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Framework for Transcoding with the Session Initiation Protocol (SIP)

The information below is for an old version of the document that is already published as an RFC.
Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 5369.
Author Gonzalo Camarillo
Last updated 2015-10-14 (Latest revision 2006-12-01)
RFC stream Internet Engineering Task Force (IETF)
Intended RFC status Informational
Additional resources Mailing list discussion
Stream WG state (None)
Document shepherd (None)
IESG IESG state Became RFC 5369 (Informational)
Action Holders
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Telechat date (None)
Responsible AD Jon Peterson
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SIPPING Working Group                                       G. Camarillo
Internet-Draft                                                  Ericsson
Expires: June 3, 2007                                  November 30, 2006

  Framework for Transcoding with the Session Initiation Protocol (SIP)

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Copyright Notice

   Copyright (C) The Internet Society (2006).


   This document defines a framework for transcoding with SIP.  This
   framework includes how to discover the need for transcoding services
   in a session and how to invoke those transcoding services.  Two
   models for transcoding services invocation are discussed: the
   conference bridge model and the third party call control model.  Both
   models meet the requirements for SIP regarding transcoding services
   invocation to support deaf, hard of hearing, and speech-impaired

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Discovery of the Need for Transcoding Services . . . . . . . .  3
   3.  Transcoding Services Invocation  . . . . . . . . . . . . . . .  4
     3.1.  Third Party Call Control Transcoding Model . . . . . . . .  5
     3.2.  Conference Bridge Transcoding Model  . . . . . . . . . . .  6
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . .  8
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  8
   6.  Contributors . . . . . . . . . . . . . . . . . . . . . . . . .  8
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . .  9
     7.1.  Normative References . . . . . . . . . . . . . . . . . . .  9
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 10
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 11
   Intellectual Property and Copyright Statements . . . . . . . . . . 12

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1.  Introduction

   Two user agents involved in a SIP [3] dialog may find it impossible
   to establish a media session due to a variety of incompatibilities.
   Assuming that both user agents understand the same session
   description format (e.g., SDP [12]), incompatibilities can be found
   at the user agent level and at the user level.  At the user agent
   level, both terminals may not support any common codec or may not
   support common media types (e.g., a text-only terminal and an audio-
   only terminal).  At the user level, a deaf person will not understand
   anything said over an audio stream.

   In order to make communications possible in the presence of
   incompatibilities, user agents need to introduce intermediaries that
   provide transcoding services to a session.  From the SIP point of
   view, the introduction of a transcoder is done in the same way to
   resolve both user level and user agent level incompatibilities.  So,
   the invocation mechanisms described in this document are generally
   applicable to any type of incompatibility related to how the
   information that needs to be communicated is encoded.

      Furthermore, although this framework focuses on transcoding, the
      mechanisms described are applicable to media manipulation in
      general.  It would be possible to use them, for example, to invoke
      a server that simply increased the volume of an audio stream.

   This document does not describe media server discovery.  That is an
   orthogonal problem that one can address using user agent provisioning
   or other methods.

   The remainder of this document is organized as follows.  Section 2
   deals with the discovery of the need for transcoding services for a
   particular session.  Section 3 introduces the third party call
   control and conference bridge transcoding invocation models, which
   are further described in Section 3.1 and Section 3.2 respectively.
   Both models meet the requirements regarding transcoding services
   invocation in RFC3351 [6] to support deaf, hard of hearing, and
   speech-impaired individuals.

2.  Discovery of the Need for Transcoding Services

   According to the one-party consent model defined in RFC 3238 [2],
   services that involve media manipulation invocation are best invoked
   by one of the end-points involved in the communication, as opposed to
   being invoked by an intermediary in the network.  Following this
   principle, one of the end-points should be the one detecting that
   transcoding is needed for a particular session.

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   In order to decide whether or not transcoding is needed, a user agent
   needs to know the capabilities of the remote user agent.  A user
   agent acting as an offerer [4] typically obtains this knowledge by
   downloading a presence document that includes media capabilities
   (e.g., Bob is available on a terminal that only supports audio) or by
   getting an SDP description of media capabilities as defined in RFC
   3264 [4].

   Presence documents are typically received in a NOTIFY [5] request as
   a result of a subscription.  SDP media capabilities descriptions are
   typically received in a 200 (OK) response to an OPTIONS request or in
   a 488 (Not Acceptable Here) response to an INVITE.

   In the absence of presence information, routing logic that involves
   parallel forking to several user agents may make it difficult (or
   impossible) for the caller to know which user agent will answer the
   next call attempt.  For example, a call attempt may reach the user's
   voice mail while the next one may reach a SIP phone where the user is
   available.  If both terminating user agents have different
   capabilities, the caller cannot know, even after the first call
   attempt, whether or not transcoding will be necessary for the
   session.  This is a well-known SIP problem that is referred to as
   HERFP (Heterogeneous Error Response Forking Problem).  Resolving
   HERFP is outside the scope of this document.

   It is recommended that an offerer does not invoke transcoding
   services before making sure that the answerer does not support the
   capabilities needed for the session.  Making wrong assumptions about
   the answerer's capabilities can lead to situations where two
   transcoders are introduced (one by the offerer and one by the
   answerer) in a session that would not need any transcoding services
   at all.

      An example of the situation above is a call between two GSM phones
      (without using transcoding-free operation).  Both phones use a GSM
      codec, but the speech is converted from GSM to PCM by the
      originating MSC (Mobile Switching Center) and from PCM back to GSM
      by the terminating MSC.

   Note that transcoding services can be symmetric (e.g., speech-to-text
   plus text-to-speech) or asymmetric (e.g., a one-way speech-to-text
   transcoding for a hearing-impaired user that can talk).

3.  Transcoding Services Invocation

   Once the need for transcoding for a particular session has been
   identified as described in Section 2, one of the user agents needs to

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   invoke transcoding services.

   As stated earlier, transcoder location is outside the scope of this
   document.  So, we assume that the user agent invoking transcoding
   services knows the URI of a server that provides them.

   Invoking transcoding services from a server (T) for a session between
   two user agents (A and B) involves establishing two media sessions;
   one between A and T and another between T and B. How to invoke T's
   services (i.e., how to establish both A-T and T-B sessions) depends
   on how we model the transcoding service.  We have considered two
   models for invoking a transcoding service.  The first is to use third
   party call control [7], also referred to as 3pcc.  The second is to
   use a (dial-in and dial-out) conference bridge that negotiates the
   appropriate media parameters on each individual leg (i.e., A-T and

   Section 3.1 analyzes the applicability of the third party call
   control model and Section 3.2 analyzes the applicability of the
   conference bridge transcoding invocation model.

3.1.  Third Party Call Control Transcoding Model

   In the 3pcc transcoding model, defined in [10], the user agent
   invoking the transcoding service has a signalling relationship with
   the transcoder and another signalling relationship with the remote
   user agent.  There is no signalling relationship between the
   transcoder and the remote user agent, as shown in Figure 1.

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          |       |
          |   T   |**
          |       |  **
          +-------+    **
            ^   *        **
            |   *          **
            |   *            **
           SIP  *              **
            |   *                **
            |   *                  **
            v   *                    **
          +-------+               +-------+
          |       |               |       |
          |   A   |<-----SIP----->|   B   |
          |       |               |       |
          +-------+               +-------+

           <-SIP-> Signalling
           ******* Media

   Figure 1: Third party call control model

   This model is suitable for advanced endpoints that are able to
   perform third party call control.  It allows end-points to invoke
   transcoding services on a stream basis.  That is, the media streams
   that need transcoding are routed through the transcoder while the
   streams that do not need it are sent directly between the endpoints.
   This model also allows to invoke one transcoder for the sending
   direction and a different one for the receiving direction of the same

   Invoking a transcoder in the middle of an ongoing session is also
   quite simple.  This is useful when session changes occur (e.g., an
   audio session is upgraded to an audio/video session) and the end-
   points cannot cope with the changes (e.g., they had common audio
   codecs but no common video codecs).

   The privacy level that is achieved using 3pcc is high, since the
   transcoder does not see the signalling between both end-points.  In
   this model, the transcoder only has access to the information that is
   strictly needed to perform its function.

3.2.  Conference Bridge Transcoding Model

   In a centralized conference, there are a number of media streams
   between the conference server and each participant of a conference.

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   For a given media type (e.g., audio) the conference server sends,
   over each individual stream, the media received over the rest of the
   streams, typically performing some mixing.  If the capabilities of
   all the endpoints participating in the conference are not the same,
   the conference server may have to send audio to different
   participants using different audio codecs.

   Consequently, we can model a transcoding service as a two-party
   conference server that may change not only the codec in use, but also
   the format of the media (e.g., audio to text).

   Using this model, T behaves as a B2BUA (Back-to-Back User Agent) and
   the whole A-T-B session is established as described in [11].
   Figure 2 shows the signalling relationships between the end-points
   and the transcoder.

          |       |**
          |   T   |  **
          |       |\   **
          +-------+ \\   **
            ^   *     \\   **
            |   *       \\   **
            |   *         SIP  **
           SIP  *           \\   **
            |   *             \\   **
            |   *               \\   **
            v   *                 \    **
          +-------+               +-------+
          |       |               |       |
          |   A   |               |   B   |
          |       |               |       |
          +-------+               +-------+

           <-SIP-> Signalling
           ******* Media

   Figure 2: Conference bridge model

   In the conferencing bridge model, the end-point invoking the
   transcoder is generally involved in less signalling exchanges than in
   the 3pcc model.  This may be an important feature for end-points
   using low bandwidth or high-delay access links (e.g., some wireless

   On the other hand, this model is less flexible than the 3pcc model.

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   It is not possible to use different transcoders for different streams
   or for different directions of a stream.

   Invoking a transcoder in the middle of an ongoing session or changing
   from one transcoder to another requires the remote end-point to
   support the Replaces [9] extension.  At present, not many user agents
   support it.

   Simple end-points that cannot perform 3pcc and thus cannot use the
   3pcc model, of course, need to use the conference bridge model.

4.  Security Considerations

   The specifications of the 3pcc and the conferencing transcoding
   models discuss security issues directly related to the implementation
   of those models.  Additionally, there are some considerations that
   apply to transcoding in general.

   In a session, a transcoder has access to at least some of the media
   exchanged between the endpoints.  In order to avoid rogue transcoders
   getting access to those media, it is recommended that endpoints
   authenticate the transcoder.  TLS [1] and S/MIME [8] can be used for
   this purpose.

   To achieve a higher degree of privacy, endpoints following the 3pcc
   transcoding model can use one transcoder in one direction and a
   different one in the other direction.  This way, no single transcoder
   has access to all the media exchanged between the endpoints.

   The fact that transcoders need to access media exchanged between the
   endpoints implies that endpoints cannot use end-to-end media security
   mechanisms.  Media encryption would not allow the transcoder to
   access the media and media integrity protection would not allow the
   transcoder to modify the media (which is obviously necessary to
   perform the transcoding function).  Nevertheless, endpoints can still
   use media security between the transcoder and themselves.

5.  IANA Considerations

   This document does not contain any IANA actions.

6.  Contributors

   This document is the result of discussions amongst the conferencing
   design team.  The members of this team include Eric Burger, Henning

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   Schulzrinne and Arnoud van Wijk.

7.  References

7.1.  Normative References

   [1]   Dierks, T. and C. Allen, "The TLS Protocol Version 1.0",
         RFC 2246, January 1999.

   [2]   Floyd, S. and L. Daigle, "IAB Architectural and Policy
         Considerations for Open Pluggable Edge Services", RFC 3238,
         January 2002.

   [3]   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.

   [4]   Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
         Session Description Protocol (SDP)", RFC 3264, June 2002.

   [5]   Roach, A., "Session Initiation Protocol (SIP)-Specific Event
         Notification", RFC 3265, June 2002.

   [6]   Charlton, N., Gasson, M., Gybels, G., Spanner, M., and A. van
         Wijk, "User Requirements for the Session Initiation Protocol
         (SIP) in Support of Deaf, Hard of Hearing and Speech-impaired
         Individuals", RFC 3351, August 2002.

   [7]   Rosenberg, J., Peterson, J., Schulzrinne, H., and G. Camarillo,
         "Best Current Practices for Third Party Call Control (3pcc) in
         the Session Initiation Protocol (SIP)", BCP 85, RFC 3725,
         April 2004.

   [8]   Ramsdell, B., "Secure/Multipurpose Internet Mail Extensions
         (S/MIME) Version 3.1 Certificate Handling", RFC 3850,
         July 2004.

   [9]   Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
         Protocol (SIP) "Replaces" Header", RFC 3891, September 2004.

   [10]  Camarillo, G., Burger, E., Schulzrinne, H., and A. van Wijk,
         "Transcoding Services Invocation in the Session Initiation
         Protocol (SIP) Using Third Party Call Control (3pcc)",
         RFC 4117, June 2005.

   [11]  Camarillo, G., "The Session Initiation Protocol (SIP)
         Conference Bridge Transcoding Model",

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         draft-ietf-sipping-transc-conf-03 (work in progress),
         June 2006.

7.2.  Informative References

   [12]  Handley, M., "SDP: Session Description Protocol",
         draft-ietf-mmusic-sdp-new-26 (work in progress), January 2006.

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Author's Address

   Gonzalo Camarillo
   Hirsalantie 11
   Jorvas  02420


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