    Network Working Group                                       G. Camarillo
    Request for Comments: 5369                                      Ericsson
    Category: Informational                                     October 2008

      Framework for Transcoding with the Session Initiation Protocol (SIP)

    Status 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.

    Abstract

       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
       individuals.

    Table of Contents

       1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . . . 2
       2.  Discovery of the Need for Transcoding Services  . . . . . . . . 2
       3.  Transcoding Services Invocation . . . . . . . . . . . . . . . . 4
         3.1.  Third-Party Call Control Transcoding Model  . . . . . . . . 4
         3.2.  Conference Bridge Transcoding Model . . . . . . . . . . . . 6
       4.  Security Considerations . . . . . . . . . . . . . . . . . . . . 7
       5.  Contributors  . . . . . . . . . . . . . . . . . . . . . . . . . 8
       6.  References  . . . . . . . . . . . . . . . . . . . . . . . . . . 8
         6.1.  Normative References  . . . . . . . . . . . . . . . . . . . 8
         6.2.  Informative References  . . . . . . . . . . . . . . . . . . 9

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    1.  Introduction

       Two user agents involved in a SIP [RFC3261] 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 [RFC4566]),
       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 increases 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 Sections 3.1 and 3.2, respectively.  Both
       models meet the requirements regarding transcoding services
       invocation in RFC 3351 [RFC3351], which 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
       [RFC3238], services that involve media manipulation invocation are
       best invoked by one of the endpoints involved in the communication,
       as opposed to being invoked by an intermediary in the network.
       Following this principle, one of the endpoints 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 [RFC3264] 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 [RFC3264].

       Presence documents are typically received in a NOTIFY request
       [RFC3265] 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
       voicemail 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
          (Global System for Mobile Communications) phones (without using
          transcoding-free operation).  Both phones use a GSM codec, but the
          speech is converted from GSM to PCM (Pulse Code Modulation) 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).

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    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
       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 [RFC3725], 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 T-B).

       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 [RFC4117], 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 endpoints 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 invoking one transcoder for the sending
       direction and a different one for the receiving direction of the same
       stream.

       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
       endpoints 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 endpoints.  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.
       For a given media type (e.g., audio) the conference server sends,

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       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 [RFC5370].
       Figure 2 shows the signalling relationships between the endpoints and
       the transcoder.

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

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

                         Figure 2: Conference Bridge Model

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

       On the other hand, this model is less flexible than the 3pcc model.
       It is not possible to use different transcoders for different streams
       or for different directions of a stream.

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       Invoking a transcoder in the middle of an ongoing session or changing
       from one transcoder to another requires the remote endpoint to
       support the Replaces [RFC3891] extension.  At present, not many user
       agents support it.

       Simple endpoints 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 [RFC5246] and S/MIME [RFC3850] 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.  Contributors

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

    6.  References

    6.1.  Normative References

       [RFC3238]  Floyd, S. and L. Daigle, "IAB Architectural and Policy
                  Considerations for Open Pluggable Edge Services",
                  RFC 3238, January 2002.

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       [RFC3261]  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.

       [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
                  with Session Description Protocol (SDP)", RFC 3264,
                  June 2002.

       [RFC3265]  Roach, A.B., "Session Initiation Protocol (SIP)-Specific
                  Event Notification", RFC 3265, June 2002.

       [RFC3351]  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.

       [RFC3725]  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.

       [RFC3850]  Ramsdell, B., "Secure/Multipurpose Internet Mail
                  Extensions (S/MIME) Version 3.1 Certificate Handling",
                  RFC 3850, July 2004.

       [RFC3891]  Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
                  Protocol (SIP) "Replaces" Header", RFC 3891,
                  September 2004.

       [RFC4117]  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.

       [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
                  (TLS) Protocol Version 1.2", RFC 5246, August 2008.

       [RFC5370]  Camarillo, G., "The Session Initiation Protocol (SIP)
                  Conference Bridge Transcoding Model", RFC 5370,
                  October 2008.

    6.2.  Informative References

       [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
                  Description Protocol", RFC 4566, July 2006.

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    Author's Address

       Gonzalo Camarillo
       Ericsson
       Hirsalantie 11
       Jorvas  02420
       Finland

       EMail: Gonzalo.Camarillo@ericsson.com

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