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Hellstrom 3 Internet-Draft Gunnar Hellstrom Accessible Communication 4 Intended status: Informational 30 October 2020 5 Expires: 3 May 2021 7 Real-time text solutions for multi-party sessions 8 draft-hellstrom-avtcore-multi-party-rtt-solutions-04 10 Abstract 12 This document specifies methods for Real-Time Text (RTT) media 13 handling in multi-party calls. The main discussed transport is to 14 carry Real-Time text by the RTP protocol in a time-sampled mode 15 according to RFC 4103. The mechanisms enable the receiving 16 application to present the received real-time text media, separated 17 per source, in different ways according to user preferences. Some 18 presentation related features are also described explaining suitable 19 variations of transmission and presentation of text. 21 Call control features are described for the SIP environment. A 22 number of alternative methods for providing the multi-party 23 negotiation, transmission and presentation are discussed and a 24 recommendation for the main ones is provided. The main solution for 25 SIP based centralized multi-party handling of real-time text is 26 achieved through a media control unit coordinating multiple RTP text 27 streams into one RTP stream. 29 Alternative methods using a single RTP stream and source 30 identification inline in the text stream are also described, one of 31 them being provided as a lower functionality fallback method for 32 endpoints with no multi-party awareness for RTT. 34 Bridging methods where the text stream is carried without the 35 contents being dealt with in detail by the bridge are also discussed. 37 Brief information is also provided for multi-party RTT in the WebRTC 38 environment. 40 The intention is to provide background for decisions, specification 41 and implementation of selected methods. 43 Status of This Memo 45 This Internet-Draft is submitted in full conformance with the 46 provisions of BCP 78 and BCP 79. 48 Internet-Drafts are working documents of the Internet Engineering 49 Task Force (IETF). Note that other groups may also distribute 50 working documents as Internet-Drafts. The list of current Internet- 51 Drafts is at https://datatracker.ietf.org/drafts/current/. 53 Internet-Drafts are draft documents valid for a maximum of six months 54 and may be updated, replaced, or obsoleted by other documents at any 55 time. It is inappropriate to use Internet-Drafts as reference 56 material or to cite them other than as "work in progress." 58 This Internet-Draft will expire on 3 May 2021. 60 Copyright Notice 62 Copyright (c) 2020 IETF Trust and the persons identified as the 63 document authors. All rights reserved. 65 This document is subject to BCP 78 and the IETF Trust's Legal 66 Provisions Relating to IETF Documents (https://trustee.ietf.org/ 67 license-info) in effect on the date of publication of this document. 68 Please review these documents carefully, as they describe your rights 69 and restrictions with respect to this document. Code Components 70 extracted from this document must include Simplified BSD License text 71 as described in Section 4.e of the Trust Legal Provisions and are 72 provided without warranty as described in the Simplified BSD License. 74 Table of Contents 76 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 77 1.1. Requirements Language . . . . . . . . . . . . . . . . . . 5 78 2. Centralized conference model . . . . . . . . . . . . . . . . 5 79 3. Requirements on multi-party RTT . . . . . . . . . . . . . . . 6 80 3.1. General requirements . . . . . . . . . . . . . . . . . . 6 81 3.2. Performance requirements . . . . . . . . . . . . . . . . 7 82 4. RTP based solutions . . . . . . . . . . . . . . . . . . . . . 8 83 4.1. Coordination of text RTP streams . . . . . . . . . . . . 8 84 4.1.1. RTP-based solutions with a central mixer . . . . . . 8 85 4.1.1.1. RTP Mixer using default RFC 4103 methods . . . . 8 86 4.1.1.2. RTP Mixer using the default method but decreased 87 transmission interval . . . . . . . . . . . . . . . 9 88 4.1.1.3. RTP Mixer with frequent transmission and indicating 89 sources in CSRC-list . . . . . . . . . . . . . . . 10 90 4.1.1.4. RTP Mixer using timestamp to identify 91 redundancy . . . . . . . . . . . . . . . . . . . . 11 92 4.1.1.5. RTP Mixer with multiple primary data in each packet 93 and individual sequence numbers . . . . . . . . . . 12 94 4.1.1.6. RTP Mixer with multiple primary data in each 95 packet . . . . . . . . . . . . . . . . . . . . . . 13 97 4.1.1.7. RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy 98 in the packets . . . . . . . . . . . . . . . . . . 14 99 4.1.1.8. RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy 100 and separate sequence number in the packets . . . . 16 101 4.1.1.9. RTP Mixer indicating participants by a control code 102 in the stream . . . . . . . . . . . . . . . . . . . 18 103 4.1.1.10. Mixing for multi-party unaware user agents . . . 20 104 4.1.2. RTP-based bridging with minor RTT media contents 105 reformatting by the bridge . . . . . . . . . . . . . 21 106 4.1.2.1. RTP Translator sending one RTT stream per 107 participant . . . . . . . . . . . . . . . . . . . . 21 108 4.1.2.2. Distributing packets in an end-to-end encryption 109 structure . . . . . . . . . . . . . . . . . . . . . 24 110 4.1.2.3. Mesh of RTP endpoints . . . . . . . . . . . . . . 25 111 4.1.2.4. Multiple RTP sessions, one for each 112 participant . . . . . . . . . . . . . . . . . . . . 25 113 5. Preferred RTP-based multi-party RTT transport method . . . . 26 114 6. Session control of RTP-based multi-party RTT sessions . . . . 26 115 6.1. Implicit RTT multi-party capability indication . . . . . 27 116 6.2. RTT multi-party capability declared by SIP media-tags . . 28 117 6.3. SDP media attribute for RTT multi-party capability 118 indication . . . . . . . . . . . . . . . . . . . . . . . 29 119 6.4. Simplified SDP media attribute for RTT multi-party 120 capability indication . . . . . . . . . . . . . . . . . . 31 121 6.5. SDP format parameter for RTT multi-party capability 122 indication . . . . . . . . . . . . . . . . . . . . . . . 31 123 6.6. A text media subtype for support of multi-party rtt . . . 33 124 6.7. Preferred capability declaration method for RTP-based 125 transport. . . . . . . . . . . . . . . . . . . . . . . . 33 126 6.8. Identification of the source of text for RTP-based 127 solutions . . . . . . . . . . . . . . . . . . . . . . . . 33 128 7. RTT bridging in WebRTC . . . . . . . . . . . . . . . . . . . 34 129 7.1. RTT bridging in WebRTC with one data channel per 130 source . . . . . . . . . . . . . . . . . . . . . . . . . 34 131 7.2. RTT bridging in WebRTC with one common data channel . . . 35 132 7.3. Preferred rtt multi-party method for WebRTC . . . . . . . 35 133 8. Presentation of multi-party text . . . . . . . . . . . . . . 36 134 8.1. Associating identities with text streams . . . . . . . . 36 135 8.2. Presentation details for multi-party aware endpoints. . . 36 136 8.2.1. Bubble style presentation . . . . . . . . . . . . . . 37 137 8.2.2. Other presentation styles . . . . . . . . . . . . . . 38 138 9. Presentation details for multi-party unaware endpoints. . . . 39 139 10. Security Considerations . . . . . . . . . . . . . . . . . . . 39 140 11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 39 141 12. Congestion considerations . . . . . . . . . . . . . . . . . . 40 142 13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 40 143 14. Change history . . . . . . . . . . . . . . . . . . . . . . . 40 144 14.1. Changes to 145 draft-hellstrom-avtcore-multi-party-rtt-solutions-04 . . 40 146 14.2. Changes to 147 draft-hellstrom-avtcore-multi-party-rtt-solutions-03 . . 40 148 14.3. Changes to 149 draft-hellstrom-avtcore-multi-party-rtt-solutions-02 . . 40 150 14.4. Changes to 151 draft-hellstrom-avtcore-multi-party-rtt-solutions-01 . . 40 152 14.5. Changes from draft-hellstrom-mmusic-multi-party-rtt-02 to 153 draft-hellstrom-avtcore-multi-party-rtt-solutions-00 . . 41 154 14.6. Changes from version 155 draft-hellstrom-mmusic-multi-party-rtt-01 to -02 . . . . 41 156 15. References . . . . . . . . . . . . . . . . . . . . . . . . . 41 157 15.1. Normative References . . . . . . . . . . . . . . . . . . 41 158 15.2. Informative References . . . . . . . . . . . . . . . . . 42 159 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 45 161 1. Introduction 163 Real-time text (RTT) is a medium in real-time conversational 164 sessions. Text entered by participants in a session is transmitted 165 in a time-sampled fashion, so that no specific user action is needed 166 to cause transmission. This gives a direct flow of text in the rate 167 it is created, that is suitable in a real-time conversational 168 setting. The real-time text medium can be combined with other media 169 in multimedia sessions. 171 Media from a number of multimedia session participants can be 172 combined in a multi-party session. The present document specifies 173 how the real-time text streams can be handled in multi-party 174 sessions. Recommendations are provided for preferred methods. 176 The description is mainly focused on the transport level, but also 177 describes a few session and presentation level aspects. 179 Transport of real-time text is specified in RFC 4103 [RFC4103] RTP 180 Payload for text conversation. It makes use of RFC 3550 [RFC3550] 181 Real Time Protocol, for transport. Robustness against network 182 transmission problems is normally achieved through redundant 183 transmission based on the principle from RFC 2198 [RFC2198], with one 184 primary and two redundant transmission of each text element. Primary 185 and redundant transmissions are combined in packets and described by 186 a redundancy header. This transport is usually used in the SIP 187 Session Initiation Protocol RFC 3261 [RFC3261] environment. 189 A very brief overview of functions for real-time text handling in 190 multi-party sessions is described in RFC 4597 [RFC4597] Conferencing 191 Scenarios, sections 4.8 and 4.10. The present specification builds 192 on that description and indicates which protocol mechanisms should be 193 used to implement multi-party handling of real-time text. 195 Real-time text can also be transported in the WebRTC environment, by 196 using WebRTC data channels according to 197 [I-D.ietf-mmusic-t140-usage-data-channel]. Multi-party aspects for 198 WebRTC solutions are briefly covered. 200 1.1. Requirements Language 202 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 203 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 204 document are to be interpreted as described in RFC 2119 [RFC2119]. 206 2. Centralized conference model 208 In the centralized conference model for SIP, introduced in RFC 4353 209 [RFC4353] "A Framework for Conferencing with the Session Initiation 210 Protocol (SIP)", one function co-ordinates the communication with 211 participants in the multi-party session. This function also controls 212 media mixer functions for the media appearing in the session. The 213 central function is common for control of all media, while the media 214 mixers may work differently for each media. 216 The central function is called the Focus UA. Many variants exist for 217 setting up sessions including the multipoint control centre. It is 218 not within scope of this description to describe these, but rather 219 the media specific handling in the mixer required to handle multi- 220 party calls with RTT. 222 The main principle for handling real-time text media in a centralized 223 conference is that one RTP session for real-time text is established 224 including the multipoint media control centre and the participating 225 endpoints which are going to have real-time text exchange with the 226 others. 228 The different possible mechanisms for mixing and transporting RTT 229 differs in the way they multiplex the text streams and how they 230 identify the sources of the streams. RFC 7667 [RFC7667] describes a 231 number of possible use cases for RTP. This specification refers to 232 different sections of RFC 7667 for further reading of the situations 233 caused by the different possible design choices. 235 The recommended method for using RTP based RTT in a centralized 236 conference model is specified in 237 [I-D.ietf-avtcore-multi-party-rtt-mix] based on the recommendations 238 in this document. 240 Real-time text can also be transported in the WebRTC environment, by 241 using WebRTC data channels according to 242 [I-D.ietf-mmusic-t140-usage-data-channel]. Ways to handle multi- 243 party calls in that environmnent are also specified. 245 3. Requirements on multi-party RTT 247 3.1. General requirements 249 The following general requirements are placed on multi-party RTT: 251 A solution shall be applicable to IMS (3GPP TS 22.173)[TS22173], 252 SIP based VoIP and Next Generation Emergency Services (NENA i3 253 [NENAi3], ETSI TS 103 479 [TS103479], RFC 6443[RFC6443]). 255 The transmission interval for text should not be longer than 500 256 milliseconds when there is anything available to send. Ref ITU-T 257 T.140 [T140]. 259 If text loss is detected or suspected, a missing text marker 260 should be inserted in the text stream. Ref ITU-T T.140 Amendment 261 1 [T140ad1]. ETSI EN 301 549 [EN301549] 263 The display of text from the members of the conversation shall be 264 arranged so that the text from each participant is clearly 265 readable, and its source and the relative timing of entered text 266 is visualized in the display. Mechanisms for looking back in the 267 contents from the current session should be provided. The text 268 should be displayed as soon as it is received. Ref ITU-T T.140 269 [T140] 271 Bridges must be multimedia capable (voice, video, text). Ref NENA 272 i3 STA-010.2. [NENAi3] 274 It MUST be possible to use real-time text in conferences both as a 275 medium of discussion between individual participants (for example, 276 for sidebar discussions in real-time text while listening to the 277 main conference audio) and for central support of the conference 278 with real-time text interpretation of speech. Ref (R7) in RFC 279 5194.[RFC5194] 281 It should be possible to protect RTT contents with usual means for 282 privacy and integrity. Ref RFC 6881 section 16. [RFC6881] 283 Conferencing procedures are documented in RFC 4579 [RFC4579]. Ref 284 NENA i3 STA-010.2.[NENAi3] 286 Conferencing applies to any kind of media stream by which users 287 may want to communicate. Ref 3GPP TS 24.147 [TS24147] 289 The framework for SIP conferences is specified in RFC 4353 290 [RFC4353]. Ref 3GPP TS 24.147 [TS24147] 292 3.2. Performance requirements 294 The mixer performance requirements can be expressed in one number, 295 extracted from the user requirements on real-time text expressed in 296 ITU-T F.700, where it is stated that for "good" usability, text 297 characters should not be delayed more than 1 second from creation to 298 presentation. For "usable" usability the figure is 2 seconds. The 299 main factor behind these limits is from when taking turns in a 300 conversation gets disturbed by a delay of when a response gets 301 visible to the receiving part. If that times get too long, the 302 receiving part gets unsure if the previous utterance was well 303 perceived and the receiving part maybe prepares for repetition. This 304 is similar to the same effect in voice communication, where the 305 usability limit is 400 ms delay. 307 Another important factor in a multi-party conference is the 308 opportunity for a participant using real-time text to provide timely 309 comments and get a chance to enter the discussion if the majority of 310 participants use voice in the conference. A complicating factor when 311 stating the requirements is that some transport methods do not cause 312 a total delay, but instead an increasing jerkiness when the number of 313 simultaneously sending participants is increased. 315 It should however be remembered that the expected number of 316 participants sending real-time text simultaneously is low. Just as 317 with voice or sign language, the capability of the participants to 318 perceive utterances from more than one participant at a time is very 319 limited. Therefore the normal case in multi-party situations is that 320 one participant at a time is the main provider of text. Others might 321 usually just provide very brief comments such as "yes" or "no" or 322 "may I comment?". Only at very rare situations two participants 323 provide more information simultaneously. 325 * The number of expected simultaneously transmitting users is 326 different for different applications. In all cases, just one 327 transmitting user is the normal case. Two simultaneously 328 transmitting participants can occasionally be expected in 329 emergency services, relay services, small unmanaged conferences 330 and group calls and large managed conferences. Three 331 simultaneously transmitting participants may appear occasionally 332 in large unmanaged conferences. The following can therefore 333 express the performance requirement. 335 * The mean delay of text passing the mixer introduced when only one 336 participant is sending text should be kept to a minimum and should 337 not be more than 400 ms. 339 * The mean delay of text passing the mixer should not be more than 1 340 second during moments when up to three users are sending text 341 simultaneously. 343 * For the very rare case that more than three participants send text 344 simultaneously, the mixer may take action to limit the introduced 345 delay of the text passing the mixer to 7 seconds e.g. by 346 discarding text from some participants and instead inserting a 347 general warning about possible text loss in the stream. 349 4. RTP based solutions 351 4.1. Coordination of text RTP streams 353 Coordinating and sending text RTP streams in the multi-party session 354 can be done in a number of ways. The most suitable methods are 355 specified here with pros and cons. 357 A receiving and presenting endpoint MUST separate text from the 358 different sources and identify and display them accordingly. 360 4.1.1. RTP-based solutions with a central mixer 362 A set of solutions can be based on the central RTP mixer. They are 363 described here and a preferred method selected. 365 4.1.1.1. RTP Mixer using default RFC 4103 methods 367 Without any extra specifications, a mixer would transmit with 300 368 milliseconds intervals, and use RFC 4103 [RFC4103] with the default 369 redundancy of one original and two redundant transmissions. The 370 source of the text would be indicated by a single member in the CSRC 371 list. Text from different sources cannot be transmitted in the same 372 packet. Therefore, from the time when the mixer sent one piece of 373 new text from one source, it will need to transmit that text again 374 twice as redundant data, before it can send text from another source. 375 The jerkiness = time between transmission of new text is 900 ms. 376 This is clearly insufficient. 378 Pros: 380 Only a capability negotiation method is needed. No other update of 381 standards are needed, just a general remark that traditional RTP- 382 mixing is used. 384 Cons: 386 Clearly insufficient mixer switching performance. 388 A bit complex handling of transmission when there is new text 389 available from more than one source. The mixer needs to send two 390 packets more with redundant text from the current source before 391 starting to send anything from the other source. 393 4.1.1.2. RTP Mixer using the default method but decreased transmission 394 interval 396 This method makes use of the default RTP-mixing method briefly 397 described in Section 4.1.1.1. The only difference is that the 398 transmission interval is decreased to 100 milliseconds when there is 399 text from more than one source available for transmission. The 400 jerkiness is 300 ms. The mean delay with two simultaneously sending 401 participants is 250 ms, and with three simultaneously sending 402 participants 500 ms. This is acceptable performance. 404 Pros: 406 Minor influence on standards 408 Can be relatively rapidly be introduced in the intended technical 409 environments. 411 Can be declared in sdp as the already existing "text/red" format with 412 a multi-party attribute for capability negotiation. 414 Cons: 416 The introduced jerkiness of new text from more than the required 417 three simultaneously sending sources is high. 419 Slightly higher risk for loss of text at bursty packet loss than for 420 the recommended transmission interval (300 ms) for RFC 4103. 422 When complete loss of packets occur (beyond recovery), it is not 423 possible to deduct from which source text was lost. 425 A bit complex handling of transmission when there is new text 426 available from more than one source. The mixer needs to send two 427 packets more with redundant text from the current source before 428 starting to send anything from the other source. 430 4.1.1.3. RTP Mixer with frequent transmission and indicating sources in 431 CSRC-list 433 An RTP media mixer combines text from participants into one RTP 434 stream, thus all using the same destination address/port combination, 435 the same RTP SSRC, and one sequence number series as described in 436 Section 7.1 and 7.3 of RTP RFC 3550 [RFC3550] about the Mixer 437 function. This method is also briefly described in RFC 7667, section 438 3.6.1 Media mixing mixer [RFC7667]. 440 The sources of the text in each RTP packet are identified by the CSRC 441 list in the RTP packets, containing the SSRC of the initial sources 442 of text. The order of the CSRC parameters is with the SSRC of the 443 source of the primary text first, followed by the SSRC of the first 444 level redundancy, and then the second level redundancy. 446 The transmission interval should be 100 milliseconds when there is 447 text to transmit from more than one source, and otherwise 300 ms. 449 The identification of the sources is made through the CSRC fields and 450 can be made more readable at the receiver through the RTCP SDES CNAME 451 and NAME packets as described in RTP[RFC3550]. 453 Information provided through the notification according to RFC 4575 454 [RFC4575] when the participant joined the conference provides also 455 suitable information and a reference to the SSRC. 457 A receiving endpoint is supposed to separate text items from the 458 different sources and identify and display them accordingly. 460 The ordered CSRC lists in the RFC 4103 [RFC4103] packets make it 461 possible to recover from loss of one and two packets in sequence and 462 assign the recovered text to the right source. For more loss, a 463 marker for possible loss should be inserted or presented. 465 The conference server needs to have authority to decrypt the payload 466 in the received RTP packets in order to be able to recover text from 467 redundant data or insert the missing text marker in the stream, and 468 repack the text in new packets. 470 Even if the format is very similar to "text/red" of RFC 4103, it 471 needs to be declared as a new media subtype, e.g. "text/rex". 473 Pros: 475 This method has low overhead and less complexity than the methods in 476 Section 4.1.1.1, Section 4.1.1.2, Section 4.1.1.4 and 477 Section 4.1.1.6. 479 When loss of packets occur, it is possible to recover text from 480 redundancy at loss of up to the number of redundancy levels carried 481 in the RFC 4103 [RFC4103] stream (normally primary and two redundant 482 levels). 484 This method can be implemented with most RTP implementations. 486 The source switching performance is sufficient for well-behaving 487 conference participants. The jerkiness is 100 ms. 489 Cons: 491 When more consecutive packet loss than the number of generations of 492 redundant data appears, it is not possible to deduct the sources of 493 the totally lost data. 495 Slightly higher risk for loss of text at bursty packet loss than for 496 the recommended transmission interval for RFC 4103. 498 Requires a different sub media format, e.g. "text/rex". This takes a 499 long time in standardisation and releases of target technical 500 environments. 502 The conference server needs to be allowed to decrypt/encrypt the 503 packet payload. This is however normal for media mixers for other 504 media. 506 4.1.1.4. RTP Mixer using timestamp to identify redundancy 508 This method has text only from one source per packet, as the original 509 RFC 4103 [RFC4103] specifies. Packets with text from different 510 sources are instead allowed to be merged. The recovery procedure in 511 the receiver will use the RTP timestamp and timestamp offsets in the 512 redundancy headers to evaluate if a piece of redundant data should be 513 recovered or not in case of packet loss. 515 In this method, the transmission interval is 100 milliseconds when 516 text from more than one source is available for transmission. 518 Pros: 520 The format of each packet is equal to what is specified in RFC 4103 521 [RFC4103]. 523 The source switching performance is sufficient. Text from five 524 participants can be transmitted simultaneously with 500 milliseconds 525 interval per source. 527 New text from five simultaneous sources can be transmitted within 500 528 milliseconds. This is sufficient. 530 Cons: 532 The recovery time in case of packet loss is long. With five 533 simultaneously sending participants, it will be 1.5 seconds. 535 The recovery procedure is complex and very different from what is 536 described in RFC 4103 [RFC4103]. 538 It is not sure that this change can be regarded to be an update to 539 RFC 4103. It may need a new media subtype. 541 4.1.1.5. RTP Mixer with multiple primary data in each packet and 542 individual sequence numbers 544 This method allows primary as well as redundant text from more than 545 one source per packet. The packet payload contains an ordered set of 546 redundant and primary data with the same number of generations of 547 redundancy as once agreed in the SDP negotiation. The data header 548 reflects these parts of the payload. The CSRC list contains one CSRC 549 member per source in the payload and in the same order. An 550 individual sequence number per source is included in the data header 551 replacing the t140 payload type number that is instead assumed to be 552 constant in this format. This allows an individual extra sequence 553 number per source with maximum value 127, suitable for checking for 554 which source loss of text appeared when recovery was not possible. 556 The data header would contain the following fields: 557 0 1 2 3 558 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 559 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 560 |F| Source-seq | timestamp offset | block length | 561 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 562 Where "Source-seq" is the sequence number per source. 564 The maximum number of members in the CSRC-list is 15, and that is 565 therefore the maximum number of sources that can be represented in 566 each packet provided that all data can be fitted into the size 567 allowable in one packet. 569 Transmission is done as soon as there is new text available, but not 570 with shorter interval than 150 ms and not longer than 300 ms while 571 there is anything to send. 573 A new media subtype is needed, e.g. "text/rex". 575 This is an SDP offer example for both traditional "text/red" 576 and multi-party "text/rex" format: 578 m=text 11000 RTP/AVP 101 100 98 579 a=rtpmap:98 t140/1000 580 a=rtpmap:100 red/1000 581 a=rtpmap:101 rex/1000 582 a=fmtp:100 98/98/98 583 a=fmtp:101 98/98/98 585 Pros: 587 The source switching performance is good. Text from 15 participants 588 can be transmitted simultaneously. 590 New text from 15 simultaneous sources can be transmitted within 300 591 milliseconds. This is good performance. 593 When more consecutive packet loss than the number of generations of 594 redundant data appears, it is still possible to deduct the sources of 595 the totally lost data, when next text from these sources arrive. 597 Cons: 599 The format of each packet is different from what is specified in RFC 600 4103 [RFC4103]. 602 The processing time in standard organisation will be long. 604 A new media subtype is needed, causing a bit complex negotiation. 606 The recovery procedure is a bit complex. 608 4.1.1.6. RTP Mixer with multiple primary data in each packet 610 This method allows primary as well as redundant text from more than 611 one source per packet. The packet payload contains an ordered set of 612 redundant and primary data with the same number of generations of 613 redundancy as once agreed in the SDP negotiation. The data header 614 reflects these parts of the payload. The CSRC list contains one CSRC 615 member per source in the payload and in the same order. 617 The maximum number of members in the CSRC-list is 15, and that is 618 therefore the maximum number of sources that can be represented in 619 each packet provided that all data can be fitted into the size 620 allowable in one packet. 622 Transmission is done as soon as there is new text available, but not 623 with shorter interval than 150 ms and not longer than 300 ms while 624 there is anything to send. 626 A new media subtype is needed, e.g. "text/rex". 628 SDP would be the same as in Section 4.1.1.6. 630 Pros: 632 The source switching performance is good. Text from 15 participants 633 can be transmitted simultaneously. 635 New text from 15 simultaneous sources can be transmitted within 150 636 milliseconds. This is good performance. 638 Cons: 640 The format of each packet is different from what is specified in RFC 641 4103 [RFC4103]. 643 A new media subtype is needed. 645 A new media subtype is needed, causing a bit complex negotiation. 647 The processing time in standard organisation will be long. 649 The recovery procedure is a bit complex [RFC4103]. 651 When more consecutive packet loss than the number of generations of 652 redundant data appears, it is not possible to deduct the sources of 653 the totally lost data. 655 4.1.1.7. RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy in the 656 packets 658 This method allows primary data from one source and redundant text 659 from other sources in each packet. The packet payload contains 660 primary data in "text/t140" format, and redundant data in RFC 5109 661 FEC [RFC5109] format called "text/ulpfec". That means that the 662 redundant data contains the sequence number and the CSRC and other 663 characteristics from the RTP header when the data was sent as 664 primary. The redundancy can be sent at a selected number of packets 665 after when it was sent as primary, in order to improve the protection 666 against bursty packet loss. The redundancy level is recommended to 667 be the same as in original RFC 4103. 669 RFC 4103 says that the protection against loss can be made by other 670 methods than plain redundancy, so this method is in line with that 671 statement. 673 Transmission is done as soon as there is new text available, but not 674 with shorter interval than 100 ms and not longer than 300 ms while 675 there is anything to send (new or redundant text). 677 When more consecutive packet loss than the number of generations of 678 redundant data appears, it is not possible to deduct the sources of 679 the totally lost data. 681 The sdp can indicate the format as "text/red" with "text/ulpfec" 682 redundant data in this way. with traditional RFC 4103 with "text/red" 683 with "text/t140" as redundant data as a fallback. 685 m=text 49170 RTP/AVP 98 101 100 102 686 a=rtpmap:98 red/1000 687 a=fmtp:98 100/102/102 688 a=rtpmap:102 ulpfec/1000 689 a=rtpmap:100 t140/1000 690 a=rtpmap:101 red/1000 691 a=fmtp:101 100/100/100 692 a=fmtp:100 cps=200 694 The "text/ulpfec" format includes an indication of how far back the 695 redundancy belongs, making it possible to cover bursty packet loss 696 better than the other formats with short transmission intervals. For 697 real-time text, it is recommended to send three packets between the 698 primary and the redundant transmissions of text. That makes the 699 transmission cover between 500 and 1500 ms of bursty packet loss. 700 The variation is because of the varying packet interval between many 701 and one simultaneously transmitting source. 703 The "text/ulpfec" format has a number of parameters. One is the 704 length of the data to be protected which in this case must be the 705 whole t140block. 707 Pros: 709 The source switching performance is good. Text from 5 participants 710 can be transmitted within 500 ms. 712 Good recovery from bursty packet loss. 714 The method is based on existing standards. No new registrations are 715 needed. 717 Cons: 719 When more consecutive packet loss than the number of generations of 720 redundant data appears, it is not possible to deduct the sources of 721 the totally lost data. 723 Even if the switching performance is good, it is not as good as for 724 the method called "RTP Mixer with multiple primary data in each 725 packet "Section 4.1.1.6. With more than 5 simultaneously sending 726 sources, there will be a noticeable delay of text of over 500 ms, 727 with 100 ms added per simultaneous source. This is however beyond 728 the requirements and would be a concern only in congestion 729 situations. 731 The recovery procedure is a bit complex [RFC5109]. 733 There is more overhead in terms of extra data and extra packets sent 734 than in the other methods. With the recommended two redundant 735 generations of data, each packet will be 36 bytes longer than with 736 traditional RFC 4103, and at each pause in transmission five extra 737 packets with only redundant data will be sent compared to two extra 738 packets for the traditional RFC 4103 case. 740 4.1.1.8. RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy and 741 separate sequence number in the packets 743 This method allows primary data from one source and redundant text 744 from other sources in each packet. The packet payload contains 745 primary data in a new "text/t140e" format, and redundant data in RFC 746 5109 FEC [RFC5109] format called "text/ulpfec". That means that the 747 redundant data contains the sequence number and the CSRC and other 748 characteristics from the RTP header when the data was sent as 749 primary. The redundancy can be sent at a selected number of packets 750 after when it was sent as primary, in order to improve the protection 751 against bursty packet loss. The redundancy level is recommended to 752 be the same as in original RFC 4103. The "text/t140e" format 753 contains a source-specific sequence number and the t140block. 755 RFC 4103 says that the protection against loss can be made by other 756 methods than plain redundancy, so this method is in line with that 757 statement. 759 Transmission is done as soon as there is new text available, but not 760 with shorter interval than 100 ms and not longer than 300 ms while 761 there is anything to send (new or redundant text). 763 When more consecutive packet loss than the number of generations of 764 redundant data appears, it is possible to deduct which sources lost 765 data when new data arrives from the sources. This is done by 766 monitoring the received source specific sequence numbers preceding 767 the text. 769 This is an example of how can indicate the format as "text/red" with 770 "text/t140e" as primary and "text/ulpfec" redundant data, with 771 traditional RFC 4103 with "text/red" with "text/t140" as redundant 772 data as a fallback. 774 m=text 49170 RTP/AVP 98 101 100 102 103 775 a=rtpmap:98 red/1000 776 a=fmtp:98 100/102/102 777 a=rtpmap:102 ulpfec/1000 778 a=rtpmap:103 t140/1000 779 a=rtpmap:100 t140e/1000 780 a=rtpmap:101 red/1000 781 a=fmtp:101 103/103/103 782 a=fmtp:100 cps=200 784 The "text/ulpfec" format includes an indication of how far back the 785 redundancy belongs, making it possible to cover bursty packet loss 786 better than the other formats with short transmission intervals. For 787 real-time text, it is recommended to send three packets between the 788 primary and the redundant transmissions of text. That makes the 789 transmission cover between 500 and 1500 ms of bursty packet loss. 790 The variation is because of the varying packet interval between many 791 and one simultaneously transmitting source. 793 The "text/ulpfec" format has a number of parameters. One is the 794 length of the data to be protected which in this case must be the 795 whole t140block. 797 Pros: 799 The source switching performance is good. Text from 5 participants 800 can be transmitted within 500 ms. 802 Good recovery from bursty packet loss. 804 The method is based on an existing standard for FEC. 806 When more consecutive packet loss than the number of generations of 807 redundant data appears, it is possible to deduct the source of the 808 lost data when new text arrives from the source. 810 Cons: 812 Even if the switching performance is good, it is not as good as for 813 the method called "RTP Mixer with multiple primary data in each 814 packet" Section 4.1.1.6. With more than 5 simultaneously sending 815 sources, there will be a noticeable delay of text of over 500 ms, 816 with 100 ms added per simultaneous source. This is however beyond 817 the requirements and would be a concern only in congestion 818 situations. 820 The recovery procedure is a bit complex [RFC5109]. 822 There is more overhead in terms of extra data and extra packets sent 823 than in the other methods. With the recommended two redundant 824 generations of data, each packet will be 40 bytes longer than with 825 traditional RFC 4103, and at each pause in transmission five extra 826 packets with only redundant data will be sent compared to two extra 827 packets for the traditional RFC 4103 case. 829 A new text media subtype "text/t140e" needs to be registered. 831 The processing time in standard organisation will be long. 833 4.1.1.9. RTP Mixer indicating participants by a control code in the 834 stream 836 Text from all participants except the receiving one is transmitted 837 from the media mixer in the same RTP session and stream, thus all 838 using the same destination address/port combination, the same RTP 839 SSRC and , one sequence number series as described in Section 7.1 and 840 7.3 of RTP RFC 3550 [RFC3550] about the Mixer function. The sources 841 of the text in each RTP packet are identified by a new defined T.140 842 control code "c" followed by a unique identification of the source in 843 UTF-8 string format. 845 The receiver can use the string for presenting the source of text. 846 This method is on the RTP level described in RFC 7667, section 3.6.1 847 Media mixing mixer [RFC7667]. 849 The inline coding of the source of text is applied in the data stream 850 itself, and an RTP mixer function is used for coordinating the 851 sources of text into one RTP stream. 853 Information uniquely identifying each user in the multi-party session 854 is placed as the parameter value "n" in the T.140 application 855 protocol function with the function code "c". The identifier shall 856 thus be formatted like this: SOS c n ST, where SOS and ST are coded 857 as specified in ITU-T T.140 [T140]. The "c" is the letter "c". The 858 n parameter value is a string uniquely identifying the source. This 859 parameter shall be kept short so that it can be repeated in the 860 transmission without concerns for network load. 862 A receiving endpoint is supposed to separate text items from the 863 different sources and identify and display them accordingly. 865 The conference server need to be allowed to decrypt/encrypt the 866 packet payload in order to check the source and repack the text. 868 Pros: 870 If loss of packets occur, it is possible to recover text from 871 redundancy at loss of up to the number of redundancy levels carried 872 in the RFC 4103 [RFC4103]stream. (normally primary and two redundant 873 levels. 875 This method can be implemented with most RTP implementations. 877 The method can also be used with other transports than RTP 879 Cons: 881 The method implies a moderate load by the need to insert the source 882 often in the stream. 884 If more consecutive packet loss than the number of generations of 885 redundant data appears, it is not possible to deduct the source of 886 the totally lost data. 888 The mixer needs to be able to generate suitable and unique source 889 identifications which are suitable as labels for the sources. 891 Requires an extension on the ITU-T T.140 standard, best made by the 892 ITU. 894 There is a risk that the control code indicating the change of source 895 is lost and the result is false source indication of text. 897 The conference server need to be allowed to decrypt/encrypt the 898 packet payload. 900 4.1.1.10. Mixing for multi-party unaware user agents 902 Multi-party real-time text contents can be transmitted to multi-party 903 unaware user agents if source labelling and formatting of the text is 904 performed by a mixer. This method has the limitations that the 905 layout of the presentation and the format of source identification is 906 purely controlled by the mixer, and that only one source at a time is 907 allowed to present in real-time. Other sources need to be stored 908 temporarily waiting for an appropriate moment to switch the source of 909 transmitted text. The mixer controls the switching of sources and 910 inserts a source identifier in text format at the beginning of text 911 after switch of source. The logic of the mixer to detect when a 912 switch is appropriate should detect a number of places in text where 913 a switch can be allowed, including new line, end of sentence, end of 914 phrase, a period of inactivity, and a word separator after a long 915 time of active transmission. 917 This method MAY be used when no support for multi-party awareness is 918 detected in the receiving endpoint.The base for his method is 919 described in RFC 7667, section 3.6.1 Media mixing mixer [RFC7667]. 921 See [I-D.ietf-avtcore-multi-party-rtt-mix] for a procedure for mixing 922 RTT for a conference-unaware endpoint. 924 Pros: 926 Can be transmitted to conference-unaware endpoints. 928 Can be used with other transports than RTP 930 Cons: 932 Does not allow full real-time presentation of more than one source at 933 a time. Text from other sources will be delayed. 935 The only realistic presentation format is a style with the text from 936 the different sources presented with a text label indicating source, 937 and the text collected in a chat style presentation but with more 938 frequent turn-taking. 940 Endpoints often have their own system for adding labels to the RTT 941 presentation. In that case there will be two levels of labels in the 942 presentation, one for the mixer and one for the sources. 944 If loss of more packets than can be recovered by the redundancy 945 appears, it is not possible to detect which source was struck by the 946 loss. It is also possible that a source switch occurred during the 947 loss, and therefore a false indication of the source of text can be 948 provided to the user after such loss. 950 Because of all these cons, this method is not recommended and should 951 be used as the main method, but only as fallback and the last resort 952 for backwards interoperability with multi-party unaware endpoints. 954 The conference server need to be allowed to decrypt/encrypt the 955 packet payload. 957 4.1.2. RTP-based bridging with minor RTT media contents reformatting by 958 the bridge 960 It may be desirable to send text in a multi-party setting in a way 961 that allows the text stream contents to be distributed without being 962 dealt with in detail in any central server. A number of such methods 963 are described. However, when writing this specification, no one of 964 these methods have a specified way of establishing the session by 965 sdp. 967 4.1.2.1. RTP Translator sending one RTT stream per participant 969 Within the RTP session, text from each participant is transmitted 970 from the RTP media translator (bridge) in a separate RTP stream, thus 971 using the same destination address/port combination, the same payload 972 type number (PT) but separate RTP SSRC parameters and sequence number 973 series as described in Section 7.1 and 7.2 of RTP RFC 3550 [RFC3550] 974 about the Translator function. The source of the text in each RTP 975 packet is identified by the SSRC parameter in the RTP packets, 976 containing the SSRC of the initial source of text. 978 A receiving and presenting endpoint is supposed to separate text 979 items from the different sources and identify and display them in a 980 suitable way. 982 This method is described in RFC 7667, section 3.5.1 Relay-transport 983 translator or 3.5.2 Media translator [RFC7667]. 985 The identification of the source is made through the SSRC. The 986 translation to a readable label can be done by mapping to information 987 from the RTCP SDES CNAME and NAME packets as described in 988 RTP[RFC3550], and also through information in the text media member 989 in the conference notification described in RFC 4575 [RFC4575]. 991 The sdp exchange for establishing this mixing type can be equal to 992 what is used for basic two-party use of RFC 4103 with just an added 993 attribute for indicating multi-party capability. 995 m=text 49170 RTP/AVP 98 103 996 a=rtpmap:98 red/1000 997 a=fmtp:98 103/103/103 998 a=rtpmap:103 t140/1000 999 a=fmtp:103 cps=150 1000 a=RTT-mixing:RTP-translator 1002 A similar answer including the same RTT-mixing attribute would 1003 indicate that multi-party coding can begin. An answer without the 1004 same RTT-mixing attribute could result in diversion to use of the 1005 mixing method for multi-party unaware endpoints Section 4.1.1.10 if 1006 more than two parties are involved in the session. 1008 The bridge can add new sources in the communication to a participant 1009 by first sending a conference notification according to RFC 4575 1010 [RFC4575] with the SSRC of the new source included in the 1011 corresponding "text" media member, or by sending an RTCP message with 1012 the new SSRC in an SDES packet. 1014 A receiver should be prepared to receive such indications of new 1015 streams being added to the multi-party session, so that the new SSRC 1016 is not taken for a change in SSRC value for an already established 1017 RTP stream. 1019 Transmission, reception, packet loss recovery and text loss 1020 indication is performed per source in the separate RTP streams in the 1021 same way as in two-party sessions with RFC 4103 [RFC4575]. 1023 Text is recommended to be sent by the bridge as soon as it is 1024 available for transmission, but not less than 250 ms after a previous 1025 transmission. This will in many cases result in close to 0 added 1026 delay by the bridge, because most RTT senders use a 300 ms 1027 transmission interval. 1029 It is sometimes said that this configuration is not supported by 1030 current media declarations in sdp. RFC 3264 [RFC3264]specifies in 1031 some places that one media description is supposed to describe just 1032 one RTP media stream. However this is not directly referencing an 1033 RTP stream, and use of multiple RTP streams in the same RTP session 1034 is recommended in many other RFCs. 1036 This confusion is clarified in RFC 5576 [RFC5576] section 3 by the 1037 following statements: 1039 "The term "media stream" does not appear in the SDP specification 1040 itself, but is used by a number of SDP extensions, for instance, 1041 Interactive Connectivity Establishment (ICE) [ICE], to denote the 1042 object described by an SDP media description. This term is 1043 unfortunately rather confusing, as the RTP specification [RFC3550] 1044 uses the term "media stream" to refer to an individual media source 1045 or RTP packet stream, identified by an SSRC, whereas an SDP media 1046 stream describes an entire RTP session, which can contain any number 1047 of RTP sources." 1049 In most cases, it will be sufficient that new sources are introduced 1050 with a conference notification or RTCP message. However, RFC 5576 1051 [RFC5576] specifies attributes which may be used to more explicitly 1052 announce new sources or restart of earlier established RTP streams. 1054 This method is encouraged by draft-ietf-avtcore-multiplex-guidelines 1055 [I-D.ietf-avtcore-multiplex-guidelines] section 5.2. 1057 Normal operation will be that the bridge receives text packets from 1058 the source and handles any text recovery and indication of loss 1059 needed before queueing the resulting clean text for transmission from 1060 the bridge to the receivers. 1062 It may however also be possible for the bridge to just convey the 1063 packet contents as received from the sources, with minor adjustments, 1064 and let the receiving endpoint handle all aspects of recovery and 1065 indication of loss, even for the source to bridge path. In that case 1066 also the sequence number must be maintained as it was at reception in 1067 the bridge. This mode needs further study before application. 1069 Pros: 1071 This method is the natural way to do multi-party bridging with RFC 1072 4103 based RTT. Only a small addition is included in the session 1073 establishment to verify capability by the parties because many 1074 implementations are done without multi-party capability. 1076 This method has moderate overhead in terms of work for the mixer, but 1077 high in terms of packet transmission rate. Five sources sending 1078 simultaneously cause the bridge to send 15 packets per second to each 1079 receiver. 1081 When loss of packets occur, it is possible to recover text from 1082 redundancy at loss of up to the number of redundancy levels carried 1083 in the RFC 4103 [RFC4103] stream(normally primary and two redundant 1084 levels). 1086 More loss than what can be recovered, can be detected and the marker 1087 for text loss can be inserted in the correct stream. 1089 It may be possible in some scenarios to keep the text encrypted 1090 through the Translator. 1092 Minimal delay. The delay can often be kept close to 0 with at least 1093 5 simultaneous sending participants. 1095 Cons: 1097 There are RTP implementations not supporting the Translator model. 1098 They will need to use the fall-back to multi-party-unaware mixing. 1099 An investigation about how common this is is needed before the method 1100 is used. 1102 The processing time in standard organisation will be long. 1104 With many simultaneous sending sources, the total rate of packets 1105 will be high, and can cause congestion. The requirement to handle 3 1106 simultaneous sources in this specification will cause 10 packets per 1107 second that is manageable in most cases, e.g. considering that audio 1108 usually use 50 packets per second. 1110 4.1.2.2. Distributing packets in an end-to-end encryption structure 1112 In order to achieve end-to-end encryption, it is possible to let the 1113 packets from the sources just pass though a central distributor, and 1114 handle the security agreements between the participants. 1115 Specifications exist for a framework with this functionality for 1116 application on RTP based conferences in 1117 [I-D.ietf-perc-private-media-framework]. The RTP flow and mixing 1118 characteristics has similarities with the method described under "RTP 1119 Translator sending one RTT stream per participant" above. RFC 4103 1120 RTP streams [RFC4103] would fit into the structure and it would 1121 provide a base for end-to-end encrypted rtt multi-party conferencing. 1123 Pros: 1125 Good security 1127 Straightforward multi-party handling. 1129 Cons: 1131 Does not operate under the usual SIP central conferencing 1132 architecture. 1134 Requires the participants to perform a lot of key handling. 1136 Is work in progress when this is written. 1138 4.1.2.3. Mesh of RTP endpoints 1140 Text from all participants are transmitted directly to all others in 1141 one RTP session, without a central bridge. The sources of the text 1142 in each RTP packet are identified by the source network address and 1143 the SSRC. 1145 This method is described in RFC 7667, section 3.4 Point to multi- 1146 point using mesh [RFC7667]. 1148 Pros: 1150 When loss of packets occur, it is possible to recover text from 1151 redundancy at loss of up to the number of redundancy levels carried 1152 in the RFC 4103 [RFC4103] stream. (normally primary and two redundant 1153 levels. 1155 This method can be implemented with most RTP implementations. 1157 Transmitted text can also be used with other transports than RTP 1159 Cons: 1161 This model is not described in IMS, NENA and EENA specifications, and 1162 does therefore not meet the requirements. 1164 Requires a drastically increasing number of connections when the 1165 number of participants increase. 1167 4.1.2.4. Multiple RTP sessions, one for each participant 1169 Text from all participants are transmitted directly to all others in 1170 one RTP session each, without a central bridge. Each session is 1171 established with a separate media description in SDP. The sources of 1172 the text in each RTP packet are identified by the source network 1173 address and the SSRC. 1175 Pros: 1177 When loss of packets occur, it is possible to recover text from 1178 redundancy at loss of up to the number of redundancy levels carried 1179 in the RFC 4103 [RFC4103] stream. (normally primary and two redundant 1180 levels. 1182 Complete loss of text can be indicated in the received stream. 1184 This method can be implemented with most RTP implementations. 1186 End-to-end encryption is achievable. 1188 Cons: 1190 This method is not described in IMS, NENA and ETSI specifications and 1191 does therefore not meet the requirements. 1193 A lot of network resources are spent on setting up separate sessions 1194 for each participant. 1196 5. Preferred RTP-based multi-party RTT transport method 1198 For RTP transport of RTT using RTP-mixer technology, one method for 1199 multi-party mixing and transport stand out as fulfilling the goals 1200 best and is therefore recommended. That is: "RTP Mixer using the 1201 default method but decreased transmission interval" Section 4.1.1.2 1203 For RTP transport in separate streams or sessions, no current 1204 recommendation can be made. A bridging method in the process of 1205 standardisation with interesting characteristics is the end-to-end 1206 encryption model "perc" Section 4.1.2.2. 1208 6. Session control of RTP-based multi-party RTT sessions 1210 General session control aspects for multi-party sessions are 1211 described in RFC 4575 [RFC4575] A Session Initiation Protocol (SIP) 1212 Event Package for Conference State, and RFC 4579 [RFC4579] Session 1213 Initiation Protocol (SIP) Call Control - Conferencing for User 1214 Agents. The nomenclature of these specifications are used here. 1216 The procedures for a multi-party aware model for RTT-transmission 1217 shall only be applied if a capability exchange for multi-party aware 1218 real-time text transmission has been completed and a supported method 1219 for multi-party real-time text transmission can be negotiated. 1221 A method for detection of conference-awareness for centralized SIP 1222 conferencing in general is specified in RFC 4579 [RFC4579]. The 1223 focus sends the "isfocus" feature tag in a SIP Contact header. This 1224 causes the conference-aware endpoint to subscribe to conference 1225 notifications from the focus. The focus then sends notifications to 1226 the endpoint about entering and disappearing conference participants 1227 and their media capabilities. The information is carried XML- 1228 formatted in a 'conference-info' block in the notification according 1229 to RFC 4575 [RFC4575]. The mechanism is described in detail in RFC 1230 4575 [RFC4575]. 1232 Before a conference media server starts sending multi-party RTT to an 1233 endpoint, a verification of its ability to handle multi-party RTT 1234 must be made. A decision on which mechanism to use for identifying 1235 text from the different participants must also be taken, implicitly 1236 or explicitly. These verifications and decisions can be done in a 1237 number of ways. The most apparent ways are specified here and their 1238 pros and cons described. One of the methods is selected to be the 1239 one to be used by implementations of the centralized conference model 1240 according to this specification. 1242 6.1. Implicit RTT multi-party capability indication 1244 Capability for RTT multi-party handling can be decided to be 1245 implicitly indicated by session control items. 1247 The focus may implicitly indicate muti-party RTT capability by 1248 including the media child with value "text" in the RFC 4575 [RFC4575] 1249 conference-info provided in conference notifications. 1251 An endpoint may implicitly indicate multi-party RTT capability by 1252 including the text media in the SDP in the session control 1253 transactions with the conference focus after the subscription to the 1254 conference has taken place. 1256 The implicit RTT capability indication means for the focus that it 1257 can handle multi-party RTT according to the preferred method 1258 indicated in the RTT multi-party methods section above. 1260 The implicit RTT capability indication means for the endpoint that it 1261 can handle multi-party RTT according to the preferred method 1262 indicated in the RTT multi-party methods section above. 1264 If the focus detects that an endpoint implicitly declared RTT multi- 1265 party capability, it SHALL provide RTT according to the preferred 1266 method. 1268 If the focus detects that the endpoint does not indicate any RTT 1269 multi-party capability, then it shall either provide RTT multi-party 1270 text in the way specified for conference-unaware endpoint above, or 1271 refuse to set up the session. 1273 If the endpoint detects that the focus has implicitly declared RTT 1274 multi-party capability, it shall be prepared to present RTT in a 1275 multi-party fashion according to the preferred method. 1277 Pros: 1279 Acceptance of implicit multi-party capability implies that no 1280 standardisation of explicit RTT multi-party capability exchange is 1281 required. 1283 Cons: 1285 If other methods for multi-party RTT are to be used in the same 1286 implementation environment as the preferred ones, then capability 1287 exchange needs to be defined for them. 1289 Cannot be used outside a strictly applied SIP central conference 1290 model. 1292 6.2. RTT multi-party capability declared by SIP media-tags 1294 Specifications for RTT multi-party capability declarations can be 1295 agreed for use as SIP media feature tags, to be exchanged during SIP 1296 call control operation according to the mechanisms in RFC 3840 1297 [RFC3840] and RFC 3841 [RFC3841]. Capability for the RTT Multi-party 1298 capability is then indicated by the media feature tag "rtt-mix", with 1299 a set of possible values for the different possible methods. 1301 The possible values in the list may for example be: 1303 rtp-mixer 1305 perc 1307 rtp-mixer indicates capability for using the RTP-mixer based 1308 presentation of multi-party text. 1310 perc indicates capability for using the perc based transmission of 1311 multi-party text. 1313 Example: Contact: 1315 ;methods="INVITE,ACK,OPTIONS,BYE,CANCEL" 1316 ;+sip.rtt-mix="rtp-mixer" 1318 If, after evaluation of the alternatives in this specification, only 1319 one mixing method is selected to be brought to implementation, then 1320 the media tag can be reduced to a single tag with no list of values. 1322 An offer-answer exchange should take place and the common method 1323 selected by the answering party shall be used in the session with 1324 that UA. 1326 When no common method is declared, then only the fallback method for 1327 multi-party unaware participants can be used, or the session dropped. 1329 If more than one text media section is included in SDP, all must be 1330 capable of using the declared RTT multi-party method. 1332 Pros: 1334 Provides a clear decision method. 1336 Can be extended with new mixing methods. 1338 Can guide call routing to a suitable capable focus. 1340 Cons: 1342 Requires standardization and IANA registration. 1344 Is not stream specific. If more than one text stream is specified, 1345 all must have the same type of multi-party capability. 1347 Cannot be used in the WebRTC environment. 1349 6.3. SDP media attribute for RTT multi-party capability indication 1351 An attribute can be specified on media level, to be used in text 1352 media SDP declarations for negotiating RTT multi-party capabilities. 1353 The attribute can have the name "rtt-mixing". 1355 More than one attribute can be included in one media description. 1357 The attribute can have a value. The value can for example be: 1359 rtp-mixer 1361 rtp-translator 1362 perc 1364 rtp-mixer indicates capability for using the RTP-mixer and CSRC-list 1365 based mixing of multi-party text. 1367 rtp-translator indicates capability for using the RTP-translator 1368 based mixing 1370 perc indicates capability for using the perc based transmission of 1371 multi-party text. 1373 An offer-answer exchange should take place and the common method 1374 selected by the answering party shall be used in the session with 1375 that endpoint. 1377 When no common method is declared, then only the fallback method for 1378 multi-party unaware endpoints can be used. 1380 Example: a=rtt-mixing:rtp-mixer 1382 If, after evaluation of the alternatives in this specification, only 1383 one mixing method is selected to be brought to implementation, then 1384 the attribute can be reduced to a single attribute with no list of 1385 values. 1387 Pros: 1389 Provides a clear decision method. 1391 Can be extended with new mixing methods. 1393 Can be used on specific text media. 1395 Can be used also for SDP-controlled WebRTC sessions with multiple 1396 streams in the same data channel. 1398 Cons: 1400 Requires standardization and IANA registration. 1402 Cannot guide SIP routing. 1404 6.4. Simplified SDP media attribute for RTT multi-party capability 1405 indication 1407 An attribute can be specified on media level, to be used in text 1408 media SDP declarations for negotiating RTT multi-party capabilities. 1409 The attribute can have a name suitable for the selected method and no 1410 value. It would be selected and used if only one method for multi- 1411 party rtt is brought forward from this specification, and the other 1412 left unspecified for now or found to be possible to negotiate in 1413 another way. 1415 An offer-answer exchange should take place and if both parties 1416 specify rtt-mixing capability with the same attribute, the selected 1417 mixing method shall be used. 1419 When no common method is declared, then only the fallback method for 1420 multi-party unaware endpoints can be used, or the session not 1421 accepted for multi-party use. 1423 Example: a=rtt-mix 1425 Pros: 1427 Provides a clear decision method. 1429 Very simple syntax and semantics. 1431 Can be used on specific text media. 1433 Cons: 1435 Requires standardization and IANA registration. 1437 If another RTT mixing method is also specified in the future, then 1438 that method may also need to specify and register its own attribute, 1439 instead of if an attribute with a parameter value is used, when only 1440 an addition of a new possible value is needed. 1442 Cannot guide SIP routing. 1444 6.5. SDP format parameter for RTT multi-party capability indication 1446 An FMTP format parameter can be specified for the RFC 4103 1447 [RFC4103]media, to be used in text media SDP declarations for 1448 negotiating RTT multi-party capabilities. The parameter can have the 1449 name "rtt-mixing", with one or more of its possible values. 1451 The possible values in the list are: 1453 rtp-mixer 1455 perc 1457 rtp-mixer indicates capability for using the RTP-mixer based mixing 1458 and presentation of multi-party text using the CSRC-list. 1460 perc indicates capability for using the perc based transmission of 1461 multi-party text. 1463 Example: a=fmtp 96 98/98/98 rtt-mixing=rtp-mixer 1465 If, after evaluation of the alternatives in this specification, only 1466 one mixing method is selected to be brought to implementation, then 1467 the parameter can be reduced to a single parameter with no list of 1468 values. 1470 An offer-answer exchange should take place and the common method 1471 selected by the answering party shall be used in the session with 1472 that UA. 1474 When no common method is declared, then only the fallback method can 1475 be used, or the session denied. 1477 Pros: 1479 Provides a clear decision method. 1481 Can be extended with new mixing methods. 1483 Can be used on specific text media. 1485 Can be used also for SDP-controlled WebRTC sessions with multiple 1486 streams in the same data channel. 1488 Cons: 1490 Requires standardization and IANA registration. 1492 May cause interop problems with current RFC4103 [RFC4103] 1493 implementations not expecting a new fmtp-parameter. 1495 Cannot guide SIP routing. 1497 6.6. A text media subtype for support of multi-party rtt 1499 Indicating a specific text media subtype in SDP is a straightforward 1500 way for negotiating multi-party capability. Especially if there are 1501 format differences from the "text/red" and "text/t140" formats of 1502 RFC4103 [RFC4103], then this is a natural way to do the negotiation 1503 for multi-party rtt. 1505 Pros: 1507 No extra efforts if a new format is needed anyway. 1509 Cons: 1511 None specific to using the format indication for negotiation of 1512 multi-party capability. But only feasible if a new format is needed 1513 anyway. 1515 6.7. Preferred capability declaration method for RTP-based transport. 1517 If the preferred transport method is one with a specific media 1518 subtype in sdp, then specification by media subtype is preferred. 1520 If this would not be the case, then the preferred capability 1521 declaration method would be the one with a specific SDP attribute for 1522 the selected mixing method Section 6.4 because it is straightforward. 1524 6.8. Identification of the source of text for RTP-based solutions 1526 The main way to identify the source of text in the RTP based solution 1527 is by the SSRC of the sending participant. In the RTP-mixer 1528 solution, this SSRC is included in the CSRC list of the transmitted 1529 packets. Further identification that may be needed for better 1530 labelling of received text may be achieved from a number of sources. 1531 It may be the RTCP SDES CNAME and NAME reports, and in the conference 1532 notification data (RFC 4575) [RFC4575]. 1534 As soon as a new member is added to the RTP session, its 1535 characteristics should be transmitted in RTCP SDES CNAME and NAME 1536 reports according to section 6.5 in RFC 3550 [RFC3550]. The 1537 information about the participant should also be included in the 1538 conference data including the text media member in a notification 1539 according to RFC 4575 [RFC4575]. 1541 The RTCP SDES report, SHOULD contain identification of the source 1542 represented by the SSRC/CSRC identifier. This identification MUST 1543 contain the CNAME field and MAY contain the NAME field and other 1544 defined fields of the SDES report. 1546 A focus UA SHOULD primarily convey SDES information received from the 1547 sources of the session members. When such information is not 1548 available, the focus UA SHOULD compose SSRC/CSRC, CNAME and NAME 1549 information from available information from the SIP session with the 1550 participant. 1552 Provision of detailed information in the NAME field has security 1553 implications, especially if provided without encryption. 1555 7. RTT bridging in WebRTC 1557 Within WebRTC, real-time text is specified to be carried in WebRTC 1558 data channels as specified in 1559 [I-D.ietf-mmusic-t140-usage-data-channel]. A few ways to handle 1560 multi-party RTT are mentioned briefly. They are repeated below. 1562 7.1. RTT bridging in WebRTC with one data channel per source 1564 A straightforward way to handle multi-party RTT is for the bridge to 1565 open one T.140 data channel per source towards the receiving 1566 participants. 1568 The stream-id forms a unique stream identification. 1570 The identification of the source is made through the Label property 1571 of the channel, and session information belonging to the source. The 1572 endpoint can compose a readable label for the presentation from this 1573 information. 1575 Pros: 1577 This is a straightforward solution. 1579 The load per source is low. 1581 Cons: 1583 With a high number of participants, the overhead of establishing and 1584 maintaining the high number of data channels required may be high, 1585 even if the load per channel is low. 1587 7.2. RTT bridging in WebRTC with one common data channel 1589 A way to handle multi-party RTT in WebRTC is for the bridge combine 1590 text from all sources into one data channel and insert the sources in 1591 the stream by a T.140 control code for source. 1593 This method is described in a corresponding section for RTP 1594 transmission above in Section 4.1.1.9. 1596 The identification of the source is made through insertion in the 1597 beginning of each text transmission from a source of a control code 1598 extension "c" followed by a string representing the source, framed by 1599 the control code start and end flags SOS and ST (See ITU-T T.140 1600 [T140]). 1602 A receiving endpoint is supposed to separate text items from the 1603 different sources and identify and display them in a suitable way. 1605 The endpoint does not always display the source identification in the 1606 received text at the place where it is received, but has the 1607 information as a guide for planning the presentation of received 1608 text. A label corresponding to the source identification is 1609 presented when needed depending on the selected presentation style. 1611 Pros: 1613 This solution has relatively low overhead on session and network 1614 level 1616 Cons: 1618 This solution has higher overhead on the media contents level than 1619 the WebRTC solution above. 1621 Standardisation of the new control code "c" in ITU-T T.140 [T140] is 1622 required. 1624 The conference server need to be allowed to decrypt/encrypt the data 1625 channel contents. 1627 7.3. Preferred rtt multi-party method for WebRTC 1629 For WebRTC, one method is to prefer because of the simplicity. So, 1630 for WebRTC, the method to implement for multi-party RTT with multi- 1631 party aware parties when no other method is explicitly agreed between 1632 implementing parties is: "RTT bridging in WebRTC with one data 1633 channel per source" Section 7.1. 1635 8. Presentation of multi-party text 1637 All session participants with RTP based transport MUST observe the 1638 SSRC/CSRC field of incoming text RTP packets, and make note of which 1639 source they came from in order to be able to present text in a way 1640 that makes it easy to read text from each participant in a session, 1641 and get information about the source of the text. 1643 In the WebRTC case, the Label parameter and other provided endpoint 1644 information should be used for the same purpose. 1646 8.1. Associating identities with text streams 1648 A source identity SHOULD be composed from available information 1649 sources and displayed together with the text as indicated in ITU-T 1650 T.140 Appendix[T140]. 1652 The source identity should primarily be the NAME field from incoming 1653 SDES packets. If this information is not available, and the session 1654 is a two-party session, then the T.140 source identity SHOULD be 1655 composed from the SIP session participant information. For multi- 1656 party sessions the source identity may be composed by local 1657 information if sufficient information is not available in the 1658 session. 1660 Applications may abbreviate the presented source identity to a 1661 suitable form for the available display. 1663 Applications may also replace received source information with 1664 internally used nicknames. 1666 8.2. Presentation details for multi-party aware endpoints. 1668 The multi-party aware endpoint should after any action for recovery 1669 of data from lost packets, separate the incoming streams and present 1670 them according to the style that the receiving application supports 1671 and the user has selected. The decisions taken for presentation of 1672 the multi-party interchange shall be purely on the receiving side. 1673 The sending application must not insert any item in the stream to 1674 influence presentation that is not requested by the sending 1675 participant. 1677 8.2.1. Bubble style presentation 1679 One often used style is to present real-time text in chunks in 1680 readable bubbles identified by labels containing names of sources. 1681 Bubbles are placed in one column in the presentation area and are 1682 closed and moved upwards in the presentation area after certain items 1683 or events, when there is also newer text from another source that 1684 would go into a new bubble. The text items that allows bubble 1685 closing are any character closing a phrase or sentence followed by a 1686 space or a timeout of a suitable time (about 10 seconds). 1688 Real-time active text sent from the local user should be presented in 1689 a separate area. When there is a reason to close a bubble from the 1690 local user, the bubble should be placed above all real-time active 1691 bubbles, so that the time order that real-time text entries were 1692 completed is visible. 1694 Scrolling is usually provided for viewing of recent or older text. 1695 When scrolling is done to an earlier point in the text, the 1696 presentation shall not move the scroll position by new received text. 1697 It must be the decision of the local user to return to automatic 1698 viewing of latest text actions. It may be useful with an indication 1699 that there is new text to read after scrolling to an earlier position 1700 has been activated. 1702 The presentation area may become too small to present all text in all 1703 real-time active bubbles. Various techniques can be applied to 1704 provide a good overview and good reading opportunity even in such 1705 situations. The active real-time bubble may have a limited number of 1706 lines and if their contents need more lines, then a scrolling 1707 opportunity within the real-time active bubble is provided. Another 1708 method can be to only show the label and the last line of the active 1709 real-time bubble contents, and make it possible to expand or compress 1710 the bubble presentation between full view and one line view. 1712 Erasures require special consideration. Erasure within a real-time 1713 active bubble is straightforward. But if erasure from one 1714 participant affects the last character before a bubble, the whole 1715 previous bubble becomes the actual bubble for real-time action by 1716 that participant and is placed below all other bubbles in the 1717 presentation area. If the border between bubbles was caused by the 1718 CRLF characters (instead of the normal "Line Separator"), only one 1719 erasure action is required to erase this bubble border. When a 1720 bubble is closed, it is moved up, above all real-time active bubbles. 1722 A three-party view is shown in this example . 1724 _________________________________________________ 1725 | |^| 1726 | |-| 1727 |[Alice] Hi, Alice here. | | 1728 | | | 1729 |[Bob] Bob as well. | | 1730 | | | 1731 |[Eve] Hi, this is Eve, calling from Paris. | | 1732 | I thought you should be here. | | 1733 | | | 1734 |[Alice] I am coming on Thursday, my | | 1735 | performance is not until Friday morning.| | 1736 | | | 1737 |[Bob] And I on Wednesday evening. | | 1738 | | | 1739 |[Alice] Can we meet on Thursday evening? | | 1740 | | | 1741 |[Eve] Yes, definitely. How about 7pm. | | 1742 | at the entrance of the restaurant | | 1743 | Le Lion Blanc? | | 1744 |[Eve] we can have dinner and then take a walk | | 1745 | | | 1746 | But I need to be back to | | 1747 | the hotel by 11 because I need | | 1748 | | | 1749 | I wou |-| 1750 |______________________________________________|v| 1751 | of course, I underst | 1752 |________________________________________________| 1754 Figure 1: Three-party call with bubble style. 1756 Figure 1: Example of a three-party call presented in the bubble 1757 style. 1759 8.2.2. Other presentation styles 1761 Other presentation styles than the bubble style may be arranged and 1762 appreciated by the users. In a video conference one way may be to 1763 have a real-time text area below the video view of each participant. 1764 Another view may be to provide one column in a presentation area for 1765 each participant and place the text entries in a relative vertical 1766 position corresponding to when text entry in them was completed. The 1767 labels can then be placed in the column header. The considerations 1768 for ending and moving and erasure of entered text discussed above for 1769 the bubble style are valid also for these styles. 1771 This figure shows how a coordinated column view MAY be presented. 1773 _____________________________________________________________________ 1774 | Bob | Eve | Alice | 1775 |____________________|______________________|_______________________| 1776 | | |I will arrive by TGV. | 1777 |My flight is to Orly| |Convenient to the main | 1778 | |Hi all, can we plan |station. | 1779 | |for the seminar? | | 1780 |Eve, will you do | | | 1781 |your presentation on| | | 1782 |Friday? |Yes, Friday at 10. | | 1783 |Fine, wo | |We need to meet befo | 1784 |___________________________________________________________________| 1786 Figure 2: A coordinated column-view of a three-party session with 1787 entries ordered in approximate time-order. 1789 9. Presentation details for multi-party unaware endpoints. 1791 Multi-party unaware endpoints are prepared only for presentation of 1792 two sources of text, the local user and a remote user. If mixing for 1793 multi-party unaware endpoints is to be supported, in order to enable 1794 some multi-party communication with such endpoint, the mixer need to 1795 plan the presentation and insert labels and line breaks before 1796 lables. Many limitations appear for this presentation mode, and it 1797 must be seen as a fallback and a last resort. 1799 A procedure for presenting RTT to a conference-unaware endpoint is 1800 included in [I-D.ietf-avtcore-multi-party-rtt-mix] 1802 10. Security Considerations 1804 The security considerations valid for RFC 4103 [RFC4103] and RFC 3550 1805 [RFC3550] are valid also for the multi-party sessions with text. 1807 11. IANA Considerations 1809 The items for indication and negotiation of capability for multi- 1810 party rtt should be registered with IANA in the specifications where 1811 they are specified in detail. 1813 12. Congestion considerations 1815 The congestion considerations described in RFC 4103 [RFC4103] are 1816 valid also for the recommended RTP-based multi-party use of the real- 1817 time text transport. A risk for congestion may appear if a number of 1818 conference participants are active transmitting text simultaneously, 1819 because the recommended RTP-based multi-party transmission method 1820 does not allow multiple sources of text to contribute to the same 1821 packet. 1823 In situations of risk for congestion, the Focus UA MAY combine 1824 packets from the same source to increase the transmission interval 1825 per source up to one second. Local conference policy in the Focus UA 1826 may be used to decide which streams shall be selected for such 1827 transmission frequency reduction. 1829 13. Acknowledgements 1831 Arnoud van Wijk for contributions to an earlier, expired draft of 1832 this memo. 1834 14. Change history 1836 14.1. Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-04 1838 Change name of simplified sdp attribute to "rtt-mix" to match a 1839 change in the draft draft-ietf-avtcore-multi-party-rtt-mix-09. 1841 14.2. Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-03 1843 Modified info on the method with RFC 4103 format and sdp attribute 1844 "rtt-mix-rtp-mixer". 1846 Increased the performance requirements section. 1848 Inserted recommendations, with emphasis on ease of implementation and 1849 ease of standardisation. 1851 14.3. Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-02 1853 Added detail in the section on RTP translator model alternative 1854 4.1.2.1. 1856 14.4. Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-01 1858 Added three more methods for RTP-mixer mixing. Two RFC 5109 FEC 1859 based and another with modified data header to detect source of 1860 completely lost text. 1862 Separated RTP-based and WebRTC based solutions. 1864 Deleted the multi-party-unaware mixing procedure appendix. It is now 1865 included in the draft draft-ietf-avtcore-multi-party-rtt-mix. Kept a 1866 section with a reference to the new place. 1868 14.5. Changes from draft-hellstrom-mmusic-multi-party-rtt-02 to draft- 1869 hellstrom-avtcore-multi-party-rtt-solutions-00 1871 Add discussion about switching performance, as discussed in avtcore 1872 on March 13. 1874 Added that a decrease of transmission interval to 100 ms increases 1875 switching performance by a factor 3, but still not sufficient. 1877 Added that the CSRC-list method also uses 100 milliseconds 1878 transmission interval. 1880 Added the method with multiple primary text in each packet. 1882 Added the timestamp-based method for rtp-mixing proposed by James 1883 Hamlin on March 14. 1885 Corrected the chat style presentation example picture. Delete a few 1886 "[mix]". 1888 14.6. Changes from version draft-hellstrom-mmusic-multi-party-rtt-01 to 1889 -02 1891 Change from a general overview to overview with clear 1892 recommendations. 1894 Splits text coordination methods in three groups. 1896 Recommends rtt-mixer with sources in CSRC-list but refers to its spec 1897 for details. 1899 Shortened Appendix with conference-unaware example. 1901 Cleaned up preferences. 1903 Inserted pictures of screen-views. 1905 15. References 1907 15.1. Normative References 1909 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1910 Requirement Levels", BCP 14, RFC 2119, 1911 DOI 10.17487/RFC2119, March 1997, 1912 . 1914 15.2. Informative References 1916 [EN301549] ETSI, "EN 301 549. Accessibility requirements for ICT 1917 products and services", November 2019, 1918 . 1922 [I-D.ietf-avtcore-multi-party-rtt-mix] 1923 Hellstrom, G., "RTP-mixer formatting of multi-party Real- 1924 time text", Work in Progress, Internet-Draft, draft-ietf- 1925 avtcore-multi-party-rtt-mix-06, 11 June 2020, 1926 . 1929 [I-D.ietf-avtcore-multiplex-guidelines] 1930 Westerlund, M., Burman, B., Perkins, C., Alvestrand, H., 1931 and R. Even, "Guidelines for using the Multiplexing 1932 Features of RTP to Support Multiple Media Streams", Work 1933 in Progress, Internet-Draft, draft-ietf-avtcore-multiplex- 1934 guidelines-12, 16 June 2020, . 1937 [I-D.ietf-mmusic-t140-usage-data-channel] 1938 Holmberg, C. and G. Hellstrom, "T.140 Real-time Text 1939 Conversation over WebRTC Data Channels", Work in Progress, 1940 Internet-Draft, draft-ietf-mmusic-t140-usage-data-channel- 1941 14, 10 April 2020, . 1944 [I-D.ietf-perc-private-media-framework] 1945 Jones, P., Benham, D., and C. Groves, "A Solution 1946 Framework for Private Media in Privacy Enhanced RTP 1947 Conferencing (PERC)", Work in Progress, Internet-Draft, 1948 draft-ietf-perc-private-media-framework-12, 5 June 2019, 1949 . 1952 [NENAi3] NENA, "NENA-STA-010.2-2016. Detailed Functional and 1953 Interface Standards for the NENA i3 Solution", October 1954 2016, . 1956 [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., 1957 Handley, M., Bolot, J.C., Vega-Garcia, A., and S. Fosse- 1958 Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, 1959 DOI 10.17487/RFC2198, September 1997, 1960 . 1962 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 1963 A., Peterson, J., Sparks, R., Handley, M., and E. 1964 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 1965 DOI 10.17487/RFC3261, June 2002, 1966 . 1968 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 1969 with Session Description Protocol (SDP)", RFC 3264, 1970 DOI 10.17487/RFC3264, June 2002, 1971 . 1973 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1974 Jacobson, "RTP: A Transport Protocol for Real-Time 1975 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1976 July 2003, . 1978 [RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, 1979 "Indicating User Agent Capabilities in the Session 1980 Initiation Protocol (SIP)", RFC 3840, 1981 DOI 10.17487/RFC3840, August 2004, 1982 . 1984 [RFC3841] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller 1985 Preferences for the Session Initiation Protocol (SIP)", 1986 RFC 3841, DOI 10.17487/RFC3841, August 2004, 1987 . 1989 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 1990 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 1991 . 1993 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 1994 Session Initiation Protocol (SIP)", RFC 4353, 1995 DOI 10.17487/RFC4353, February 2006, 1996 . 1998 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 1999 Session Initiation Protocol (SIP) Event Package for 2000 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 2001 2006, . 2003 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 2004 (SIP) Call Control - Conferencing for User Agents", 2005 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 2006 . 2008 [RFC4597] Even, R. and N. Ismail, "Conferencing Scenarios", 2009 RFC 4597, DOI 10.17487/RFC4597, August 2006, 2010 . 2012 [RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error 2013 Correction", RFC 5109, DOI 10.17487/RFC5109, December 2014 2007, . 2016 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 2017 Time Text over IP Using the Session Initiation Protocol 2018 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 2019 . 2021 [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific 2022 Media Attributes in the Session Description Protocol 2023 (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009, 2024 . 2026 [RFC6443] Rosen, B., Schulzrinne, H., Polk, J., and A. Newton, 2027 "Framework for Emergency Calling Using Internet 2028 Multimedia", RFC 6443, DOI 10.17487/RFC6443, December 2029 2011, . 2031 [RFC6881] Rosen, B. and J. Polk, "Best Current Practice for 2032 Communications Services in Support of Emergency Calling", 2033 BCP 181, RFC 6881, DOI 10.17487/RFC6881, March 2013, 2034 . 2036 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 2037 DOI 10.17487/RFC7667, November 2015, 2038 . 2040 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 2041 multimedia application text conversation", February 1998, 2042 . 2044 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 2045 Protocol for multimedia application text conversation", 2046 February 2000, 2047 . 2049 [TS103479] ETSI, "TS 103 479. Emergency communications (EMTEL); Core 2050 elements for network independent access to emergency 2051 services", December 2019, . 2055 [TS22173] 3GPP, "IP Multimedia Core Network Subsystem (IMS) 2056 Multimedia Telephony Service and supplementary services; 2057 Stage 1", 3GPP TS 22.173 17.1.0, 20 December 2019, 2058 . 2060 [TS24147] 3GPP, "Conferencing using the IP Multimedia (IM) Core 2061 Network (CN) subsystem; Stage 3", 3GPP TS 24.147 16.0.0, 2062 19 December 2019, 2063 . 2065 Author's Address 2067 Gunnar Hellstrom 2068 Gunnar Hellstrom Accessible Communication 2069 Esplanaden 30 2070 SE-136 70 Vendelso 2071 Sweden 2073 Phone: +46 708 204 288 2074 Email: gunnar.hellstrom@ghaccess.se