Internet Engineering Task Force                                   SIP WG
Internet Draft
SIPPING Working Group                                       G. Camarillo
Internet-Draft                                                  Ericsson
draft-ietf-sipping-transc-framework-00.txt
February 6, 2004
Expires: August, 2004

     Framework for Transcoding with August 22, 2005                               February 21, 2005

 Conference Establishment Using Request-Contained Lists in the Session
                       Initiation Protocol

STATUS OF THIS MEMO (SIP)
               draft-ietf-sipping-transc-framework-01.txt

Status of this Memo

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   Copyright (C) The Internet Society (2005).

Abstract

   This document defines a framework for transcoding with SIP.  This
   framework includes how to discover the need of transcoding services
   in a session and how to invoke those transcoding services.  Two
   models for transcoding services invocation are discussed; 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 hearing, and speech-impaired
   individuals.

Table of Contents

   1

   1.  Introduction ........................................ . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2
   2.  Discovery of the Need for Transcoding Services ......    3 . . . . . . . .  3
   3.  Transcoding Services Invocation .....................  . . . . . . . . . . . . . . .  4
     3.1   Third Party Call Control Transcoding Model .......... . . . . . . . .  5
     3.2   Conference Bridge Transcoding Model .................    5
   4  . . . . . . . . . . .  6
   4.  Security Considerations .............................  . . . . . . . . . . . . . . . . . . .  7
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  8
   5
   6.  Contributors ........................................ . . . . . . . . . . . . . . . . . . . . . . . . .  8
   6          Authors' Addresses ..................................
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . .  8
   7          Bibliography ........................................
   7.1   Normative References . . . . . . . . . . . . . . . . . . . .  8

1
   7.2   Informational References . . . . . . . . . . . . . . . . . .  9
       Author's Address . . . . . . . . . . . . . . . . . . . . . . .  9
       Intellectual Property and Copyright Statements . . . . . . . . 10

1.  Introduction

   Two user agents involved in a SIP [1] [2] 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), SDP [9]), 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 of transcoding services for a
   particular session.  Section 3.2 3 introduces the third party call
   control and conference bridge transcoding invocation model, and models, which
   are further described in Section 3.1 introduces the third
   party call control model. and Section 3.2 respectively.
   Both models meet the requirements regarding transcoding services
   invocation in RFC3351 [2] [4] to support deaf, hard of hearing and
   speech-impaired individuals.

2

2.  Discovery of the Need for Transcoding Services

   According to the one-party consent model defined in RFC 3238 [3], [1] ,
   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.

   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 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]. [3].

   Presence documents are typically received in a NOTIFY 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.

   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 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

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 we said 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 [5], 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 (draft-ietf-sipping-
   transc-3pcc), [7], 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.

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

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

                Figure 1: Third party call control model

   This model is suitable for advanced end points 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 end points.
   This model also allows to invoke 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 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 no 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

      OPEN ISSUE: this section outlines how to use the URI-list
      mechanism for INVITEs specified in [8] to invoke a transcoder.
      Some people think that having an even simpler mechanism to perform
      transcoding invocation would be useful.  We need to decide whether
      we are happy with the current solution or we want to use a
      different mechanism.

   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,
   over each individual stream, the media received over the rest of the
   streams, typically performing some mixing.  If the capabilities of
   all the end-points participating in the conference are not the same,
   the conference server may have to send audio to different
   participants using different audio codecs.

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

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

   Figure 1: Third Party Call Control Model

   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,

   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 and the whole A-T-B session is
   established as described in [6]. [draft-camarillo-sipping-transc-b2bua].
   Figure 2 shows the signalling relationships between the end-points
   and the transcoder.

   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-poing using
   low bandwidth or high-delay access links (e.g., some wireless

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

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

                   Figure 2: Conference Bridge Control Model 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-poing 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.

   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 [7] [6] 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

4.  Security Considerations

   TBD.

5.  IANA Considerations

   This document does not introduce contain any new security considerations.

5 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
   Schulzrinne and Arnoud van Wijk.

6 Authors' Addresses

   Gonzalo Camarillo
   Ericsson
   Advanced Signalling Research Lab.
   FIN-02420 Jorvas
   Finland
   electronic mail:  Gonzalo.Camarillo@ericsson.com

7 Bibliography

7.  References

7.1  Normative References

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

   [2]  Rosenberg, H. J., Schulzrinne, G. H., Camarillo, A. R. G., Johnston, J. A.,
        Peterson, R. J., Sparks, M. R., Handley, M. and E. Schooler, "SIP: session
   initiation protocol,"
        Session Initiation Protocol", RFC 3261, Internet Engineering Task Force, June 2002.

   [2] N.

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

   [4]  Charlton, M. N., Gasson, G. M., Gybels, M. G., Spanner, M. and A. van
        Wijk, "User requirements Requirements for the session initiation protocol Session Initiation Protocol
        (SIP) in
   support Support of deaf, hard Deaf, Hard of hearing Hearing and speech-impaired individuals," Speech-impaired
        Individuals", RFC 3351, Internet Engineering Task Force, Aug. August 2002.

   [3] S. Floyd and L. Daigle, "IAB architectural and policy
   considerations for open pluggable edge services," RFC 3238, Internet
   Engineering Task Force, Jan. 2002.

   [4] J. Rosenberg and H. Schulzrinne, "An offer/answer model with
   session description protocol (SDP)," RFC 3264, Internet Engineering
   Task Force, June 2002.

   [5] J. Rosenberg, J. Peterson, H. Schulzrinne,

   [5]  Rosenberg, J., Peterson, J., Schulzrinne, H. and G. Camarillo,
        "Best current practices Current Practices for third party call control Third Party Call Control (3pcc) in
        the session
   initiation protocol," Internet Draft draft-ietf-sipping-3pcc-06,
   Internet Engineering Task Force, Jan. Session Initiation Protocol (SIP)", BCP 85, RFC 3725, April
        2004.  Work in progress.

   [6] G. Camarillo, "The session initiation protocol conference bridge
   transcoding model," Internet Draft draft-camarillo-sipping-transc-
   b2bua-00, Internet Engineering Task Force, Aug. 2003.  Work in
   progress.

   [7] B.  Mahy, R., Biggs, R. W. Dean, B. and R. Mahy, Dean, "The session inititation
   protocol Session Initiation
        Protocol (SIP) Engineering Task Force, Aug. 2003.  Work "Replaces" Header", RFC 3891, September 2004.

   [7]  Camarillo, G., "Transcoding Services Invocation in progress. the Session
        Initiation Protocol (SIP)  Using Third Party Call Control
        (3pcc)", draft-ietf-sipping-transc-3pcc-02 (work in progress),
        September 2004.

   [8]  Camarillo, G. and A. Johnston, "Conference Establishment Using
        Request-Contained Lists in the Session  Initiation Protocol
        (SIP)", draft-ietf-sipping-uri-list-conferencing-01 (work in
        progress), October 2004.

7.2  Informational References

   [9]  Handley, M. and V. Jacobson, "SDP: Session Description
        Protocol", RFC 2327, April 1998.

Author's Address

   Gonzalo Camarillo
   Ericsson
   Hirsalantie 11
   Jorvas  02420
   Finland

   EMail: Gonzalo.Camarillo@ericsson.com

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