Internet Engineering Task Force Sipping WG Internet Draft Taegyu Kang November 11, 2004 Doyoung Kim Expires MAY 2005 YoungsunKim ETRI Intelligent Transcoding Gateway Model for Transcoding with the Session Initiation Protocol draft-taegyukang-sipping-transc-itg-00.txt 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/1id-abstracts.txt The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on April 17, 2005. Copyright Notice Copyright (C) The Internet Society (2004). All Rights Reserved. Abstract This document introduces a new model, Intelligent Transcoding Gateway, in Framework[6] for transcoding with the Session Initiation Protocol. The model might be applied for more general-purpose services satisfying the requirements of multimedia applications Taegyu Kang Expires - May 2005 [Page 1] SIPPING November 2004 without an additional INVITE, meanwhile the existing two models, The third party call control model and The conference bridge model, are applied for the specific application requirements in support of deaf, hard of hearing and speech-impaired individuals[2]. Taegyu Kang Expires - May 2005 [Page 2] SIPPING November 2004 Table of Contents 1. Introduction...................................................4 2. Background of Intelligent Transcoding Gateway Model............4 3. Intelligent Transcoding Gateway Model..........................4 4. The Comparison of Models.......................................5 5. Security Considerations........................................6 6. References.....................................................6 Acknowledgment....................................................7 Authors' Addresses................................................7 Intellectual Property Statement...................................7 Disclaimer of Validity and Copyright Statement....................8 Taegyu Kang Expires - May 2005 [Page 3] SIPPING November 2004 1. Introduction Simple homogeneous terminals and special purpose terminals are the majority of the existing network terminals. Communication media is getting more diverse in terms of the types of network terminals. However, the era of new communication media types through the several kinds of network terminals such as hardware IP phones(POTS style, PDA style, mobile phone style), software IP phones(special purpose software phone, messenger phone, ITSP phone), POTS phones, ISDN phones and mobile phones is coming. We believe that transcoding services are the key solution for supporting these types of communication in heterogeneous networks and different capability terminals. The transcoding function converges wireline network, wireless network, internet, and broadcasting network with a horizontal structure. We can converge the networks using the Intelligent Transcoding Gateway Model for transcoding with Session Initiation Protocol[1]. 2. Background of Intelligent Transcoding Gateway Model The IETF sipping WG has been developing two models for transcoding services: Third Party Call Control Transcoding Model[4][7] and Conference Bridge Transcoding Model[5]. The Third Party Call Control Transcoding Model is suitable for advanced end points that are able to perform third party call control. The Conference Bridge Transcoding Model is suitable for a centralized conference. These two models are useful for a specific application invoked rarely to support deaf, hard of hearing and speech-impaired individuals[2]. We need another model that is useful for general applications[8] invoked frequently. The transcoding function is needed in cases of interwoking between different networks with different codecs and communicating between different applications (text and speech). So, the Intelligent Transcoding Gateway(ITG) Model is introduced in this document. 3. Intelligent Transcoding Gateway Model The Intelligent Transcoding Gateway Model has a transcoding function in a transit node between the calling party and the called party. The Intelligent Transcoding Gateway Model T receives codec information Taegyu Kang Expires - May 2005 [Page 4] SIPPING November 2004 from A and sends additional codec information to B with SIP/SDP. We assume that T has other codec that is not supported by A or B. +-------+ +-------+ +-------+ | |<----SIP---->| |<----SIP---->| | | A | | T | | B | | a |----Media----|a-b,a-c|----Media----| c | +-------+ +-------+ +-------+ Figure 1: Intelligent Transcoding Gateway Model We can explain cases where A supports codec a, B supports codec c, and T supports transcoding function with a to b and a to c. A caller party sends an INVITE message with SDP[3] codec a to T. After receiving from A, T sends an INVITE message with SDP A(a)+T(b+c) to called party B. After receiving of the INVITE message from T, called party B responds with an OK message to T with SDP B(c). T sends an OK message with SDP TA(a) to caller party A. This procedure can provide to call setup and media communication between the caller party with codec a and the called party with codec c. The Intelligent Transcoding Gateway Model can be used for heterogeneous networks and applications. The examples of heterogeneous networks are ones between PSTN and 3GPP/3GPP2, PSTN and Internet Telephony, or Internet Telephony and 3GPP/3GPP2. The examples of heterogeneous applications are to communicate in real time between voice and text, video/voice and voice, or video/voice and text. 4. The Comparison of Models The models are compared with signaling traffic overhead. The Conference bridge model and the Third party transcoding server model need an additional signaling procedure for transcoding service whenever they are invoked. The additional signaling procedure includes an additional INVITE procedure with SIP/SDP between invoking point and serving point. Conference bridge model and Third party transcoding server model are useful for environments that rarely occur for the transcoding service. The Intelligent Transcoding Gateway Model has an overhead of additional codec information insertion at the INVITE message without an additional INVITE. The overhead for codec information insertion at T is less than that for an additional INVITE signaling procedure. Taegyu Kang Expires - May 2005 [Page 5] SIPPING November 2004 The Intelligent Transcoding Gateway Model is the best solution from the point of ITSP(Internet Telephony Service Provider). In an ITSP environment, all call control messages should be controlled by the ITSP server, ITSP would service to terminals with all kinds of codecs. ITSP's media server has a transcoding function and call control by itself. This model has the advantage of no additional messages between nodes and no modification for call set up standard. The transcoding technology with the Intelligent Transcoding Gateway Model is one of the solutions for network convergence of PSTN, mobile network, internet telephony, and digital broadcasting TV. 5. Security Considerations This document does not introduce any new security considerations. 6. 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. [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] J. Rosenberg and H. Schulzrinne, "An offer/answer model with session description protocol (SDP)," RFC 3264, Internet Engineering Task Force, June 2002. [4] J. Rosenberg, J. Peterson, H. Schulzrinne, and G. Camarillo, "Best current practices for third party call control in the session initiation protocol," RFC 3725, Internet Engineering Task Force, Jan. 2004. Work in progress. [5] 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. [6] G. Camarillo, "Framework for Transcoding with the Session Initiation Protocol," Internet Draft draft-ietf-transc- framework-00.txt, Internet Engineering Task Force, Feb. 2004. Work in progress. [7] G. Camarillo, E. Burger, H. Schulzrinne, A. van Wijk, "Transcoding Services Invocation in the Session Initiation Protocol (SIP) Using Third Party Call Control (3pcc)," Sep 2004 Work in progress. Taegyu Kang Expires - May 2005 [Page 6] SIPPING November 2004 [8] A. van Wijk, "Framework of requirements for real-time text conversation using SIP," Internet Draft draft-vanwijk-sipping- toip-00, Internet Engineering Task Force, Jan. 2004. Work in progress Acknowledgment Author's Addresses Tae-Gyu Kang ETRI VoIP Service Technology Team 161 Kajeong Yousung Taejeon 305-350 South Korea Electronic mail: tgkang@etri.re.kr Do-Young Kim VoIP Service Technology Team 161 Kajeong Yousung Taejeon 305-350 South Korea Electronic mail: dyk@etri.re.kr Young-Sun Kim Broadband convergence Network Service Research Group 161 Kajeong Yousung Taejeon 305-350 South Korea Electronic mail: sunkim@etri.re.kr 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 procedures with respect to rights in RFC 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 Taegyu Kang Expires - May 2005 [Page 7] SIPPING November 2004 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. 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. Taegyu Kang Expires - May 2005 [Page 8]