Internet Engineering Task Force SIP WG Internet Draft G. Camarillo Ericsson E. Burger SnowShore Networks H. Schulzrinne Columbia University A. van Wijk Viataal draft-ietf-sipping-transc-3pcc-00.txt February 3, 2004 Expires: August, 2004 Transcoding Services Invocation in the Session Initiation Protocol Using Third Party Call Control 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 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 To view the list Internet-Draft Shadow Directories, see 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 SIP February 3, 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 ............................. 15 5 Authors' Addresses .................................. 15 6 Bibliography ........................................ 16 G. Camarillo et. al. [Page 2]
Internet Draft SIP February 3, 2004 1 Introduction The framework for transcoding with SIP [1] describes how two SIP UAs 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 SIP February 3, 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. (1) INVITE SDP A m=audio 20000 RTP/AVP 0 G. Camarillo et. al. [Page 4]
Internet Draft SIP February 3, 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 c=IN IP4 A.domain.com (2) INVITE SDP A+B m=audio 20000 RTP/AVP 0 c=IN IP4 A.domain.com m=text 40000 RTP/AVP 96 c=IN IP4 B.domain.com a=rtpmap:96 t140/1000 (3) 200 OK SDP TA+TB m=audio 30000 RTP/AVP 0 c=IN IP4 T.domain.com m=text 30002 RTP/AVP 96 c=IN IP4 T.domain.com a=rtpmap:96 t140/1000 G. Camarillo et. al. [Page 5]
Internet Draft SIP February 3, 2004 (5) 200 OK SDP TA m=audio 30000 RTP/AVP 0 c=IN IP4 T.domain.com Four media streams (i.e., two bi-directional streams) have been established at this point: 1. Audio from A to T.domain.com:30000 2. Text from T to B.domain.com:40000 3. Text from B to T.domain.com:30002 4. Audio from T to A.domain.com:20000 When either A or B decide to terminate the session, B will send a BYE to T 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: (2) INVITE SDP A+B G. Camarillo et. al. [Page 6]
Internet Draft SIP February 3, 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 m=audio 20000 RTP/AVP 0 c=IN IP4 0.0.0.0 m=text 40000 RTP/AVP 96 c=IN IP4 B.domain.com a=rtpmap:96 t140/1000 (3) 200 OK SDP TA+TB m=audio 30000 RTP/AVP 0 c=IN IP4 T.domain.com G. Camarillo et. al. [Page 7]
Internet Draft SIP February 3, 2004 m=text 30002 RTP/AVP 96 c=IN IP4 T.domain.com a=rtpmap:96 t140/1000 (5) 200 OK SDP TA m=audio 30000 RTP/AVP 0 c=IN IP4 T.domain.com (6) ACK SDP A m=audio 20000 RTP/AVP 0 c=IN IP4 A.domain.com (8) 200 OK SDP TA+TB m=audio 30004 RTP/AVP 0 c=IN IP4 T.domain.com m=text 30006 RTP/AVP 96 c=IN IP4 T.domain.com a=rtpmap:96 t140/1000 (9) INVITE SDP TA m=audio 30004 RTP/AVP 0 c=IN IP4 T.domain.com (10) 200 OK SDP A m=audio 20002 RTP/AVP 0 c=IN IP4 A.domain.com (12) ACK SDP A+B m=audio 20002 RTP/AVP 0 c=IN IP4 A.domain.com G. Camarillo et. al. [Page 8]
Internet Draft SIP February 3, 2004 m=text 40000 RTP/AVP 96 c=IN IP4 B.domain.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.domain.com:30004 2. Text from T to B.domain.com:40000 3. Text from B to T.domain.com:30006 4. Audio from T to A.domain.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 [3], 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 [4]. FID allows requesting that a stream is sent to two different transport addresses in parallel, as shown below: a=group:FID 1 2 G. Camarillo et. al. [Page 9]
Internet Draft SIP February 3, 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 m=audio 20000 RTP/AVP 0 c=IN IP4 A.domain.com a=mid:1 m=audio 30000 RTP/AVP 0 c=IN IP4 T.domain.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 SIP February 3, 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 a 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.domain.com m=audio 20000 RTP/AVP 0 c=IN IP4 A.domain.com a=recvonly m=text 20002 RTP/AVP 96 c=IN IP4 A.domain.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.domain.com a=rtpmap:96 t140/1000 a=sendonly m=audio 20000 RTP/AVP 0 c=IN IP4 0.0.0.0 G. Camarillo et. al. [Page 11]
Internet Draft SIP February 3, 2004 a=recvonly (2) INVITE SDP AT2 m=text 20002 RTP/AVP 96 c=IN IP4 A.domain.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.domain.com a=rtpmap:96 t140/1000 a=recvonly m=audio 30002 RTP/AVP 0 c=IN IP4 T1.domain.com a=sendonly (5) 200 OK SDP T2A+T2B m=text 40000 RTP/AVP 96 c=IN IP4 T2.domain.com a=rtpmap:96 t140/1000 a=sendonly m=audio 40002 RTP/AVP 0 c=IN IP4 T2.domain.com a=recvonly (7) INVITE SDP T1B+T2B m=audio 30002 RTP/AVP 0 c=IN IP4 T1.domain.com a=sendonly m=audio 40002 RTP/AVP 0 c=IN IP4 T2.domain.com a=recvonly G. Camarillo et. al. [Page 12]
Internet Draft SIP February 3, 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 SIP February 3, 2004 (8) 200 OK SDP BT1+BT2 m=audio 50000 RTP/AVP 0 c=IN IP4 B.domain.com a=recvonly m=audio 50002 RTP/AVP 0 c=IN IP4 B.domain.com a=sendonly (11) 200 OK SDP T1A+T1B m=text 30000 RTP/AVP 96 c=IN IP4 T1.domain.com a=rtpmap:96 t140/1000 a=recvonly m=audio 30002 RTP/AVP 0 c=IN IP4 T1.domain.com a=sendonly (12) 200 OK SDP T2A+T2B m=text 40000 RTP/AVP 96 c=IN IP4 T2.domain.com a=rtpmap:96 t140/1000 a=sendonly m=audio 40002 RTP/AVP 0 c=IN IP4 T2.domain.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.domain.com a=rtpmap:96 t140/1000 a=sendonly m=audio 50000 RTP/AVP 0 c=IN IP4 B.domain.com a=recvonly G. Camarillo et. al. [Page 14]
Internet Draft SIP February 3, 2004 (17) ACK SDP AT2+BT2 m=text 20002 RTP/AVP 96 c=IN IP4 A.domain.com a=rtpmap:96 t140/1000 a=recvonly m=audio 50002 RTP/AVP 0 c=IN IP4 B.domain.com a=sendonly Four media streams have been established at this point: 1. Text from A to T1.domain.com:30000 2. Audio from T1 to B.domain.com:50000 3. Audio from B to T2.domain.com:40002 4. Text from T2 to A.domain.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. 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 Authors' Addresses Gonzalo Camarillo Ericsson G. Camarillo et. al. [Page 15]
Internet Draft SIP February 3, 2004 Advanced Signalling Research Lab. FIN-02420 Jorvas Finland electronic mail: Gonzalo.Camarillo@ericsson.com Eric W. Burger SnowShore Networks, Inc. Chelmsford, MA USA electronic mail: eburger@snowshore.com 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 6 Bibliography [1] 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. [2] J. Rosenberg, J. Peterson, H. Schulzrinne, and G. Camarillo, "Best current practices for third party call control in the session initiation protocol," Internet Draft draft-ietf-sipping-3pcc-06, Internet Engineering Task Force, Jan. 2004. Work in progress. [3] 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. [4] G. Camarillo, J. Holler, G. Eriksson, and H. Schulzrinne, "Grouping of m lines in SDP," internet draft, Internet Engineering Task Force, Feb. 2002. Work in progress. G. Camarillo et. al. [Page 16]
Internet Draft SIP February 3, 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 The IETF takes no position regarding the validity or scope of any intellectual property or other rights that might be claimed to pertain to the implementation or use of the technology described in G. Camarillo et. al. [Page 17]
Internet Draft SIP February 3, 2004 has made any effort to identify any such rights. Information on the IETF's procedures with respect to rights in standards-track and standards-related documentation can be found in BCP-11. Copies of claims of rights made available for publication 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 implementors or users of this specification can be obtained from the IETF Secretariat. The IETF invites any interested party to bring to its attention any copyrights, patents or patent applications, or other proprietary rights which may cover technology that may be required to practice this standard. Please address the information to the IETF Executive Director. Full Copyright Statement Copyright (c) The Internet Society (2004). All Rights Reserved. This document and translations of it may be copied and furnished to others, and derivative works that comment on or otherwise explain it or assist in its implementation may be prepared, copied, published and distributed, in whole or in part, without restriction of any kind, provided that the above copyright notice and this paragraph are included on all such copies and derivative works. However, this document itself may not be modified in any way, such as by removing the copyright notice or references to the Internet Society or other Internet organizations, except as needed for the purpose of developing Internet standards in which case the procedures for copyrights defined in the Internet Standards process must be followed, or as required to translate it into languages other than English. The limited permissions granted above are perpetual and will not be revoked by the Internet Society or its successors or assigns. This document and the information contained herein is provided on an "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIMS 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. G. Camarillo et. al. [Page 18]