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'10') == Outdated reference: draft-ietf-sipping-transc-conf has been published as RFC 5370 == Outdated reference: draft-ietf-mmusic-sdp-new has been published as RFC 4566 Summary: 9 errors (**), 0 flaws (~~), 4 warnings (==), 7 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 1 SIPPING Working Group G. Camarillo 2 Internet-Draft Ericsson 3 Expires: June 3, 2007 November 30, 2006 5 Framework for Transcoding with the Session Initiation Protocol (SIP) 6 draft-ietf-sipping-transc-framework-05.txt 8 Status of this Memo 10 By submitting this Internet-Draft, each author represents that any 11 applicable patent or other IPR claims of which he or she is aware 12 have been or will be disclosed, and any of which he or she becomes 13 aware will be disclosed, in accordance with Section 6 of BCP 79. 15 Internet-Drafts are working documents of the Internet Engineering 16 Task Force (IETF), its areas, and its working groups. Note that 17 other groups may also distribute working documents as Internet- 18 Drafts. 20 Internet-Drafts are draft documents valid for a maximum of six months 21 and may be updated, replaced, or obsoleted by other documents at any 22 time. It is inappropriate to use Internet-Drafts as reference 23 material or to cite them other than as "work in progress." 25 The list of current Internet-Drafts can be accessed at 26 http://www.ietf.org/ietf/1id-abstracts.txt. 28 The list of Internet-Draft Shadow Directories can be accessed at 29 http://www.ietf.org/shadow.html. 31 This Internet-Draft will expire on June 3, 2007. 33 Copyright Notice 35 Copyright (C) The Internet Society (2006). 37 Abstract 39 This document defines a framework for transcoding with SIP. This 40 framework includes how to discover the need for transcoding services 41 in a session and how to invoke those transcoding services. Two 42 models for transcoding services invocation are discussed: the 43 conference bridge model and the third party call control model. Both 44 models meet the requirements for SIP regarding transcoding services 45 invocation to support deaf, hard of hearing, and speech-impaired 46 individuals. 48 Table of Contents 50 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 51 2. Discovery of the Need for Transcoding Services . . . . . . . . 3 52 3. Transcoding Services Invocation . . . . . . . . . . . . . . . 4 53 3.1. Third Party Call Control Transcoding Model . . . . . . . . 5 54 3.2. Conference Bridge Transcoding Model . . . . . . . . . . . 6 55 4. Security Considerations . . . . . . . . . . . . . . . . . . . 8 56 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 57 6. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 8 58 7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 9 59 7.1. Normative References . . . . . . . . . . . . . . . . . . . 9 60 7.2. Informative References . . . . . . . . . . . . . . . . . . 10 61 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 11 62 Intellectual Property and Copyright Statements . . . . . . . . . . 12 64 1. Introduction 66 Two user agents involved in a SIP [3] dialog may find it impossible 67 to establish a media session due to a variety of incompatibilities. 68 Assuming that both user agents understand the same session 69 description format (e.g., SDP [12]), incompatibilities can be found 70 at the user agent level and at the user level. At the user agent 71 level, both terminals may not support any common codec or may not 72 support common media types (e.g., a text-only terminal and an audio- 73 only terminal). At the user level, a deaf person will not understand 74 anything said over an audio stream. 76 In order to make communications possible in the presence of 77 incompatibilities, user agents need to introduce intermediaries that 78 provide transcoding services to a session. From the SIP point of 79 view, the introduction of a transcoder is done in the same way to 80 resolve both user level and user agent level incompatibilities. So, 81 the invocation mechanisms described in this document are generally 82 applicable to any type of incompatibility related to how the 83 information that needs to be communicated is encoded. 85 Furthermore, although this framework focuses on transcoding, the 86 mechanisms described are applicable to media manipulation in 87 general. It would be possible to use them, for example, to invoke 88 a server that simply increased the volume of an audio stream. 90 This document does not describe media server discovery. That is an 91 orthogonal problem that one can address using user agent provisioning 92 or other methods. 94 The remainder of this document is organized as follows. Section 2 95 deals with the discovery of the need for transcoding services for a 96 particular session. Section 3 introduces the third party call 97 control and conference bridge transcoding invocation models, which 98 are further described in Section 3.1 and Section 3.2 respectively. 99 Both models meet the requirements regarding transcoding services 100 invocation in RFC3351 [6] to support deaf, hard of hearing, and 101 speech-impaired individuals. 103 2. Discovery of the Need for Transcoding Services 105 According to the one-party consent model defined in RFC 3238 [2], 106 services that involve media manipulation invocation are best invoked 107 by one of the end-points involved in the communication, as opposed to 108 being invoked by an intermediary in the network. Following this 109 principle, one of the end-points should be the one detecting that 110 transcoding is needed for a particular session. 112 In order to decide whether or not transcoding is needed, a user agent 113 needs to know the capabilities of the remote user agent. A user 114 agent acting as an offerer [4] typically obtains this knowledge by 115 downloading a presence document that includes media capabilities 116 (e.g., Bob is available on a terminal that only supports audio) or by 117 getting an SDP description of media capabilities as defined in RFC 118 3264 [4]. 120 Presence documents are typically received in a NOTIFY [5] request as 121 a result of a subscription. SDP media capabilities descriptions are 122 typically received in a 200 (OK) response to an OPTIONS request or in 123 a 488 (Not Acceptable Here) response to an INVITE. 125 In the absence of presence information, routing logic that involves 126 parallel forking to several user agents may make it difficult (or 127 impossible) for the caller to know which user agent will answer the 128 next call attempt. For example, a call attempt may reach the user's 129 voice mail while the next one may reach a SIP phone where the user is 130 available. If both terminating user agents have different 131 capabilities, the caller cannot know, even after the first call 132 attempt, whether or not transcoding will be necessary for the 133 session. This is a well-known SIP problem that is referred to as 134 HERFP (Heterogeneous Error Response Forking Problem). Resolving 135 HERFP is outside the scope of this document. 137 It is recommended that an offerer does not invoke transcoding 138 services before making sure that the answerer does not support the 139 capabilities needed for the session. Making wrong assumptions about 140 the answerer's capabilities can lead to situations where two 141 transcoders are introduced (one by the offerer and one by the 142 answerer) in a session that would not need any transcoding services 143 at all. 145 An example of the situation above is a call between two GSM phones 146 (without using transcoding-free operation). Both phones use a GSM 147 codec, but the speech is converted from GSM to PCM by the 148 originating MSC (Mobile Switching Center) and from PCM back to GSM 149 by the terminating MSC. 151 Note that transcoding services can be symmetric (e.g., speech-to-text 152 plus text-to-speech) or asymmetric (e.g., a one-way speech-to-text 153 transcoding for a hearing-impaired user that can talk). 155 3. Transcoding Services Invocation 157 Once the need for transcoding for a particular session has been 158 identified as described in Section 2, one of the user agents needs to 159 invoke transcoding services. 161 As stated earlier, transcoder location is outside the scope of this 162 document. So, we assume that the user agent invoking transcoding 163 services knows the URI of a server that provides them. 165 Invoking transcoding services from a server (T) for a session between 166 two user agents (A and B) involves establishing two media sessions; 167 one between A and T and another between T and B. How to invoke T's 168 services (i.e., how to establish both A-T and T-B sessions) depends 169 on how we model the transcoding service. We have considered two 170 models for invoking a transcoding service. The first is to use third 171 party call control [7], also referred to as 3pcc. The second is to 172 use a (dial-in and dial-out) conference bridge that negotiates the 173 appropriate media parameters on each individual leg (i.e., A-T and 174 T-B). 176 Section 3.1 analyzes the applicability of the third party call 177 control model and Section 3.2 analyzes the applicability of the 178 conference bridge transcoding invocation model. 180 3.1. Third Party Call Control Transcoding Model 182 In the 3pcc transcoding model, defined in [10], the user agent 183 invoking the transcoding service has a signalling relationship with 184 the transcoder and another signalling relationship with the remote 185 user agent. There is no signalling relationship between the 186 transcoder and the remote user agent, as shown in Figure 1. 188 +-------+ 189 | | 190 | T |** 191 | | ** 192 +-------+ ** 193 ^ * ** 194 | * ** 195 | * ** 196 SIP * ** 197 | * ** 198 | * ** 199 v * ** 200 +-------+ +-------+ 201 | | | | 202 | A |<-----SIP----->| B | 203 | | | | 204 +-------+ +-------+ 206 <-SIP-> Signalling 207 ******* Media 209 Figure 1: Third party call control model 211 This model is suitable for advanced endpoints that are able to 212 perform third party call control. It allows end-points to invoke 213 transcoding services on a stream basis. That is, the media streams 214 that need transcoding are routed through the transcoder while the 215 streams that do not need it are sent directly between the endpoints. 216 This model also allows to invoke one transcoder for the sending 217 direction and a different one for the receiving direction of the same 218 stream. 220 Invoking a transcoder in the middle of an ongoing session is also 221 quite simple. This is useful when session changes occur (e.g., an 222 audio session is upgraded to an audio/video session) and the end- 223 points cannot cope with the changes (e.g., they had common audio 224 codecs but no common video codecs). 226 The privacy level that is achieved using 3pcc is high, since the 227 transcoder does not see the signalling between both end-points. In 228 this model, the transcoder only has access to the information that is 229 strictly needed to perform its function. 231 3.2. Conference Bridge Transcoding Model 233 In a centralized conference, there are a number of media streams 234 between the conference server and each participant of a conference. 236 For a given media type (e.g., audio) the conference server sends, 237 over each individual stream, the media received over the rest of the 238 streams, typically performing some mixing. If the capabilities of 239 all the endpoints participating in the conference are not the same, 240 the conference server may have to send audio to different 241 participants using different audio codecs. 243 Consequently, we can model a transcoding service as a two-party 244 conference server that may change not only the codec in use, but also 245 the format of the media (e.g., audio to text). 247 Using this model, T behaves as a B2BUA (Back-to-Back User Agent) and 248 the whole A-T-B session is established as described in [11]. 249 Figure 2 shows the signalling relationships between the end-points 250 and the transcoder. 252 +-------+ 253 | |** 254 | T | ** 255 | |\ ** 256 +-------+ \\ ** 257 ^ * \\ ** 258 | * \\ ** 259 | * SIP ** 260 SIP * \\ ** 261 | * \\ ** 262 | * \\ ** 263 v * \ ** 264 +-------+ +-------+ 265 | | | | 266 | A | | B | 267 | | | | 268 +-------+ +-------+ 270 <-SIP-> Signalling 271 ******* Media 273 Figure 2: Conference bridge model 275 In the conferencing bridge model, the end-point invoking the 276 transcoder is generally involved in less signalling exchanges than in 277 the 3pcc model. This may be an important feature for end-points 278 using low bandwidth or high-delay access links (e.g., some wireless 279 accesses). 281 On the other hand, this model is less flexible than the 3pcc model. 283 It is not possible to use different transcoders for different streams 284 or for different directions of a stream. 286 Invoking a transcoder in the middle of an ongoing session or changing 287 from one transcoder to another requires the remote end-point to 288 support the Replaces [9] extension. At present, not many user agents 289 support it. 291 Simple end-points that cannot perform 3pcc and thus cannot use the 292 3pcc model, of course, need to use the conference bridge model. 294 4. Security Considerations 296 The specifications of the 3pcc and the conferencing transcoding 297 models discuss security issues directly related to the implementation 298 of those models. Additionally, there are some considerations that 299 apply to transcoding in general. 301 In a session, a transcoder has access to at least some of the media 302 exchanged between the endpoints. In order to avoid rogue transcoders 303 getting access to those media, it is recommended that endpoints 304 authenticate the transcoder. TLS [1] and S/MIME [8] can be used for 305 this purpose. 307 To achieve a higher degree of privacy, endpoints following the 3pcc 308 transcoding model can use one transcoder in one direction and a 309 different one in the other direction. This way, no single transcoder 310 has access to all the media exchanged between the endpoints. 312 The fact that transcoders need to access media exchanged between the 313 endpoints implies that endpoints cannot use end-to-end media security 314 mechanisms. Media encryption would not allow the transcoder to 315 access the media and media integrity protection would not allow the 316 transcoder to modify the media (which is obviously necessary to 317 perform the transcoding function). Nevertheless, endpoints can still 318 use media security between the transcoder and themselves. 320 5. IANA Considerations 322 This document does not contain any IANA actions. 324 6. Contributors 326 This document is the result of discussions amongst the conferencing 327 design team. The members of this team include Eric Burger, Henning 328 Schulzrinne and Arnoud van Wijk. 330 7. References 332 7.1. Normative References 334 [1] Dierks, T. and C. Allen, "The TLS Protocol Version 1.0", 335 RFC 2246, January 1999. 337 [2] Floyd, S. and L. Daigle, "IAB Architectural and Policy 338 Considerations for Open Pluggable Edge Services", RFC 3238, 339 January 2002. 341 [3] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., 342 Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: 343 Session Initiation Protocol", RFC 3261, June 2002. 345 [4] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with 346 Session Description Protocol (SDP)", RFC 3264, June 2002. 348 [5] Roach, A., "Session Initiation Protocol (SIP)-Specific Event 349 Notification", RFC 3265, June 2002. 351 [6] Charlton, N., Gasson, M., Gybels, G., Spanner, M., and A. van 352 Wijk, "User Requirements for the Session Initiation Protocol 353 (SIP) in Support of Deaf, Hard of Hearing and Speech-impaired 354 Individuals", RFC 3351, August 2002. 356 [7] Rosenberg, J., Peterson, J., Schulzrinne, H., and G. Camarillo, 357 "Best Current Practices for Third Party Call Control (3pcc) in 358 the Session Initiation Protocol (SIP)", BCP 85, RFC 3725, 359 April 2004. 361 [8] Ramsdell, B., "Secure/Multipurpose Internet Mail Extensions 362 (S/MIME) Version 3.1 Certificate Handling", RFC 3850, 363 July 2004. 365 [9] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation 366 Protocol (SIP) "Replaces" Header", RFC 3891, September 2004. 368 [10] Camarillo, G., Burger, E., Schulzrinne, H., and A. van Wijk, 369 "Transcoding Services Invocation in the Session Initiation 370 Protocol (SIP) Using Third Party Call Control (3pcc)", 371 RFC 4117, June 2005. 373 [11] Camarillo, G., "The Session Initiation Protocol (SIP) 374 Conference Bridge Transcoding Model", 375 draft-ietf-sipping-transc-conf-03 (work in progress), 376 June 2006. 378 7.2. Informative References 380 [12] Handley, M., "SDP: Session Description Protocol", 381 draft-ietf-mmusic-sdp-new-26 (work in progress), January 2006. 383 Author's Address 385 Gonzalo Camarillo 386 Ericsson 387 Hirsalantie 11 388 Jorvas 02420 389 Finland 391 Email: Gonzalo.Camarillo@ericsson.com 393 Intellectual Property Statement 395 The IETF takes no position regarding the validity or scope of any 396 Intellectual Property Rights or other rights that might be claimed to 397 pertain to the implementation or use of the technology described in 398 this document or the extent to which any license under such rights 399 might or might not be available; nor does it represent that it has 400 made any independent effort to identify any such rights. 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