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'22' on line 997 looks like a reference -- Missing reference section? '24' on line 1179 looks like a reference Summary: 3 errors (**), 0 flaws (~~), 1 warning (==), 35 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 1 SIPPING Workgroup A. van Wijk, Editor 2 Internet Draft G. Gybels, Editor 3 Category: Informational August 30, 2006 4 Expires: March 3, 2007 6 Framework for real-time text over IP using the Session Initiation 7 Protocol (SIP) 8 draft-ietf-sipping-toip-07.txt 10 Status of this Memo 12 By submitting this Internet-Draft, each author represents that any 13 applicable patent or other IPR claims of which he or she is aware have 14 been or will be disclosed, and any of which he or she becomes aware 15 will be disclosed, in accordance with Section 6 of BCP 79. 17 Internet-Drafts are working documents of the Internet Engineering Task 18 Force (IETF), its areas, and its working groups. Note that other 19 groups may also distribute working documents as Internet-Drafts. 21 Internet-Drafts are draft documents valid for a maximum of six months 22 and may be updated, replaced, or obsoleted by other documents at any 23 time. It is inappropriate to use Internet-Drafts as reference 24 material or to cite them other than as "work in progress." 26 The list of current Internet-Drafts can be accessed at 27 http://www.ietf.org/ietf/1id-abstracts.txt. 29 The list of Internet-Draft Shadow Directories can be accessed at 30 http://www.ietf.org/shadow.html. 32 This Internet-Draft will expire on March 3, 2007. 34 Copyright Notice 36 Copyright (C) The Internet Society (2006). 38 Abstract 40 This document lists the essential requirements for real-time Text- 41 over-IP (ToIP) and defines a framework for implementation of all 42 required functions based on the Session Initiation Protocol (SIP) and 43 the Real-Time Transport Protocol (RTP). This includes interworking 44 between Text-over-IP and existing text telephony on the PSTN and other 45 networks. 47 Table of Contents 49 1. Introduction....................................................2 50 2. Scope...........................................................3 51 3. Terminology.....................................................3 52 4. Definitions.....................................................4 53 5. Requirements....................................................6 54 5.1 General requirements for ToIP................................6 55 5.2 Detailed requirements for ToIP...............................7 57 van Wijk, et al. Expires March 3, 2007 [Page 1] 58 5.2.1 Session set-up and control requirements..................7 59 5.2.2 Transport requirements...................................8 60 5.2.3 Transcoding service requirements.........................9 61 5.2.4 Presentation and User control requirements..............10 62 5.2.5 Interworking requirements...............................11 63 5.2.5.1 PSTN Interworking requirements......................12 64 5.2.5.2 Cellular Interworking requirements..................12 65 5.2.5.3 Instant Messaging Interworking requirements.........12 66 6. Implementation Framework.......................................13 67 6.1 General implementation framework............................13 68 6.2 Detailed implementation framework...........................13 69 6.2.1 Session control and set-up..............................13 70 6.2.1.1 Pre-session set-up..................................13 71 6.2.1.2 Session Negotiations................................14 72 6.2.2 Transport...............................................15 73 6.2.3 Transcoding services....................................15 74 6.2.4 Presentation and User control functions.................16 75 6.2.4.1 Progress and status information.....................16 76 6.2.4.2 Alerting............................................16 77 6.2.4.3 Text presentation...................................16 78 6.2.4.4 File storage........................................16 79 6.2.5 Interworking functions..................................16 80 6.2.5.1 PSTN Interworking...................................18 81 6.2.5.2 Mobile Interworking.................................19 82 6.2.5.2.1 Cellular "No-gain"..............................19 83 6.2.5.2.2 Cellular Text Telephone Modem (CTM).............19 84 6.2.5.2.3 Cellular "Baudot mode"..........................19 85 6.2.5.2.4 Mobile data channel mode........................19 86 6.2.5.2.5 Mobile ToIP.....................................20 87 6.2.5.3 Instant Messaging Interworking......................20 88 6.2.5.4 Multi-functional Combination gateways...............21 89 6.2.5.5 Character set transcoding...........................21 90 7. Further recommendations for implementers and service providers.22 91 7.1 Access to Emergency services................................22 92 7.2 Home Gateways or Analog Terminal Adapters...................22 93 7.3 User Mobility...............................................23 94 7.4 Firewalls and NATs..........................................23 95 7.5 Quality of Service..........................................23 96 8. IANA Considerations............................................23 97 9. Security Considerations........................................23 98 10. Authors' Addresses.............................................23 99 11. Contributors...................................................24 100 12. References.....................................................24 101 12.1 Normative references........................................24 102 12.2 Informative references......................................26 104 1. Introduction 106 For many years, real-time text has been in use as a medium for 107 conversational, interactive dialogue between users in a similar way 108 to how voice telephony is used. Such interactive text is different 109 from messaging and semi-interactive solutions like Instant Messaging 111 van Wijk, et al. Expires March 3, 2007 [Page 2] 112 in that it offers an equivalent conversational experience to users 113 who cannot, or do not wish to, use voice. It therefore meets a 114 different set of requirements from other text-based solutions already 115 available on IP networks. 117 Traditionally, deaf, hard of hearing and speech-impaired people are 118 amongst the most prolific users of real-time, conversational, 119 text but, because of its interactivity, it is becoming popular amongst 120 mainstream users as well. Real-time text conversation can be combined 121 with other conversational media like video or voice. 123 This document describes how existing IETF protocols can be used to 124 implement a Text-over-IP solution (ToIP). This ToIP framework is 125 specifically designed to be compatible with Voice-over-IP (VoIP), 126 Video-over-IP and Multimedia-over-IP (MoIP) environments, as well as 127 meeting the requirements of deaf, hard of hearing and speech-impaired 128 users as described in RFC3351 [2] and of mainstream users. 130 ToIP also offers an IP equivalent of analog text telephony services as 131 used by deaf, hard of hearing, speech-impaired and mainstream users. 133 The Session Initiation Protocol (SIP) [3] is the protocol of choice 134 for control of Multimedia communications and Voice-over-IP (VoIP) in 135 particular. It offers all the necessary control and signalling 136 required for the ToIP framework. 138 The Real-Time Transport Protocol (RTP) [4] is the protocol of choice 139 for real-time data transmission, and its use for real-time text 140 payloads is described in RFC4103 [5]. 142 This document defines a framework for ToIP to be used either by itself 143 or as part of integrated, multi-media services, including Total 144 Conversation [6]. 146 2. Scope 148 This document defines a framework for the implementation of real-time 149 ToIP, either stand-alone or as a part of multimedia services, 150 including Total Conversation [6]. It provides the: 151 a. requirements for real-time text; 152 b. requirements for ToIP interworking; 153 c. description of ToIP implementation using SIP and RTP; 154 d. description of ToIP interworking with other text services. 156 3. Terminology 158 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 159 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 160 "OPTIONAL" in this document are to be interpreted as described in 161 BCP 14, RFC 2119 [7] and indicate requirement levels for compliant 162 implementations. 164 van Wijk, et al. Expires March 3, 2007 [Page 3] 165 4. Definitions 167 Audio bridging: a function of an audio media bridge server, gateway or 168 relay service that sends to each destination the combination of audio 169 from all participants in a conference excluding the participant(s) at 170 that destination. At the RTP level, this is an instance of the mixer 171 function as defined in RFC 3550 [4]. 173 Cellular: a telecommunication network that has wireless access and can 174 support voice and data services over very large geographical areas. 175 Also called Mobile. 177 Full duplex: media is sent independently in both directions. 179 Half duplex: media can only be sent in one direction at a time or, 180 if an attempt to send information in both directions is made, errors 181 may be introduced into the presented media. 183 Interactive text: another term for real-time text, as defined below. 185 Real-time text: a term for real time transmission of text in a 186 character-by-character fashion for use in conversational services, 187 often as a text equivalent to voice based conversational services. 188 Conversational text is defined in the ITU-T Framework for multimedia 189 services, Recommendation F.700 [25]. 191 Text gateway: a function that transcodes between different forms of 192 text transport methods, e.g., between ToIP in IP networks and Baudot 193 or ITU-T V.21 text telephony in the PSTN. 195 Textphone: also "text telephone". A terminal device that allows end- 196 to-end real-time text communication using analog transmission. A 197 variety of PSTN textphone protocols exists world-wide. A textphone can 198 often be combined with a voice telephone, or include voice 199 communication functions for simultaneous or alternating use of text 200 and voice in a call. 202 Text bridging: a function of the text media bridge server, gateway 203 (including transcoding gateways) or relay service analogous to that of 204 audio bridging as defined above, except that text is the medium of 205 conversation. 207 Text relay service: a third-party or intermediary that enables 208 communications between deaf, hard of hearing and speech-impaired 209 people and voice telephone users by translating between voice and 210 real-time text in a call. 212 Text telephony: analog textphone service. 214 Total Conversation: a multimedia service offering real time 215 conversation in video, real-time text and voice according to 217 van Wijk, et al. Expires March 3, 2007 [Page 4] 218 interoperable standards. All media streams flow in real time. (See 219 ITU-T F.703 "Multimedia conversational services" [6].) 221 Transcoding service: a service provided by a third-party User Agent 222 that transcodes one stream into another. Transcoding can be done by 223 human operators, in an automated manner, or by a combination of both 224 methods. Within this document the term particularly applies to 225 conversion between different types of media. A text relay service is 226 an example of a transcoding service that converts between real-time 227 text and audio. 229 TTY: originally, an abbreviation for "teletype". Often used in North 230 America as an alternative designation for a text telephone or 231 textphone. Also called TDD, Telecommunication Device for the Deaf. 233 Video relay service: a service that enables communications between 234 deaf and hard of hearing people and hearing persons with voice 235 telephones by translating between sign language and spoken language in 236 a call. 238 Acronyms: 240 2G Second generation cellular (mobile) 241 2.5G Enhanced second generation cellular (mobile) 242 3G Third generation cellular (mobile) 243 ATA Analog Telephone Adaptor 244 CDMA Code Division Multiple Access 245 CLI Calling Line Identification 246 CTM Cellular Text Telephone Modem 247 ENUM E.164 number storage in DNS (see RFC3761) 248 GSM Global System for Mobile Communications 249 ISDN Integrated Services Digital Network 250 ITU-T International Telecommunications Union-Telecommunications 251 Standardisation Sector 252 NAT Network Address Translation 253 PSTN Public Switched Telephone Network 254 RTP Real Time Transport Protocol 255 SDP Session Description Protocol 256 SIP Session Initiation Protocol 257 SRTP Secure Real Time Transport Protocol 258 TDD Telecommunication Device for the Deaf 259 TDMA Time Division Multiple Access 260 TTY Analog textphone (Teletypewriter) 261 ToIP Real-time Text over Internet Protocol 262 URI Uniform Resource Identifier 263 UTF-8 Universal Transfer Format-8 264 VCO/HCO Voice Carry Over/Hearing Carry Over 265 VoIP Voice over Internet Protocol 267 van Wijk, et al. Expires March 3, 2007 [Page 5] 268 5. Requirements 270 The framework described in section 6 defines a real-time text-based 271 conversational service that is the text equivalent of voice based 272 telephony. This section describes the requirements that the framework 273 is designed to meet and the functionality it should offer. 275 5.1 General requirements for ToIP 277 Any framework for ToIP must be designed to meet the requirements of 278 RFC3351 [2]. A basic requirement is that it must provide a 279 standardized way for offering real-time text-based, conversational 280 services that can be used as an equivalent to voice telephony by deaf, 281 hard of hearing speech-impaired and mainstream users. 283 It is important to understand that real-time text conversations are 284 significantly different from other text-based communications like 285 email or Instant Messaging. Real-time text conversations deliver an 286 equivalent mode to voice conversations by providing transmission of 287 text character by character as it is entered, so that the conversation 288 can be followed closely and immediate interaction take place. 290 Store-and-forward systems like email or messaging on mobile networks 291 or non-streaming systems like instant messaging are unable to provide 292 that functionality. In particular, they do not allow for smooth 293 communication through a Text Relay Service. 295 In order to make ToIP the text equivalent of voice services, ToIP 296 needs to offer equivalent features in terms of conversationality to 297 those provided by voice. To achieve that, ToIP needs to: 299 a. offer real-time transport and presentation of the conversation; 300 b. provide simultaneous transmission in both directions; 301 c. support both point-to-point and multipoint communication; 302 d. allow other media, like audio and video, to be used in conjunction 303 with ToIP; 304 e. ensure that the real-time text service is always available. 306 Real-time text is a useful subset of Total Conversation as defined in 307 ITU-T F.703 [6]. Total Conversation allows participants to use 308 multiple modes of communication during the conversation, either at the 309 same time or by switching between modes, e.g., between real-time text 310 and audio. 312 Deaf, hard-of-hearing and mainstream users may invoke ToIP services 313 for many different reasons: 315 - because they are in a noisy environment, e.g., in a machine room of 316 a factory where listening is difficult; 317 - because they are busy with another call and want to participate in 318 two calls at the same time; 320 van Wijk, et al. Expires March 3, 2007 [Page 6] 321 - for implementing text and/or speech recording services (e.g., text 322 documentation/ audio recording) for legal purposes, for clarity or 323 for flexibility; 324 - to overcome language barriers through speech translation and/or 325 transcoding services; 326 - because of hearing loss, deafness or tinnitus as a result of the 327 aging process or for any other reason, creating a need to replace or 328 complement voice with real-time text in conversational sessions. 330 In many of the above examples, real-time text may accompany speech. 331 The text could be displayed side by side, or in a manner similar to 332 subtitling in broadcasting environments, or in any other suitable 333 manner. This could occur with users who are hard of hearing and also 334 for mixed media calls with both hearing and deaf people participating 335 in the call. 337 A ToIP user may wish to call another ToIP user, join a conference 338 session involving several users, or initiate or join a multimedia 339 session, such as a Total Conversation session. 341 5.2 Detailed requirements for ToIP 343 The following sections list individual requirements for ToIP. Each 344 requirement has been given a unique identifier (R1, R2, etc). Section 345 6 (Implementation Framework) describes how to implement ToIP based on 346 these requirements and using existing protocols and techniques. 348 The requirements are organized under the following headings: 349 - session set-up and session control; 350 - transport; 351 - use of transcoding services; 352 - presentation and user control; 353 - interworking. 355 5.2.1 Session set-up and control requirements 357 Conversations could be started using a mode other than real-time text. 358 Simultaneous or alternating voice and real-time text is used by a 359 large number of people who can send voice but must receive text (due 360 to a hearing impairment), or who can hear but must send text (due to a 361 speech impairment). 363 R1: It SHOULD be possible to start conversations in any mode (real- 364 time text, voice, video) or combination of modes. 366 R2: It MUST be possible for the users to switch to real-time text, or 367 add real-time text as an additional modality, during the conversation. 369 R3: Systems supporting ToIP MUST allow users to select any of the 370 supported conversation modes at any time, including in mid- 371 conversation. 373 van Wijk, et al. Expires March 3, 2007 [Page 7] 374 R4: Systems SHOULD allow the user to specify a preferred mode of 375 communication in each direction, with the ability to fall back to 376 alternatives that the user has indicated are acceptable. 378 R5: If the user requests simultaneous use of real-time text and audio, 379 and this is not possible because of constraints in the network, the 380 system SHOULD try to establish text only communication. 382 R6: If the user has expressed a preference for real-time text, 383 establishment of a connection including real-time text MUST have 384 priority over other outcomes of the session setup. 386 R7: It MUST be possible to use real-time text in conferences both as a 387 medium of discussion between individual participants (for example, for 388 sidebar discussions in real-time text while listening to the main 389 conference audio) and for central support of the conference with 390 real-time text interpretation of speech. 392 R8: Session set up and negotiation of modalities MUST allow users to 393 specify the language of the real-time text to be used. (It is 394 RECOMMENDED that similar functionality be provided for the video part 395 of the conversation, i.e. to specify the sign language being used). 397 R9: Where certain session services are available for the audio media 398 part of a session, these functions MUST also be supported for the 399 real-time text media part of the same session. For example, call 400 transfer must act on all media in the session. 402 5.2.2 Transport requirements 404 ToIP will often be used to access a relay service [I], allowing real- 405 time text users to communicate with voice users. With relay services, 406 as well as in direct user-to-user conversation, it is crucial that 407 text characters are sent as soon as possible after they are entered. 408 While buffering may be done to improve efficiency, the delays SHOULD 409 be kept minimal. In particular, buffering of whole lines of text will 410 not meet character delay requirements. 412 R10: Characters must be transmitted soon after entry of each character 413 so that the maximum delay requirement can be met. An end-to-end delay 414 time of one second is regarded as good, while users note and 415 appreciate shorter delays, down to 300ms. A delay of up to two seconds 416 is possible to use. 418 R11: Real-time text transmission from a terminal SHALL be performed 419 character by character as entered, or in small groups of characters, 420 so that no character is delayed from entry to transmission by more 421 than 300 milliseconds. 423 R12: It MUST be possible to transmit characters at a rate sufficient 424 to support fast human typing as well as speech-to-text methods of 426 van Wijk, et al. Expires March 3, 2007 [Page 8] 427 generating real-time text. A rate of 30 characters per second is 428 regarded as sufficient. 430 R13: A ToIP service MUST be able to deal with international character 431 sets. 433 R14: Where it is possible, loss or corruption of real-time text during 434 transport SHOULD be detected and the user should be informed. 436 R15: Transport of real-time text SHOULD be as robust as possible, so 437 as to minimize loss of characters. 439 R16: It SHOULD be possible to send and receive real-time text 440 simultaneously. 442 5.2.3 Transcoding service requirements 444 If the User Agents of different participants indicate that there is an 445 incompatibility between their capabilities to support certain media 446 types, e.g. one User Agent only offering T.140 over IP as described in 447 RFC4103 [5] and the other one only supporting audio, the user might 448 want to invoke a transcoding service. 450 Some users may indicate their preferred modality to be audio while 451 others may indicate real-time text. In this case, transcoding services 452 might be needed for text-to-speech (TTS) and speech-to-text (STT). 453 Other examples of possible scenarios for including a relay service in 454 the conversation are: text bridging after conversion from speech, 455 audio bridging after conversion from real-time text, etc. 457 A number of requirements, motivations and implementation guidelines 458 for relay service invocation can be found in RFC 3351 [2]. 460 R17: It MUST be possible for users to invoke a transcoding service 461 where such service is available. 463 R18: It MUST be possible for users to indicate their preferred 464 modality (e.g. ToIP). 466 R19: It MUST be possible to negotiate the requirements for transcoding 467 services in real time in the process of setting up a call. 469 R20: It MUST be possible to negotiate the requirements for transcoding 470 services in mid-call, for the immediate addition of those services to 471 the call. 473 R21: Communication between the end participants SHOULD continue after 474 the addition or removal of a text relay service, and the effect of the 475 change should be limited in the users' perception to the direct effect 476 of having or not having the transcoding service in the connection. 478 van Wijk, et al. Expires March 3, 2007 [Page 9] 479 R22: When setting up a session, it MUST be possible for a user to 480 specify the type of relay service requested (e.g., speech to text or 481 text to speech). The specification of a type of relay MUST include a 482 language specifier. 484 R23: It SHOULD be possible to route the session to a preferred relay 485 service even if the user invokes the session from another region or 486 network than that usually used. 488 R24: It is RECOMMENDED that ToIP implementations make the invocation 489 and use of relay services as easy as possible. 491 5.2.4 Presentation and User control requirements 493 A user should never be in doubt about the status of the session, even 494 if the user is unable to make use of the audio or visual indication. 495 For example, tactile indications could be used by deafblind 496 individuals. 498 R25: User Agents for ToIP services MUST have alerting methods (e.g., 499 for incoming sessions) that can be used by deaf and hard of hearing 500 people or provide a range of alternative, but equivalent, alerting 501 methods that can be selected by all users, regardless of their 502 abilities. 504 R26: Where real-time text is used in conjunction with other media, 505 exposure of user control functions through the User Interface needs to 506 be done in an equivalent manner for all supported media. For example, 507 it must be possible for the user to select between audio, visual or 508 tactile prompts, or all must be supplied. 510 R27: If available, identification of the originating party (for 511 example in the form of a URI or a CLI) MUST be clearly presented to 512 the user in a form suitable for the user BEFORE the session invitation 513 is answered. 515 R28: When a session invitation involving ToIP originates from a PSTN 516 text telephone (e.g. transcoded via a text gateway), this SHOULD be 517 indicated to the user. The ToIP client MAY adjust the presentation of 518 the real-time text to the user as a consequence. 520 R29: An indication SHOULD be given to the user when real-time text is 521 available during the call, even if it is not invoked at call setup 522 (e.g. when only voice and/or video is used initially). 524 R30: The user MUST be informed of any change in modalities. 526 R31: Users MUST be presented with appropriate session progress 527 information at all times. 529 R32: Systems for ToIP SHOULD support an answering machine function, 530 equivalent to answering machines on telephony networks. 532 R33: If an answering machine function is supported, it MUST support at 533 least 160 characters for the greeting message. It MUST support 534 incoming text message storage of a minimum of 4096 characters, 535 although systems MAY support much larger storage. It is RECOMMENDED 536 that systems support storage of at least 20 incoming messages of up to 537 16000 characters per message. 539 R34: When the answering machine is activated, user alerting SHOULD 540 still take place. The user SHOULD be allowed to monitor the auto- 541 answer progress and where this is provided the user SHOULD be allowed 542 to intervene during any stage of the answering machine procedure and 543 take control of the session. 545 R35: It SHOULD be possible to save the text portion of a conversation. 547 R36: The presentation of the conversation SHOULD be done in such a way 548 that users can easily identify which party generated any given portion 549 of text. 551 R37: ToIP SHOULD handle characters such as new line, erasure and 552 alerting during a session as specified in ITU-T T.140 [9]. 554 5.2.5 Interworking requirements 556 There is a range of existing real-time text services. There is also a 557 range of network technologies that could support real-time text 558 services. 560 Real-time/interactive texting facilities exist already in various 561 forms and on various networks. In the PSTN, they are commonly referred 562 to as text telephony. 564 Text gateways are used for converting between different protocols for 565 text conversation. They can be used between networks or within 566 networks where different transport technologies are used. 568 R38: ToIP SHOULD provide interoperability with text conversation 569 features in other networks, for instance the PSTN. 571 R39: When communicating via a gateway to other networks and protocols, 572 the ToIP service SHOULD support the functionality for alternating or 573 simultaneous use of modalities as offered by the interworking network. 575 R40: Calling party identification information, such as CLI, MUST be 576 passed by gateways and converted to an appropriate form if required. 578 R41: When interworking with other networks and services, the ToIP 579 service SHOULD provide buffering mechanisms to deal with delays in 580 call setup, differences in transmission speeds and/or to interwork 581 with half duplex services. 583 5.2.5.1 PSTN Interworking requirements 585 Analog text telephony is used in many countries, mainly by deaf, hard 586 of hearing and speech-impaired individuals. 588 R42: ToIP services MUST provide interworking with PSTN legacy text 589 telephony devices. 591 R43: When interworking with PSTN legacy text telephony services, 592 alternating text and voice function MAY be supported. (Called "voice 593 carry over (VCO) and hearing carry over (HCO)"). 595 5.2.5.2 Cellular Interworking requirements 597 As mobile communications have been adopted widely, various solutions 598 for real-time texting while on the move were developed. ToIP services 599 should provide interworking with such services as well. 601 Alternative means of transferring the Text telephony data have been 602 developed when TTY services over cellular were mandated by the FCC in 603 the USA. They are the a) "No-gain" codec solution, and b) the Cellular 604 Text Telephony Modem (CTM) solution [8] both collectively called 605 "Baudot mode" solution in the USA. 607 The GSM and 3G standards from 3GPP make use of the CTM modem in the 608 voice channel for text telephony. However, implementations also exist 609 that use the data channel to provide such functionality. Interworking 610 with these solutions should be done using text gateways that set up 611 the data channel connection at the GSM side and provide ToIP at the 612 other side. 614 R44: a ToIP service SHOULD provide interworking with mobile text 615 conversation services. 617 5.2.5.3 Instant Messaging Interworking requirements 619 Many people use Instant Messaging to communicate via the Internet 620 using text. Instant Messaging usually transfers blocks of text rather 621 than streaming as is used by ToIP. Usually a specific action is 622 required by the user to activate transmission, such as pressing the 623 ENTER key or a send button. As such, it is not a replacement for ToIP 624 and in particular does not meet the needs for real time conversations 625 including those of deaf, hard of hearing and speech-impaired users as 626 defined in RFC 3351 [2]. It is less suitable for communications 627 through a relay service [I]. 629 The streaming nature of ToIP provides a more direct conversational 630 user experience and, when given the choice, users may prefer ToIP. 632 R45: a ToIP service MAY provide interworking with Instant Messaging 633 services. 635 6. Implementation Framework 637 This section describes an implementation framework for ToIP that meets 638 the requirements and offers the functionality as set out in section 5. 639 The framework presented here uses existing standards that are already 640 commonly used for voice based conversational services on IP networks. 642 6.1 General implementation framework 644 This framework specifies the use of the Session Initiation Protocol 645 (SIP) [3] to set up, control and tear down the connections between 646 ToIP users whilst the media is transported using the Real-Time 647 Transport Protocol (RTP) [4] as described in RFC 4103 [5]. 649 RFC 4504 describes how to implement support for real-time text in SIP 650 telephony devices [23]. 652 6.2 Detailed implementation framework 654 6.2.1 Session control and set-up 656 ToIP services MUST use the Session Initiation Protocol (SIP) [3] for 657 setting up, controlling and terminating sessions for real-time text 658 conversation with one or more participants and possibly including 659 other media like video or audio. The session description protocol 660 (SDP) used in SIP to describe the session is used to express the 661 attributes of the session and to negotiate a set of compatible media 662 types. 664 SIP [3] allows participants to negotiate all media including real-time 665 text conversation [5]. ToIP services can provide the ability to set up 666 conversation sessions from any location as well as provision for 667 privacy and security through the application of standard SIP 668 techniques. 670 6.2.1.1 Pre-session set-up 672 The requirements of the user to be reached at a consistent address and 673 to store preferences for evaluation at session setup are met by pre- 674 session setup actions. That includes storing of registration 675 information in the SIP registrar, to provide information about how a 676 user can be contacted. This will allow sessions to be set up rapidly 677 and with proper routing and addressing. 679 The need to use real-time text as a medium of communications can be 680 expressed by users during registration time. Two situations need to be 681 considered in the pre-session setup environment: 683 a. User Preferences: It MUST be possible for a user to indicate a 684 preference for real-time text by registering that preference with a 685 SIP server that is part of the ToIP service. 687 b. Server support of User Preferences: SIP servers that support ToIP 688 services MUST have the capability to act on calling user 689 preferences for real-time text in order to accept or reject the 690 session. The actions taken can be based on the called users 691 preferences defined as part of the pre-session setup registration. 692 For example, if the user is called by another party, and it is 693 determined that a transcoding server is needed, the session should 694 be re-directed or otherwise handled accordingly. 696 The ability to include a transcoding service MUST NOT require user 697 registration in any specific SIP registrar, but MAY require 698 authorisation of the SIP registrar to invoke the service. 700 A point-to-point session takes place between two parties. For ToIP, 701 one or both of the communicating parties will indicate real-time text 702 as a possible or preferred medium for conversation using SIP in the 703 session setup. 705 The following features MAY be implemented to facilitate the session 706 establishment using ToIP: 708 a. Caller Preferences: SIP headers (e.g., Contact) [11] can be used to 709 show that real-time text is the medium of choice for 710 communications. 712 b. Called Party Preferences [12]: The called party being passive can 713 formulate a clear rule indicating how a session should be handled 714 either using real-time text as a preferred medium or not, and 715 whether a designated SIP proxy needs to handle this session or it 716 will be handled in the SIP User Agent. 718 c. SIP Server support for User Preferences: It is RECOMMENDED that SIP 719 servers also handle the incoming sessions in accordance with 720 preferences expressed for real-time text. The SIP Server can also 721 enforce ToIP policy rules for communications (e.g. use of the 722 transcoding server for ToIP). 724 6.2.1.2 Session Negotiations 726 The Session Description Protocol (SDP) used in SIP [3] provides the 727 capabilities to indicate real-time text as a medium in the session 728 setup. RFC 4103 [5] uses the RTP payload types "text/red" and 729 "text/t140" for support of ToIP which can be indicated in the SDP as a 730 part of the SIP INVITE, OK and SIP/200/ACK media negotiations. In 731 addition, SIPs offer/answer model [13] can also be used in conjunction 732 with other capabilities including the use of a transcoding server for 733 enhanced session negotiations [14,15,16]. 735 6.2.2 Transport 737 ToIP services MUST support the Real-Time Transport Protocol (RTP) [4] 738 according to the specification of RFC 4103 [4] for the transport of 739 real-time text between participants. 741 RFC 4103 describes the transmission of T.140 [9] real-time text on IP 742 networks. 744 In order to enable the use of international character sets, the 745 transmission format for real-time text conversation SHALL be UTF-8 746 [17], in accordance with ITU-T T.140. 748 If real-time text is detected to be missing after transmission, there 749 SHOULD be a "text loss" indication in the real-time text as specified 750 in T.140 Addendum 1 [9]. 752 The redundancy method of RFC 4103 [5] SHOULD be used to significantly 753 increase the reliability of the real-time text transmission. A 754 redundancy level using 2 generations gives very reliable results and 755 is therefore strongly RECOMMENDED. 757 Real-time text capability is announced in SDP by a declaration similar 758 to this example: 760 m=text 11000 RTP/AVP 100 98 761 a=rtpmap:98 t140/1000 762 a=rtpmap:100 red/1000 763 a=fmtp:100 98/98/98 765 By having this single coding and transmission scheme for real-time 766 text defined in the SIP session control environment, the opportunity 767 for interoperability is optimized. However, if good reasons exist, 768 other transport mechanisms MAY be offered and used for the T.140 coded 769 text provided that proper negotiation is introduced, but RFC 4103 [5] 770 transport MUST be used as both the default and the fallback transport. 772 6.2.3 Transcoding services 774 Invocation of a transcoding service MAY happen automatically when the 775 session is being set up based on any valid indication or negotiation 776 of supported or preferred media types. A transcoding framework 777 document using SIP [14] describes invoking relay services, where the 778 relay acts as a conference bridge or uses the third party control 779 mechanism. ToIP implementations SHOULD support this transcoding 780 framework. 782 6.2.4 Presentation and User control functions 784 6.2.4.1 Progress and status information 786 Session progress information SHOULD use simple language so that as 787 many users as possible can understand it. The use of jargon or 788 ambiguous terminology SHOULD be avoided. It is RECOMMENDED that text 789 information be used together with icons to symbolise the session 790 progress information. 792 In summary, it SHOULD be possible to observe indicators about: 793 - Incoming session 794 - Availability of real-time text, voice and video channels 795 - Session progress 796 - Incoming real-time text 797 - Any loss in incoming real-time text 798 - Typed and transmitted real-time text. 800 6.2.4.2 Alerting 802 For users who cannot use the audible alerter for incoming sessions, it 803 is RECOMMENDED to include a tactile as well as a visual indicator. 805 Among the alerting options are alerting by the User Agent's User 806 Interface and specific alerting User Agents registered to the same 807 registrar as the main User Agent. 809 It should be noted that external alerting systems exist and one common 810 interface for triggering the alerting action is a contact closure 811 between two conductors. 813 6.2.4.3 Text presentation 815 Requirement R32 states that, in the display of text conversations, 816 users must be able to distinguish easily between different speakers. 817 This could be done using color, positioning of the text (i.e. incoming 818 real-time text and outgoing real-time text in different display 819 areas), by in-band identifiers of the parties or by a combination of 820 any of these techniques. 822 6.2.4.4 File storage 824 Requirement R31 recommends that ToIP systems allow the user to save 825 text conversations. This SHOULD be done using a standard file format. 826 For example: a UTF-8 text file in XHTML format [18] including 827 timestamps, party names (or addresses) and the conversation text. 829 6.2.5 Interworking functions 831 A number of systems for real-time text conversation already exist as 832 well as a number of message oriented text communication systems. 834 Interoperability is of interest between ToIP and some of these 835 systems. 837 Interoperation of half-duplex and full-duplex protocols, and between 838 protocols that have different data rates, may require text buffering. 839 Some intelligence will be needed to determine when to change direction 840 when operating in half-duplex mode. Identification may be required of 841 half-duplex operation either at the "user" level (ie. users must 842 inform each other) or at the "protocol" level (where an indication 843 must be sent back to the Gateway). However, special care needs to be 844 taken to provide the best possible real-time performance. 846 Buffering schemes SHOULD be dimensioned to adjust for receiving at 30 847 characters per second and transmitting at 6 characters per second for 848 up to 4 minutes (i.e. less than 3000 characters). 850 When converting between simultaneous voice and text on the IP side, 851 and alternating voice and text on the other side of a gateway, a 852 conflict can occur if the IP user transmits both audio and text at the 853 same time. In such situations, text transmission SHOULD have 854 precedence, so that while text is transmitted, audio is lost. 856 Transcoding of text to and from other coding formats may need to take 857 place in gateways between ToIP and other forms of text conversation, 858 for example to connect to a PSTN text telephone. 860 Session set-up through gateways to other networks may require the use 861 of specially formatted addresses or other mechanisms for invoking 862 those gateways. 864 ToIP interworking requires a method to invoke a text gateway. These 865 text gateways act as User Agents at the IP side. The capabilities of 866 the gateway during the call will be determined by the call 867 capabilities of the terminal that is using the gateway. For example, a 868 PSTN textphone is generally only able to receive voice and real-time 869 text, so the gateway will only allow ToIP and audio. 871 Examples of possible scenarios for invocation of the text gateway are: 873 a. PSTN textphone users dial a prefix number before dialing out. 874 b. Separate real-time text subscriptions, linked to the phone number 875 or terminal identifier/ IP address. 876 c. Real-time text capability indicators. 877 d. Real-time text preference indicators. 878 e. Listen for V.18 modem modulation text activity in all PSTN calls 879 and routing of the call to an appropriate gateway. 880 f. Call transfer request by the called user. 881 g. Placing a call via the web, and using one of the methods described 882 here 883 h. A text gateway with its own telephone number and/or SIP address. 884 (This requires user interaction with the gateway to place a call). 886 i. ENUM address analysis and number plan. 887 j. Number or address analysis leads to a gateway for all PSTN calls. 889 6.2.5.1 PSTN Interworking 891 Analog text telephony is cumbersome because of incompatible national 892 implementations where interworking was never considered. A large 893 number of these implementations have been documented in ITU-T V.18 894 [19], which also defines the modem detection sequences for the 895 different text protocols. The modem type identification may in rare 896 cases take considerable time depending on user actions. 898 To resolve analog textphone incompatibilities, text telephone gateways 899 are needed to transcode incoming analog signals into T.140 and vice 900 versa. The modem capability exchange time can be reduced by the text 901 telephone gateways initially assuming the analog text telephone 902 protocol used in the region where the gateway is located. For example, 903 in the USA, Baudot [II] might be tried as the initial protocol. If 904 negotiation for Baudot fails, the full V.18 modem capability exchange 905 will take place. In the UK, ITU-T V.21 [III] might be the first 906 choice. 908 In particular transmission of real-time text on PSTN networks takes 909 place using a variety of codings and modulations, including ITU-T V.21 910 [III], Baudot [II], DTMF, V.23 [IV] and others. Many difficulties have 911 arisen as a result of this variety in text telephony protocols and the 912 ITU-T V.18 [19] standard was developed to address some of these 913 issues. 915 ITU-T V.18 [19] offers a native text telephony method plus it defines 916 interworking with current protocols. In the interworking mode, it will 917 recognise one of the older protocols and fall back to that 918 transmission method when required. 920 Text gateways MUST use the ITU-T V.18 [19] standard at the PSTN side. 921 A text gateway MUST act as a SIP User Agent on the IP side and support 922 RFC 4103 real-time text transport. 924 While ToIP allows receiving and sending real-time text simultaneously 925 and is displayed on a split screen, many analog text telephones 926 require users to take turns typing. This is because many text 927 telephones operate strictly half duplex. Only one can transmit text at 928 a time. The users apply strict turn-taking rules. 930 There are several text telephones which communicate in full duplex, 931 but merge transmitted text and received text in the same line in the 932 same display window. Here too the users apply strict turn taking 933 rules. 935 Native V.18 text telephones support full duplex and separate display 936 from reception and transmission so that the full duplex capability can 937 be used fully. Such devices could use the ToIP split screen as well, 938 but almost all text telephones use a restricted character set and many 939 use low text transmission speeds (4 to 7 characters per second). 941 That is why it is important for the ToIP user to know that he or she 942 is connected with an analog text telephone. The session description 943 [10] SHOULD contain an indication that the other endpoint for the call 944 is a PSTN textphone (e.g. connected via an ATA or through a text 945 gateway). This means that the textphone user may be used to formal 946 turn taking during the call. 948 6.2.5.2 Mobile Interworking 950 Mobile wireless (or Cellular) circuit switched connections provide a 951 digital real-time transport service for voice or data. The access 952 technologies include GSM, CDMA, TDMA, iDen and various 3G 953 technologies as well as WiFi or WiMAX. 955 ToIP may be supported over the cellular wireless packet switched 956 service. It interfaces to the Internet. 958 The following sections describe how mobile text telephony is 959 supported. 961 6.2.5.2.1 Cellular "No-gain" 963 The "No-gain" text telephone transporting technology uses specially 964 modified EFR [20] and EVR [21] speech vocoders in mobile terminals 965 used to provide a text telephony call. It provides full duplex 966 operation and supports alternating voice and text ("VCO/HCO"). It is 967 dedicated to CDMA and TDMA mobile technologies and the US Baudot (i.e. 968 45 bit/s) type of text telephones. 970 6.2.5.2.2 Cellular Text Telephone Modem (CTM) 972 CTM [8] is a technology independent modem technology that provides the 973 transport of text telephone characters at up to 10 characters/sec 974 using modem signals that can be carried by many voice codecs and uses 975 a highly redundant encoding technique to overcome the fading and cell 976 changing losses. 978 6.2.5.2.3 Cellular "Baudot mode" 980 This term is often used by cellular terminal suppliers for a cellular 981 phone mode that allows TTYs to operate into a cellular phone and to 982 communicate with a fixed line TTY. Thus it is a common name for the 983 "No-Gain" and the CTM solutions when applied to the Baudot type 984 textphones. 986 6.2.5.2.4 Mobile data channel mode 988 Many mobile terminals allow the use of the circuit switched data 989 channel to transfer data in real-time. Data rates of 9600 bit/s are 990 usually supported on the 2G mobile network. Gateways provide 991 interoperability with PSTN textphones. 993 6.2.5.2.5 Mobile ToIP 995 ToIP could be supported over mobile wireless packet switched services 996 that interface to the Internet. For 3GPP 3G services, ToIP support is 997 described in 3G TS 26.235 [22]. 999 6.2.5.3 Instant Messaging Interworking 1001 Text gateways MAY be used to allow interworking between Instant 1002 Messaging systems and ToIP solutions. Because Instant Messaging is 1003 based on blocks of text, rather than on a continuous stream of 1004 characters like ToIP, gateways MUST transcode between the two formats. 1005 Text gateways for interworking between Instant Messaging and ToIP MUST 1006 apply a procedure for bridging the different conversational formats of 1007 real-time text versus text messaging. The following advice may improve 1008 user experience for both parties in a call through a messaging 1009 gateway. 1011 a. Concatenate individual characters originating at the ToIP side into 1012 blocks of text. 1014 b. When the length of the concatenated message becomes longer than 50 1015 characters, the buffered text SHOULD be transmitted to the Instant 1016 Messaging side as soon as any non-alphanumerical character is 1017 received from the ToIP side. 1019 c. When a new line indicator is received from the ToIP side, the 1020 buffered characters up to that point, including the carriage return 1021 and/or line feed characters, SHOULD be transmitted to the Instant 1022 Messaging side. 1024 d. When the ToIP side has been idle for at least 5 seconds, all 1025 buffered text up to that point SHOULD be transmitted to the Instant 1026 Messaging side. 1028 e. Text Gateways must be capable to maintain the real-time performance 1029 for ToIP while providing the interworking services. 1031 It is RECOMMENDED that during the session, both users be constantly 1032 updated on the progress of the text input. Many Instant Messaging 1033 protocols signal that a user is typing to the other party in the 1034 conversation. Text gateways between such Instant Messaging protocols 1035 and ToIP MUST provide this signalling to the Instant Messaging side 1036 when characters start being received, or at the beginning of the 1037 conversation. 1039 At the ToIP side, an indicator of writing the Instant Message MUST be 1040 present where the Instant Messaging protocol provides one. For 1041 example, the real-time text user MAY see ". . . waiting for replying 1042 IM. . . " and when 5 seconds have passed another . (dot) can be shown. 1044 Those solutions will reduce the difficulties between streaming and 1045 blocked text services. 1047 Even though the text gateway can connect Instant Messaging and ToIP, 1048 the best solution is to take advantage of the fact that the user 1049 interfaces and the user communities for instant messaging and ToIP 1050 telephony are very similar. After all, the character input, the 1051 character display, Internet connectivity and SIP stack can be the same 1052 for Instant Messaging (SIMPLE) and ToIP. Thus, the user may simply use 1053 different applications for ToIP and text messaging in the same 1054 terminal. 1056 Devices that implement Instant Messaging SHOULD implement ToIP as 1057 described in this document so that a more complete text communication 1058 service can be provided. 1060 6.2.5.4 Multi-functional Combination gateways 1062 In practice many interworking gateways will be implemented as gateways 1063 that combine different functions. As such, a text gateway could be 1064 built to have modems to interwork with the PSTN and support both 1065 Instant Messaging as well as ToIP. Such interworking functions are 1066 called Combination gateways. 1068 Combination gateways could provide interworking between all of their 1069 supported text based functions. For example, a Text gateway that has 1070 modems to interwork with the PSTN and that support both Instant 1071 Messaging and ToIP could support the following interworking functions: 1073 - PSTN text telephony to ToIP. 1074 - PSTN text telephony to Instant Messaging. 1075 - Instant Messaging to ToIP. 1077 6.2.5.5 Character set transcoding 1079 Gateways between the ToIP network and other networks MAY need to 1080 transcode text streams. ToIP makes use of the ISO 10646 character set. 1081 Most PSTN textphones use a 7-bit character set, or a character set 1082 that is converted to a 7-bit character set by the V.18 modem. 1084 When transcoding between character sets and T.140 in gateways, special 1085 consideration MUST be given to the national variants of the 7 bit 1086 codes, with national characters mapping into different codes in the 1087 ISO 10646 code space. The national variant to be used could be 1088 selectable by the user on a per call basis, or be configured as a 1089 national default for the gateway. 1091 The indicator of missing text in T.140, specified in T.140 amendment 1092 1, cannot be represented in the 7 bit character codes. Therefore the 1093 indicator of missing text SHOULD be transcoded to the ' (apostrophe) 1094 character in legacy text telephone systems, where this character 1095 exists. For legacy systems where the ' character does not exist, the . 1096 (full stop) character SHOULD be used instead. 1098 7. Further recommendations for implementers and service providers 1100 7.1 Access to Emergency services 1102 It must be possible to place an emergency call using ToIP and it must 1103 be possible to use a relay service in such call. The emergency service 1104 provided to users utilising the real-time text medium must be 1105 equivalent to the emergency service provided to users utilising speech 1106 or other media. 1108 A text gateway must be able to route real-time text calls to emergency 1109 service providers when any of the recognised emergency numbers that 1110 support text communications for the country or region are called e.g. 1111 "911" in USA and "112" in Europe. Routing real-time text calls to 1112 emergency services may require the use of a transcoding service. 1114 A text gateway with cellular wireless packet switched services must be 1115 able to route real-time text calls to emergency service providers when 1116 any of the recognized emergency numbers that support real-time text 1117 communication for the country is called. 1119 7.2 Home Gateways or Analog Terminal Adapters 1121 Analog terminal adapters (ATA) using SIP based IP communication and 1122 RJ-11 connectors for connecting traditional PSTN devices SHOULD enable 1123 connection of legacy PSTN text telephones [23]. 1125 These adapters SHOULD contain V.18 modem functionality, voice handling 1126 functionality, and conversion functions to/from SIP based ToIP with 1127 T.140 transported according to RFC 4103 [4], in a similar way as it 1128 provides interoperability for voice sessions. 1130 If a session is set up and text/t140 capability is not declared by the 1131 destination endpoint (by the end-point terminal or the text gateway in 1132 the network at the end-point), a method for invoking a transcoding 1133 server SHALL be used. If no such server is available, the signals from 1134 the textphone MAY be transmitted in the voice channel as audio with 1135 high quality of service. 1137 NOTE: It is preferred that such analog terminal adaptors do use RFC 1138 4103 [5] on board and thus act as a text gateway. Sending textphone 1139 signals over the voice channel is undesirable due to possible 1140 filtering and compression and packet loss between the end-points. This 1141 can result in character loss in the textphone conversation or even not 1142 allowing the textphones to connect to each other. 1144 7.3 User Mobility 1146 ToIP User Agents SHOULD use the same mechanisms as other SIP User 1147 Agents to resolve mobility issues. It is RECOMMENDED that users use a 1148 SIP address, resolved by a SIP registrar, to enable basic user 1149 mobility. Further mechanisms are defined for all session types for 3G 1150 IP multimedia systems. 1152 7.4 Firewalls and NATs 1154 ToIP uses the same signalling and transport protocols as VoIP. Hence, 1155 the same firewall and NAT solutions and network functionality that 1156 apply to VoIP MUST also apply to ToIP. 1158 7.5 Quality of Service 1160 Where Quality of Service (QoS) mechanisms are used, the real-time text 1161 streams should be assigned appropriate QoS characteristics, so that 1162 the performance requirements can be met and the real-time text stream 1163 is not degraded unfavourably in comparison to voice performance in 1164 congested situations. 1166 8. IANA Considerations 1168 There are no IANA considerations for this specification. 1170 9. Security Considerations 1172 User confidentiality and privacy need to be met as described in SIP 1173 [3]. For example, nothing should reveal the fact that the ToIP user 1174 might be a person with a hearing or speech impairment. ToIP is after 1175 all a mainstream communication medium for all users. It is up to the 1176 ToIP user to make his or her hearing or speech impairment public. If a 1177 transcoding server is being used, this SHOULD be transparent. 1178 Encryption SHOULD be used on end-to-end or hop-by-hop basis as 1179 described in SIP [3] and SRTP [24]. 1181 Authentication MUST be provided for users in addition to message 1182 integrity and access control. 1184 Protection against Denial-of-service (DoS) attacks needs to be 1185 provided considering the case that the ToIP users might need 1186 transcoding servers. 1188 10. Authors' Addresses 1190 Guido Gybels 1191 Department of New Technologies 1192 RNID, 19-23 Featherstone Street 1193 London EC1Y 8SL, UK 1194 Email: guido.gybels@rnid.org.uk 1195 Tel +44-20-7294 3713 1196 Txt +44-20-7608 0511 1197 Fax +44-20-7296 8069 1199 Arnoud A. T. van Wijk 1200 Foundation for an Information and Communication Network for the Deaf 1201 and Hard of Hearing 1202 "AnnieS" 1203 http://www.annies.nl/ 1204 Email: arnoud@annies.nl 1206 11. Contributors 1208 The following people contributed to this document: Willem Dijkstra, 1209 Barry Dingle, Gunnar Hellstrom, Radhika R. Roy, Henry Sinnreich and 1210 Gregg C Vanderheiden. 1212 The content and concepts within are a product of the SIPPING Working 1213 Group. Tom Taylor (Nortel) acted as independent reviewer and 1214 contributed significantly to the structure and content of this 1215 document. 1217 12. References 1219 12.1 Normative references 1221 1. S. Bradner, "Intellectual Property Rights in IETF Technology", 1222 BCP 79, RFC 3979, IETF, March 2005. 1224 2. Charlton, Gasson, Gybels, Spanner, van Wijk, "User Requirements 1225 for the Session Initiation Protocol (SIP) in Support of Deaf, Hard 1226 of Hearing and Speech-impaired Individuals", RFC 3351, IETF, 1227 August 2002. 1229 3. J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J. 1230 Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: Session 1231 Initiation Protocol", RFC 3621, IETF, June 2002. 1233 4. H. Schulzrinne, S.Casner, R. Frederick, V. Jacobsone, "RTP: A 1234 Transport Protocol for Real-Time Applications", RFC 3550, IETF, 1235 July 2003. 1237 5. G. Hellstrom, P. Jones, "RTP Payload for Text Conversation", 1238 RFC 4103, IETF, June 2005. 1240 6. ITU-T Recommendation F.703,"Multimedia Conversational Services", 1241 November 2000. 1243 7. S. Bradner, "Key words for use in RFCs to Indicate Requirement 1244 Levels", BCP 14, RFC 2119, IETF, March 1997 1246 8. 3GPP TS 26.226 "Cellular Text Telephone Modem Description" (CTM). 1248 9. ITU-T Recommendation T.140, "Protocol for Multimedia Application 1249 Text Conversation" (February 1998) and Addendum 1 (February 2000). 1251 10. M. Handley, V. Jacobson, C. Perkins, "SDP: Session Description 1252 Protocol", RFC 4566, IETF, July 2006. 1254 11. J. Rosenberg, H. Schulzrinne, P. Kyzivat, "Indicating User Agent 1255 Capabilities in the Session Initiation Protocol (SIP)", RFC 3840, 1256 IETF, August 2004 1258 12. J. Rosenberg, H. Schulzrinne, P. Kyzivat, "Caller Preferences for 1259 the Session Initiation Protocol (SIP)", RFC 3841, IETF, 1260 August 2004 1262 13. J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the 1263 Session Description Protocol (SDP)", RFC 3624, IETF, June 2002. 1265 14. G. Camarillo, "Framework for Transcoding with the Session 1266 Initiation Protocol" IETF May 2006 - Work in progress. 1268 15. G. Camarillo, H. Schulzrinne, E. Burger, and A. van Wijk, 1269 "Transcoding Services Invocation in the Session Initiation 1270 Protocol (SIP) Using Third Party Call Control (3pcc)" RFC 4117, 1271 IETF, June 2005. 1273 16. G. Camarillo, "The SIP Conference Bridge Transcoding Model," IETF, 1274 January 2006 - Work in Progress. 1276 17. Yergeau, F., "UTF-8, a transformation format of ISO 10646", 1277 RFC 3629, IETF,November 2003. 1279 18. "XHTML 1.0: The Extensible HyperText Markup Language: A 1280 Reformulation of HTML 4 in XML 1.0", W3C Recommendation. Available 1281 at http://www.w3.org/TR/xhtml1. 1283 19. ITU-T Recommendation V.18,"Operational and Interworking 1284 Requirements for DCEs operating in Text Telephone Mode", 1285 November 2000. 1287 20. TIA/EIA/IS-823-A "TTY/TDD Extension to TIA/EIA-136-410 Enhanced 1288 Full Rate Speech Codec (must used in conjunction with 1289 TIA/EIA/IS-840)" 1291 21. TIA/EIA/IS-127-2 "Enhanced Variable Rate Codec, Speech Service 1292 Option 3 for Wideband Spread Spectrum Digital Systems. 1293 Addendum 2." 1295 22. "IP Multimedia default codecs". 3GPP TS 26.235 1297 23. H. Sinnreich, S. Lass, and C. Stredicke, " SIP Telephony Device 1298 Requirements and Configuration" RFC 4504, IETF, May 2006. 1300 24. Baugher, McGrew, Carrara, Naslund, Norrman, "The Secure Real Time 1301 Transport Protocol (SRTP)", RFC 3711, IETF, March 2004. 1303 25. ITU-T Recommendation F.700,"Framework Recommendation for 1304 Multimedia Services", November 2000. 1306 12.2 Informative references 1308 I. European Telecommunications Standards Institute (ETSI), "Human 1309 Factors (HF); Guidelines for Telecommunication Relay Services for 1310 Text Telephones". TR 101 806, June 2000. 1312 II. TIA/EIA/825 "A Frequency Shift Keyed Modem for Use on the Public 1313 Switched Telephone Network." (The specification for 45.45 and 50 1314 bit/s TTY modems.) 1316 III. International Telecommunication Union (ITU), "300 bits per second 1317 duplex modem standardized for use in the general switched 1318 telephone network". ITU-T Recommendation V.21, November 1988. 1320 IV. International Telecommunication Union (ITU), "600/1200-baud modem 1321 standardized for use in the general switched telephone network". 1322 ITU-T Recommendation V.23, November 1988. 1324 Full Copyright Statement 1326 Copyright (C) The Internet Society (2006). 1328 This document is subject to the rights, licenses and restrictions 1329 contained in BCP 78, and except as set forth therein, the authors 1330 retain all their rights. 1332 This document and the information contained herein are provided on an 1333 "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS 1334 OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET 1335 ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED, 1336 INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE 1337 INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED 1338 WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. 1340 Intellectual Property 1342 The IETF takes no position regarding the validity or scope of any 1343 Intellectual Property Rights or other rights that might be claimed to 1344 pertain to the implementation or use of the technology described in 1345 this document or the extent to which any license under such rights 1346 might or might not be available; nor does it represent that it has 1347 made any independent effort to identify any such rights. 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