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INFORMATIONAL
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Network Working Group                                     E. Burger, Ed.Request for Comments: 4240                                   J. Van DykeCategory: Informational                                       A. Spitzer                                             Brooktrout Technology, Inc.                                                           December 2005Basic Network Media Services with SIPStatus of This Memo   This memo provides information for the Internet community.  It does   not specify an Internet standard of any kind.  Distribution of this   memo is unlimited.Copyright Notice   Copyright (C) The Internet Society (2005).Abstract   In SIP-based networks, there is a need to provide basic network media   services.  Such services include network announcements, user   interaction, and conferencing services.  These services are basic   building blocks, from which one can construct interesting   applications.  In order to have interoperability between servers   offering these building blocks (also known as Media Servers) and   application developers, one needs to be able to locate and invoke   such services in a well defined manner.   This document describes a mechanism for providing an interoperable   interface between Application Servers, which provide application   services to SIP-based networks, and Media Servers, which provide the   basic media processing building blocks.Burger, et al.               Informational                      [Page 1]

RFC 4240                   SIP Media Services              December 2005Table of Contents1. Overview ........................................................21.1. Conventions Used in This Document ..........................32. Mechanism .......................................................33. Announcement Service ............................................53.1. Operation ..................................................83.2. Protocol Diagram ...........................................93.3. Formal Syntax ..............................................94. Prompt and Collect Service .....................................114.1. Formal Syntax for Prompt and Collect Service ..............125. Conference Service .............................................135.1. Protocol Diagram ..........................................145.2. Formal Syntax .............................................166. IANA Considerations ............................................177. The User Part ..................................................178. Security Considerations ........................................209. Contributors ...................................................2010. Acknowledgements ..............................................2011. References ....................................................2111.1. Normative References .....................................2111.2. Informative References ...................................221.  Overview   In SIP-based media networks (RFC 3261 [10]), there is a need to   provide basic network media services.  Such services include playing   announcements, initiating a media mixing session (conference), and   prompting and collecting information with a user.   These services are basic in nature, are few in number, and   fundamentally have not changed in 25 years of enhanced telephony   services.  Moreover, given their elemental nature, one would not   expect them to change in the future.   Multifunction media servers provide network media services to clients   using server protocols such as SIP, often in conjunction with markup   languages such as VoiceXML [20] and MSCML [21].  This document   describes how to identify to a multifunction media server what sort   of session the client is requesting, without modifying the SIP   protocol.   It is critically important to note that the mechanism described here   in no way modifies the SIP protocol, the meaning, or definition of a   SIP Request URI, or does it put any restrictions, in any way, on   devices that do not implement this convention.Burger, et al.               Informational                      [Page 2]

RFC 4240                   SIP Media Services              December 2005   Announcements are media played to the user.  Announcements can be   static media files, media files generated in real-time, media streams   generated in real-time, multimedia objects, or combinations of the   above.   Media mixing is the act of mixing different RTP streams, as described   inRFC 3550 [13].  Note that the service described here suffices for   simple mixing of media for a basic conferencing service.  This   service does not address enhanced conferencing services, such as   floor control, gain control, muting, subconferences, etc.  MSCML [21]   addresses enhanced conferencing.  However, that is beyond the scope   of this document.  Interested readers should read conferencing-   framework [22] for details on the IETF SIP conferencing framework.   Prompt and collect is where the server prompts the user for some   information, as in an announcement, and then collects the user's   response.  This can be a one-step interaction, for example by playing   an announcement, "Please enter your pass code", followed by   collecting a string of digits.  It can also be a more complex   interaction, specified, for example, by VoiceXML [20] or MSCML [21].1.1.  Conventions Used in This DocumentRFC 2119 [6] the interpretations for the key words "MUST", "MUST   NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT",   "RECOMMENDED", "MAY", and "OPTIONAL" found in this document.2.  Mechanism   In the context of SIP control of media servers, we take advantage of   the fact that the standard SIP URI has a user part.  Multifunction   media servers do not have users.  Thus we use the user address, or   the left-hand-side of the URI, as a service indicator.   The use of the user part of the SIP Request URI has a number of   useful properties:   o  There is no change to core SIP.   o  Only devices that choose to conform to this standard have to      implement it.   o  This document only applies to multifunction SIP-controlled media      servers.   o  This document has no impact on non-multifunction SIP-controlled      media servers.   o  The mechanism described in this document has absolutely no impact      on SIP devices other than media servers.Burger, et al.               Informational                      [Page 3]

RFC 4240                   SIP Media Services              December 2005   The last bullet point is crucial.  In particular, the user part   convention described here places absolutely no restrictions on any   SIP user agent, proxy, back-to-back user agent (B2BUA), or any future   device.  The user parts defined here only apply to multifunction   media servers that chose to implement the convention.  With the   exception of a conforming media server, these user names and   conventions have no impact on the user part namespace.  They do not   restrict the use of these user names at devices other than a   multifunction media server.   Note that the set of services is small, well defined, and well   contained.  The section The User Part (Section 7) discusses the   issues with using a fixed set of user-space names.   For per-service security, the media server SHOULD use the security   protocols described inRFC 3261 [10].   The media server MAY issue 401 challenges for authentication.  The   media server SHOULD support the sips: scheme for the announcement   service.  The media server MUST support the sips: scheme for the   dialog and conference services.  The level of authentication to   require for each service is a matter of local policy.   The media server, upon receiving an INVITE, notes the service   indicator.  Depending on the service indicator, the media server will   either honor the request or return a failure response code.   The service indicator is the concatenation of the service name and an   optional service instance identifier, separated by an equal sign.   PerRFC 3261 [10], the service indicator is case insensitive.  The   service name MUST be from the set alphanumeric characters plus dash   (US-ASCII %2C).  The service name MUST NOT include an equal sign   (US-ASCII %3D).   The service name MAY have long- and short-forms, as SIP does for   headers.   A given service indicator MAY have an associated set of parameters.   Such parameters MUST follow the convention set out for SIP URI   parameters.  That is, a semi-colon separated list of keyword=value   pairs.   Certain services may have an association with a unique service   instance on the media server.  For example, a given media server can   host multiple, separate conference sessions.  To identify unique   service instances, a unique identifier modifies the service name.Burger, et al.               Informational                      [Page 4]

RFC 4240                   SIP Media Services              December 2005   The unique identifier MUST meet the rules for a legal user part of a   SIP URI.  An equal sign, US-ASCII %3D, MUST separate the service   indicator from the unique identifier.   Note that since the service indicator is case insensitive, the   service instance identifier is also case insensitive.   The requesting client issues a SIP INVITE to the media server,   specifying the requested service and any appropriate parameters.   If the media server can perform the requested service, it does so,   following the processing steps described in the service definition   document.   If the media server cannot perform the requested service or does not   recognize the service indicator, it MUST respond with the response   code 488 NOT ACCEPTABLE HERE.  This is appropriate, as 488 refers to   a problem with the user part of the URI.  Moreover, 606 is not   appropriate, as some other media server may be able to satisfy the   request.RFC 3261 [10] describes the 488 and 606 response codes.   Some services require a unique identifier.  Most services   automatically create a service instance upon the first INVITE with   the given identifier.  However, if a service requires an existing   service instance, and no such service instance exists on the media   server, the media server MUST respond with the response code 404 NOT   FOUND.  This is appropriate as the service itself exists on the media   server, but the particular service instance does not.  It is as if   the user was not home.3.  Announcement Service   A network announcement is the delivery of a multimedia resource, such   as a prompt file, to a terminal device.  Note the multimedia resource   may be any multimedia object that the media server supports.  This   service can play a single object with multiple streams, such as a   video and audio prompt.  However, this service cannot play multiple   objects on the same SIP dialog.   There are two types of network announcements.  The differentiating   characteristic between the two types is whether the network fully   sets up the SIP dialog before playing the announcement.  The analog   in the Public Switched Telephone Network (PSTN) is whether answer   supervision is supplied (i.e., does the announcement server answer   the call prior to delivering the announcement?).Burger, et al.               Informational                      [Page 5]

RFC 4240                   SIP Media Services              December 2005   Playing an announcement after call setup is straightforward.  First,   the requesting device issues an INVITE to the media server requesting   the announcement service.  The media server negotiates the SDP and   responds with a 200 OK.  After receiving the ACK from the requesting   device, the media server plays the requested object and issues a BYE   to the requesting device.   If the media server supports announcements, but it cannot find the   referenced URI, it MUST respond with the 404 response code and SHOULD   send the reason phrase "Announcement content not found".   If the media server receives an INVITE for the announcement service   without a "play=" parameter, it MUST respond with the response code   400 and SHOULD send the reason phrase "Mandatory play parameter   missing".   If there is an error retrieving the announcement, the media server   MUST respond with a 400 response code and SHOULD send the reason   phrase "Announcement content could not be retrieved".  In addition   the media  server SHOULD include a Warning header with appropriate   explanatory text explaining what failed.   The Request URI fully describes the announcement service through the   use of the user part of the address and additional URI parameters.   The user portion of the address, "annc", specifies the announcement   service on the media server.  The service has several associated URI   parameters that control the content and delivery of the announcement.   These parameters are described below:   play      Specifies the resource or announcement sequence to be played.   repeat      Specifies how many times the media server should repeat the      announcement or sequence named by the "play=" parameter.  The      value "forever" means the repeat should be effectively unbounded.      In this case, it is RECOMMENDED the media server implements some      local policy, such as limiting what "forever" means, to ensure      errant clients do not create a denial of service attack.   delay      Specifies a delay interval between announcement repetitions.  The      delay is measured in milliseconds.   duration      Specifies the maximum duration of the announcement.  The media      server will discontinue the announcement and end the call if theBurger, et al.               Informational                      [Page 6]

RFC 4240                   SIP Media Services              December 2005      maximum duration has been reached.  The duration is measured in      milliseconds.   locale      Specifies the language and optionally country variant of the      announcement sequence named in the "play=" parameter.RFC 3066      [9] specifies the locale tag.  The locale tag is usually a two- or      three-letter code per ISO 639-1 [11].  The country variant is also      often a two-letter code per ISO 3166-1 [12].  These elements are      concatenated with a single under bar (%x5F) character, such as      "en_CA".  If only the language is specified, such as locale=en,      the choice of country variant is an implementation matter.      Implementations SHOULD provide the best possible match between the      requested locale and the available languages in the event the      media server cannot honor the locale request precisely.  For      example, if the request has locale=ca_FR, but the media server      only has fr_FR available, the media server should use the fr_FR      variant.  Implementations SHOULD provide a default locale to use      if no language variants are available.   param[n]      Provides a mechanism for passing values that are to be substituted      into an announcement sequence.  Up to 9 parameters ("param1="      through "param9=") may be specified.  The mechanics of      announcement sequences are beyond the scope of this document.   extension      Provides a mechanism for extending the parameter set.  If the      media server receives an extension it does not understand, it MUST      silently ignore the extension parameter and value.   The "play=" parameter is mandatory and MUST be present.  All other   parameters are OPTIONAL.   NOTE: Some encodings are not self-describing.  Thus, the   implementation relies on filename extension conventions for   determining the media type.   Note thatRFC 3261 [10] implies that proxies are supposed to pass   parameters through unchanged.  However, be aware that non-conforming   proxies may strip Request-URI parameters.  That said, given the   likely scenarios for the mechanisms presented in this document, this   should not be an issue.  Most likely, the proxy inserting the   parameters is the last proxy before the media server.  If the service   provider deploys a proxy for load balancing or service location   purposes, the service provider should ensure that its choice of proxy   preserves parameters.Burger, et al.               Informational                      [Page 7]

RFC 4240                   SIP Media Services              December 2005   The form of the SIP Request URI for announcements is as follows.   Note that the backslash, CRLF, and spacing before the "play=" in the   example is for readability purposes only.   sip:annc@ms2.example.net; \       play=http://audio.example.net/allcircuitsbusy.g711   sip:annc@ms2.example.net; \       play=file://fileserver.example.net//geminii/yourHoroscope.wav3.1.  Operation   The scenarios below assume there is a SIP Proxy, application server,   or media gateway controller between the caller and the media server.   However, the announcement service works as described below even if   the caller invokes the service directly.  We chose to discuss the   proxy case, as it will be the most common case.   The caller issues an INVITE to the serving SIP Proxy.  The SIP Proxy   determines what audio prompt to play to the caller.  The proxy   responds to the caller with 100 TRYING.   It is important to note that the mechanism described here in no way   modifies the behavior of SIP [10].  In particular, this convention   does not modify SDP negotiation [18].   The proxy issues an INVITE to the media server, requesting the   appropriate prompt to play coded in the play= parameter.  The media   server responds with 200 OK.  The proxy relays the 200 OK to the   caller.  The caller then issues an ACK.  The proxy then relays the   ACK to the media server.   With the call established, the media server plays the requested   prompt.  When the media server completes the play of the prompt, it   issues a BYE to the proxy.  The proxy then issues a BYE to the   caller.Burger, et al.               Informational                      [Page 8]

RFC 4240                   SIP Media Services              December 20053.2.  Protocol Diagram   Caller                   Proxy                 Media Server     |   INVITE               |                        |     |----------------------->|   INVITE               |     |   100 TRYING           |----------------------->|     |<-----------------------|   200 OK               |     |   200 OK               |<-----------------------|     |<-----------------------|                        |     |   ACK                  |                        |     |----------------------->|   ACK                  |     |                        |----------------------->|     |                        |                        |     |              Play Announcement (RTP)            |     |<================================================|     |                        |                        |     |                        |   BYE                  |     |   BYE                  |<-----------------------|     |<-----------------------|                        |     |   200 OK               |                        |     |----------------------->|    200 OK              |     |                        |----------------------->|     |                        |                        |3.3.  Formal Syntax   The following syntax specification uses the augmented Backus-Naur   Form (BNF) as described inRFC 4234 [7].   ANNC-URL        = sip-ind annc-ind "@" hostport                       annc-parameters uri-parameters   sip-ind         = "sip:" / "sips:"   annc-ind        = "annc"   annc-parameters = ";" play-param [ ";" content-param ]                                    [ ";" delay-param]                                    [ ";" duration-param ]                                    [ ";" repeat-param ]                                    [ ";" locale-param ]                                    [ ";" variable-params ]                                    [ ";" extension-params ]   play-param      = "play=" prompt-url   content-param   = "content-type=" MIME-type   delay-param     = "delay=" delay-valueBurger, et al.               Informational                      [Page 9]

RFC 4240                   SIP Media Services              December 2005   delay-value     = 1*DIGIT   duration-param  = "duration=" duration-value   duration-value  = 1*DIGIT   repeat-param    = "repeat=" repeat-value   repeat-value    = 1*DIGIT / "forever"   locale-param    = "locale=" token                        ; perRFC 3066, usually                        ; ISO639-1_ISO3166-1                        ; e.g., en, en_US, en_UK, etc.   variable-params = param-name "=" variable-value   param-name      = "param" DIGIT ; e.g., "param1"   variable-value  = 1*(ALPHA / DIGIT)   extension-params = extension-param [ ";" extension-params ]   extension-param  = token "=" token   "uri-parameters" is the SIP Request-URI parameter list as described   inRFC 3261 [10].  All parameters of the Request URI are part of the   URI matching algorithm.   The MIME-type is the MIME [1] [2] [3] [4] [5] content type for the   announcement, such as audio/basic, audio/G729, audio/mpeg,   video/mpeg, and so on.   A number of MIME registrations, which could be used here, have   parameters, for instance, video/DV.  To accommodate this, and retain   compatibility with the SIP URI structure, the MIME-type parameter   separator (semicolon, %3b) and value separator (equal, %d3) MUST be   escaped.  For example:   sip:annc@ms.example.net; \       play=file://fs.example.net//clips/my-intro.dvi; \       content-type=video/mpeg%3bencode%d3314M-25/625-50   The locale-value consists of a tag as specified inRFC 3066 [9].   The definition of hostport is as specified byRFC 3261 [10].Burger, et al.               Informational                     [Page 10]

RFC 4240                   SIP Media Services              December 2005   The syntax of prompt-url consists of a URL scheme as specified byRFC3986 [8] or a special token indicating a provisioned announcement   sequence.  For example, the URL scheme MAY include any of the   following.   o  http/https   o  ftp   o  file (referencing a local or NFS (RFC 3530 [16]) object)   o  nfs (RFC 2224 [14])   If a provisioned announcement sequence is to be played, the value of   prompt-url will have the following form:   prompt-url      = "/provisioned/" announcement-id   announcement-id = 1*(ALPHA / DIGIT)   Note that the scheme "/provisioned/" was chosen because of a   hesitation to register a "provisioned:" URI scheme.   This document is strictly focused on the SIP interface for the   announcement service and, as such, does not detail how announcement   sequences are provisioned or defined.   Note that the media type of the object the prompt-url refers to can   be most anything, including audio file formats, text file formats, or   URI lists.  See the Prompt and Collect Service (Section 4) section   for more on this topic.4.  Prompt and Collect Service   This service is also known as a voice dialog.  It establishes an   aural dialog with the user.   The dialog service follows the model of the announcement service.   However, the service indicator is "dialog".  The dialog service takes   a parameter, voicexml=, indicating the URI of the VoiceXML script to   execute.   sip:dialog@mediaserver.example.net; \       voicexml=http://vxmlserver.example.net/cgi-bin/script.vxml   A Media Server MAY accept additional SIP request URI parameters and   deliver them to the VoiceXML interpreter session as session   variables.   Although not good VoiceXML programming practice, VoiceXML scripts   might contain sensitive information, such as a user's pass code in aBurger, et al.               Informational                     [Page 11]

RFC 4240                   SIP Media Services              December 2005   DTMF grammar.  Thus, the media server MUST support the https scheme   for the voicexml parameter for secure fetching of scripts.  Likewise,   dynamic grammars often do have user-identifying information.  As   such, the VoiceXML browser implementation on the media server MUST   support https fetching of grammars and subsequent documents.   Returned information often is sensitive.  For example, the   information could be financial information or instructions.  Thus,   the media server MUST support https posting of results.4.1.  Formal Syntax for Prompt and Collect Service   The following syntax specification uses the augmented Backus-Naur   Form (BNF) as described inRFC 4234 [7].   DIALOG-URL        = sip-ind dialog-ind "@" hostport                          dialog-parameters   sip-ind           = "sip:" / "sips:"   dialog-ind        = "dialog"   dialog-parameters = ";" dialog-param [ vxml-parameters ]                                        [ uri-parameters ]   dialog-param      = "voicexml=" vxml-url   vxml-parameters   = vxml-param [ vxml-parameters ]   vxml-param        = ";" vxml-keyword "=" vxml-value   vxml-keyword      = token   vxml-value        = token   The vxml-url is the URI of the VoiceXML script.  If present, other   parameters get passed to the VoiceXML interpreter session with the   assigned vxml-keyword vxml-value pairs.  Note that all vxml-keywords   MUST have values.   If there is a vxml-keyword without a corresponding vxml-value, the   media server MUST reject the request with a 400 BAD REQUEST response   code.  In addition, the media server MUST state "Missing VXML Value"   in the reason phrase.   The media server presents the parameters as environment variables in   the connection object.  Specifically, the parameter appears in the   connection.sip tree.Burger, et al.               Informational                     [Page 12]

RFC 4240                   SIP Media Services              December 2005   If the Media Server does not support the passing of keyword-value   pairs to the VoiceXML interpreter session, it MUST ignore the   parameters.   "uri-parameters" is the SIP Request-URI parameter list as described   inRFC 3261 [10].  All parameters in the parameter list, whether they   come from uri-parameters or from vxml-keyworks, are part of the URI   matching algorithm.5.  Conference Service   One identifies mixing sessions through their SIP request URIs.  To   create a mixing session, one sends an INVITE to a request URI that   represents the session.  If the URI does not already exist on the   media server and the requested resources are available, the media   server creates a new mixing session.  If there is an existing URI for   the session, then the media server interprets it as a request for the   new session to join the existing session.  The form of the SIP   request URI for conferencing is:   sip:conf=uniqueIdentifier@mediaserver.example.net   The left-hand side of the request URI is actually the username of the   request in the request URI and the To header.  The host portion of   the URI identifies a particular media server.  The "conf" user name   conveys to the media server that this is a request for the mixing   service.  The uniqueIdentifier can be any value that is compliant   with the SIP URI specification.  It is the responsibility of the   conference control application to ensure the identifier is unique   within the scope of any potential conflict.   In the terminology of the conferencing framework [22], this URI   convention tells the media server that the application server is   requesting it to act as a Focus.  The conf-id value identifies the   particular focus instance.   As a focus in the conferencing framework, the media server MUST   support the ";isfocus" parameter in the Request URI.  Note, however,   that the presence or absence of the ";isfocus" parameter has no   protocol impact at the media server.   It is worth noting that the conference URI shared between the   application and media servers provides enhanced security, as the SIP   control interface does not have to be exposed to participants.  It   also allows the assignment of a specific media server to be delayed   as long as possible, thereby simplifying resource management.Burger, et al.               Informational                     [Page 13]

RFC 4240                   SIP Media Services              December 2005   One can add additional legs to the conference by INVITEing them to   the above-mentioned request URI.  Per the matching rules ofRFC 3261   [10], the conf-id parameter is part of the matching string.   Conversely, one can remove legs by issuing a BYE in the corresponding   dialog.  The mixing session, and thus the conference-specific request   URI, remains active so long as there is at least one SIP dialog   associated with the given request URI.   If the Request-URI has "conf" as the user part, but does not have a   conf-id parameter, the media server MUST respond with a 404 NOT   FOUND.      NOTE: The media server could create a unique conference instance      and return the conf-id string to the User Agent Clinet (UAC) if      there is no conf-id present.  However, such an operation may have      other operational issues, such as permissions and billing.  Thus      an application server or proxy is a better place to do such an      operation.  Moreover, such action would make the media server into      a Conference Factory in the terminology of conference-framework      [22].  That is not the appropriate behavior for a media server.   Since some conference use cases, such as business conferencing, have   billing implications, the media server SHOULD authenticate the   application server or proxy.  At a minimum, the media server MUST   implement sips:.5.1.  Protocol Diagram   This diagram shows the establishment of a three-way conference.  This   section is informative.  It is only one method of establishing a   conference.  This example shows a simple back-to-back user agent.   The conference-framework [22] describes additional parameters and   behaviors of the Application Server.  For example, the first INVITE   from P1 to the Application Server would include the ";isfocus"   parameter; the Application Server would act as a Conference Factory;   and so on.  However, none of that protocol machinery has an impact on   the operation of the Application Server to Media Server interface,   which is the focus of this protocol document.Burger, et al.               Informational                     [Page 14]

RFC 4240                   SIP Media Services              December 2005    P1       P2        P3         Application Server     Media Server     |       |        |                  |                   |     |  INVITE sip:public-conf@as.example.net                |     |---------------------------------->|                   |     |       |        |   INVITE sip:conf=123@ms.example.net |     |       |        |                  |------------------>|     |       |        |                  | 200 OK            |     |  200 OK        |                  |<------------------|     |<----------------------------------|                   |     |  ACK  |        |                  |                   |     |---------------------------------->| ACK               |     |       |        |                  |------------------>|     |       |        | RTP w/ P1        |                   |     |<=====================================================>|     |       |        |                  |                   |     |  INVITE sip:public-conf@as.example.net                |     |       |-------------------------->|                   |     |       |        |   INVITE sip:conf=123@ms.example.net |     |       |        |                  |------------------>|     |       |        |                  | 200 OK            |     |       | 200 OK |                  |<------------------|     |       |<--------------------------|                   |     |       |  ACK   |                  |                   |     |       |-------------------------->| ACK               |     |       |        |                  |------------------>|     |       |        |                  |                   |     |       |        | RTP w/ P1+P2-P2  |                   |     |       |<=============================================>|     |       |        | RTP w/ P1+P2-P1  |                   |     |<=====================================================>|     |       |        |                  |                   |     |  INVITE sip:public-conf@as.example.net                |     |       |        |----------------->|                   |     |       |        |   INVITE sip:conf=123@ms.example.net |     |       |        |                  |------------------>|     |       |        |                  | 200 OK            |     |       |        | 200 OK           |<------------------|     |       |        |<-----------------|                   |     |       |        |  ACK             |                   |     |       |        |----------------->| ACK               |     |       |        |                  |------------------>|     |       |        |                  |                   |     |       |        | RTP w/ P1+P2+P3-P3                   |     |       |        |<====================================>|     |       |        | RTP w/ P1+P2+P3-P2                   |     |       |<=============================================>|     |       |        | RTP w/ P1+P2+P3-P1                   |     |<=====================================================>|Burger, et al.               Informational                     [Page 15]

RFC 4240                   SIP Media Services              December 2005     |       |        |                  |                   |     |       |        |                  |                   |   Using the terminology of conference-framework [22], the Application   Server is the Conference Factory, and the Media Server is the   Conference Focus.   Note that the above call flow does not show any 100 TRYING messages   that would typically flow from the Application Server to the UACs;   nor does it show the ACKs from the UACs to the Application Server or   from the Application Server to the Media Server.   Each leg can drop out either under the supervision of the UAC, by the   UAC sending a BYE, or under the supervision of the Application   Server, by the Application Server issuing a BYE.  In either case, the   Application Server will either issue a BYE on behalf of the UAC or   issue it directly to the Media Server, corresponding to the   respective disconnect case.   It is left as a trivial exercise to the reader for how the   Application Server can mute legs, create side conferences, and so   forth.   Note that the Application Server is a server to the participants   (UACs).  However, the Application Server is a client for mixing   services to the Media Server.5.2.  Formal Syntax   The following syntax specification uses the augmented Backus-Naur   Form (BNF) as described inRFC 4234 [7].   CONF-URL        = sip-ind conf-ind "=" instance-id "@" hostport                     [ uri-parameters ]   sip-ind         = "sip:" / "sips:"   conf-ind        = "conf"   instance-id     = token   "uri-parameters" is the SIP Request-URI parameter list as described   inRFC 3261 [10].  All parameters in the parameter list are part of   the URI matching algorithm.Burger, et al.               Informational                     [Page 16]

RFC 4240                   SIP Media Services              December 20056.  IANA Considerations   The IANA has registered the following parameters in the SIP/SIPS URI   Parameters registry, following the specification required policy ofRFC 3969 [19]:   Parameter Name    Predefined Values    Reference   --------------    -----------------    ---------   play                      noRFC 4240   repeat                    noRFC 4240   delay                     noRFC 4240   duration                  noRFC 4240   locale                    noRFC 4240   param[n]                  noRFC 4240   extension                 noRFC 42407.  The User Part   There has been considerable discussion about the wisdom of using   fixed user parts in a request URI.  The most common objection is that   the user part should be opaque and a local matter.  The other   objection is that using a fixed user part removes those specified   user addresses from the user address space.   We address the latter issue first.  The common example is the   Postmaster address defined byRFC 2821 [15].  The objection is that   by using the Postmaster token for something special, one removes that   token for anyone.  Thus, the Postmaster General of the United States,   for example, cannot have the mail address Postmaster@usps.gov.   However, one may debate whether this is a significant limitation.   This document explicitly addresses this issue.  The user names   described in the text (namely annc, ivr, dialog, and conf) are   available for whatever local use a given SIP user agent or proxy   wishes for them.  What this document does is give special meaning for   these user names at media servers that implement this specification.   If a media server chooses not to implement this specification,   nothing breaks.  If a user wishes to use one of the user names   described in this document at their SIP user agent, nothing breaks   and their user agent will work as expected.   The key point is, one cannot confuse the namespace at a Media Server   with the namespace for an organization.  For example, let us take the   case where a network offers services for "Ann Charles".  She likes to   use the name "annc", and thus she would like to use   "sip:annc@example.net".  We offer there is ABSOLUTELY NO NAME   COLLISION WHATSOEVER.  Why is this so?  This is so because   sip:annc@example.net will resolve to the specific user at a specificBurger, et al.               Informational                     [Page 17]

RFC 4240                   SIP Media Services              December 2005   device for Ann.  As an example, example.net's SIP Proxy Server   resolves sip:annc@example.net to annc@anns-phone.example.net.   Conversely, one directs requests for the media service annc directly   to the Media Server, e.g., sip:annc@ms21.ap.example.net.  Moreover,   by definition, requests for Ann Charles, or anything other than the   announcement service, will NEVER be directly sent to the Media   Server.  If that were not true, no phone in the world could use the   user part "eburger", as eburger is a reserved user part in the   Brooktrout domain.  Clearly, this is not the case.   If one wishes to make their media server accessible to the global   Internet, but retain one of the Media Server-specific user names in   the domain, a SIP Proxy can easily translate whatever opaque name one   chooses to the Media Server-specific user name.  For example, if a   domain wishes to offer services for the above mentioned Ann Charles   at sip:annc@example.com, they can offer the announcement service at   sip:my-special-announcement-service@example.com.  The former address,   sip:annc@example.com, would resolve to the actual device where annc   resides.  The latter would resolve to the media server announcement   server address, sip:annc@mediaserver.example.com, as an example.   Note that this convention makes it easier to provision this service.   With a fixed mapping at the multifunction media server, there are   less provisioning data elements to get wrong.   Here is another way of looking at this issue.  Unix reserves the   special user "root".  Just about all Unix machines have a user root,   who has an address "root@a-specific-machine.example.com", where   "a-specific-machine" is the fully-qualified domain name (FQDN) of a   particular instance of a machine.  There are very well-defined   semantics for the "root" user.   Even though most every Unix machine has a "root" user, often there is   no mapping for a "root" user in a domain, such as "root@example.com".   Conversely, there is no restriction on creating an MX record for   "root@example.com".  That choice is fully up to the administrative   authority for the domain.   The "users" proposed by this document, "annc", "conf", and "dialog"   are all users at a Media Server, just as the "root", "bin", and   "nobody" users are "users" at a Unix host.   After much discussion, with input from the W3C URI work group, we   considered obfuscating the user name by prepending "__sip-" to the   user name.  However, as explained above, this obfuscation is not   necessary.  There is a fundamental difference between a user name at   a device and a user name at an MX record (SMTP) or Address-of-Record   (SIP).  Again, there is no possibility that the name on the device   may "leak out" into the SIP routing network.Burger, et al.               Informational                     [Page 18]

RFC 4240                   SIP Media Services              December 2005   The most important thing to note about this convention is that the   left-hand side of the request URI is opaque to the network.  The only   network elements that need to know about the convention are the Media   Server and client.  Even proxies doing mapping resolution, as in the   example above for public announcement services, do not need to be   aware of the convention.  The convention is purely a matter of   provisioning.   Some have proposed that such naming be a pure matter of local   convention.  For example, the thesis of the informational RFCRFC3087 [17] is that you can address services using a request URI.   However, some have taken the examples in the document to an extreme.   Namely, that the only way to address services is via arbitrary,   opaque, long user parts.  Clearly, it is possible to provision the   service names, rather than fixed names.  While this can work in a   closed network, where the Application Servers and Media Servers are   in the same administrative domain, this does not work across domains,   such as in the Internet.  This is because the client of the media   service has to know the local name for each service / domain pair.   This is particularly onerous for situations where there is an ad hoc   relationship between the application and the media service.  Without   a well-known relationship between service and service address, how   would the client locate the service?   One very important result of using the user part as the service   descriptor is that we can use all of the standard SIP machinery,   without modification.  For example, Media Servers with different   capabilities can SIP Register their capabilities as users.  For   example, a VoiceXML-only device will register the "dialog" user,   while a multi-purpose Media Server will register all of the users.   Note that this is why the URI to play is a parameter.  Doing   otherwise would overburden a normal SIP proxy or redirect server.   Conversely, having the conference ID be part of the user part gives   an indication that requests get routed similarly (as opposed to   requiring a Globally Routable User Agent URI (GRUU), which would   restrict routing to the same device).   Likewise, this scheme lets us leverage the standard SIP proxy   behavior of using an intelligent redirect server or proxy server to   provide high-available services.  For example, two Media Servers can   register with a SIP redirect server for the annc user.  If one of the   Media Servers fails, the registration will expire and all requests   for the announcement service ("calls to the annc user") will get sent   to the surviving Media Server.Burger, et al.               Informational                     [Page 19]

RFC 4240                   SIP Media Services              December 20058.  Security Considerations   Exposing network services with well-known addresses may not be   desirable.  The Media Server SHOULD authenticate and authorize   requesting endpoints per local policy.   Some interactions in this document result in the transfer of   confidential information.  Moreover, many of the interactions require   integrity protection.  Thus, the Media Server MUST implement the   sips: scheme.  In addition, application developers are RECOMMENDED to   use the security services offered by the Media Server to ensure the   integrity and confidentiality of their user's data, as appropriate.   Untrusted network elements could use the convention described here   for providing information services.  Many extant billing arrangements   are for completed calls.  Successful call completion occurs with a   2xx result code.  This can be an issue for the early media   announcement service.  This is one of the reasons why the early media   announcement service is deprecated.   Services such as repeating an announcement forever create the   possibility for denial of service attacks.  The media server SHOULD   have local policies to deal with this, such as time-limiting how long   "forever" is, analyzing where multiple requests come from,   implementing white-lists for such a service, and so on.9.  Contributors   Jeff Van Dyke and Andy Spitzer of SnowShore did just about all of the   work developing netann, in conjunction with many application   developers, media server manufacturers, and service providers, some   of whom are listed in the Acknowledgements section.  All I did was do   the theory and write it up.  That also means all of the mistakes are   mine, as well.10.  Acknowledgements   We would like to thank Kevin Summers and Ravindra Kabre of Sonus   Networks for their constructive comments, as well as Jonathan   Rosenberg of Dynamicsoft and Tim Melanchuk of Convedia for their   encouragement.  In addition, the discussion at the Las Vegas Interim   Workgroup Meeting in 2002 was invaluable for clearing up the issues   surrounding the left-hand-side of the request URI.  Christer Holmberg   helped tune the language of the multimedia announcement service.   Orit Levin from Radvision gave a close read on the most recent   version of the document.  Pete Danielsen from Lucent has consistently   provided excellent reviews of the many different versions of this   document.Burger, et al.               Informational                     [Page 20]

RFC 4240                   SIP Media Services              December 2005   Pascal Jalet provided the theoretical underpinning and David Rio   provided the experimental evidence for why the conference identifier   belongs in the user part of the request-URI.   I am particularly indebted to Alan Johnston for his review of this   document and ensuring its conformance with the SIP conference control   work in the IETF.   Mary Barnes, as usual, found the holes and showed how to fix them.   The authors would like to give a special thanks to Walter O'Connor   for doing much of the initial implementation.   Note that at the time of this writing, there are 7 known independent   server implementations that are interoperable with 23 known client   implementations.  Our apologies if we did not count your   implementation.11.  References11.1.  Normative References   [1]   Freed, N. and N. Borenstein, "Multipurpose Internet Mail         Extensions (MIME) Part One: Format of Internet Message Bodies",RFC 2045, November 1996.   [2]   Freed, N. and N. Borenstein, "Multipurpose Internet Mail         Extensions (MIME) Part Two: Media Types",RFC 2046,         November 1996.   [3]   Moore, K., "MIME (Multipurpose Internet Mail Extensions) Part         Three: Message Header Extensions for Non-ASCII Text",RFC 2047,         November 1996.   [4]   Freed, N., Klensin, J., and J. Postel, "Multipurpose Internet         Mail Extensions (MIME) Part Four: Registration Procedures",BCP 13,RFC 2048, November 1996.   [5]   Freed, N. and N. Borenstein, "Multipurpose Internet Mail         Extensions (MIME) Part Five: Conformance Criteria and         Examples",RFC 2049, November 1996.   [6]   Bradner, S., "Key words for use in RFCs to Indicate Requirement         Levels",BCP 14,RFC 2119, March 1997.   [7]   Crocker, D. and P. Overell, "Augmented BNF for Syntax         Specifications: ABNF",RFC 4234, October 2005.Burger, et al.               Informational                     [Page 21]

RFC 4240                   SIP Media Services              December 2005   [8]   Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform         Resource Identifier (URI): Generic Syntax", STD 66,RFC 3986,         January 2005.   [9]   Alvestrand, H., "Tags for the Identification of Languages",BCP 47,RFC 3066, January 2001.   [10]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,         Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:         Session Initiation Protocol",RFC 3261, June 2002.   [11]  International Organization for Standardization, "Codes for the         representation of names of languages -- Part 1: Alpha-2 code",         ISO Standard 639-1, July 2002.   [12]  International Organization for Standardization, "Codes for the         representation of names of countries and their subdivisions --         Part 1: Country codes", ISO Standard 3166-1, October 1997.11.2.  Informative References   [13]  Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,         "RTP: A Transport Protocol for Real-Time Applications", STD 64,RFC 3550, July 2003.   [14]  Callaghan, B., "NFS URL Scheme",RFC 2224, October 1997.   [15]  Klensin, J., "Simple Mail Transfer Protocol",RFC 2821,         April 2001.   [16]  Shepler, S., Callaghan, B., Robinson, D., Thurlow, R., Beame,         C., Eisler, M., and D. Noveck, "Network File System (NFS)         version 4 Protocol",RFC 3530, April 2003.   [17]  Campbell, B. and R. Sparks, "Control of Service Context using         SIP Request-URI",RFC 3087, April 2001.   [18]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with         Session Description Protocol (SDP)",RFC 3264, June 2002.   [19]  Camarillo, G., "The Internet Assigned Number Authority (IANA)         Uniform Resource Identifier (URI) Parameter Registry for the         Session Initiation Protocol (SIP)",BCP 99,RFC 3969,         December 2004.Burger, et al.               Informational                     [Page 22]

RFC 4240                   SIP Media Services              December 2005   [20]  Burnett, D., Hunt, A., McGlashan, S., Porter, B., Lucas, B.,         Ferrans, J., Rehor, K., Carter, J., Danielsen, P., and S.         Tryphonas, "Voice Extensible Markup Language (VoiceXML) Version         2.0", W3C REC REC-voicexml20-20040316, March 2004.   [21]  Van Dyke, J., Burger, E., Ed., and A. Spitzer, "Media Server         Control Markup Language (MSCML) and Protocol", Work in         Progress, December 2004.   [22]  Rosenberg, J., "A Framework for Conferencing with the Session         Initiation Protocol", Work in Progress, October 2004.Authors' Addresses   Eric Burger   Brooktrout Technology, Inc.   18 Keewaydin Dr.   Salem, NH  03079   USA   EMail: eburger@brooktrout.com   Jeff Van Dyke   Brooktrout Technology, Inc.   18 Keewaydin Dr.   Salem, NH  03079   USA   EMail: jvandyke@brooktrout.com   Andy Spitzer   Brooktrout Technology, Inc.   18 Keewaydin Dr.   Salem, NH  03079   USA   EMail: woof@brooktrout.comBurger, et al.               Informational                     [Page 23]

RFC 4240                   SIP Media Services              December 2005Full Copyright Statement   Copyright (C) The Internet Society (2005).   This document is subject to the rights, licenses and restrictions   contained inBCP 78, and except as set forth therein, the authors   retain all their rights.   This document and the information contained herein are provided on an   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS   OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET   ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,   INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE   INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.Intellectual Property   The IETF takes no position regarding the validity or scope of any   Intellectual Property Rights or other rights that might be claimed to   pertain to the implementation or use of the technology described in   this document or the extent to which any license under such rights   might or might not be available; nor does it represent that it has   made any independent effort to identify any such rights.  Information   on the procedures with respect to rights in RFC documents can be   found inBCP 78 andBCP 79.   Copies of IPR disclosures made to the IETF Secretariat and any   assurances of licenses to be made available, or the result of an   attempt made to obtain a general license or permission for the use of   such proprietary rights by implementers or users of this   specification can be obtained from the IETF on-line IPR repository athttp://www.ietf.org/ipr.   The IETF invites any interested party to bring to its attention any   copyrights, patents or patent applications, or other proprietary   rights that may cover technology that may be required to implement   this standard.  Please address the information to the IETF at ietf-   ipr@ietf.org.Acknowledgement   Funding for the RFC Editor function is currently provided by the   Internet Society.Burger, et al.               Informational                     [Page 24]

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