[Docs] [txt|pdf] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits]

Versions: (draft-camarillo-sipping-transc-3pcc) 00 01 02 RFC 4117

Internet Engineering Task Force                                   SIP WG
Internet Draft                                              G. Camarillo
                                                                Ericsson
                                                               E. Burger
                                                              Brooktrout
                                                          H. Schulzrinne
                                                     Columbia University
                                                             A. van Wijk
                                                                 Viataal
draft-ietf-sipping-transc-3pcc-02.txt
September 17, 2004
Expires: March, 2005


       Transcoding Services Invocation in the Session Initiation
          Protocol (SIP) Using Third Party Call Control (3pcc)

STATUS OF THIS MEMO

   By submitting this Internet-Draft, I certify that any applicable
   patent or other IPR claims of which I am aware have been disclosed,
   and any of which I become aware will be disclosed, in accordance with
   RFC 3668.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that other
   groups may also distribute working documents as Internet-Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time. It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt.

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.


Abstract

   This document describes how to invoke transcoding services using SIP
   and third party call control. This way of invocation meets the
   requirements for SIP regarding transcoding services invocation to
   support deaf, hard of hearing and speech-impaired individuals.





G. Camarillo et. al.                                          [Page 1]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004





                           Table of Contents



   1          Introduction ........................................    3
   2          General Overview ....................................    3
   3          Third Party Call Control Flows ......................    3
   3.1        Terminology .........................................    4
   3.2        Callee's Invocation .................................    4
   3.3        Caller's Invocation .................................    9
   3.4        Receiving the Original Stream .......................    9
   3.5        Transcoding Services in Parallel ....................   11
   3.6        Transcoding Services in Serial ......................   15
   4          Security Considerations .............................   17
   5          IANA Considerations .................................   17
   6          Authors' Addresses ..................................   17
   7          Normative References ................................   17
   8          Informative References ..............................   18






























G. Camarillo et. al.                                          [Page 2]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


1 Introduction

   The framework for transcoding with SIP [4] describes how two SIP [1]
   UAs (User Agents) can discover imcompatibilities that prevent them
   from establishing a session (e.g., lack of support for a common codec
   or for a common media type). When such incompatibilities are found,
   the UAs need to invoke transcoding services to successfully establish
   the session. 3pcc (third party call control) [2] is one way to
   perform such invocation.

2 General Overview

   In the 3pcc model for transcoding invocation, a transcoding server
   that provides a particular transcoding service (e.g., speech-to-text)
   is identified by a URI. A UA that wishes to invoke that service sends
   an INVITE request to that URI establishing a number of media streams.
   The way the transcoder manipulates and manages the contents of those
   media streams (e.g., the text received over the text stream is
   transformed into speech and sent over the audio stream) is service
   specific.

        All the call flows in this document use SDP. The same call
        flows could be used with another session description
        protocol that provided similar session description
        capabilities.

3 Third Party Call Control Flows

   Given two UAs (A and B) and a transcoding server (T), the invocation
   of a transcoding service consists of establishing two sessions; A-T
   and T-B. How these sessions are established depends on which party,
   the caller (A) or the callee (B), invokes the transcoding services.
   Section 3.2 deals with callee invocation and Section 3.3 deals with
   caller invocation.

   In all our 3pcc flows we have followed a general principle; a 200
   (OK) response from the transcoding service has to be received before
   contacting the callee. This tries to ensure that the transcoding
   service will be available when the callee accepts the session.

   Still, the transcoding service does not know the exact type of
   transcoding it will be performing until the callee accepts the
   session. So, there are always chances of failing to provide
   transcoding services after the callee has accepted the session. A
   system with tough requirements could use preconditions to avoid this
   situation. When preconditions are used, the callee is not alerted
   until everything is ready for the session.




G. Camarillo et. al.                                          [Page 3]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


3.1 Terminology

   All the flows in this document follow the naming convention below:

        SDP A: A session description generated by A. It contains, among
             other things, the transport address/es (IP address and port
             number) where A wants to receive media for each particular
             stream.

        SDP B: A session description generated by B. It contains, among
             other things, the transport address/es where B wants to
             receive media for each particular stream.

        SDP A+B: A session description that contains, among other
             things, the transport address/es where A wants to receive
             media and the transport address/es where B wants to receive
             media.

        SDP TA: A session description generated by T and intended for A.
             It contains, among other things, the transport address/es
             where T wants to receive media from A.

        SDP TB: A session description generated by T and intended for B.
             It contains, among other things, the transport address/es
             where T wants to receive media from B.

        SDP TA+TB: A session description generated by T that contains,
             among other things, the transport address/es where T wants
             to receive media from A and the transport address/es where
             T wants to receive media from B.

3.2 Callee's Invocation

   In this scenario, B receives an INVITE from A and B decides to
   introduce T in the session. Figure 1 shows the call flow for this
   scenario.


   In Figure 1, A can both hear and speak and B is a deaf user with a
   speech impairment. A proposes to establish a session that consists of
   an audio stream (1). B wants to send and receive only text, so it
   invokes a transcoding service T that will perform both speech-to-text
   and text-to-speech conversions (2). The session descriptions of
   Figure 1 are partially shown below.







G. Camarillo et. al.                                          [Page 4]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


      A                            T                            B

      |                            |                            |
      |--------------------(1) INVITE SDP A-------------------->|
      |                            |                            |
      |                            |<---(2) INVITE SDP A+B------|
      |                            |                            |
      |                            |---(3) 200 OK SDP TA+TB---->|
      |                            |                            |
      |                            |<---------(4) ACK-----------|
      |                            |                            |
      |<-------------------(5) 200 OK SDP TA--------------------|
      |                            |                            |
      |------------------------(6) ACK------------------------->|
      |                            |                            |
      | ************************** | ************************** |
      |*          MEDIA           *|*          MEDIA           *|
      | ************************** | ************************** |
      |                            |                            |



   Figure 1: Callee's invocation of a transcoding service


   (1) INVITE SDP A

          m=audio 20000 RTP/AVP 0
          c=IN IP4 A.example.com


   (2) INVITE SDP A+B

          m=audio 20000 RTP/AVP 0
          c=IN IP4 A.example.com
          m=text 40000 RTP/AVP 96
          c=IN IP4 B.example.com
          a=rtpmap:96 t140/1000



   (3) 200 OK SDP TA+TB

          m=audio 30000 RTP/AVP 0
          c=IN IP4 T.example.com
          m=text 30002 RTP/AVP 96
          c=IN IP4 T.example.com
          a=rtpmap:96 t140/1000



G. Camarillo et. al.                                          [Page 5]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


   (5) 200 OK SDP TA

          m=audio 30000 RTP/AVP 0
          c=IN IP4 T.example.com



   Four media streams (i.e., two bi-directional streams) have been
   established at this point:

        1.   Audio from A to T.example.com:30000

        2.   Text from T to B.example.com:40000

        3.   Text from B to T.example.com:30002

        4.   Audio from T to A.example.com:20000

   When either A or B decide to terminate the session, they send a BYE
   indicating that the session is over.

   If the first INVITE (1) received by B is empty (no session
   description), the call flow is slightly different. Figure 2 shows the
   messages involved.


   B may have different reasons for invoking T before knowing A's
   session description. B may want to hide its capabilities, and
   therefore it wants to return a session description with all the
   codecs B supports plus all the codecs T supports. Or T may provide
   recording services (besides transcoding), and B wants T to record the
   conversation, regardless of whether or not transcoding is needed.

   This scenario (Figure 2) is a bit more complex than the previous one.
   In INVITE (2), B still does not have SDP A, so it cannot provide T
   with that information. When B finally receives SDP A in (6), it has
   to send it to T. B sends an empty INVITE to T (7) and gets a 200 OK
   with SDP TA+TB (8). In general, this SDP TA+TB can be different than
   the one that was sent in (3). That is why B needs to send the updated
   SDP TA to A in (9). A then sends a possibly updated SDP A (10) and B
   sends it to T in (12). On the other hand, if T happens to return the
   same SDP TA+TB in (8) as in (3), B can skip messages (9), (10), and
   (11). So, implementors of transcoding services are encouraged to
   return the same session description in (8) as in (3) in this type of
   scenario. The session descriptions of this flow are shown below:






G. Camarillo et. al.                                          [Page 6]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


      A                            T                            B

      |                            |                            |
      |----------------------(1) INVITE------------------------>|
      |                            |                            |
      |                            |<-----(2) INVITE SDP B------|
      |                            |                            |
      |                            |---(3) 200 OK SDP TA+TB---->|
      |                            |                            |
      |                            |<---------(4) ACK-----------|
      |                            |                            |
      |<-------------------(5) 200 OK SDP TA--------------------|
      |                            |                            |
      |-----------------------(6) ACK SDP A-------------------->|
      |                            |                            |
      |                            |<-------(7) INVITE----------|
      |                            |                            |
      |                            |---(8) 200 OK SDP TA+TB---->|
      |                            |                            |
      |<-----------------(9) INVITE SDP TA----------------------|
      |                            |                            |
      |------------------(10) 200 OK SDP A--------------------->|
      |                            |                            |
      |<-----------------------(11) ACK-------------------------|
      |                            |                            |
      |                            |<-----(12) ACK SDP A+B------|
      |                            |                            |
      | ************************** | ************************** |
      |*          MEDIA           *|*          MEDIA           *|
      | ************************** | ************************** |


   Figure 2: Callee's invocation after initial INVITE without SDP


   (2) INVITE SDP A+B

          m=audio 20000 RTP/AVP 0
          c=IN IP4 0.0.0.0
          m=text 40000 RTP/AVP 96
          c=IN IP4 B.example.com
          a=rtpmap:96 t140/1000



   (3) 200 OK SDP TA+TB

          m=audio 30000 RTP/AVP 0
          c=IN IP4 T.example.com


G. Camarillo et. al.                                          [Page 7]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


          m=text 30002 RTP/AVP 96
          c=IN IP4 T.example.com
          a=rtpmap:96 t140/1000



   (5) 200 OK SDP TA

          m=audio 30000 RTP/AVP 0
          c=IN IP4 T.example.com



   (6) ACK SDP A

          m=audio 20000 RTP/AVP 0
          c=IN IP4 A.example.com



   (8) 200 OK SDP TA+TB

          m=audio 30004 RTP/AVP 0
          c=IN IP4 T.example.com
          m=text 30006 RTP/AVP 96
          c=IN IP4 T.example.com
          a=rtpmap:96 t140/1000



   (9) INVITE SDP TA

          m=audio 30004 RTP/AVP 0
          c=IN IP4 T.example.com



   (10) 200 OK SDP A

          m=audio 20002 RTP/AVP 0
          c=IN IP4 A.example.com



   (12) ACK SDP A+B

          m=audio 20002 RTP/AVP 0
          c=IN IP4 A.example.com



G. Camarillo et. al.                                          [Page 8]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


          m=text 40000 RTP/AVP 96
          c=IN IP4 B.example.com
          a=rtpmap:96 t140/1000



   Four media streams (i.e., two bi-directional streams) have been
   established at this point:

        1.   Audio from A to T.example.com:30004

        2.   Text from T to B.example.com:40000

        3.   Text from B to T.example.com:30006

        4.   Audio from T to A.example.com:20002

3.3 Caller's Invocation

   In this scenario, A wishes to establish a session with B using a
   transcoding service. A uses 3pcc to set up the session between T and
   B. The call flow we provide here is slightly different than the ones
   in [2]. In [2], the controller establishes a session between two user
   agents, which are the ones deciding the characteristics of the
   streams. Here, A wants to establish a session between T and B, but A
   wants to decide how many and which types of streams are established.
   That is why A sends its session description in the first INVITE (1)
   to T, as opposed to the media-less initial INVITE recommended by [2].
   Figure 3 shows the call flow for this scenario.


   We do not include the session descriptions of this flow, since they
   are very similar to the ones in Figure 2. In this flow, if T returns
   the same SDP TA+TB in (8) as in (2), messages (9), (10), and (11) can
   be skipped.

3.4 Receiving the Original Stream

   Sometimes, as pointed out in the requirements for SIP in support of
   deaf, hard of hearing, and speech-impaired individuals [5], a user
   wants to receive both the original stream (e.g., audio) and the
   transcoded stream (e.g., the output of the speech-to-text
   conversion). There are various possible solutions for this problem.
   One solution consists of using the SDP group attribute with FID
   semantics [3]. FID allows requesting that a stream is sent to two
   different transport addresses in parallel, as shown below:





G. Camarillo et. al.                                          [Page 9]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004



      A                            T                            B

      |                            |                            |
      |-------(1) INVITE SDP A---->|                            |
      |                            |                            |
      |<----(2) 200 OK SDP TA+TB---|                            |
      |                            |                            |
      |----------(3) ACK---------->|                            |
      |                            |                            |
      |--------------------(4) INVITE SDP TA------------------->|
      |                            |                            |
      |<--------------------(5) 200 OK SDP B--------------------|
      |                            |                            |
      |-------------------------(6) ACK------------------------>|
      |                            |                            |
      |--------(7) INVITE--------->|                            |
      |                            |                            |
      |<---(8) 200 OK SDP TA+TB  --|                            |
      |                            |                            |
      |--------------------(9) INVITE SDP TA------------------->|
      |                            |                            |
      |<-------------------(10) 200 OK SDP B--------------------|
      |                            |                            |
      |-------------------------(11) ACK----------------------->|
      |                            |                            |
      |------(12) ACK SDP A+B----->|                            |
      |                            |                            |
      | ************************** | ************************** |
      |*          MEDIA           *|*          MEDIA           *|
      | ************************** | ************************** |
      |                            |                            |


   Figure 3: Caller's invocation of a transcoding service


            a=group:FID 1 2
            m=audio 20000 RTP/AVP 0
            c=IN IP4 A.example.com
            a=mid:1
            m=audio 30000 RTP/AVP 0
            c=IN IP4 T.example.com
            a=mid:2



   The problem with this solution is that the majority of the SIP user
   agents do not support FID. Moreover, only a small fraction of the few


G. Camarillo et. al.                                         [Page 10]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


   UAs that do support FID, support sending simultaneous copies of the
   same media stream at the same time. In addition, FID forces both
   copies of the stream to use the same codec.

   So, we recommend that T (instead of one of the user agent) replicates
   the media stream. The transcoder T receiving the following session
   description performs speech-to-text and text-to-speech conversions
   between the first audio stream and the text stream. In addition, T
   copies the first audio stream to the second audio stream and sends it
   to A.

            m=audio 40000 RTP/AVP 0
            c=IN IP4 B.example.com
            m=audio 20000 RTP/AVP 0
            c=IN IP4 A.example.com
            a=recvonly
            m=text 20002 RTP/AVP 96
            c=IN IP4 A.example.com
            a=rtpmap:96 t140/1000



3.5 Transcoding Services in Parallel

   Transcoding services sometimes consist of human relays (e.g., a
   person performing speech-to-text and text-to-speech conversions for a
   session). If the same person is involved in both conversions (i.e.,
   from A to B and from B to A), he or she has access to all the
   conversation. In order to provide some degree of privacy, sometimes
   two different persons are allocated to do the job (i.e., one person
   handles A->B and the other B->A). This type of disposition is also
   useful for automated transcoding services, where one machine converts
   text to synthetic speech (text-to-speech) and a different machine
   performs voice recognition (speech-to-text).

   The scenario just described involves four different sessions; A-T1,
   T1-B, B-T2 and T2-A. Figure 4 shows the call flow where A invokes T1
   and T2.


   (1) INVITE SDP AT1

          m=text 20000 RTP/AVP 96
          c=IN IP4 A.example.com
          a=rtpmap:96 t140/1000
          a=sendonly
          m=audio 20000 RTP/AVP 0
          c=IN IP4 0.0.0.0
          a=recvonly


G. Camarillo et. al.                                         [Page 11]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004




   (2) INVITE SDP AT2

          m=text 20002 RTP/AVP 96
          c=IN IP4 A.example.com
          a=rtpmap:96 t140/1000
          a=recvonly
          m=audio 20000 RTP/AVP 0
          c=IN IP4 0.0.0.0
          a=sendonly



   (3) 200 OK SDP T1A+T1B

          m=text 30000 RTP/AVP 96
          c=IN IP4 T1.example.com
          a=rtpmap:96 t140/1000
          a=recvonly
          m=audio 30002 RTP/AVP 0
          c=IN IP4 T1.example.com
          a=sendonly



   (5) 200 OK SDP T2A+T2B

          m=text 40000 RTP/AVP 96
          c=IN IP4 T2.example.com
          a=rtpmap:96 t140/1000
          a=sendonly
          m=audio 40002 RTP/AVP 0
          c=IN IP4 T2.example.com
          a=recvonly



   (7) INVITE SDP T1B+T2B

          m=audio 30002 RTP/AVP 0
          c=IN IP4 T1.example.com
          a=sendonly
          m=audio 40002 RTP/AVP 0
          c=IN IP4 T2.example.com
          a=recvonly





G. Camarillo et. al.                                         [Page 12]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004



  A                          T1                     T2            B

  |                          |                      |             |
  |----(1) INVITE SDP AT1--->|                      |             |
  |                          |                      |             |
  |----------------(2) INVITE SDP AT2-------------->|             |
  |                          |                      |             |
  |<-(3) 200 OK SDP T1A+T1B--|                      |             |
  |                          |                      |             |
  |---------(4) ACK--------->|                      |             |
  |                          |                      |             |
  |<---------------(5) 200 OK SDP T2A+T2B-----------|             |
  |                          |                      |             |
  |----------------------(6) ACK------------------->|             |
  |                          |                      |             |
  |-----------------------(7) INVITE SDP T1B+T2B----------------->|
  |                          |                      |             |
  |<----------------------(8) 200 OK SDP BT1+BT2------------------|
  |                          |                      |             |
  |------(9) INVITE--------->|                      |             |
  |                          |                      |             |
  |-------------------(10) INVITE------------------>|             |
  |                          |                      |             |
  |<-(11) 200 OK SDP T1A+T1B-|                      |             |
  |                          |                      |             |
  |<------------(12) 200 OK SDP T2A+T2B-------------|             |
  |                          |                      |             |
  |------------------(13) INVITE SDP T1B+T2B--------------------->|
  |                          |                      |             |
  |<-----------------(14) 200 OK SDP BT1+BT2----------------------|
  |                          |                      |             |
  |--------------------------(15) ACK---------------------------->|
  |                          |                      |             |
  |---(16) ACK SDP AT1+BT1-->|                      |             |
  |                          |                      |             |
  |------------(17) ACK SDP AT2+BT2---------------->|             |
  |                          |                      |             |
  | ************************ | ********************************** |
  |*          MEDIA         *|*               MEDIA              *|
  | ************************ | ********************************** |
  |                          |                      |             |
  | ***********************************************   ***********
  |*                      MEDIA                    *|*   MEDIA   *|
  | *********************************************** | *********** |
  |                          |                      |             |


   Figure 4: Transcoding services in parallel


G. Camarillo et. al.                                         [Page 13]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


   (8) 200 OK SDP BT1+BT2

          m=audio 50000 RTP/AVP 0
          c=IN IP4 B.example.com
          a=recvonly
          m=audio 50002 RTP/AVP 0
          c=IN IP4 B.example.com
          a=sendonly



   (11) 200 OK SDP T1A+T1B

          m=text 30000 RTP/AVP 96
          c=IN IP4 T1.example.com
          a=rtpmap:96 t140/1000
          a=recvonly
          m=audio 30002 RTP/AVP 0
          c=IN IP4 T1.example.com
          a=sendonly



   (12) 200 OK SDP T2A+T2B

          m=text 40000 RTP/AVP 96
          c=IN IP4 T2.example.com
          a=rtpmap:96 t140/1000
          a=sendonly
          m=audio 40002 RTP/AVP 0
          c=IN IP4 T2.example.com
          a=recvonly



   Since T1 have returned the same SDP in (11) as in (3) and T2 has
   returned the same SDP in (12) as in (5), messages (13), (14) and (15)
   can be skipped.

   (16) ACK SDP AT1+BT1

          m=text 20000 RTP/AVP 96
          c=IN IP4 A.example.com
          a=rtpmap:96 t140/1000
          a=sendonly
          m=audio 50000 RTP/AVP 0
          c=IN IP4 B.example.com
          a=recvonly



G. Camarillo et. al.                                         [Page 14]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


   (17) ACK SDP AT2+BT2

          m=text 20002 RTP/AVP 96
          c=IN IP4 A.example.com
          a=rtpmap:96 t140/1000
          a=recvonly
          m=audio 50002 RTP/AVP 0
          c=IN IP4 B.example.com
          a=sendonly



   Four media streams have been established at this point:

        1.   Text from A to T1.example.com:30000

        2.   Audio from T1 to B.example.com:50000

        3.   Audio from B to T2.example.com:40002

        4.   Text from T2 to A.example.com:20002

   Note that B, the user agent server, needs to support two media
   streams; one sendonly and the other recvonly. At present, some user
   agents, although they support a single sendrecv media stream, they do
   not support a different media line per direction. Implementers are
   encouraged to build support for this feature.

3.6 Transcoding Services in Serial

   In a distributed environment, a complex transcoding service (e.g.,
   English text to Spanish speech) is often provided by several servers.
   For example, one server performs English text to Spanish text
   translation, and its output is feed into a server that performs
   text-to-speech conversion. The flow in Figure 5 shows how A invokes
   T1 and T2.















G. Camarillo et. al.                                         [Page 15]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004



  A                           T1                    T2            B

  |                           |                     |             |
  |----(1) INVITE SDP A-----> |                     |             |
  |                           |                     |             |
  |<-(2) 200 OK SDP T1A+T1T2- |                     |             |
  |                           |                     |             |
  |----------(3) ACK--------> |                     |             |
  |                           |                     |             |
  |-----------(4) INVITE SDP T1T2------------------>|             |
  |                           |                     |             |
  |<-----------(5) 200 OK SDP T2T1+T2B--------------|             |
  |                           |                     |             |
  |---------------------(6) ACK-------------------->|             |
  |                           |                     |             |
  |---------------------------(7) INVITE SDP T2B----------------->|
  |                           |                     |             |
  |<--------------------------(8) 200 OK SDP B--------------------|
  |                           |                     |             |
  |--------------------------------(9) ACK----------------------->|
  |                           |                     |             |
  |---(10) INVITE-----------> |                     |             |
  |                           |                     |             |
  |------------------(11) INVITE------------------->|             |
  |                           |                     |             |
  |<-(12) 200 OK SDP T1A+T1T2-|                     |             |
  |                           |                     |             |
  |<-------------(13) 200 OK SDP T2T1+T2B-----------|             |
  |                           |                     |             |
  |---(14) ACK SDP T1T2+B---> |                     |             |
  |                           |                     |             |
  |-----------------------(15) INVITE SDP T2B-------------------->|
  |                           |                     |             |
  |<----------------------(16) 200 OK SDP B-----------------------|
  |                           |                     |             |
  |----------------(17) ACK SDP T1T2+B------------->|             |
  |                           |                     |             |
  |----------------------------(18) ACK-------------------------->|
  |                           |                     |             |
  | ************************* | *******************   *********** |
  |*         MEDIA           *|*       MEDIA       *|*   MEDIA   *|
  | ************************* | ******************* | *********** |
  |                           |                     |             |


   Figure 5: Transcoding services in serial




G. Camarillo et. al.                                         [Page 16]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


4 Security Considerations

   This document describes how to use third party call control to invoke
   transcoding services. It does not introduce new security
   considerations besides the ones discussed in [2].

5 IANA Considerations

   This document has no actions for IANA.

6 Authors' Addresses

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

   Eric Burger
   Brooktrout Technology, Inc.
   18 Keewaydin Way
   Salem, NH 03079
   USA
   electronic mail:  eburger@ieee.org

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University 1214 Amsterdam Avenue, MC 0401
   New York, NY 10027
   USA
   electronic mail:  schulzrinne@cs.columbia.edu

   Arnoud van Wijk
   Viataal
   Research & Development
   Afdeling RDS
   Theerestraat 42
   5271 GD Sint-Michielsgestel
   The Netherlands
   electronic mail:  a.vwijk@viataal.nl

7 Normative References

   [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.



G. Camarillo et. al.                                         [Page 17]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


   [2] J. Rosenberg, J. Peterson, H. Schulzrinne, and G. Camarillo,
   "Best current practices for third party call control (3pcc) in the
   session initiation protocol (SIP)," RFC 3725, Internet Engineering
   Task Force, Apr. 2004.

   [3] G. Camarillo, G. Eriksson, J. Holler, and H. Schulzrinne,
   "Grouping of media lines in the session description protocol (SDP),"
   RFC 3388, Internet Engineering Task Force, Dec. 2002.

8 Informative References

   [4] G. Camarillo, "Framework for transcoding with the session
   initiation protocol," Internet Draft draft-camarillo-sipping-transc-
   framework-00, Internet Engineering Task Force, Aug. 2003.  Work in
   progress.

   [5] 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.


   Intellectual Property Statement

   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 IETF's procedures with respect to rights in IETF Documents can
   be found in BCP 78 and BCP 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 at
   http://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.






G. Camarillo et. al.                                         [Page 18]

Internet Draft          3pcc Transcoding in SIP       September 17, 2004


   Disclaimer of Validity

   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.


   Copyright Statement

   Copyright (C) The Internet Society (2004). This document is subject
   to the rights, licenses and restrictions contained in BCP 78, and
   except as set forth therein, the authors retain all their rights.


   Acknowledgment

   Funding for the RFC Editor function is currently provided by the
   Internet Society.





























G. Camarillo et. al.                                         [Page 19]


Html markup produced by rfcmarkup 1.109, available from https://tools.ietf.org/tools/rfcmarkup/