RTCWEB Working Group                                          C. PerkinsInternet-Draft                                     University of GlasgowIntended status: Standards Track                           M. WesterlundExpires: September 18, 2016                                     Ericsson                                                                  J. Ott                                                        Aalto University                                                          March 17, 2016  Web Real-Time Communication (WebRTC): Media Transport and Use of RTP                     draft-ietf-rtcweb-rtp-usage-26Abstract   The Web Real-Time Communication (WebRTC) framework provides support   for direct interactive rich communication using audio, video, text,   collaboration, games, etc. between two peers' web-browsers.  This   memo describes the media transport aspects of the WebRTC framework.   It specifies how the Real-time Transport Protocol (RTP) is used in   the WebRTC context, and gives requirements for which RTP features,   profiles, and extensions need to be supported.Status of This Memo   This Internet-Draft is submitted in full conformance with the   provisions of BCP 78 and BCP 79.   Internet-Drafts are working documents of the Internet Engineering   Task Force (IETF).  Note that other groups may also distribute   working documents as Internet-Drafts.  The list of current Internet-   Drafts is at http://datatracker.ietf.org/drafts/current/.   Internet-Drafts are draft documents valid for a maximum of six months   and may be updated, replaced, or obsoleted by other documents at any   time.  It is inappropriate to use Internet-Drafts as reference   material or to cite them other than as "work in progress."   This Internet-Draft will expire on September 18, 2016.Copyright Notice   Copyright (c) 2016 IETF Trust and the persons identified as the   document authors.  All rights reserved.   This document is subject to BCP 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 documentsPerkins, et al.        Expires September 18, 2016               [Page 1]Internet-Draft               RTP for WebRTC                   March 2016   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.Table of Contents   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3   2.  Rationale . . . . . . . . . . . . . . . . . . . . . . . . . .   4   3.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4   4.  WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . .   5     4.1.  RTP and RTCP  . . . . . . . . . . . . . . . . . . . . . .   5     4.2.  Choice of the RTP Profile . . . . . . . . . . . . . . . .   7     4.3.  Choice of RTP Payload Formats . . . . . . . . . . . . . .   8     4.4.  Use of RTP Sessions . . . . . . . . . . . . . . . . . . .  10     4.5.  RTP and RTCP Multiplexing . . . . . . . . . . . . . . . .  10     4.6.  Reduced Size RTCP . . . . . . . . . . . . . . . . . . . .  11     4.7.  Symmetric RTP/RTCP  . . . . . . . . . . . . . . . . . . .  11     4.8.  Choice of RTP Synchronisation Source (SSRC) . . . . . . .  12     4.9.  Generation of the RTCP Canonical Name (CNAME) . . . . . .  12     4.10. Handling of Leap Seconds  . . . . . . . . . . . . . . . .  13   5.  WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . .  13     5.1.  Conferencing Extensions and Topologies  . . . . . . . . .  14       5.1.1.  Full Intra Request (FIR)  . . . . . . . . . . . . . .  15       5.1.2.  Picture Loss Indication (PLI) . . . . . . . . . . . .  15       5.1.3.  Slice Loss Indication (SLI) . . . . . . . . . . . . .  16       5.1.4.  Reference Picture Selection Indication (RPSI) . . . .  16       5.1.5.  Temporal-Spatial Trade-off Request (TSTR) . . . . . .  16       5.1.6.  Temporary Maximum Media Stream Bit Rate Request               (TMMBR) . . . . . . . . . . . . . . . . . . . . . . .  16     5.2.  Header Extensions . . . . . . . . . . . . . . . . . . . .  17       5.2.1.  Rapid Synchronisation . . . . . . . . . . . . . . . .  17       5.2.2.  Client-to-Mixer Audio Level . . . . . . . . . . . . .  17       5.2.3.  Mixer-to-Client Audio Level . . . . . . . . . . . . .  18       5.2.4.  Media Stream Identification . . . . . . . . . . . . .  18       5.2.5.  Coordination of Video Orientation . . . . . . . . . .  18   6.  WebRTC Use of RTP: Improving Transport Robustness . . . . . .  19     6.1.  Negative Acknowledgements and RTP Retransmission  . . . .  19     6.2.  Forward Error Correction (FEC)  . . . . . . . . . . . . .  20   7.  WebRTC Use of RTP: Rate Control and Media Adaptation  . . . .  20     7.1.  Boundary Conditions and Circuit Breakers  . . . . . . . .  21     7.2.  Congestion Control Interoperability and Legacy Systems  .  22   8.  WebRTC Use of RTP: Performance Monitoring . . . . . . . . . .  22   9.  WebRTC Use of RTP: Future Extensions  . . . . . . . . . . . .  23   10. Signalling Considerations . . . . . . . . . . . . . . . . . .  23   11. WebRTC API Considerations . . . . . . . . . . . . . . . . . .  25   12. RTP Implementation Considerations . . . . . . . . . . . . . .  27Perkins, et al.        Expires September 18, 2016               [Page 2]Internet-Draft               RTP for WebRTC                   March 2016     12.1.  Configuration and Use of RTP Sessions  . . . . . . . . .  27       12.1.1.  Use of Multiple Media Sources Within an RTP Session   27       12.1.2.  Use of Multiple RTP Sessions . . . . . . . . . . . .  28       12.1.3.  Differentiated Treatment of RTP Streams  . . . . . .  33     12.2.  Media Source, RTP Streams, and Participant            Identification . . . . . . . . . . . . . . . . . . . . .  35       12.2.1.  Media Source Identification  . . . . . . . . . . . .  35       12.2.2.  SSRC Collision Detection . . . . . . . . . . . . . .  36       12.2.3.  Media Synchronisation Context  . . . . . . . . . . .  37   13. Security Considerations . . . . . . . . . . . . . . . . . . .  37   14. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  39   15. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  39   16. References  . . . . . . . . . . . . . . . . . . . . . . . . .  39     16.1.  Normative References . . . . . . . . . . . . . . . . . .  39     16.2.  Informative References . . . . . . . . . . . . . . . . .  44   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  461.  Introduction   The Real-time Transport Protocol (RTP) [RFC3550] provides a framework   for delivery of audio and video teleconferencing data and other real-   time media applications.  Previous work has defined the RTP protocol,   along with numerous profiles, payload formats, and other extensions.   When combined with appropriate signalling, these form the basis for   many teleconferencing systems.   The Web Real-Time communication (WebRTC) framework provides the   protocol building blocks to support direct, interactive, real-time   communication using audio, video, collaboration, games, etc., between   two peers' web-browsers.  This memo describes how the RTP framework   is to be used in the WebRTC context.  It proposes a baseline set of   RTP features that are to be implemented by all WebRTC Endpoints,   along with suggested extensions for enhanced functionality.   This memo specifies a protocol intended for use within the WebRTC   framework, but is not restricted to that context.  An overview of the   WebRTC framework is given in [I-D.ietf-rtcweb-overview].   The structure of this memo is as follows.  Section 2 outlines our   rationale in preparing this memo and choosing these RTP features.   Section 3 defines terminology.  Requirements for core RTP protocols   are described in Section 4 and suggested RTP extensions are described   in Section 5.  Section 6 outlines mechanisms that can increase   robustness to network problems, while Section 7 describes congestion   control and rate adaptation mechanisms.  The discussion of mandated   RTP mechanisms concludes in Section 8 with a review of performance   monitoring and network management tools.  Section 9 gives some   guidelines for future incorporation of other RTP and RTP ControlPerkins, et al.        Expires September 18, 2016               [Page 3]Internet-Draft               RTP for WebRTC                   March 2016   Protocol (RTCP) extensions into this framework.  Section 10 describes   requirements placed on the signalling channel.  Section 11 discusses   the relationship between features of the RTP framework and the WebRTC   application programming interface (API), and Section 12 discusses RTP   implementation considerations.  The memo concludes with security   considerations (Section 13) and IANA considerations (Section 14).2.  Rationale   The RTP framework comprises the RTP data transfer protocol, the RTP   control protocol, and numerous RTP payload formats, profiles, and   extensions.  This range of add-ons has allowed RTP to meet various   needs that were not envisaged by the original protocol designers, and   to support many new media encodings, but raises the question of what   extensions are to be supported by new implementations.  The   development of the WebRTC framework provides an opportunity to review   the available RTP features and extensions, and to define a common   baseline RTP feature set for all WebRTC Endpoints.  This builds on   the past 20 years of RTP development to mandate the use of extensions   that have shown widespread utility, while still remaining compatible   with the wide installed base of RTP implementations where possible.   RTP and RTCP extensions that are not discussed in this document can   be implemented by WebRTC Endpoints if they are beneficial for new use   cases.  However, they are not necessary to address the WebRTC use   cases and requirements identified in [RFC7478].   While the baseline set of RTP features and extensions defined in this   memo is targeted at the requirements of the WebRTC framework, it is   expected to be broadly useful for other conferencing-related uses of   RTP.  In particular, it is likely that this set of RTP features and   extensions will be appropriate for other desktop or mobile video   conferencing systems, or for room-based high-quality telepresence   applications.3.  Terminology   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this   document are to be interpreted as described in [RFC2119].  The RFC   2119 interpretation of these key words applies only when written in   ALL CAPS.  Lower- or mixed-case uses of these key words are not to be   interpreted as carrying special significance in this memo.   We define the following additional terms:Perkins, et al.        Expires September 18, 2016               [Page 4]Internet-Draft               RTP for WebRTC                   March 2016   WebRTC MediaStream:  The MediaStream concept defined by the W3C in      the WebRTC API [W3C.WD-mediacapture-streams-20130903].  A      MediaStream consists of zero or more MediaStreamTracks.   MediaStreamTrack:  Part of the MediaStream concept defined by the W3C      in the WebRTC API [W3C.WD-mediacapture-streams-20130903].  A      MediaStreamTrack is an individual stream of media from any type of      media source like a microphone or a camera, but also conceptual      sources, like a audio mix or a video composition, are possible.   Transport-layer Flow:  A uni-directional flow of transport packets      that are identified by having a particular 5-tuple of source IP      address, source port, destination IP address, destination port,      and transport protocol used.   Bi-directional Transport-layer Flow:  A bi-directional transport-      layer flow is a transport-layer flow that is symmetric.  That is,      the transport-layer flow in the reverse direction has a 5-tuple      where the source and destination address and ports are swapped      compared to the forward path transport-layer flow, and the      transport protocol is the same.   This document uses the terminology from   [I-D.ietf-avtext-rtp-grouping-taxonomy] and   [I-D.ietf-rtcweb-overview].  Other terms are used according to their   definitions from the RTP Specification [RFC3550].  Especially note   the following frequently used terms: RTP Stream, RTP Session, and   Endpoint.4.  WebRTC Use of RTP: Core Protocols   The following sections describe the core features of RTP and RTCP   that need to be implemented, along with the mandated RTP profiles.   Also described are the core extensions providing essential features   that all WebRTC Endpoints need to implement to function effectively   on today's networks.4.1.  RTP and RTCP   The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be   implemented as the media transport protocol for WebRTC.  RTP itself   comprises two parts: the RTP data transfer protocol, and the RTP   control protocol (RTCP).  RTCP is a fundamental and integral part of   RTP, and MUST be implemented and used in all WebRTC Endpoints.   The following RTP and RTCP features are sometimes omitted in limited   functionality implementations of RTP, but are REQUIRED in all WebRTC   Endpoints:Perkins, et al.        Expires September 18, 2016               [Page 5]Internet-Draft               RTP for WebRTC                   March 2016   o  Support for use of multiple simultaneous SSRC values in a single      RTP session, including support for RTP endpoints that send many      SSRC values simultaneously, following [RFC3550] and      [I-D.ietf-avtcore-rtp-multi-stream].  The RTCP optimisations for      multi-SSRC sessions defined in      [I-D.ietf-avtcore-rtp-multi-stream-optimisation] MAY be supported;      if supported the usage MUST be signalled.   o  Random choice of SSRC on joining a session; collision detection      and resolution for SSRC values (see also Section 4.8).   o  Support for reception of RTP data packets containing CSRC lists,      as generated by RTP mixers, and RTCP packets relating to CSRCs.   o  Sending correct synchronisation information in the RTCP Sender      Reports, to allow receivers to implement lip-synchronisation; see      Section 5.2.1 regarding support for the rapid RTP synchronisation      extensions.   o  Support for multiple synchronisation contexts.  Participants that      send multiple simultaneous RTP packet streams SHOULD do so as part      of a single synchronisation context, using a single RTCP CNAME for      all streams and allowing receivers to play the streams out in a      synchronised manner.  For compatibility with potential future      versions of this specification, or for interoperability with non-      WebRTC devices through a gateway, receivers MUST support multiple      synchronisation contexts, indicated by the use of multiple RTCP      CNAMEs in an RTP session.  This specification mandates the usage      of a single CNAME when sending RTP Streams in some circumstances,      see Section 4.9.   o  Support for sending and receiving RTCP SR, RR, SDES, and BYE      packet types.  Note that support for other RTCP packet types is      OPTIONAL, unless mandated by other parts of this specification.      Note that additional RTCP Packet types are used by the RTP/SAVPF      Profile (Section 4.2) and the other RTCP extensions (Section 5).      WebRTC endpoints that implement the SDP bundle negotiation      extension will use the SDP grouping framework 'mid' attribute to      identify media streams.  Such endpoints MUST implement the RTCP      SDES MID item described in      [I-D.ietf-mmusic-sdp-bundle-negotiation].   o  Support for multiple endpoints in a single RTP session, and for      scaling the RTCP transmission interval according to the number of      participants in the session; support for randomised RTCP      transmission intervals to avoid synchronisation of RTCP reports;      support for RTCP timer reconsideration (Section 6.3.6 ofPerkins, et al.        Expires September 18, 2016               [Page 6]Internet-Draft               RTP for WebRTC                   March 2016      [RFC3550]) and reverse reconsideration (Section 6.3.4 of      [RFC3550]).   o  Support for configuring the RTCP bandwidth as a fraction of the      media bandwidth, and for configuring the fraction of the RTCP      bandwidth allocated to senders, e.g., using the SDP "b=" line      [RFC4566][RFC3556].   o  Support for the reduced minimum RTCP reporting interval described      in Section 6.2 of [RFC3550].  When using the reduced minimum RTCP      reporting interval, the fixed (non-reduced) minimum interval MUST      be used when calculating the participant timeout interval (see      Sections 6.2 and 6.3.5 of [RFC3550]).  The delay before sending      the initial compound RTCP packet can be set to zero (see      Section 6.2 of [RFC3550] as updated by      [I-D.ietf-avtcore-rtp-multi-stream]).   o  Support for discontinuous transmission.  RTP allows endpoints to      pause and resume transmission at any time.  When resuming, the RTP      sequence number will increase by one, as usual, while the increase      in the RTP timestamp value will depend on the duration of the      pause.  Discontinuous transmission is most commonly used with some      audio payload formats, but is not audio specific, and can be used      with any RTP payload format.   o  Ignore unknown RTCP packet types and RTP header extensions.  This      is to ensure robust handling of future extensions, middlebox      behaviours, etc., that can result in not signalled RTCP packet      types or RTP header extensions being received.  If a compound RTCP      packet is received that contains a mixture of known and unknown      RTCP packet types, the known packets types need to be processed as      usual, with only the unknown packet types being discarded.   It is known that a significant number of legacy RTP implementations,   especially those targeted at VoIP-only systems, do not support all of   the above features, and in some cases do not support RTCP at all.   Implementers are advised to consider the requirements for graceful   degradation when interoperating with legacy implementations.   Other implementation considerations are discussed in Section 12.4.2.  Choice of the RTP Profile   The complete specification of RTP for a particular application domain   requires the choice of an RTP Profile.  For WebRTC use, the Extended   Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF) [RFC5124], as   extended by [RFC7007], MUST be implemented.  The RTP/SAVPF profile is   the combination of basic RTP/AVP profile [RFC3551], the RTP profilePerkins, et al.        Expires September 18, 2016               [Page 7]Internet-Draft               RTP for WebRTC                   March 2016   for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP   profile (RTP/SAVP) [RFC3711].   The RTCP-based feedback extensions [RFC4585] are needed for the   improved RTCP timer model.  This allows more flexible transmission of   RTCP packets in response to events, rather than strictly according to   bandwidth, and is vital for being able to report congestion signals   as well as media events.  These extensions also allow saving RTCP   bandwidth, and an endpoint will commonly only use the full RTCP   bandwidth allocation if there are many events that require feedback.   The timer rules are also needed to make use of the RTP conferencing   extensions discussed in Section 5.1.      Note: The enhanced RTCP timer model defined in the RTP/AVPF      profile is backwards compatible with legacy systems that implement      only the RTP/AVP or RTP/SAVP profile, given some constraints on      parameter configuration such as the RTCP bandwidth value and "trr-      int" (the most important factor for interworking with RTP/(S)AVP      endpoints via a gateway is to set the trr-int parameter to a value      representing 4 seconds, see Section 6.1 in      [I-D.ietf-avtcore-rtp-multi-stream]).   The secure RTP (SRTP) profile extensions [RFC3711] are needed to   provide media encryption, integrity protection, replay protection and   a limited form of source authentication.  WebRTC Endpoints MUST NOT   send packets using the basic RTP/AVP profile or the RTP/AVPF profile;   they MUST employ the full RTP/SAVPF profile to protect all RTP and   RTCP packets that are generated (i.e., implementations MUST use SRTP   and SRTCP).  The RTP/SAVPF profile MUST be configured using the   cipher suites, DTLS-SRTP protection profiles, keying mechanisms, and   other parameters described in [I-D.ietf-rtcweb-security-arch].4.3.  Choice of RTP Payload Formats   Mandatory to implement audio codecs and RTP payload formats for   WebRTC endpoints are defined in [I-D.ietf-rtcweb-audio].  Mandatory   to implement video codecs and RTP payload formats for WebRTC   endpoints are defined in [I-D.ietf-rtcweb-video].  WebRTC endpoints   MAY additionally implement any other codec for which an RTP payload   format and associated signalling has been defined.   WebRTC Endpoints cannot assume that the other participants in an RTP   session understand any RTP payload format, no matter how common.  The   mapping between RTP payload type numbers and specific configurations   of particular RTP payload formats MUST be agreed before those payload   types/formats can be used.  In an SDP context, this can be done using   the "a=rtpmap:" and "a=fmtp:" attributes associated with an "m="Perkins, et al.        Expires September 18, 2016               [Page 8]Internet-Draft               RTP for WebRTC                   March 2016   line, along with any other SDP attributes needed to configure the RTP   payload format.   Endpoints can signal support for multiple RTP payload formats, or   multiple configurations of a single RTP payload format, as long as   each unique RTP payload format configuration uses a different RTP   payload type number.  As outlined in Section 4.8, the RTP payload   type number is sometimes used to associate an RTP packet stream with   a signalling context.  This association is possible provided unique   RTP payload type numbers are used in each context.  For example, an   RTP packet stream can be associated with an SDP "m=" line by   comparing the RTP payload type numbers used by the RTP packet stream   with payload types signalled in the "a=rtpmap:" lines in the media   sections of the SDP.  This leads to the following considerations:      If RTP packet streams are being associated with signalling      contexts based on the RTP payload type, then the assignment of RTP      payload type numbers MUST be unique across signalling contexts.      If the same RTP payload format configuration is used in multiple      contexts, then a different RTP payload type number has to be      assigned in each context to ensure uniqueness.      If the RTP payload type number is not being used to associate RTP      packet streams with a signalling context, then the same RTP      payload type number can be used to indicate the exact same RTP      payload format configuration in multiple contexts.   A single RTP payload type number MUST NOT be assigned to different   RTP payload formats, or different configurations of the same RTP   payload format, within a single RTP session (note that the "m=" lines   in an SDP bundle group [I-D.ietf-mmusic-sdp-bundle-negotiation] form   a single RTP session).   An endpoint that has signalled support for multiple RTP payload   formats MUST be able to accept data in any of those payload formats   at any time, unless it has previously signalled limitations on its   decoding capability.  This requirement is constrained if several   types of media (e.g., audio and video) are sent in the same RTP   session.  In such a case, a source (SSRC) is restricted to switching   only between the RTP payload formats signalled for the type of media   that is being sent by that source; see Section 4.4.  To support rapid   rate adaptation by changing codec, RTP does not require advance   signalling for changes between RTP payload formats used by a single   SSRC that were signalled during session set-up.   If performing changes between two RTP payload types that use   different RTP clock rates, an RTP sender MUST follow thePerkins, et al.        Expires September 18, 2016               [Page 9]Internet-Draft               RTP for WebRTC                   March 2016   recommendations in Section 4.1 of [RFC7160].  RTP receivers MUST   follow the recommendations in Section 4.3 of [RFC7160] in order to   support sources that switch between clock rates in an RTP session   (these recommendations for receivers are backwards compatible with   the case where senders use only a single clock rate).4.4.  Use of RTP Sessions   An association amongst a set of endpoints communicating using RTP is   known as an RTP session [RFC3550].  An endpoint can be involved in   several RTP sessions at the same time.  In a multimedia session, each   type of media has typically been carried in a separate RTP session   (e.g., using one RTP session for the audio, and a separate RTP   session using a different transport-layer flow for the video).   WebRTC Endpoints are REQUIRED to implement support for multimedia   sessions in this way, separating each RTP session using different   transport-layer flows for compatibility with legacy systems (this is   sometimes called session multiplexing).   In modern day networks, however, with the widespread use of network   address/port translators (NAT/NAPT) and firewalls, it is desirable to   reduce the number of transport-layer flows used by RTP applications.   This can be done by sending all the RTP packet streams in a single   RTP session, which will comprise a single transport-layer flow (this   will prevent the use of some quality-of-service mechanisms, as   discussed in Section 12.1.3).  Implementations are therefore also   REQUIRED to support transport of all RTP packet streams, independent   of media type, in a single RTP session using a single transport layer   flow, according to [I-D.ietf-avtcore-multi-media-rtp-session] (this   is sometimes called SSRC multiplexing).  If multiple types of media   are to be used in a single RTP session, all participants in that RTP   session MUST agree to this usage.  In an SDP context,   [I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to signal such a   bundle of RTP packet streams forming a single RTP session.   Further discussion about the suitability of different RTP session   structures and multiplexing methods to different scenarios can be   found in [I-D.ietf-avtcore-multiplex-guidelines].4.5.  RTP and RTCP Multiplexing   Historically, RTP and RTCP have been run on separate transport layer   flows (e.g., two UDP ports for each RTP session, one port for RTP and   one port for RTCP).  With the increased use of Network Address/Port   Translation (NAT/NAPT) this has become problematic, since maintaining   multiple NAT bindings can be costly.  It also complicates firewall   administration, since multiple ports need to be opened to allow RTP   traffic.  To reduce these costs and session set-up times,Perkins, et al.        Expires September 18, 2016              [Page 10]Internet-Draft               RTP for WebRTC                   March 2016   implementations are REQUIRED to support multiplexing RTP data packets   and RTCP control packets on a single transport-layer flow [RFC5761].   Such RTP and RTCP multiplexing MUST be negotiated in the signalling   channel before it is used.  If SDP is used for signalling, this   negotiation MUST use the mechanism defined in [RFC5761].   Implementations can also support sending RTP and RTCP on separate   transport-layer flows, but this is OPTIONAL to implement.  If an   implementation does not support RTP and RTCP sent on separate   transport-layer flows, it MUST indicate that using the mechanism   defined in [I-D.ietf-mmusic-mux-exclusive].   Note that the use of RTP and RTCP multiplexed onto a single   transport-layer flow ensures that there is occasional traffic sent on   that port, even if there is no active media traffic.  This can be   useful to keep NAT bindings alive [RFC6263].4.6.  Reduced Size RTCP   RTCP packets are usually sent as compound RTCP packets, and [RFC3550]   requires that those compound packets start with an Sender Report (SR)   or Receiver Report (RR) packet.  When using frequent RTCP feedback   messages under the RTP/AVPF Profile [RFC4585] these statistics are   not needed in every packet, and unnecessarily increase the mean RTCP   packet size.  This can limit the frequency at which RTCP packets can   be sent within the RTCP bandwidth share.   To avoid this problem, [RFC5506] specifies how to reduce the mean   RTCP message size and allow for more frequent feedback.  Frequent   feedback, in turn, is essential to make real-time applications   quickly aware of changing network conditions, and to allow them to   adapt their transmission and encoding behaviour.  Implementations   MUST support sending and receiving non-compound RTCP feedback packets   [RFC5506].  Use of non-compound RTCP packets MUST be negotiated using   the signalling channel.  If SDP is used for signalling, this   negotiation MUST use the attributes defined in [RFC5506].  For   backwards compatibility, implementations are also REQUIRED to support   the use of compound RTCP feedback packets if the remote endpoint does   not agree to the use of non-compound RTCP in the signalling exchange.4.7.  Symmetric RTP/RTCP   To ease traversal of NAT and firewall devices, implementations are   REQUIRED to implement and use Symmetric RTP [RFC4961].  The reason   for using symmetric RTP is primarily to avoid issues with NATs and   Firewalls by ensuring that the send and receive RTP packet streams,   as well as RTCP, are actually bi-directional transport-layer flows.   This will keep alive the NAT and firewall pinholes, and help indicate   consent that the receive direction is a transport-layer flow thePerkins, et al.        Expires September 18, 2016              [Page 11]Internet-Draft               RTP for WebRTC                   March 2016   intended recipient actually wants.  In addition, it saves resources,   specifically ports at the endpoints, but also in the network as NAT   mappings or firewall state is not unnecessary bloated.  The amount of   per flow QoS state kept in the network is also reduced.4.8.  Choice of RTP Synchronisation Source (SSRC)   Implementations are REQUIRED to support signalled RTP synchronisation   source (SSRC) identifiers.  If SDP is used, this MUST be done using   the "a=ssrc:" SDP attribute defined in Section 4.1 and Section 5 of   [RFC5576] and the "previous-ssrc" source attribute defined in   Section 6.2 of [RFC5576]; other per-SSRC attributes defined in   [RFC5576] MAY be supported.   While support for signalled SSRC identifiers is mandated, their use   in an RTP session is OPTIONAL.  Implementations MUST be prepared to   accept RTP and RTCP packets using SSRCs that have not been explicitly   signalled ahead of time.  Implementations MUST support random SSRC   assignment, and MUST support SSRC collision detection and resolution,   according to [RFC3550].  When using signalled SSRC values, collision   detection MUST be performed as described in Section 5 of [RFC5576].   It is often desirable to associate an RTP packet stream with a non-   RTP context.  For users of the WebRTC API a mapping between SSRCs and   MediaStreamTracks are provided per Section 11.  For gateways or other   usages it is possible to associate an RTP packet stream with an "m="   line in a session description formatted using SDP.  If SSRCs are   signalled this is straightforward (in SDP the "a=ssrc:" line will be   at the media level, allowing a direct association with an "m=" line).   If SSRCs are not signalled, the RTP payload type numbers used in an   RTP packet stream are often sufficient to associate that packet   stream with a signalling context (e.g., if RTP payload type numbers   are assigned as described in Section 4.3 of this memo, the RTP   payload types used by an RTP packet stream can be compared with   values in SDP "a=rtpmap:" lines, which are at the media level in SDP,   and so map to an "m=" line).4.9.  Generation of the RTCP Canonical Name (CNAME)   The RTCP Canonical Name (CNAME) provides a persistent transport-level   identifier for an RTP endpoint.  While the Synchronisation Source   (SSRC) identifier for an RTP endpoint can change if a collision is   detected, or when the RTP application is restarted, its RTCP CNAME is   meant to stay unchanged for the duration of a RTCPeerConnection   [W3C.WD-webrtc-20130910], so that RTP endpoints can be uniquely   identified and associated with their RTP packet streams within a set   of related RTP sessions.Perkins, et al.        Expires September 18, 2016              [Page 12]Internet-Draft               RTP for WebRTC                   March 2016   Each RTP endpoint MUST have at least one RTCP CNAME, and that RTCP   CNAME MUST be unique within the RTCPeerConnection.  RTCP CNAMEs   identify a particular synchronisation context, i.e., all SSRCs   associated with a single RTCP CNAME share a common reference clock.   If an endpoint has SSRCs that are associated with several   unsynchronised reference clocks, and hence different synchronisation   contexts, it will need to use multiple RTCP CNAMEs, one for each   synchronisation context.   Taking the discussion in Section 11 into account, a WebRTC Endpoint   MUST NOT use more than one RTCP CNAME in the RTP sessions belonging   to single RTCPeerConnection (that is, an RTCPeerConnection forms a   synchronisation context).  RTP middleboxes MAY generate RTP packet   streams associated with more than one RTCP CNAME, to allow them to   avoid having to resynchronize media from multiple different endpoints   part of a multi-party RTP session.   The RTP specification [RFC3550] includes guidelines for choosing a   unique RTP CNAME, but these are not sufficient in the presence of NAT   devices.  In addition, long-term persistent identifiers can be   problematic from a privacy viewpoint (Section 13).  Accordingly, a   WebRTC Endpoint MUST generate a new, unique, short-term persistent   RTCP CNAME for each RTCPeerConnection, following [RFC7022], with a   single exception; if explicitly requested at creation an   RTCPeerConnection MAY use the same CNAME as as an existing   RTCPeerConnection within their common same-origin context.   An WebRTC Endpoint MUST support reception of any CNAME that matches   the syntax limitations specified by the RTP specification [RFC3550]   and cannot assume that any CNAME will be chosen according to the form   suggested above.4.10.  Handling of Leap Seconds   The guidelines regarding handling of leap seconds to limit their   impact on RTP media play-out and synchronization given in [RFC7164]   SHOULD be followed.5.  WebRTC Use of RTP: Extensions   There are a number of RTP extensions that are either needed to obtain   full functionality, or extremely useful to improve on the baseline   performance, in the WebRTC context.  One set of these extensions is   related to conferencing, while others are more generic in nature.   The following subsections describe the various RTP extensions   mandated or suggested for use within WebRTC.Perkins, et al.        Expires September 18, 2016              [Page 13]Internet-Draft               RTP for WebRTC                   March 20165.1.  Conferencing Extensions and Topologies   RTP is a protocol that inherently supports group communication.   Groups can be implemented by having each endpoint send its RTP packet   streams to an RTP middlebox that redistributes the traffic, by using   a mesh of unicast RTP packet streams between endpoints, or by using   an IP multicast group to distribute the RTP packet streams.  These   topologies can be implemented in a number of ways as discussed in   [I-D.ietf-avtcore-rtp-topologies-update].   While the use of IP multicast groups is popular in IPTV systems, the   topologies based on RTP middleboxes are dominant in interactive video   conferencing environments.  Topologies based on a mesh of unicast   transport-layer flows to create a common RTP session have not seen   widespread deployment to date.  Accordingly, WebRTC Endpoints are not   expected to support topologies based on IP multicast groups or to   support mesh-based topologies, such as a point-to-multipoint mesh   configured as a single RTP session (Topo-Mesh in the terminology of   [I-D.ietf-avtcore-rtp-topologies-update]).  However, a point-to-   multipoint mesh constructed using several RTP sessions, implemented   in WebRTC using independent RTCPeerConnections   [W3C.WD-webrtc-20130910], can be expected to be used in WebRTC, and   needs to be supported.   WebRTC Endpoints implemented according to this memo are expected to   support all the topologies described in   [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send   and receive unicast RTP packet streams to and from some peer device,   provided that peer can participate in performing congestion control   on the RTP packet streams.  The peer device could be another RTP   endpoint, or it could be an RTP middlebox that redistributes the RTP   packet streams to other RTP endpoints.  This limitation means that   some of the RTP middlebox-based topologies are not suitable for use   in WebRTC.  Specifically:   o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,      since they make the use of RTCP for congestion control and quality      of service reports problematic (see Section 3.8 of      [I-D.ietf-avtcore-rtp-topologies-update]).   o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology      SHOULD NOT be used because its safe use requires a congestion      control algorithm or RTP circuit breaker that handles point to      multipoint, which has not yet been standardised.   The following topology can be used, however it has some issues worth   noting:Perkins, et al.        Expires September 18, 2016              [Page 14]Internet-Draft               RTP for WebRTC                   March 2016   o  Content modifying MCUs with RTCP termination (Topo-RTCP-      terminating-MCU) MAY be used.  Note that in this RTP Topology, RTP      loop detection and identification of active senders is the      responsibility of the WebRTC application; since the clients are      isolated from each other at the RTP layer, RTP cannot assist with      these functions (see section 3.9 of      [I-D.ietf-avtcore-rtp-topologies-update]).   The RTP extensions described in Section 5.1.1 to Section 5.1.6 are   designed to be used with centralised conferencing, where an RTP   middlebox (e.g., a conference bridge) receives a participant's RTP   packet streams and distributes them to the other participants.  These   extensions are not necessary for interoperability; an RTP endpoint   that does not implement these extensions will work correctly, but   might offer poor performance.  Support for the listed extensions will   greatly improve the quality of experience and, to provide a   reasonable baseline quality, some of these extensions are mandatory   to be supported by WebRTC Endpoints.   The RTCP conferencing extensions are defined in Extended RTP Profile   for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/   AVPF) [RFC4585] and the memo on Codec Control Messages (CCM) in RTP/   AVPF [RFC5104]; they are fully usable by the Secure variant of this   profile (RTP/SAVPF) [RFC5124].5.1.1.  Full Intra Request (FIR)   The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1   of the Codec Control Messages [RFC5104].  It is used to make the   mixer request a new Intra picture from a participant in the session.   This is used when switching between sources to ensure that the   receivers can decode the video or other predictive media encoding   with long prediction chains.  WebRTC Endpoints that are sending media   MUST understand and react to FIR feedback messages they receive,   since this greatly improves the user experience when using   centralised mixer-based conferencing.  Support for sending FIR   messages is OPTIONAL.5.1.2.  Picture Loss Indication (PLI)   The Picture Loss Indication message is defined in Section 6.3.1 of   the RTP/AVPF profile [RFC4585].  It is used by a receiver to tell the   sending encoder that it lost the decoder context and would like to   have it repaired somehow.  This is semantically different from the   Full Intra Request above as there could be multiple ways to fulfil   the request.  WebRTC Endpoints that are sending media MUST understand   and react to PLI feedback messages as a loss tolerance mechanism.   Receivers MAY send PLI messages.Perkins, et al.        Expires September 18, 2016              [Page 15]Internet-Draft               RTP for WebRTC                   March 20165.1.3.  Slice Loss Indication (SLI)   The Slice Loss Indication message is defined in Section 6.3.2 of the   RTP/AVPF profile [RFC4585].  It is used by a receiver to tell the   encoder that it has detected the loss or corruption of one or more   consecutive macro blocks, and would like to have these repaired   somehow.  It is RECOMMENDED that receivers generate SLI feedback   messages if slices are lost when using a codec that supports the   concept of macro blocks.  A sender that receives an SLI feedback   message SHOULD attempt to repair the lost slice(s).5.1.4.  Reference Picture Selection Indication (RPSI)   Reference Picture Selection Indication (RPSI) messages are defined in   Section 6.3.3 of the RTP/AVPF profile [RFC4585].  Some video encoding   standards allow the use of older reference pictures than the most   recent one for predictive coding.  If such a codec is in use, and if   the encoder has learnt that encoder-decoder synchronisation has been   lost, then a known as correct reference picture can be used as a base   for future coding.  The RPSI message allows this to be signalled.   Receivers that detect that encoder-decoder synchronisation has been   lost SHOULD generate an RPSI feedback message if codec being used   supports reference picture selection.  A RTP packet stream sender   that receives such an RPSI message SHOULD act on that messages to   change the reference picture, if it is possible to do so within the   available bandwidth constraints, and with the codec being used.5.1.5.  Temporal-Spatial Trade-off Request (TSTR)   The temporal-spatial trade-off request and notification are defined   in Sections 3.5.2 and 4.3.2 of [RFC5104].  This request can be used   to ask the video encoder to change the trade-off it makes between   temporal and spatial resolution, for example to prefer high spatial   image quality but low frame rate.  Support for TSTR requests and   notifications is OPTIONAL.5.1.6.  Temporary Maximum Media Stream Bit Rate Request (TMMBR)   The TMMBR feedback message is defined in Sections 3.5.4 and 4.2.1 of   the Codec Control Messages [RFC5104].  This request and its   notification message are used by a media receiver to inform the   sending party that there is a current limitation on the amount of   bandwidth available to this receiver.  There can be various reasons   for this: for example, an RTP mixer can use this message to limit the   media rate of the sender being forwarded by the mixer (without doing   media transcoding) to fit the bottlenecks existing towards the other   session participants.  WebRTC Endpoints that are sending media are   REQUIRED to implement support for TMMBR messages, and MUST followPerkins, et al.        Expires September 18, 2016              [Page 16]Internet-Draft               RTP for WebRTC                   March 2016   bandwidth limitations set by a TMMBR message received for their SSRC.   The sending of TMMBR requests is OPTIONAL.5.2.  Header Extensions   The RTP specification [RFC3550] provides the capability to include   RTP header extensions containing in-band data, but the format and   semantics of the extensions are poorly specified.  The use of header   extensions is OPTIONAL in WebRTC, but if they are used, they MUST be   formatted and signalled following the general mechanism for RTP   header extensions defined in [RFC5285], since this gives well-defined   semantics to RTP header extensions.   As noted in [RFC5285], the requirement from the RTP specification   that header extensions are "designed so that the header extension may   be ignored" [RFC3550] stands.  To be specific, header extensions MUST   only be used for data that can safely be ignored by the recipient   without affecting interoperability, and MUST NOT be used when the   presence of the extension has changed the form or nature of the rest   of the packet in a way that is not compatible with the way the stream   is signalled (e.g., as defined by the payload type).  Valid examples   of RTP header extensions might include metadata that is additional to   the usual RTP information, but that can safely be ignored without   compromising interoperability.5.2.1.  Rapid Synchronisation   Many RTP sessions require synchronisation between audio, video, and   other content.  This synchronisation is performed by receivers, using   information contained in RTCP SR packets, as described in the RTP   specification [RFC3550].  This basic mechanism can be slow, however,   so it is RECOMMENDED that the rapid RTP synchronisation extensions   described in [RFC6051] be implemented in addition to RTCP SR-based   synchronisation.   This header extension uses the [RFC5285] generic header extension   framework, and so needs to be negotiated before it can be used.5.2.2.  Client-to-Mixer Audio Level   The Client to Mixer Audio Level extension [RFC6464] is an RTP header   extension used by an endpoint to inform a mixer about the level of   audio activity in the packet to which the header is attached.  This   enables an RTP middlebox to make mixing or selection decisions   without decoding or detailed inspection of the payload, reducing the   complexity in some types of mixers.  It can also save decoding   resources in receivers, which can choose to decode only the most   relevant RTP packet streams based on audio activity levels.Perkins, et al.        Expires September 18, 2016              [Page 17]Internet-Draft               RTP for WebRTC                   March 2016   The Client-to-Mixer Audio Level [RFC6464] header extension MUST be   implemented.  It is REQUIRED that implementations are capable of   encrypting the header extension according to [RFC6904] since the   information contained in these header extensions can be considered   sensitive.  The use of this encryption is RECOMMENDED, however usage   of the encryption can be explicitly disabled through API or   signalling.   This header extension uses the [RFC5285] generic header extension   framework, and so needs to be negotiated before it can be used.5.2.3.  Mixer-to-Client Audio Level   The Mixer to Client Audio Level header extension [RFC6465] provides   an endpoint with the audio level of the different sources mixed into   a common source stream by a RTP mixer.  This enables a user interface   to indicate the relative activity level of each session participant,   rather than just being included or not based on the CSRC field.  This   is a pure optimisation of non critical functions, and is hence   OPTIONAL to implement.  If this header extension is implemented, it   is REQUIRED that implementations are capable of encrypting the header   extension according to [RFC6904] since the information contained in   these header extensions can be considered sensitive.  It is further   RECOMMENDED that this encryption is used, unless the encryption has   been explicitly disabled through API or signalling.   This header extension uses the [RFC5285] generic header extension   framework, and so needs to be negotiated before it can be used.5.2.4.  Media Stream Identification   WebRTC endpoints that implement the SDP bundle negotiation extension   will use the SDP grouping framework 'mid' attribute to identify media   streams.  Such endpoints MUST implement the RTP MID header extension   described in [I-D.ietf-mmusic-sdp-bundle-negotiation].   This header extension uses the [RFC5285] generic header extension   framework, and so needs to be negotiated before it can be used.5.2.5.  Coordination of Video Orientation   WebRTC endpoints that send or receive video MUST implement the   coordination of video orientation (CVO) RTP header extension as   described in Section 4 of [I-D.ietf-rtcweb-video].   This header extension uses the [RFC5285] generic header extension   framework, and so needs to be negotiated before it can be used.Perkins, et al.        Expires September 18, 2016              [Page 18]Internet-Draft               RTP for WebRTC                   March 20166.  WebRTC Use of RTP: Improving Transport Robustness   There are tools that can make RTP packet streams robust against   packet loss and reduce the impact of loss on media quality.  However,   they generally add some overhead compared to a non-robust stream.   The overhead needs to be considered, and the aggregate bit-rate MUST   be rate controlled to avoid causing network congestion (see   Section 7).  As a result, improving robustness might require a lower   base encoding quality, but has the potential to deliver that quality   with fewer errors.  The mechanisms described in the following sub-   sections can be used to improve tolerance to packet loss.6.1.  Negative Acknowledgements and RTP Retransmission   As a consequence of supporting the RTP/SAVPF profile, implementations   can send negative acknowledgements (NACKs) for RTP data packets   [RFC4585].  This feedback can be used to inform a sender of the loss   of particular RTP packets, subject to the capacity limitations of the   RTCP feedback channel.  A sender can use this information to optimise   the user experience by adapting the media encoding to compensate for   known lost packets.   RTP packet stream senders are REQUIRED to understand the Generic NACK   message defined in Section 6.2.1 of [RFC4585], but MAY choose to   ignore some or all of this feedback (following Section 4.2 of   [RFC4585]).  Receivers MAY send NACKs for missing RTP packets.   Guidelines on when to send NACKs are provided in [RFC4585].  It is   not expected that a receiver will send a NACK for every lost RTP   packet, rather it needs to consider the cost of sending NACK   feedback, and the importance of the lost packet, to make an informed   decision on whether it is worth telling the sender about a packet   loss event.   The RTP Retransmission Payload Format [RFC4588] offers the ability to   retransmit lost packets based on NACK feedback.  Retransmission needs   to be used with care in interactive real-time applications to ensure   that the retransmitted packet arrives in time to be useful, but can   be effective in environments with relatively low network RTT (an RTP   sender can estimate the RTT to the receivers using the information in   RTCP SR and RR packets, as described at the end of Section 6.4.1 of   [RFC3550]).  The use of retransmissions can also increase the forward   RTP bandwidth, and can potentially caused increased packet loss if   the original packet loss was caused by network congestion.  Note,   however, that retransmission of an important lost packet to repair   decoder state can have lower cost than sending a full intra frame.   It is not appropriate to blindly retransmit RTP packets in response   to a NACK.  The importance of lost packets and the likelihood of themPerkins, et al.        Expires September 18, 2016              [Page 19]Internet-Draft               RTP for WebRTC                   March 2016   arriving in time to be useful needs to be considered before RTP   retransmission is used.   Receivers are REQUIRED to implement support for RTP retransmission   packets [RFC4588] sent using SSRC multiplexing, and MAY also support   RTP retransmission packets sent using session multiplexing.  Senders   MAY send RTP retransmission packets in response to NACKs if support   for the RTP retransmission payload format has been negotiated, and if   the sender believes it is useful to send a retransmission of the   packet(s) referenced in the NACK.  Senders do not need to retransmit   every NACKed packet.6.2.  Forward Error Correction (FEC)   The use of Forward Error Correction (FEC) can provide an effective   protection against some degree of packet loss, at the cost of steady   bandwidth overhead.  There are several FEC schemes that are defined   for use with RTP.  Some of these schemes are specific to a particular   RTP payload format, others operate across RTP packets and can be used   with any payload format.  It needs to be noted that using redundant   encoding or FEC will lead to increased play out delay, which needs to   be considered when choosing FEC schemes and their parameters.   WebRTC endpoints MUST follow the recommendations for FEC use given in   [I-D.ietf-rtcweb-fec].  WebRTC endpoints MAY support other types of   FEC, but these MUST be negotiated before they are used.7.  WebRTC Use of RTP: Rate Control and Media Adaptation   WebRTC will be used in heterogeneous network environments using a   variety of link technologies, including both wired and wireless   links, to interconnect potentially large groups of users around the   world.  As a result, the network paths between users can have widely   varying one-way delays, available bit-rates, load levels, and traffic   mixtures.  Individual endpoints can send one or more RTP packet   streams to each participant, and there can be several participants.   Each of these RTP packet streams can contain different types of   media, and the type of media, bit rate, and number of RTP packet   streams as well as transport-layer flows can be highly asymmetric.   Non-RTP traffic can share the network paths with RTP transport-layer   flows.  Since the network environment is not predictable or stable,   WebRTC Endpoints MUST ensure that the RTP traffic they generate can   adapt to match changes in the available network capacity.   The quality of experience for users of WebRTC is very dependent on   effective adaptation of the media to the limitations of the network.   Endpoints have to be designed so they do not transmit significantly   more data than the network path can support, except for very shortPerkins, et al.        Expires September 18, 2016              [Page 20]Internet-Draft               RTP for WebRTC                   March 2016   time periods, otherwise high levels of network packet loss or delay   spikes will occur, causing media quality degradation.  The limiting   factor on the capacity of the network path might be the link   bandwidth, or it might be competition with other traffic on the link   (this can be non-WebRTC traffic, traffic due to other WebRTC flows,   or even competition with other WebRTC flows in the same session).   An effective media congestion control algorithm is therefore an   essential part of the WebRTC framework.  However, at the time of this   writing, there is no standard congestion control algorithm that can   be used for interactive media applications such as WebRTC's flows.   Some requirements for congestion control algorithms for   RTCPeerConnections are discussed in [I-D.ietf-rmcat-cc-requirements].   If a standardized congestion control algorithm that satisfies these   requirements is developed in the future, this memo will need to be be   updated to mandate its use.7.1.  Boundary Conditions and Circuit Breakers   WebRTC Endpoints MUST implement the RTP circuit breaker algorithm   that is described in [I-D.ietf-avtcore-rtp-circuit-breakers].  The   RTP circuit breaker is designed to enable applications to recognise   and react to situations of extreme network congestion.  However,   since the RTP circuit breaker might not be triggered until congestion   becomes extreme, it cannot be considered a substitute for congestion   control, and applications MUST also implement congestion control to   allow them to adapt to changes in network capacity.  The congestion   control algorithm will have to be proprietary until a standardized   congestion control algorithm is available.  Any future RTP congestion   control algorithms are expected to operate within the envelope   allowed by the circuit breaker.   The session establishment signalling will also necessarily establish   boundaries to which the media bit-rate will conform.  The choice of   media codecs provides upper- and lower-bounds on the supported bit-   rates that the application can utilise to provide useful quality, and   the packetisation choices that exist.  In addition, the signalling   channel can establish maximum media bit-rate boundaries using, for   example, the SDP "b=AS:" or "b=CT:" lines and the RTP/AVPF Temporary   Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of   this memo).  Signalled bandwidth limitations, such as SDP "b=AS:" or   "b=CT:" lines received from the peer, MUST be followed when sending   RTP packet streams.  A WebRTC Endpoint receiving media SHOULD signal   its bandwidth limitations.  These limitations have to be based on   known bandwidth limitations, for example the capacity of the edge   links.Perkins, et al.        Expires September 18, 2016              [Page 21]Internet-Draft               RTP for WebRTC                   March 20167.2.  Congestion Control Interoperability and Legacy Systems   All endpoints that wish to interwork with WebRTC MUST implement RTCP   and provide congestion feedback via the defined RTCP reporting   mechanisms.   When interworking with legacy implementations that support RTCP using   the RTP/AVP profile [RFC3551], congestion feedback is provided in   RTCP RR packets every few seconds.  Implementations that have to   interwork with such endpoints MUST ensure that they keep within the   RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers]   constraints to limit the congestion they can cause.   If a legacy endpoint supports RTP/AVPF, this enables negotiation of   important parameters for frequent reporting, such as the "trr-int"   parameter, and the possibility that the endpoint supports some useful   feedback format for congestion control purpose such as TMMBR   [RFC5104].  Implementations that have to interwork with such   endpoints MUST ensure that they stay within the RTP circuit breaker   [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the   congestion they can cause, but might find that they can achieve   better congestion response depending on the amount of feedback that   is available.   With proprietary congestion control algorithms issues can arise when   different algorithms and implementations interact in a communication   session.  If the different implementations have made different   choices in regards to the type of adaptation, for example one sender   based, and one receiver based, then one could end up in situation   where one direction is dual controlled, when the other direction is   not controlled.  This memo cannot mandate behaviour for proprietary   congestion control algorithms, but implementations that use such   algorithms ought to be aware of this issue, and try to ensure that   effective congestion control is negotiated for media flowing in both   directions.  If the IETF were to standardise both sender- and   receiver-based congestion control algorithms for WebRTC traffic in   the future, the issues of interoperability, control, and ensuring   that both directions of media flow are congestion controlled would   also need to be considered.8.  WebRTC Use of RTP: Performance Monitoring   As described in Section 4.1, implementations are REQUIRED to generate   RTCP Sender Report (SR) and Reception Report (RR) packets relating to   the RTP packet streams they send and receive.  These RTCP reports can   be used for performance monitoring purposes, since they include basic   packet loss and jitter statistics.Perkins, et al.        Expires September 18, 2016              [Page 22]Internet-Draft               RTP for WebRTC                   March 2016   A large number of additional performance metrics are supported by the   RTCP Extended Reports (XR) framework, see [RFC3611][RFC6792].  At the   time of this writing, it is not clear what extended metrics are   suitable for use in WebRTC, so there is no requirement that   implementations generate RTCP XR packets.  However, implementations   that can use detailed performance monitoring data MAY generate RTCP   XR packets as appropriate.  The use of RTCP XR packets SHOULD be   signalled; implementations MUST ignore RTCP XR packets that are   unexpected or not understood.9.  WebRTC Use of RTP: Future Extensions   It is possible that the core set of RTP protocols and RTP extensions   specified in this memo will prove insufficient for the future needs   of WebRTC.  In this case, future updates to this memo have to be made   following the Guidelines for Writers of RTP Payload Format   Specifications [RFC2736], How to Write an RTP Payload Format   [I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP   Control Protocol [RFC5968], and SHOULD take into account any future   guidelines for extending RTP and related protocols that have been   developed.   Authors of future extensions are urged to consider the wide range of   environments in which RTP is used when recommending extensions, since   extensions that are applicable in some scenarios can be problematic   in others.  Where possible, the WebRTC framework will adopt RTP   extensions that are of general utility, to enable easy implementation   of a gateway to other applications using RTP, rather than adopt   mechanisms that are narrowly targeted at specific WebRTC use cases.10.  Signalling Considerations   RTP is built with the assumption that an external signalling channel   exists, and can be used to configure RTP sessions and their features.   The basic configuration of an RTP session consists of the following   parameters:   RTP Profile:  The name of the RTP profile to be used in session.  The      RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate      on basic level, as can their secure variants RTP/SAVP [RFC3711]      and RTP/SAVPF [RFC5124].  The secure variants of the profiles do      not directly interoperate with the non-secure variants, due to the      presence of additional header fields for authentication in SRTP      packets and cryptographic transformation of the payload.  WebRTC      requires the use of the RTP/SAVPF profile, and this MUST be      signalled.  Interworking functions might transform this into the      RTP/SAVP profile for a legacy use case, by indicating to thePerkins, et al.        Expires September 18, 2016              [Page 23]Internet-Draft               RTP for WebRTC                   March 2016      WebRTC Endpoint that the RTP/SAVPF is used and configuring a trr-      int value of 4 seconds.   Transport Information:  Source and destination IP address(s) and      ports for RTP and RTCP MUST be signalled for each RTP session.  In      WebRTC these transport addresses will be provided by ICE [RFC5245]      that signals candidates and arrives at nominated candidate address      pairs.  If RTP and RTCP multiplexing [RFC5761] is to be used, such      that a single port, i.e. transport-layer flow, is used for RTP and      RTCP flows, this MUST be signalled (see Section 4.5).   RTP Payload Types, media formats, and format parameters:  The mapping      between media type names (and hence the RTP payload formats to be      used), and the RTP payload type numbers MUST be signalled.  Each      media type MAY also have a number of media type parameters that      MUST also be signalled to configure the codec and RTP payload      format (the "a=fmtp:" line from SDP).  Section 4.3 of this memo      discusses requirements for uniqueness of payload types.   RTP Extensions:  The use of any additional RTP header extensions and      RTCP packet types, including any necessary parameters, MUST be      signalled.  This signalling is to ensure that a WebRTC Endpoint's      behaviour, especially when sending, of any extensions is      predictable and consistent.  For robustness, and for compatibility      with non-WebRTC systems that might be connected to a WebRTC      session via a gateway, implementations are REQUIRED to ignore      unknown RTCP packets and RTP header extensions (see also      Section 4.1).   RTCP Bandwidth:  Support for exchanging RTCP Bandwidth values to the      endpoints will be necessary.  This SHALL be done as described in      "Session Description Protocol (SDP) Bandwidth Modifiers for RTP      Control Protocol (RTCP) Bandwidth" [RFC3556] if using SDP, or      something semantically equivalent.  This also ensures that the      endpoints have a common view of the RTCP bandwidth.  A common view      of the RTCP bandwidth among different endpoints is important, to      prevent differences in RTCP packet timing and timeout intervals      causing interoperability problems.   These parameters are often expressed in SDP messages conveyed within   an offer/answer exchange.  RTP does not depend on SDP or on the   offer/answer model, but does require all the necessary parameters to   be agreed upon, and provided to the RTP implementation.  Note that in   WebRTC it will depend on the signalling model and API how these   parameters need to be configured but they will be need to either be   set in the API or explicitly signalled between the peers.Perkins, et al.        Expires September 18, 2016              [Page 24]Internet-Draft               RTP for WebRTC                   March 201611.  WebRTC API Considerations   The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and   Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses   the concept of a MediaStream that consists of zero or more   MediaStreamTracks.  A MediaStreamTrack is an individual stream of   media from any type of media source like a microphone or a camera,   but also conceptual sources, like a audio mix or a video composition,   are possible.  The MediaStreamTracks within a MediaStream might need   to be synchronized during play back.   A MediaStreamTrack's realisation in RTP in the context of an   RTCPeerConnection consists of a source packet stream identified with   an SSRC within an RTP session part of the RTCPeerConnection.  The   MediaStreamTrack can also result in additional packet streams, and   thus SSRCs, in the same RTP session.  These can be dependent packet   streams from scalable encoding of the source stream associated with   the MediaStreamTrack, if such a media encoder is used.  They can also   be redundancy packet streams, these are created when applying Forward   Error Correction (Section 6.2) or RTP retransmission (Section 6.1) to   the source packet stream.   It is important to note that the same media source can be feeding   multiple MediaStreamTracks.  As different sets of constraints or   other parameters can be applied to the MediaStreamTrack, each   MediaStreamTrack instance added to a RTCPeerConnection SHALL result   in an independent source packet stream, with its own set of   associated packet streams, and thus different SSRC(s).  It will   depend on applied constraints and parameters if the source stream and   the encoding configuration will be identical between different   MediaStreamTracks sharing the same media source.  If the encoding   parameters and constraints are the same, an implementation could   choose to use only one encoded stream to create the different RTP   packet streams.  Note that such optimisations would need to take into   account that the constraints for one of the MediaStreamTracks can at   any moment change, meaning that the encoding configurations might no   longer be identical and two different encoder instances would then be   needed.   The same MediaStreamTrack can also be included in multiple   MediaStreams, thus multiple sets of MediaStreams can implicitly need   to use the same synchronisation base.  To ensure that this works in   all cases, and does not force an endpoint to disrupt the media by   changing synchronisation base and CNAME during delivery of any   ongoing packet streams, all MediaStreamTracks and their associated   SSRCs originating from the same endpoint need to be sent using the   same CNAME within one RTCPeerConnection.  This is motivating the use   of a single CNAME in Section 4.9.Perkins, et al.        Expires September 18, 2016              [Page 25]Internet-Draft               RTP for WebRTC                   March 2016      The requirement on using the same CNAME for all SSRCs that      originate from the same endpoint, does not require a middlebox      that forwards traffic from multiple endpoints to only use a single      CNAME.   Different CNAMEs normally need to be used for different   RTCPeerConnection instances, as specified in Section 4.9.  Having two   communication sessions with the same CNAME could enable tracking of a   user or device across different services (see Section 4.4.1 of   [I-D.ietf-rtcweb-security] for details).  A web application can   request that the CNAMEs used in different RTCPeerConnections (within   a same-orign context) be the same, this allows for synchronization of   the endpoint's RTP packet streams across the different   RTCPeerConnections.      Note: this doesn't result in a tracking issue, since the creation      of matching CNAMEs depends on existing tracking within a single      origin.   The above will currently force a WebRTC Endpoint that receives a   MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing   on any RTCPeerConnection to perform resynchronisation of the stream.   Since the sending party needs to change the CNAME to the one it uses,   this implies it has to use a local system clock as timebase for the   synchronisation.  Thus, the relative relation between the timebase of   the incoming stream and the system sending out needs to be defined.   This relation also needs monitoring for clock drift and likely   adjustments of the synchronisation.  The sending entity is also   responsible for congestion control for its sent streams.  In cases of   packet loss the loss of incoming data also needs to be handled.  This   leads to the observation that the method that is least likely to   cause issues or interruptions in the outgoing source packet stream is   a model of full decoding, including repair etc., followed by encoding   of the media again into the outgoing packet stream.  Optimisations of   this method are clearly possible and implementation specific.   A WebRTC Endpoint MUST support receiving multiple MediaStreamTracks,   where each of the different MediaStreamTracks (and their sets of   associated packet streams) uses different CNAMEs.  However,   MediaStreamTracks that are received with different CNAMEs have no   defined synchronisation.      Note: The motivation for supporting reception of multiple CNAMEs      is to allow for forward compatibility with any future changes that      enable more efficient stream handling when endpoints relay/forward      streams.  It also ensures that endpoints can interoperate with      certain types of multi-stream middleboxes or endpoints that are      not WebRTC.Perkins, et al.        Expires September 18, 2016              [Page 26]Internet-Draft               RTP for WebRTC                   March 2016   Javascript Session Establishment Protocol [I-D.ietf-rtcweb-jsep]   specifies that the binding between the WebRTC MediaStreams,   MediaStreamTracks and the SSRC is done as specified in "Cross Session   Stream Identification in the Session Description Protocol"   [I-D.ietf-mmusic-msid].  The MSID document [I-D.ietf-mmusic-msid]   also defines, in section 4.1, how to map unknown source packet stream   SSRCs to MediaStreamTracks and MediaStreams.  This later is relevant   to handle some cases of legacy interoperability.  Commonly the RTP   Payload Type of any incoming packets will reveal if the packet stream   is a source stream or a redundancy or dependent packet stream.  The   association to the correct source packet stream depends on the   payload format in use for the packet stream.   Finally this specification puts a requirement on the WebRTC API to   realize a method for determining the CSRC list (Section 4.1) as well   as the Mixer-to-Client audio levels (Section 5.2.3) (when supported)   and the basic requirements for this is further discussed in   Section 12.2.1.12.  RTP Implementation Considerations   The following discussion provides some guidance on the implementation   of the RTP features described in this memo.  The focus is on a WebRTC   Endpoint implementation perspective, and while some mention is made   of the behaviour of middleboxes, that is not the focus of this memo.12.1.  Configuration and Use of RTP Sessions   A WebRTC Endpoint will be a simultaneous participant in one or more   RTP sessions.  Each RTP session can convey multiple media sources,   and can include media data from multiple endpoints.  In the   following, some ways in which WebRTC Endpoints can configure and use   RTP sessions are outlined.12.1.1.  Use of Multiple Media Sources Within an RTP Session   RTP is a group communication protocol, and every RTP session can   potentially contain multiple RTP packet streams.  There are several   reasons why this might be desirable:   Multiple media types:  Outside of WebRTC, it is common to use one RTP      session for each type of media source (e.g., one RTP session for      audio sources and one for video sources, each sent over different      transport layer flows).  However, to reduce the number of UDP      ports used, the default in WebRTC is to send all types of media in      a single RTP session, as described in Section 4.4, using RTP and      RTCP multiplexing (Section 4.5) to further reduce the number of      UDP ports needed.  This RTP session then uses only one bi-Perkins, et al.        Expires September 18, 2016              [Page 27]Internet-Draft               RTP for WebRTC                   March 2016      directional transport-layer flow, but will contain multiple RTP      packet streams, each containing a different type of media.  A      common example might be an endpoint with a camera and microphone      that sends two RTP packet streams, one video and one audio, into a      single RTP session.   Multiple Capture Devices:  A WebRTC Endpoint might have multiple      cameras, microphones, or other media capture devices, and so might      want to generate several RTP packet streams of the same media      type.  Alternatively, it might want to send media from a single      capture device in several different formats or quality settings at      once.  Both can result in a single endpoint sending multiple RTP      packet streams of the same media type into a single RTP session at      the same time.   Associated Repair Data:  An endpoint might send a RTP packet stream      that is somehow associated with another stream.  For example, it      might send an RTP packet stream that contains FEC or      retransmission data relating to another stream.  Some RTP payload      formats send this sort of associated repair data as part of the      source packet stream, while others send it as a separate packet      stream.   Layered or Multiple Description Coding:  An endpoint can use a      layered media codec, for example H.264 SVC, or a multiple      description codec, that generates multiple RTP packet streams,      each with a distinct RTP SSRC, within a single RTP session.   RTP Mixers, Translators, and Other Middleboxes:  An RTP session, in      the WebRTC context, is a point-to-point association between an      endpoint and some other peer device, where those devices share a      common SSRC space.  The peer device might be another WebRTC      Endpoint, or it might be an RTP mixer, translator, or some other      form of media processing middlebox.  In the latter cases, the      middlebox might send mixed or relayed RTP streams from several      participants, that the WebRTC Endpoint will need to render.  Thus,      even though a WebRTC Endpoint might only be a member of a single      RTP session, the peer device might be extending that RTP session      to incorporate other endpoints.  WebRTC is a group communication      environment and endpoints need to be capable of receiving,      decoding, and playing out multiple RTP packet streams at once,      even in a single RTP session.12.1.2.  Use of Multiple RTP Sessions   In addition to sending and receiving multiple RTP packet streams   within a single RTP session, a WebRTC Endpoint might participate inPerkins, et al.        Expires September 18, 2016              [Page 28]Internet-Draft               RTP for WebRTC                   March 2016   multiple RTP sessions.  There are several reasons why a WebRTC   Endpoint might choose to do this:   To interoperate with legacy devices:  The common practice in the non-      WebRTC world is to send different types of media in separate RTP      sessions, for example using one RTP session for audio and another      RTP session, on a separate transport layer flow, for video.  All      WebRTC Endpoints need to support the option of sending different      types of media on different RTP sessions, so they can interwork      with such legacy devices.  This is discussed further in      Section 4.4.   To provide enhanced quality of service:  Some network-based quality      of service mechanisms operate on the granularity of transport      layer flows.  If it is desired to use these mechanisms to provide      differentiated quality of service for some RTP packet streams,      then those RTP packet streams need to be sent in a separate RTP      session using a different transport-layer flow, and with      appropriate quality of service marking.  This is discussed further      in Section 12.1.3.   To separate media with different purposes:  An endpoint might want to      send RTP packet streams that have different purposes on different      RTP sessions, to make it easy for the peer device to distinguish      them.  For example, some centralised multiparty conferencing      systems display the active speaker in high resolution, but show      low resolution "thumbnails" of other participants.  Such systems      might configure the endpoints to send simulcast high- and low-      resolution versions of their video using separate RTP sessions, to      simplify the operation of the RTP middlebox.  In the WebRTC      context this is currently possible by establishing multiple WebRTC      MediaStreamTracks that have the same media source in one (or more)      RTCPeerConnection.  Each MediaStreamTrack is then configured to      deliver a particular media quality and thus media bit-rate, and      will produce an independently encoded version with the codec      parameters agreed specifically in the context of that      RTCPeerConnection.  The RTP middlebox can distinguish packets      corresponding to the low- and high-resolution streams by      inspecting their SSRC, RTP payload type, or some other information      contained in RTP payload, RTP header extension or RTCP packets,      but it can be easier to distinguish the RTP packet streams if they      arrive on separate RTP sessions on separate transport-layer flows.   To directly connect with multiple peers:  A multi-party conference      does not need to use an RTP middlebox.  Rather, a multi-unicast      mesh can be created, comprising several distinct RTP sessions,      with each participant sending RTP traffic over a separate RTP      session (that is, using an independent RTCPeerConnection object)Perkins, et al.        Expires September 18, 2016              [Page 29]Internet-Draft               RTP for WebRTC                   March 2016      to every other participant, as shown in Figure 1.  This topology      has the benefit of not requiring an RTP middlebox node that is      trusted to access and manipulate the media data.  The downside is      that it increases the used bandwidth at each sender by requiring      one copy of the RTP packet streams for each participant that are      part of the same session beyond the sender itself.   +---+     +---+   | A |<--->| B |   +---+     +---+     ^         ^      \       /       \     /        v   v        +---+        | C |        +---+            Figure 1: Multi-unicast using several RTP sessions      The multi-unicast topology could also be implemented as a single      RTP session, spanning multiple peer-to-peer transport layer      connections, or as several pairwise RTP sessions, one between each      pair of peers.  To maintain a coherent mapping of the relationship      between RTP sessions and RTCPeerConnection objects it is recommend      that this is implemented as several individual RTP sessions.  The      only downside is that endpoint A will not learn of the quality of      any transmission happening between B and C, since it will not see      RTCP reports for the RTP session between B and C, whereas it would      if all three participants were part of a single RTP session.      Experience with the Mbone tools (experimental RTP-based multicast      conferencing tools from the late 1990s) has showed that RTCP      reception quality reports for third parties can be presented to      users in a way that helps them understand asymmetric network      problems, and the approach of using separate RTP sessions prevents      this.  However, an advantage of using separate RTP sessions is      that it enables using different media bit-rates and RTP session      configurations between the different peers, thus not forcing B to      endure the same quality reductions if there are limitations in the      transport from A to C as C will.  It is believed that these      advantages outweigh the limitations in debugging power.   To indirectly connect with multiple peers:  A common scenario in      multi-party conferencing is to create indirect connections to      multiple peers, using an RTP mixer, translator, or some other type      of RTP middlebox.  Figure 2 outlines a simple topology that might      be used in a four-person centralised conference.  The middleboxPerkins, et al.        Expires September 18, 2016              [Page 30]Internet-Draft               RTP for WebRTC                   March 2016      acts to optimise the transmission of RTP packet streams from      certain perspectives, either by only sending some of the received      RTP packet stream to any given receiver, or by providing a      combined RTP packet stream out of a set of contributing streams.   +---+      +-------------+      +---+   | A |<---->|             |<---->| B |   +---+      | RTP mixer,  |      +---+              | translator, |              | or other    |   +---+      | middlebox   |      +---+   | C |<---->|             |<---->| D |   +---+      +-------------+      +---+                Figure 2: RTP mixer with only unicast paths      There are various methods of implementation for the middlebox.  If      implemented as a standard RTP mixer or translator, a single RTP      session will extend across the middlebox and encompass all the      endpoints in one multi-party session.  Other types of middlebox      might use separate RTP sessions between each endpoint and the      middlebox.  A common aspect is that these RTP middleboxes can use      a number of tools to control the media encoding provided by a      WebRTC Endpoint.  This includes functions like requesting the      breaking of the encoding chain and have the encoder produce a so      called Intra frame.  Another is limiting the bit-rate of a given      stream to better suit the mixer view of the multiple down-streams.      Others are controlling the most suitable frame-rate, picture      resolution, the trade-off between frame-rate and spatial quality.      The middlebox has the responsibility to correctly perform      congestion control, source identification, manage synchronisation      while providing the application with suitable media optimisations.      The middlebox also has to be a trusted node when it comes to      security, since it manipulates either the RTP header or the media      itself (or both) received from one endpoint, before sending it on      towards the endpoint(s), thus they need to be able to decrypt and      then re-encrypt the RTP packet stream before sending it out.      RTP Mixers can create a situation where an endpoint experiences a      situation in-between a session with only two endpoints and      multiple RTP sessions.  Mixers are expected to not forward RTCP      reports regarding RTP packet streams across themselves.  This is      due to the difference in the RTP packet streams provided to the      different endpoints.  The original media source lacks information      about a mixer's manipulations prior to sending it the different      receivers.  This scenario also results in that an endpoint'sPerkins, et al.        Expires September 18, 2016              [Page 31]Internet-Draft               RTP for WebRTC                   March 2016      feedback or requests go to the mixer.  When the mixer can't act on      this by itself, it is forced to go to the original media source to      fulfil the receivers request.  This will not necessarily be      explicitly visible to any RTP and RTCP traffic, but the      interactions and the time to complete them will indicate such      dependencies.      Providing source authentication in multi-party scenarios is a      challenge.  In the mixer-based topologies, endpoints source      authentication is based on, firstly, verifying that media comes      from the mixer by cryptographic verification and, secondly, trust      in the mixer to correctly identify any source towards the      endpoint.  In RTP sessions where multiple endpoints are directly      visible to an endpoint, all endpoints will have knowledge about      each others' master keys, and can thus inject packets claimed to      come from another endpoint in the session.  Any node performing      relay can perform non-cryptographic mitigation by preventing      forwarding of packets that have SSRC fields that came from other      endpoints before.  For cryptographic verification of the source,      SRTP would require additional security mechanisms, for example      TESLA for SRTP [RFC4383], that are not part of the base WebRTC      standards.   To forward media between multiple peers:  It is sometimes desirable      for an endpoint that receives an RTP packet stream to be able to      forward that RTP packet stream to a third party.  The are some      obvious security and privacy implications in supporting this, but      also potential uses.  This is supported in the W3C API by taking      the received and decoded media and using it as media source that      is re-encoding and transmitted as a new stream.      At the RTP layer, media forwarding acts as a back-to-back RTP      receiver and RTP sender.  The receiving side terminates the RTP      session and decodes the media, while the sender side re-encodes      and transmits the media using an entirely separate RTP session.      The original sender will only see a single receiver of the media,      and will not be able to tell that forwarding is happening based on      RTP-layer information since the RTP session that is used to send      the forwarded media is not connected to the RTP session on which      the media was received by the node doing the forwarding.      The endpoint that is performing the forwarding is responsible for      producing an RTP packet stream suitable for onwards transmission.      The outgoing RTP session that is used to send the forwarded media      is entirely separate to the RTP session on which the media was      received.  This will require media transcoding for congestion      control purpose to produce a suitable bit-rate for the outgoing      RTP session, reducing media quality and forcing the forwardingPerkins, et al.        Expires September 18, 2016              [Page 32]Internet-Draft               RTP for WebRTC                   March 2016      endpoint to spend the resource on the transcoding.  The media      transcoding does result in a separation of the two different legs      removing almost all dependencies, and allowing the forwarding      endpoint to optimise its media transcoding operation.  The cost is      greatly increased computational complexity on the forwarding node.      Receivers of the forwarded stream will see the forwarding device      as the sender of the stream, and will not be able to tell from the      RTP layer that they are receiving a forwarded stream rather than      an entirely new RTP packet stream generated by the forwarding      device.12.1.3.  Differentiated Treatment of RTP Streams   There are use cases for differentiated treatment of RTP packet   streams.  Such differentiation can happen at several places in the   system.  First of all is the prioritization within the endpoint   sending the media, which controls, both which RTP packet streams that   will be sent, and their allocation of bit-rate out of the current   available aggregate as determined by the congestion control.   It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will   allow the application to indicate relative priorities for different   MediaStreamTracks.  These priorities can then be used to influence   the local RTP processing, especially when it comes to congestion   control response in how to divide the available bandwidth between the   RTP packet streams.  Any changes in relative priority will also need   to be considered for RTP packet streams that are associated with the   main RTP packet streams, such as redundant streams for RTP   retransmission and FEC.  The importance of such redundant RTP packet   streams is dependent on the media type and codec used, in regards to   how robust that codec is to packet loss.  However, a default policy   might to be to use the same priority for redundant RTP packet stream   as for the source RTP packet stream.   Secondly, the network can prioritize transport-layer flows and sub-   flows, including RTP packet streams.  Typically, differential   treatment includes two steps, the first being identifying whether an   IP packet belongs to a class that has to be treated differently, the   second consisting of the actual mechanism to prioritize packets.   Three common methods for classifying IP packets are:   DiffServ:  The endpoint marks a packet with a DiffServ code point to      indicate to the network that the packet belongs to a particular      class.   Flow based:  Packets that need to be given a particular treatment are      identified using a combination of IP and port address.Perkins, et al.        Expires September 18, 2016              [Page 33]Internet-Draft               RTP for WebRTC                   March 2016   Deep Packet Inspection:  A network classifier (DPI) inspects the      packet and tries to determine if the packet represents a      particular application and type that is to be prioritized.   Flow-based differentiation will provide the same treatment to all   packets within a transport-layer flow, i.e., relative prioritization   is not possible.  Moreover, if the resources are limited it might not   be possible to provide differential treatment compared to best-effort   for all the RTP packet streams used in a WebRTC session.  The use of   flow-based differentiation needs to be coordinated between the WebRTC   system and the network(s).  The WebRTC endpoint needs to know that   flow-based differentiation might be used to provide the separation of   the RTP packet streams onto different UDP flows to enable a more   granular usage of flow based differentiation.  The used flows, their   5-tuples and prioritization will need to be communicated to the   network so that it can identify the flows correctly to enable   prioritization.  No specific protocol support for this is specified.   DiffServ assumes that either the endpoint or a classifier can mark   the packets with an appropriate DSCP so that the packets are treated   according to that marking.  If the endpoint is to mark the traffic   two requirements arise in the WebRTC context: 1) The WebRTC Endpoint   has to know which DSCP to use and that it can use them on some set of   RTP packet streams. 2) The information needs to be propagated to the   operating system when transmitting the packet.  Details of this   process are outside the scope of this memo and are further discussed   in "DSCP and other packet markings for RTCWeb QoS"   [I-D.ietf-tsvwg-rtcweb-qos].   Deep Packet Inspectors will, despite the SRTP media encryption, still   be fairly capable at classifying the RTP streams.  The reason is that   SRTP leaves the first 12 bytes of the RTP header unencrypted.  This   enables easy RTP stream identification using the SSRC and provides   the classifier with useful information that can be correlated to   determine for example the stream's media type.  Using packet sizes,   reception times, packet inter-spacing, RTP timestamp increments and   sequence numbers, fairly reliable classifications are achieved.   For packet based marking schemes it might be possible to mark   individual RTP packets differently based on the relative priority of   the RTP payload.  For example video codecs that have I, P, and B   pictures could prioritise any payloads carrying only B frames less,   as these are less damaging to loose.  However, depending on the QoS   mechanism and what markings that are applied, this can result in not   only different packet drop probabilities but also packet reordering,   see [I-D.ietf-tsvwg-rtcweb-qos] and [I-D.ietf-dart-dscp-rtp] for   further discussion.  As a default policy all RTP packets related to a   RTP packet stream ought to be provided with the same prioritization;Perkins, et al.        Expires September 18, 2016              [Page 34]Internet-Draft               RTP for WebRTC                   March 2016   per-packet prioritization is outside the scope of this memo, but   might be specified elsewhere in future.   It is also important to consider how RTCP packets associated with a   particular RTP packet stream need to be marked.  RTCP compound   packets with Sender Reports (SR), ought to be marked with the same   priority as the RTP packet stream itself, so the RTCP-based round-   trip time (RTT) measurements are done using the same transport-layer   flow priority as the RTP packet stream experiences.  RTCP compound   packets containing RR packet ought to be sent with the priority used   by the majority of the RTP packet streams reported on.  RTCP packets   containing time-critical feedback packets can use higher priority to   improve the timeliness and likelihood of delivery of such feedback.12.2.  Media Source, RTP Streams, and Participant Identification12.2.1.  Media Source Identification   Each RTP packet stream is identified by a unique synchronisation   source (SSRC) identifier.  The SSRC identifier is carried in each of   the RTP packets comprising a RTP packet stream, and is also used to   identify that stream in the corresponding RTCP reports.  The SSRC is   chosen as discussed in Section 4.8.  The first stage in   demultiplexing RTP and RTCP packets received on a single transport   layer flow at a WebRTC Endpoint is to separate the RTP packet streams   based on their SSRC value; once that is done, additional   demultiplexing steps can determine how and where to render the media.   RTP allows a mixer, or other RTP-layer middlebox, to combine encoded   streams from multiple media sources to form a new encoded stream from   a new media source (the mixer).  The RTP packets in that new RTP   packet stream can include a Contributing Source (CSRC) list,   indicating which original SSRCs contributed to the combined source   stream.  As described in Section 4.1, implementations need to support   reception of RTP data packets containing a CSRC list and RTCP packets   that relate to sources present in the CSRC list.  The CSRC list can   change on a packet-by-packet basis, depending on the mixing operation   being performed.  Knowledge of what media sources contributed to a   particular RTP packet can be important if the user interface   indicates which participants are active in the session.  Changes in   the CSRC list included in packets needs to be exposed to the WebRTC   application using some API, if the application is to be able to track   changes in session participation.  It is desirable to map CSRC values   back into WebRTC MediaStream identities as they cross this API, to   avoid exposing the SSRC/CSRC name space to WebRTC applications.   If the mixer-to-client audio level extension [RFC6465] is being used   in the session (see Section 5.2.3), the information in the CSRC listPerkins, et al.        Expires September 18, 2016              [Page 35]Internet-Draft               RTP for WebRTC                   March 2016   is augmented by audio level information for each contributing source.   It is desirable to expose this information to the WebRTC application   using some API, after mapping the CSRC values to WebRTC MediaStream   identities, so it can be exposed in the user interface.12.2.2.  SSRC Collision Detection   The RTP standard requires RTP implementations to have support for   detecting and handling SSRC collisions, i.e., resolve the conflict   when two different endpoints use the same SSRC value (see section 8.2   of [RFC3550]).  This requirement also applies to WebRTC Endpoints.   There are several scenarios where SSRC collisions can occur:   o  In a point-to-point session where each SSRC is associated with      either of the two endpoints and where the main media carrying SSRC      identifier will be announced in the signalling channel, a      collision is less likely to occur due to the information about      used SSRCs.  If SDP is used, this information is provided by      Source-Specific SDP Attributes [RFC5576].  Still, collisions can      occur if both endpoints start using a new SSRC identifier prior to      having signalled it to the peer and received acknowledgement on      the signalling message.  The Source-Specific SDP Attributes      [RFC5576] contains a mechanism to signal how the endpoint resolved      the SSRC collision.   o  SSRC values that have not been signalled could also appear in an      RTP session.  This is more likely than it appears, since some RTP      functions use extra SSRCs to provide their functionality.  For      example, retransmission data might be transmitted using a separate      RTP packet stream that requires its own SSRC, separate to the SSRC      of the source RTP packet stream [RFC4588].  In those cases, an      endpoint can create a new SSRC that strictly doesn't need to be      announced over the signalling channel to function correctly on      both RTP and RTCPeerConnection level.   o  Multiple endpoints in a multiparty conference can create new      sources and signal those towards the RTP middlebox.  In cases      where the SSRC/CSRC are propagated between the different endpoints      from the RTP middlebox collisions can occur.   o  An RTP middlebox could connect an endpoint's RTCPeerConnection to      another RTCPeerConnection from the same endpoint, thus forming a      loop where the endpoint will receive its own traffic.  While it is      clearly considered a bug, it is important that the endpoint is      able to recognise and handle the case when it occurs.  This case      becomes even more problematic when media mixers, and so on, are      involved, where the stream received is a different stream but      still contains this client's input.Perkins, et al.        Expires September 18, 2016              [Page 36]Internet-Draft               RTP for WebRTC                   March 2016   These SSRC/CSRC collisions can only be handled on RTP level as long   as the same RTP session is extended across multiple   RTCPeerConnections by a RTP middlebox.  To resolve the more generic   case where multiple RTCPeerConnections are interconnected,   identification of the media source(s) part of a MediaStreamTrack   being propagated across multiple interconnected RTCPeerConnection   needs to be preserved across these interconnections.12.2.3.  Media Synchronisation Context   When an endpoint sends media from more than one media source, it   needs to consider if (and which of) these media sources are to be   synchronized.  In RTP/RTCP, synchronisation is provided by having a   set of RTP packet streams be indicated as coming from the same   synchronisation context and logical endpoint by using the same RTCP   CNAME identifier.   The next provision is that the internal clocks of all media sources,   i.e., what drives the RTP timestamp, can be correlated to a system   clock that is provided in RTCP Sender Reports encoded in an NTP   format.  By correlating all RTP timestamps to a common system clock   for all sources, the timing relation of the different RTP packet   streams, also across multiple RTP sessions can be derived at the   receiver and, if desired, the streams can be synchronized.  The   requirement is for the media sender to provide the correlation   information; it is up to the receiver to use it or not.13.  Security Considerations   The overall security architecture for WebRTC is described in   [I-D.ietf-rtcweb-security-arch], and security considerations for the   WebRTC framework are described in [I-D.ietf-rtcweb-security].  These   considerations also apply to this memo.   The security considerations of the RTP specification, the RTP/SAVPF   profile, and the various RTP/RTCP extensions and RTP payload formats   that form the complete protocol suite described in this memo apply.   It is not believed there are any new security considerations   resulting from the combination of these various protocol extensions.   The Extended Secure RTP Profile for Real-time Transport Control   Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides   handling of fundamental issues by offering confidentiality, integrity   and partial source authentication.  A mandatory to implement and use   media security solution is created by combining this secured RTP   profile and DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of   [I-D.ietf-rtcweb-security-arch].Perkins, et al.        Expires September 18, 2016              [Page 37]Internet-Draft               RTP for WebRTC                   March 2016   RTCP packets convey a Canonical Name (CNAME) identifier that is used   to associate RTP packet streams that need to be synchronised across   related RTP sessions.  Inappropriate choice of CNAME values can be a   privacy concern, since long-term persistent CNAME identifiers can be   used to track users across multiple WebRTC calls.  Section 4.9 of   this memo mandates generation of short-term persistent RTCP CNAMES,   as specified in RFC7022, resulting in untraceable CNAME values that   alleviate this risk.   Some potential denial of service attacks exist if the RTCP reporting   interval is configured to an inappropriate value.  This could be done   by configuring the RTCP bandwidth fraction to an excessively large or   small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some   similar mechanism, or by choosing an excessively large or small value   for the RTP/AVPF minimal receiver report interval (if using SDP, this   is the "a=rtcp-fb:... trr-int" parameter) [RFC4585].  The risks are   as follows:   1.  the RTCP bandwidth could be configured to make the regular       reporting interval so large that effective congestion control       cannot be maintained, potentially leading to denial of service       due to congestion caused by the media traffic;   2.  the RTCP interval could be configured to a very small value,       causing endpoints to generate high rate RTCP traffic, potentially       leading to denial of service due to the non-congestion controlled       RTCP traffic; and   3.  RTCP parameters could be configured differently for each       endpoint, with some of the endpoints using a large reporting       interval and some using a smaller interval, leading to denial of       service due to premature participant timeouts due to mismatched       timeout periods which are based on the reporting interval (this       is a particular concern if endpoints use a small but non-zero       value for the RTP/AVPF minimal receiver report interval (trr-int)       [RFC4585], as discussed in Section 6.1 of       [I-D.ietf-avtcore-rtp-multi-stream]).   Premature participant timeout can be avoided by using the fixed (non-   reduced) minimum interval when calculating the participant timeout   (see Section 4.1 of this memo and Section 6.1 of   [I-D.ietf-avtcore-rtp-multi-stream]).  To address the other concerns,   endpoints SHOULD ignore parameters that configure the RTCP reporting   interval to be significantly longer than the default five second   interval specified in [RFC3550] (unless the media data rate is so low   that the longer reporting interval roughly corresponds to 5% of the   media data rate), or that configure the RTCP reporting interval small   enough that the RTCP bandwidth would exceed the media bandwidth.Perkins, et al.        Expires September 18, 2016              [Page 38]Internet-Draft               RTP for WebRTC                   March 2016   The guidelines in [RFC6562] apply when using variable bit rate (VBR)   audio codecs such as Opus (see Section 4.3 for discussion of mandated   audio codecs).  The guidelines in [RFC6562] also apply, but are of   lesser importance, when using the client-to-mixer audio level header   extensions (Section 5.2.2) or the mixer-to-client audio level header   extensions (Section 5.2.3).  The use of the encryption of the header   extensions are RECOMMENDED, unless there are known reasons, like RTP   middleboxes performing voice activity based source selection or third   party monitoring that will greatly benefit from the information, and   this has been expressed using API or signalling.  If further evidence   are produced to show that information leakage is significant from   audio level indications, then use of encryption needs to be mandated   at that time.   In multi-party communication scenarios using RTP Middleboxes, a lot   of trust is placed on these middleboxes to preserve the sessions   security.  The middlebox needs to maintain the confidentiality,   integrity and perform source authentication.  As discussed in   Section 12.1.1 the middlebox can perform checks that prevents any   endpoint participating in a conference to impersonate another.  Some   additional security considerations regarding multi-party topologies   can be found in [I-D.ietf-avtcore-rtp-topologies-update].14.  IANA Considerations   This memo makes no request of IANA.   Note to RFC Editor: this section is to be removed on publication as   an RFC.15.  Acknowledgements   The authors would like to thank Bernard Aboba, Harald Alvestrand,   Cary Bran, Ben Campbell, Alissa Cooper, Spencer Dawkins, Charles   Eckel, Alex Eleftheriadis, Christian Groves, Chris Inacio, Cullen   Jennings, Olle Johansson, Suhas Nandakumar, Dan Romascanu, Jim   Spring, Martin Thomson, and the other members of the IETF RTCWEB   working group for their valuable feedback.16.  References16.1.  Normative References   [I-D.ietf-avtcore-multi-media-rtp-session]              Westerlund, M., Perkins, C., and J. Lennox, "Sending              Multiple Types of Media in a Single RTP Session", draft-              ietf-avtcore-multi-media-rtp-session-13 (work in              progress), December 2015.Perkins, et al.        Expires September 18, 2016              [Page 39]Internet-Draft               RTP for WebRTC                   March 2016   [I-D.ietf-avtcore-rtp-circuit-breakers]              Perkins, C. and V. Varun, "Multimedia Congestion Control:              Circuit Breakers for Unicast RTP Sessions", draft-ietf-              avtcore-rtp-circuit-breakers-13 (work in progress),              February 2016.   [I-D.ietf-avtcore-rtp-multi-stream]              Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,              "Sending Multiple RTP Streams in a Single RTP Session",              draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),              December 2015.   [I-D.ietf-avtcore-rtp-multi-stream-optimisation]              Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,              "Sending Multiple RTP Streams in a Single RTP Session:              Grouping RTCP Reception Statistics and Other Feedback",              draft-ietf-avtcore-rtp-multi-stream-optimisation-12 (work              in progress), March 2016.   [I-D.ietf-avtcore-rtp-topologies-update]              Westerlund, M. and S. Wenger, "RTP Topologies", draft-              ietf-avtcore-rtp-topologies-update-10 (work in progress),              July 2015.   [I-D.ietf-mmusic-mux-exclusive]              Holmberg, C., "Indicating Exclusive Support of RTP/RTCP              Multiplexing using SDP", draft-ietf-mmusic-mux-              exclusive-03 (work in progress), February 2016.   [I-D.ietf-mmusic-sdp-bundle-negotiation]              Holmberg, C., Alvestrand, H., and C. Jennings,              "Negotiating Media Multiplexing Using the Session              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-              negotiation-27 (work in progress), February 2016.   [I-D.ietf-rtcweb-audio]              Valin, J. and C. Bran, "WebRTC Audio Codec and Processing              Requirements", draft-ietf-rtcweb-audio-10 (work in              progress), February 2016.   [I-D.ietf-rtcweb-fec]              Uberti, J., "WebRTC Forward Error Correction              Requirements", draft-ietf-rtcweb-fec-02 (work in              progress), October 2015.Perkins, et al.        Expires September 18, 2016              [Page 40]Internet-Draft               RTP for WebRTC                   March 2016   [I-D.ietf-rtcweb-overview]              Alvestrand, H., "Overview: Real Time Protocols for              Browser-based Applications", draft-ietf-rtcweb-overview-15              (work in progress), January 2016.   [I-D.ietf-rtcweb-security]              Rescorla, E., "Security Considerations for WebRTC", draft-              ietf-rtcweb-security-08 (work in progress), February 2015.   [I-D.ietf-rtcweb-security-arch]              Rescorla, E., "WebRTC Security Architecture", draft-ietf-              rtcweb-security-arch-11 (work in progress), March 2015.   [I-D.ietf-rtcweb-video]              Roach, A., "WebRTC Video Processing and Codec              Requirements", draft-ietf-rtcweb-video-06 (work in              progress), June 2015.   [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>.   [RFC2736]  Handley, M. and C. Perkins, "Guidelines for Writers of RTP              Payload Format Specifications", BCP 36, RFC 2736,              DOI 10.17487/RFC2736, December 1999,              <http://www.rfc-editor.org/info/rfc2736>.   [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>.   [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>.   [RFC3556]  Casner, S., "Session Description Protocol (SDP) Bandwidth              Modifiers for RTP Control Protocol (RTCP) Bandwidth",              RFC 3556, DOI 10.17487/RFC3556, July 2003,              <http://www.rfc-editor.org/info/rfc3556>.   [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>.Perkins, et al.        Expires September 18, 2016              [Page 41]Internet-Draft               RTP for WebRTC                   March 2016   [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>.   [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>.   [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>.   [RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",              BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,              <http://www.rfc-editor.org/info/rfc4961>.   [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>.   [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>.   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP              Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July              2008, <http://www.rfc-editor.org/info/rfc5285>.   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size              Real-Time Transport Control Protocol (RTCP): Opportunities              and Consequences", RFC 5506, DOI 10.17487/RFC5506, April              2009, <http://www.rfc-editor.org/info/rfc5506>.   [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>.   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer              Security (DTLS) Extension to Establish Keys for the Secure              Real-time Transport Protocol (SRTP)", RFC 5764,              DOI 10.17487/RFC5764, May 2010,              <http://www.rfc-editor.org/info/rfc5764>.Perkins, et al.        Expires September 18, 2016              [Page 42]Internet-Draft               RTP for WebRTC                   March 2016   [RFC6051]  Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP              Flows", RFC 6051, DOI 10.17487/RFC6051, November 2010,              <http://www.rfc-editor.org/info/rfc6051>.   [RFC6464]  Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time              Transport Protocol (RTP) Header Extension for Client-to-              Mixer Audio Level Indication", RFC 6464,              DOI 10.17487/RFC6464, December 2011,              <http://www.rfc-editor.org/info/rfc6464>.   [RFC6465]  Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-              time Transport Protocol (RTP) Header Extension for Mixer-              to-Client Audio Level Indication", RFC 6465,              DOI 10.17487/RFC6465, December 2011,              <http://www.rfc-editor.org/info/rfc6465>.   [RFC6562]  Perkins, C. and JM. Valin, "Guidelines for the Use of              Variable Bit Rate Audio with Secure RTP", RFC 6562,              DOI 10.17487/RFC6562, March 2012,              <http://www.rfc-editor.org/info/rfc6562>.   [RFC6904]  Lennox, J., "Encryption of Header Extensions in the Secure              Real-time Transport Protocol (SRTP)", RFC 6904,              DOI 10.17487/RFC6904, April 2013,              <http://www.rfc-editor.org/info/rfc6904>.   [RFC7007]  Terriberry, T., "Update to Remove DVI4 from the              Recommended Codecs for the RTP Profile for Audio and Video              Conferences with Minimal Control (RTP/AVP)", RFC 7007,              DOI 10.17487/RFC7007, August 2013,              <http://www.rfc-editor.org/info/rfc7007>.   [RFC7022]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,              "Guidelines for Choosing RTP Control Protocol (RTCP)              Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,              September 2013, <http://www.rfc-editor.org/info/rfc7022>.   [RFC7160]  Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple              Clock Rates in an RTP Session", RFC 7160,              DOI 10.17487/RFC7160, April 2014,              <http://www.rfc-editor.org/info/rfc7160>.   [RFC7164]  Gross, K. and R. Brandenburg, "RTP and Leap Seconds",              RFC 7164, DOI 10.17487/RFC7164, March 2014,              <http://www.rfc-editor.org/info/rfc7164>.Perkins, et al.        Expires September 18, 2016              [Page 43]Internet-Draft               RTP for WebRTC                   March 2016   [W3C.WD-mediacapture-streams-20130903]              Burnett, D., Bergkvist, A., Jennings, C., and A.              Narayanan, "Media Capture and Streams", World Wide Web              Consortium WD WD-mediacapture-streams-20130903, September              2013, <http://www.w3.org/TR/2013/              WD-mediacapture-streams-20130903>.   [W3C.WD-webrtc-20130910]              Bergkvist, A., Burnett, D., Jennings, C., and A.              Narayanan, "WebRTC 1.0: Real-time Communication Between              Browsers", World Wide Web Consortium WD WD-webrtc-              20130910, September 2013,              <http://www.w3.org/TR/2013/WD-webrtc-20130910>.16.2.  Informative References   [I-D.ietf-avtcore-multiplex-guidelines]              Westerlund, M., Perkins, C., and H. Alvestrand,              "Guidelines for using the Multiplexing Features of RTP to              Support Multiple Media Streams", draft-ietf-avtcore-              multiplex-guidelines-03 (work in progress), October 2014.   [I-D.ietf-avtext-rtp-grouping-taxonomy]              Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and              B. Burman, "A Taxonomy of Semantics and Mechanisms for              Real-Time Transport Protocol (RTP) Sources", draft-ietf-              avtext-rtp-grouping-taxonomy-08 (work in progress), July              2015.   [I-D.ietf-dart-dscp-rtp]              Black, D. and P. Jones, "Differentiated Services              (DiffServ) and Real-time Communication", draft-ietf-dart-              dscp-rtp-10 (work in progress), November 2014.   [I-D.ietf-mmusic-msid]              Alvestrand, H., "WebRTC MediaStream Identification in the              Session Description Protocol", draft-ietf-mmusic-msid-11              (work in progress), October 2015.   [I-D.ietf-payload-rtp-howto]              Westerlund, M., "How to Write an RTP Payload Format",              draft-ietf-payload-rtp-howto-14 (work in progress), May              2015.   [I-D.ietf-rmcat-cc-requirements]              Jesup, R. and Z. Sarker, "Congestion Control Requirements              for Interactive Real-Time Media", draft-ietf-rmcat-cc-              requirements-09 (work in progress), December 2014.Perkins, et al.        Expires September 18, 2016              [Page 44]Internet-Draft               RTP for WebRTC                   March 2016   [I-D.ietf-rtcweb-jsep]              Uberti, J., Jennings, C., and E. Rescorla, "Javascript              Session Establishment Protocol", draft-ietf-rtcweb-jsep-13              (work in progress), March 2016.   [I-D.ietf-tsvwg-rtcweb-qos]              Jones, P., Dhesikan, S., Jennings, C., and D. Druta, "DSCP              and other packet markings for WebRTC QoS", draft-ietf-              tsvwg-rtcweb-qos-14 (work in progress), March 2016.   [RFC3611]  Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,              "RTP Control Protocol Extended Reports (RTCP XR)",              RFC 3611, DOI 10.17487/RFC3611, November 2003,              <http://www.rfc-editor.org/info/rfc3611>.   [RFC4383]  Baugher, M. and E. Carrara, "The Use of Timed Efficient              Stream Loss-Tolerant Authentication (TESLA) in the Secure              Real-time Transport Protocol (SRTP)", RFC 4383,              DOI 10.17487/RFC4383, February 2006,              <http://www.rfc-editor.org/info/rfc4383>.   [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>.   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific              Media Attributes in the Session Description Protocol              (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,              <http://www.rfc-editor.org/info/rfc5576>.   [RFC5968]  Ott, J. and C. Perkins, "Guidelines for Extending the RTP              Control Protocol (RTCP)", RFC 5968, DOI 10.17487/RFC5968,              September 2010, <http://www.rfc-editor.org/info/rfc5968>.   [RFC6263]  Marjou, X. and A. Sollaud, "Application Mechanism for              Keeping Alive the NAT Mappings Associated with RTP / RTP              Control Protocol (RTCP) Flows", RFC 6263,              DOI 10.17487/RFC6263, June 2011,              <http://www.rfc-editor.org/info/rfc6263>.   [RFC6792]  Wu, Q., Ed., Hunt, G., and P. Arden, "Guidelines for Use              of the RTP Monitoring Framework", RFC 6792,              DOI 10.17487/RFC6792, November 2012,              <http://www.rfc-editor.org/info/rfc6792>.Perkins, et al.        Expires September 18, 2016              [Page 45]Internet-Draft               RTP for WebRTC                   March 2016   [RFC7478]  Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-              Time Communication Use Cases and Requirements", RFC 7478,              DOI 10.17487/RFC7478, March 2015,              <http://www.rfc-editor.org/info/rfc7478>.Authors' Addresses   Colin Perkins   University of Glasgow   School of Computing Science   Glasgow  G12 8QQ   United Kingdom   Email: csp@csperkins.org   URI:   https://csperkins.org/   Magnus Westerlund   Ericsson   Farogatan 6   SE-164 80 Kista   Sweden   Phone: +46 10 714 82 87   Email: magnus.westerlund@ericsson.com   Joerg Ott   Aalto University   School of Electrical Engineering   Espoo  02150   Finland   Email: jorg.ott@aalto.fiPerkins, et al.        Expires September 18, 2016              [Page 46]