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Internet Engineering Task Force                                   SIP WG
Internet Draft                                              G. Camarillo
                                                                Ericsson
draft-camarillo-sipping-transc-framework-00.txt
August 28, 2003
Expires: February, 2004


     Framework for Transcoding with the Session Initiation Protocol

STATUS OF THIS MEMO

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
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   http://www.ietf.org/ietf/1id-abstracts.txt

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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; 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.










G. Camarillo                                                  [Page 1]

Internet Draft                    SIP                    August 28, 2003





                           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 .................    5
   4          Security Considerations .............................    8
   5          Contributors ........................................    8
   6          Authors' Addresses ..................................    8
   7          Bibliography ........................................    8



































G. Camarillo                                                  [Page 2]

Internet Draft                    SIP                    August 28, 2003


1 Introduction

   Two user agents involved in a SIP [1] 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), 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 be able to understand what
   it is 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.
   Therefore, 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 introduces the conference bridge transcoding invocation
   model, and Section 3.1 introduces the third party call control model.
   Both models meet the requirements regarding transcoding services
   invocation in RFC3351 [2] 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 [3],
   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



G. Camarillo                                                  [Page 3]

Internet Draft                    SIP                    August 28, 2003


   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].
   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 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 previously, transcoder location is outside the scope of
   this document. Therefore, 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



G. Camarillo                                                  [Page 4]

Internet Draft                    SIP                    August 28, 2003


   use a (dial-in and possibly 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-camarillo-sipping-
   transc-3pcc), 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.


   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

   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.



G. Camarillo                                                  [Page 5]

Internet Draft                    SIP                    August 28, 2003










       +-------+
       |       |
       |   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, T behaves as a B2BUA and the whole A-T-B session is
   established as described in (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



G. Camarillo                                                  [Page 6]

Internet Draft                    SIP                    August 28, 2003










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


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



   Figure 2: Conference Bridge Control Model


   low bandwidth or high-delay access links (e.g., some wireless
   accesses).

   However, 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 [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.



G. Camarillo                                                  [Page 7]

Internet Draft                    SIP                    August 28, 2003


4 Security Considerations

   This document does not introduce any new security considerations.

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 Authors' Addresses

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

7 Bibliography

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

   [2] N. Charlton, M. Gasson, G. Gybels, M. Spanner, 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, Internet Engineering Task Force, Aug. 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. L. Peterson, H. Schulzrinne, and G. Camarillo,
   "Best current practices for third party call control in the session
   initiation protocol," internet draft, Internet Engineering Task
   Force, July 2003.  Work in progress.

   [6] B. Biggs, R. W. Dean, and R. Mahy, "The session inititation
   protocol (SIP) Engineering Task Force, Aug. 2003.  Work in progress.





G. Camarillo                                                  [Page 8]

Internet Draft                    SIP                    August 28, 2003


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G. Camarillo                                                  [Page 9]

Internet Draft                    SIP                    August 28, 2003





















































G. Camarillo                                                 [Page 10]


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