott, Joe Pallas, Maria   Papadopouli, Sujal Patel, Ema Patki, Alagu Periyannan, Colin Perkins,   Pekka Pessi, Igor Plotnikov, Pete Resnick, Peter Saint-Andre, Holger   Schmidt, Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff   Smith, Alexander Sokolsky, Dale Stammen, John Francis Stracke, Geetha   Srikantan, Scott Taylor, David Walker, Stephan Wenger, Dale R.   Worley, and Byungjo Yoon, and especially Flemming Andreasen.Schulzrinne, et al.          Standards Track                  [Page 316]

RFC 7826                        RTSP 2.0                   December 2016Contributors   The following people have made written contributions that were   included in the specification:   o  Tom Marshall contributed text on the usage of 3rr status codes.   o  Thomas Zheng contributed text on the usage of the Range in PLAY      responses and proposed an earlier version of the PLAY_NOTIFY      method.   o  Sean Sheedy contributed text on the timeout behavior of RTSP      messages and connections, the 463 (Destination Prohibited) status      code, and proposed an earlier version of the PLAY_NOTIFY method.   o  Greg Sherwood proposed an earlier version of the PLAY_NOTIFY      method.   o  Fredrik Lindholm contributed text about the RTSP security      framework.   o  John Lazzaro contributed the text for RTP over Independent TCP.   o  Aravind Narasimhan contributed by rewriting "Media-Transport      Alternatives" (Appendix C) and making editorial improvements on a      number of places in the specification.   o  Torbjorn Einarsson has done some editorial improvements of the      text.Schulzrinne, et al.          Standards Track                  [Page 317]

RFC 7826                        RTSP 2.0                   December 2016Authors' Addresses   Henning Schulzrinne   Columbia University   1214 Amsterdam Avenue   New York, NY  10027   United States of America   Email: schulzrinne@cs.columbia.edu   Anup Rao   Cisco   United States of America   Email: anrao@cisco.com   Rob Lanphier   San Francisco, CA   United States of America   Email: robla@robla.net   Magnus Westerlund   Ericsson   Faeroegatan 2   Stockholm  SE-164 80   Sweden   Email: magnus.westerlund@ericsson.com   Martin Stiemerling (editor)   University of Applied Sciences Darmstadt   Haardtring 100   64295 Darmstadt   Germany   Email: mls.ietf@gmail.com   URI:http://www.stiemerling.orgSchulzrinne, et al.          Standards Track                  [Page 318]