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(See the Legal Provisions document at https://trustee.ietf.org/license-info for more information.) -- The document date (11 April 2021) is 1108 days in the past. Is this intentional? Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) -- Possible downref: Non-RFC (?) normative reference: ref. 'T140' -- Possible downref: Non-RFC (?) normative reference: ref. 'T140ad1' Summary: 0 errors (**), 0 flaws (~~), 3 warnings (==), 4 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 AVTCore G. Hellstrom 3 Internet-Draft Gunnar Hellstrom Accessible Communication 4 Updates: 4103 (if approved) 11 April 2021 5 Intended status: Standards Track 6 Expires: 13 October 2021 8 RTP-mixer formatting of multi-party Real-time text 9 draft-ietf-avtcore-multi-party-rtt-mix-14 11 Abstract 13 Enhancements for RFC 4103 real-time text mixing are provided in this 14 document, suitable for a centralized conference model that enables 15 source identification and rapidly interleaved transmission of text 16 from different sources. The intended use is for real-time text 17 mixers and participant endpoints capable of providing an efficient 18 presentation or other treatment of a multi-party real-time text 19 session. The specified mechanism builds on the standard use of the 20 CSRC list in the RTP packet for source identification. The method 21 makes use of the same "text/t140" and "text/red" formats as for two- 22 party sessions. 24 Solutions using multiple RTP streams in the same RTP session are 25 briefly mentioned, as they could have some benefits over the RTP- 26 mixer model. The possibility to implement the solution in a wide 27 range of existing RTP implementations made the RTP-mixer model be 28 selected to be fully specified in this document. 30 A capability exchange is specified so that it can be verified that a 31 mixer and a participant can handle the multi-party coded real-time 32 text stream using the RTP-mixer method. The capability is indicated 33 by use of an SDP media attribute "rtt-mixer". 35 The document updates RFC 4103 "RTP Payload for Text Conversation". 37 A specification of how a mixer can format text for the case when the 38 endpoint is not multi-party aware is also provided. 40 Status of This Memo 42 This Internet-Draft is submitted in full conformance with the 43 provisions of BCP 78 and BCP 79. 45 Internet-Drafts are working documents of the Internet Engineering 46 Task Force (IETF). Note that other groups may also distribute 47 working documents as Internet-Drafts. The list of current Internet- 48 Drafts is at https://datatracker.ietf.org/drafts/current/. 50 Internet-Drafts are draft documents valid for a maximum of six months 51 and may be updated, replaced, or obsoleted by other documents at any 52 time. It is inappropriate to use Internet-Drafts as reference 53 material or to cite them other than as "work in progress." 55 This Internet-Draft will expire on 13 October 2021. 57 Copyright Notice 59 Copyright (c) 2021 IETF Trust and the persons identified as the 60 document authors. All rights reserved. 62 This document is subject to BCP 78 and the IETF Trust's Legal 63 Provisions Relating to IETF Documents (https://trustee.ietf.org/ 64 license-info) in effect on the date of publication of this document. 65 Please review these documents carefully, as they describe your rights 66 and restrictions with respect to this document. Code Components 67 extracted from this document must include Simplified BSD License text 68 as described in Section 4.e of the Trust Legal Provisions and are 69 provided without warranty as described in the Simplified BSD License. 71 Table of Contents 73 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 74 1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 75 1.2. Selected solution and considered alternatives . . . . . . 6 76 1.3. Intended application . . . . . . . . . . . . . . . . . . 9 77 2. Overview of the two specified solutions and selection of 78 method . . . . . . . . . . . . . . . . . . . . . . . . . 10 79 2.1. The RTP-mixer based solution for multi-party aware 80 endpoints . . . . . . . . . . . . . . . . . . . . . . . . 10 81 2.2. Mixing for multi-party unaware endpoints . . . . . . . . 10 82 2.3. Offer/answer considerations . . . . . . . . . . . . . . . 11 83 2.4. Actions depending on capability negotiation result . . . 11 84 3. Details for the RTP-mixer based multi-party aware mixing 85 method . . . . . . . . . . . . . . . . . . . . . . . . . 12 86 3.1. Use of fields in the RTP packets . . . . . . . . . . . . 12 87 3.2. Initial transmission of a BOM character . . . . . . . . . 12 88 3.3. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 12 89 3.4. Transmission interval . . . . . . . . . . . . . . . . . . 13 90 3.5. Only one source per packet . . . . . . . . . . . . . . . 13 91 3.6. Do not send received text to the originating source . . . 13 92 3.7. Clean incoming text . . . . . . . . . . . . . . . . . . . 13 93 3.8. Redundant transmission principles . . . . . . . . . . . . 13 94 3.9. Interleaving text from different sources . . . . . . . . 14 95 3.10. Text placement in packets . . . . . . . . . . . . . . . . 14 96 3.11. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 15 97 3.12. Creation of the redundancy . . . . . . . . . . . . . . . 15 98 3.13. Timer offset fields . . . . . . . . . . . . . . . . . . . 15 99 3.14. Other RTP header fields . . . . . . . . . . . . . . . . . 16 100 3.15. Pause in transmission . . . . . . . . . . . . . . . . . . 16 101 3.16. RTCP considerations . . . . . . . . . . . . . . . . . . . 16 102 3.17. Reception of multi-party contents . . . . . . . . . . . . 16 103 3.18. Performance considerations . . . . . . . . . . . . . . . 18 104 3.19. Security for session control and media . . . . . . . . . 19 105 3.20. SDP offer/answer examples . . . . . . . . . . . . . . . . 19 106 3.21. Packet sequence example from interleaved transmission . . 20 107 3.22. Maximum character rate "CPS" . . . . . . . . . . . . . . 23 108 4. Presentation level considerations . . . . . . . . . . . . . . 23 109 4.1. Presentation by multi-party aware endpoints . . . . . . . 24 110 4.2. Multi-party mixing for multi-party unaware endpoints . . 26 111 5. Relation to Conference Control . . . . . . . . . . . . . . . 31 112 5.1. Use with SIP centralized conferencing framework . . . . . 32 113 5.2. Conference control . . . . . . . . . . . . . . . . . . . 32 114 6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 32 115 6.1. Gateway considerations with Textphones (e.g. TTYs). . . 32 116 6.2. Gateway considerations with WebRTC. . . . . . . . . . . . 32 117 7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 33 118 8. Congestion considerations . . . . . . . . . . . . . . . . . . 34 119 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 34 120 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 34 121 10.1. Registration of the "rtt-mixer" SDP media attribute . . 34 122 11. Security Considerations . . . . . . . . . . . . . . . . . . . 35 123 12. Change history . . . . . . . . . . . . . . . . . . . . . . . 35 124 12.1. Changes included in 125 draft-ietf-avtcore-multi-party-rtt-mix-14 . . . . . . . 35 126 12.2. Changes included in 127 draft-ietf-avtcore-multi-party-rtt-mix-13 . . . . . . . 36 128 12.3. Changes included in 129 draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . . 36 130 12.4. Changes included in 131 draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 36 132 12.5. Changes included in 133 draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 36 134 12.6. Changes included in 135 draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 37 136 12.7. Changes included in 137 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 37 138 12.8. Changes included in 139 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 37 140 12.9. Changes included in 141 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 37 142 12.10. Changes included in 143 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 38 144 12.11. Changes included in 145 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 38 147 12.12. Changes included in 148 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 38 149 12.13. Changes included in 150 draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 39 151 12.14. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 39 152 12.15. Changes from 153 draft-hellstrom-avtcore-multi-party-rtt-source-03 to 154 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 40 155 12.16. Changes from 156 draft-hellstrom-avtcore-multi-party-rtt-source-02 to 157 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 40 158 12.17. Changes from 159 draft-hellstrom-avtcore-multi-party-rtt-source-01 to 160 -02 . . . . . . . . . . . . . . . . . . . . . . . . . . 40 161 12.18. Changes from 162 draft-hellstrom-avtcore-multi-party-rtt-source-00 to 163 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 41 164 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 41 165 13.1. Normative References . . . . . . . . . . . . . . . . . . 41 166 13.2. Informative References . . . . . . . . . . . . . . . . . 43 167 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 43 169 1. Introduction 171 "RTP Payload for Text Conversation" [RFC4103] specifies use of RTP 172 [RFC3550] for transmission of real-time text (RTT) and the "text/ 173 t140" format. It also specifies a redundancy format "text/red" for 174 increased robustness. The "text/red" format is registered in 175 [RFC4102]. 177 Real-time text is usually provided together with audio and sometimes 178 with video in conversational sessions. 180 A requirement related to multi-party sessions from the presentation 181 level standard T.140 for real-time text is: "The display of text from 182 the members of the conversation should be arranged so that the text 183 from each participant is clearly readable, and its source and the 184 relative timing of entered text is visualized in the display." 186 Another requirement is that the mixing procedure must not introduce 187 delays in the text streams that are experienced to be disturbing the 188 real-time experience of the receiving users. 190 Use of RTT is increasing, and specifically, use in emergency calls is 191 increasing. Emergency call use requires multi-party mixing. RFC 192 4103 "RTP Payload for Text Conversation" mixer implementations can 193 use traditional RTP functions for source identification, but the 194 performance of the mixer when giving turns for the different sources 195 to transmit is limited when using the default transmission 196 characteristics with redundancy. 198 The redundancy scheme of [RFC4103] enables efficient transmission of 199 earlier transmitted redundant text in packets together with new text. 200 However the redundancy header format has no source indicators for the 201 redundant transmissions. The redundant parts in a packet must 202 therefore be from the same source as the new text. The recommended 203 transmission is one new and two redundant generations of text 204 (T140blocks) in each packet and the recommended transmission interval 205 for two-party use is 300 ms. 207 Real-time text mixers for multi-party sessions need to include the 208 source with each transmitted group of text from a conference 209 participant so that the text can be transmitted interleaved with text 210 groups from different sources in the rate they are created. This 211 enables the text groups to be presented by endpoints in suitable 212 grouping with other text from the same source. 214 The presentation can then be arranged so that text from different 215 sources can be presented in real-time and easily read. At the same 216 time it is possible for a reading user to perceive approximately when 217 the text was created in real time by the different parties. The 218 transmission and mixing is intended to be done in a general way so 219 that presentation can be arranged in a layout decided by the 220 endpoint. 222 There are existing implementations of RFC 4103 in endpoints without 223 the updates from this document. These will not be able to receive 224 and present real-time text mixed for multi-party aware endpoints. 226 A negotiation mechanism is therefore needed for verification if the 227 parties are able to handle a common method for multi-party 228 transmission and agreeing on using that method. 230 A fall-back mixing procedure is also needed for cases when the 231 negotiation result indicates that a receiving endpoint is not capable 232 of handling the mixed format. Multi-party unaware endpoints would 233 possibly otherwise present all received multi-party mixed text as if 234 it came from the same source regardless of any accompanying source 235 indication coded in fields in the packet. Or they may have any other 236 undesirable way of acting on the multi-party content. The fall-back 237 method is called the mixing procedure for multi-party unaware 238 endpoints. The fall-back method is naturally not expected to meet 239 all performance requirements placed on the mixing procedure for 240 multi-party aware endpoints. 242 The document updates [RFC4103] by introducing an attribute for 243 indicating capability for the RTP-mixer based multi-party mixing case 244 and rules for source indications and interleaving of text from 245 different sources. 247 1.1. Terminology 249 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 250 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 251 "OPTIONAL" in this document are to be interpreted as described in BCP 252 14 [RFC2119] [RFC8174] when, and only when, they appear in all 253 capitals, as shown above. 255 The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP- 256 mixer, RTP-translator as defined in [RFC3550] 258 The term "T140block" is defined in [RFC4103] to contain one or more 259 T.140 code elements. 261 "TTY" stands for a text telephone type used in North America. 263 "WebRTC" stands for web based communication specified by W3C and 264 IETF. 266 "DTLS-SRTP" stands for security specified in [RFC5764]. 268 "multi-party aware" stands for an endpoint receiving real-time text 269 from multiple sources through a common conference mixer being able to 270 present the text in real-time separated by source and presented so 271 that a user can get an impression of the approximate relative timing 272 of text from different parties. 274 "multi-party unaware" stands for an endpoint not itself being able to 275 separate text from different sources when received through a common 276 conference mixer. 278 1.2. Selected solution and considered alternatives 280 A number of alternatives were considered when searching an efficient 281 and easily implemented multi-party method for real-time text. This 282 section explains a few of them briefly. 284 Multiple RTP streams, one per participant. 285 One RTP stream per source would be sent in the same RTP session 286 with the "text/red" format. From some points of view, use of 287 multiple RTP streams, one for each source, sent in the same RTP 288 session would be efficient, and would use exactly the same packet 289 format as [RFC4103] and the same payload type. A couple of 290 relevant scenarios using multiple RTP-streams are specified in 291 "RTP Topologies" [RFC7667]. One possibility of special interest 292 is the Selective Forwarding Middlebox (SFM) topology specified in 293 RFC 7667 section 3.7 that could enable end to end encryption. In 294 contrast to audio and video, real-time text is only transmitted 295 when the users actually transmit information. Thus an SFM 296 solution would not need to exclude any party from transmission 297 under normal conditions. In order to allow the mixer to convey 298 the packets with the payload preserved and encrypted, an SFM 299 solution would need to act on some specific characteristics of the 300 "text/red" format. The redundancy headers are part of the 301 payload, so the receiver would need to just assume that the 302 payload type number in the redundancy header is for "text/t140". 303 The characters per second parameter (CPS) would need to act per 304 stream. The relation between the SSRC and the source would need 305 to be conveyed in some specified way, e.g. in the CSRC. Recovery 306 and loss detection would preferably be based on sequence number 307 gap detection. Thus sequence number gaps in the incoming stream 308 to the mixer would need to be reflected in the stream to the 309 participant and no new gaps created by the mixer. However, the 310 RTP implementation in both mixers and endpoints need to support 311 multiple streams in the same RTP session in order to use this 312 mechanism. For best deployment opportunity, it should be possible 313 to upgrade existing endpoint solutions to be multi-party aware 314 with a reasonable effort. There is currently a lack of support 315 for multi-stream RTP in certain implementation technologies. This 316 fact made this solution only briefly mentioned in this document as 317 an option for further study. 319 RTP-mixer based method for multi-party aware endpoints. 320 The "text/red" format in RFC 4103 is sent with shorter 321 transmission interval with the RTP-mixer method and indicating 322 source in CSRC. The "text/red" format with "text/t140" payload in 323 a single RTP stream can be sent when text is available from the 324 call participants instead of at the regular 300 ms. The source is 325 indicated in the CSRC field. Transmission of packets with text 326 from different sources can then be done smoothly while 327 simultaneous transmission occurs as long as it is not limited by 328 the maximum character rate "CPS". With ten participants sending 329 text simultaneously, the switching and transmission performance is 330 good. With more simultaneously sending participants, and 331 receivers with default capacity there will be a noticeable 332 jerkiness and delay in text presentation. The jerkiness will be 333 more expressed the more participants who send text simultaneously. 334 Two seconds jerkiness will be noticeable and slightly unpleasant, 335 but corresponds in time to what typing humans often cause by 336 hesitation or changing position while typing. A benefit of this 337 method is that no new packet format needs to be introduced and 338 implemented. Since simultaneous typing by more than two parties 339 is very rare, this method can be used successfully with good 340 performance. Recovery of text in case of packet loss is based on 341 analysis of timestamps of received redundancy versus earlier 342 received text. Negotiation is based on a new SDP media attribute 343 "rtt-mixer". This method is selected to be the main one specified 344 in this document. 346 Multiple sources per packet. 347 A new "text" media subtype would be specified with up to 15 348 sources in each packet. The mechanism would make use of the RTP 349 mixer model specified in RTP [RFC3550]. Text from up to 15 350 sources can be included in each packet. Packets are normally sent 351 every 300 ms. The mean delay will be 150 ms. The sources are 352 indicated in strict order in the CSRC list of the RTP packets. A 353 new redundancy packet format is specified. This method would 354 result in good performance, but would require standardisation and 355 implementation of new releases in the target technologies that 356 would take more time than desirable to complete. It was therefore 357 not selected to be included in this document. 359 Mixing for multi-party unaware endpoints 360 Presentation of text from multiple parties is prepared by the 361 mixer in one single stream. It is desirable to have a method that 362 does not require any modifications in existing user devices 363 implementing RFC 4103 for RTT without explicit support of multi- 364 party sessions. This is possible by having the mixer insert a new 365 line and a text formatted source label before each switch of text 366 source in the stream. Switch of source can only be done in places 367 in the text where it does not disturb the perception of the 368 contents. Text from only one source can be presented in real time 369 at a time. The delay will therefore be varying. The method also 370 has other limitations, but is included in this document as a 371 fallback method. In calls where parties take turns properly by 372 ending their entries with a new line, the limitations will have 373 limited influence on the user experience. while only two parties 374 send text, these two will see the text in real time with no delay. 375 This method is specified as a fallback method in this document. 377 RTT transport in WebRTC 378 Transport of real-time text in the WebRTC technology is specified 379 to use the WebRTC data channel in [RFC8865]. That specification 380 contains a section briefly describing its use in multi-party 381 sessions. The focus of this document is RTP transport. 382 Therefore, even if the WebRTC transport provides good multi-party 383 performance, it is just mentioned in this document in relation to 384 providing gateways with multi-party capabilities between RTP and 385 WebRTC technologies. 387 1.3. Intended application 389 The method for multi-party real-time text specified in this document 390 is primarily intended for use in transmission between mixers and 391 endpoints in centralised mixing configurations. It is also 392 applicable between mixers. An often mentioned application is for 393 emergency service calls with real-time text and voice, where a 394 calltaker wants to make an attended handover of a call to another 395 agent, and stay observing the session. Multimedia conference 396 sessions with support for participants to contribute in text is 397 another application. Conferences with central support for speech-to- 398 text conversion is yet another mentioned application. 400 In all these applications, normally only one participant at a time 401 will send long text utterances. In some cases, one other participant 402 will occasionally contribute with a longer comment simultaneously. 403 That may also happen in some rare cases when text is interpreted to 404 text in another language in a conference. Apart from these cases, 405 other participants are only expected to contribute with very brief 406 utterings while others are sending text. 408 Users expect that the text they send is presented in real-time in a 409 readable way to the other participants even if they send 410 simultaneously with other users and even when they make brief edit 411 operations of their text by backspacing and correcting their text. 413 Text is supposed to be human generated, by some text input means, 414 such as typing on a keyboard or using speech-to-text technology. 415 Occasional small cut-and-paste operations may appear even if that is 416 not the initial purpose of real-time text. 418 The real-time characteristics of real-time text is essential for the 419 participants to be able to contribute to a conversation. If the text 420 is too much delayed from typing a letter to its presentation, then, 421 in some conference situations, the opportunity to comment will be 422 gone and someone else will grab the turn. A delay of more than one 423 second in such situations is an obstacle for good conversation. 425 2. Overview of the two specified solutions and selection of method 427 This section contains a brief introduction of the two methods 428 specified in this document. 430 2.1. The RTP-mixer based solution for multi-party aware endpoints 432 This method specifies negotiated use of the RFC 4103 format for 433 multi-party transmission in a single RTP stream. The main purpose of 434 this document is to specify a method for true multi-party real-time 435 text mixing for multi-party aware endpoints that can be widely 436 deployed. The RTP-mixer based method makes use of the current format 437 for real-time text in [RFC4103]. It is an update of RFC 4103 by a 438 clarification on one way to use it in the multi-party situation. 439 That is done by completing a negotiation for this kind of multi-party 440 capability and by interleaving packets from different sources. The 441 source is indicated in the CSRC element in the RTP packets. Specific 442 considerations are made to be able to recover text after packet loss. 444 The detailed procedures for the RTP-mixer based multi-party aware 445 case are specified in Section 3. 447 Please use [RFC4103] as reference when reading the specification. 449 2.2. Mixing for multi-party unaware endpoints 451 A method is also specified in this document for cases when the 452 endpoint participating in a multi-party call does not itself 453 implement any solution, or not the same, as the mixer. The method 454 requires the mixer to insert text dividers and readable labels and 455 only send text from one source at a time until a suitable point 456 appears for source change. This solution is a fallback method with 457 functional limitations. It acts on the presentation level. 459 A party acting as a mixer, which has not negotiated any method for 460 true multi-party RTT handling, but negotiated a "text/red" or "text/ 461 t140" format in a session with a participant SHOULD in order to 462 maintain interoperability, if nothing else is specified for the 463 application, format transmitted text to that participant to be 464 suitable to present on a multi-party unaware endpoint as further 465 specified in Section 4.2. 467 2.3. Offer/answer considerations 469 RTP Payload for Text Conversation [RFC4103] specifies use of RTP 470 [RFC3550], and a redundancy format "text/red" for increased 471 robustness of real-time text transmission. This document updates 472 [RFC4103] by introducing a capability negotiation for handling multi- 473 party real-time text, a way to indicate the source of transmitted 474 text, and rules for efficient timing of the transmissions interleaved 475 from different sources. 477 The capability negotiation for the "RTP-mixer based multi-party 478 method" is based on use of the SDP media attribute "rtt-mixer". 480 Both parties SHALL indicate their capability in a session setup or 481 modification, and evaluate the capability of the counterpart. 483 The syntax is as follows: 484 "a=rtt-mixer" 486 If any other method for RTP-based multi-party real-time text gets 487 specified, it is assumed that it will be recognized by some specific 488 SDP feature exchange. 490 It is possible to both indicate capability for the RTP-mixer based 491 method and another method. An answer MUST NOT accept more than one 492 method. 494 2.4. Actions depending on capability negotiation result 496 A transmitting party SHALL send text according to the RTP-mixer based 497 multi-party method only when the negotiation for that method was 498 successful and when it conveys text for another source. In all other 499 cases, the packets SHALL be populated and interpreted as for a two- 500 party session. 502 A party which has negotiated the "rtt-mixer" SDP media attribute MUST 503 populate the CSRC-list and format the packets according to Section 3 504 if it acts as an rtp-mixer and sends multi-party text. 506 A party which has negotiated the "rtt-mixer" SDP media attribute MUST 507 interpret the contents of the "CC" field, the CSRC-list and the 508 packets according to Section 3 in received RTP packets in the 509 corresponding RTP stream. 511 A party which has not successfully completed the negotiation of the 512 "rtt-mixer" SDP media attribute MUST NOT transmit packets interleaved 513 from different sources in the same RTP stream as specified in 514 Section 3. If the party is a mixer and did declare the "rtt-mixer" 515 SDP media attribute, it SHOULD perform the procedure for multi-party 516 unaware endpoints. If the party is not a mixer, it SHOULD transmit 517 according to [RFC4103]. 519 3. Details for the RTP-mixer based multi-party aware mixing method 521 3.1. Use of fields in the RTP packets 523 The CC field SHALL show the number of members in the CSRC list, which 524 SHALL be one (1) in transmissions from a mixer when conveying text 525 from other sources in a multi-party session, and otherwise 0. 527 When text is conveyed by a mixer during a multi-party session, a CSRC 528 list SHALL be included in the packet. The single member in the CSRC- 529 list SHALL contain the SSRC of the source of the T140blocks in the 530 packet. 532 When redundancy is used, the RECOMMENDED level of redundancy is to 533 use one primary and two redundant generations of T140blocks. In some 534 cases, a primary or redundant T140block is empty, but is still 535 represented by a member in the redundancy header. 537 From other aspects, the contents of the RTP packets are equal to what 538 is specified in [RFC4103]. 540 3.2. Initial transmission of a BOM character 542 As soon as a participant is known to participate in a session with 543 another entity and is available for text reception, a Unicode BOM 544 character SHALL be sent to it by the other entity according to the 545 procedures in this section. If the transmitter is a mixer, then the 546 source of this character SHALL be indicated to be the mixer itself. 548 Note that the BOM character SHALL be transmitted with the same 549 redundancy procedures as any other text. 551 3.3. Keep-alive 553 After that, the transmitter SHALL send keep-alive traffic to the 554 receiver(s) at regular intervals when no other traffic has occurred 555 during that interval, if that is decided for the actual connection. 556 It is RECOMMENDED to use the keep-alive solution from [RFC6263]. The 557 consent check of [RFC7675] is a possible alternative if it is used 558 anyway for other reasons. 560 3.4. Transmission interval 562 A "text/red" or "text/t140" transmitter in a mixer SHALL send packets 563 distributed in time as long as there is something (new or redundant 564 T140blocks) to transmit. The maximum transmission interval SHALL 565 then be 330 ms, when no other limitations cause a longer interval to 566 be temporarily used. It is RECOMMENDED to send the next packet to a 567 receiver as soon as new text to that receiver is available, as long 568 as the maximum character rate ("CPS") to the receiver is not exceeded 569 during any 10 second interval. The intention of these time intervals 570 is to keep the latency low and network load limited while keeping a 571 good protection against text loss in bursty packet loss conditions. 572 The main purpose of the 330 ms interval is for timing of redundant 573 transmission, when no new text from the same source is available. 575 If the "CPS" value is reached, longer transmission intervals SHALL be 576 applied and only part of the text queued for transmission sent at end 577 of each transmission interval, until the transmission rate falls 578 under the "CPS" value again. See also Section 8 580 For a transmitter not acting in a mixer, the transmission interval 581 principles from [RFC4103] apply, and the transmission interval SHALL 582 be 300 ms. 584 3.5. Only one source per packet 586 New text and redundant copies of earlier text from one source SHALL 587 be transmitted in the same packet if available for transmission at 588 the same time. Text from different sources MUST NOT be transmitted 589 in the same packet. 591 3.6. Do not send received text to the originating source 593 Text received by a mixer from a participant SHOULD NOT be included in 594 transmission from the mixer to that participant, because the normal 595 behavior of the endpoint is to present locally produced locally. 597 3.7. Clean incoming text 599 A mixer SHALL handle reception, recovery from packet loss, deletion 600 of superfluous redundancy, marking of possible text loss and deletion 601 of 'BOM' characters from each participant before queueing received 602 text for transmission to receiving participants. 604 3.8. Redundant transmission principles 606 A transmitting party using redundancy SHALL send redundant 607 repetitions of T140blocks already transmitted in earlier packets. 609 The number of redundant generations of T140blocks to include in 610 transmitted packets SHALL be deduced from the SDP negotiation. It 611 SHALL be set to the minimum of the number declared by the two parties 612 negotiating a connection. It is RECOMMENDED to declare and transmit 613 one original and two redundant generations of the T140blocks, because 614 that provides good protection against text loss in case of packet 615 loss, and low overhead. 617 3.9. Interleaving text from different sources 619 When text from more than one source is available for transmission 620 from a mixer, the mixer SHALL let the sources take turns in having 621 their text transmitted. 623 The source with the oldest text received in the mixer or oldest 624 redundant text SHALL be next in turn to get all its available unsent 625 text transmitted. Any redundant repetitions of earlier transmitted 626 text not yet sent the intended number of times SHALL be included as 627 redundant retransmission in the transmission. 629 3.10. Text placement in packets 631 The mixer SHALL compose and transmit an RTP packet to a receiver when 632 one of the following conditions has occurred: 634 * There is unsent text available for transmission to that receiver. 636 * 330 ms has passed since already transmitted text was queued for 637 transmission as redundant text. 639 At time of transmission, the mixer SHALL populate the RTP packet with 640 all T140blocks queued for transmission originating from the source in 641 turn for transmission as long as this is not in conflict with the 642 allowed number of characters per second ("CPS") or the maximum packet 643 size. In this way, the latency of the latest received text is kept 644 low even in moments of simultaneous transmission from many sources. 646 Redundant text SHALL also be included. See Section 3.12 648 The SSRC of the source SHALL be placed as the only member in the 649 CSRC-list. 651 Note: The CSRC-list in an RTP packet only includes the participant 652 whose text is included in text blocks. It is not the same as the 653 total list of participants in a conference. With audio and video 654 media, the CSRC-list would often contain all participants who are not 655 muted whereas text participants that don't type are completely silent 656 and thus are not represented in RTP packet CSRC-lists. 658 3.11. Empty T140blocks 660 If no unsent T140blocks were available for a source at the time of 661 populating a packet, but T140blocks are available which have not yet 662 been sent the full intended number of redundant transmissions, then 663 the primary T140block for that source is composed of an empty 664 T140block, and populated (without taking up any length) in a packet 665 for transmission. The corresponding SSRC SHALL be placed as usual in 666 its place in the CSRC-list. 668 The first packet in the session, the first after a source switch and 669 the first after a pause SHALL be poulated with the available 670 T140blocks for the source in turn to be sent as primary, and empty 671 T140blocks for the agreed number of redundancy generations. 673 3.12. Creation of the redundancy 675 The primary T140block from a source in the latest transmitted packet 676 is saved for populating the first redundant T140block for that source 677 in next transmission of text from that source. The first redundant 678 T140block for that source from the latest transmission is saved for 679 populating the second redundant T140block in next transmission of 680 text from that source. 682 Usually this is the level of redundancy used. If a higher number of 683 redundancy is negotiated, then the procedure SHALL be maintained 684 until all available redundant levels of T140blocks are placed in the 685 packet. If a receiver has negotiated a lower number of "text/red" 686 generations, then that level SHALL be the maximum used by the 687 transmitter. 689 The T140blocks saved for transmission as redundant data are assigned 690 a planned transmission time 330 ms after the current time, but SHOULD 691 be transmitted earlier if new text for the same source gets in turn 692 for transmission before that time. 694 3.13. Timer offset fields 696 The timestamp offset values SHALL be inserted in the redundancy 697 header, with the time offset from the RTP timestamp in the packet 698 when the corresponding T140block was sent as primary. 700 The timestamp offsets are expressed in the same clock tick units as 701 the RTP timestamp. 703 The timestamp offset values for empty T140blocks have no relevance 704 but SHOULD be assigned realistic values. 706 3.14. Other RTP header fields 708 The number of members in the CSRC list ( 0 or 1) SHALL be placed in 709 the "CC" header field. Only mixers place value 1 in the "CC" field. 710 A value of "0" indicates that the source is the transmitting device 711 itself and that the source is indicated by the SSRC field. This 712 value is used by endpoints, and by mixers sending data that it is 713 source of itself. 715 The current time SHALL be inserted in the timestamp. 717 The SSRC of the mixer for the RTT session SHALL be inserted in the 718 SSRC field of the RTP header. 720 The M-bit SHALL be handled as specified in [RFC4103]. 722 3.15. Pause in transmission 724 When there is no new T140block to transmit, and no redundant 725 T140block that has not been retransmitted the intended number of 726 times from any source, the transmission process SHALL be stopped 727 until either new T140blocks arrive, or a keep-alive method calls for 728 transmission of keep-alive packets. 730 3.16. RTCP considerations 732 A mixer SHALL send RTCP reports with SDES, CNAME and NAME information 733 about the sources in the multi-party call. This makes it possible 734 for participants to compose a suitable label for text from each 735 source. 737 Integrity SHALL be considered when composing these fields. They 738 contain name and address information that may be sensitive to 739 transmit in its entirety e.g. to unauthenticated participants. 740 Similar considerations SHALL be taken as for other media. 742 3.17. Reception of multi-party contents 744 The "text/red" receiver included in an endpoint with presentation 745 functions will receive RTP packets in the single stream from the 746 mixer, and SHALL distribute the T140blocks for presentation in 747 presentation areas for each source. Other receiver roles, such as 748 gateways or chained mixers are also feasible, and requires 749 consideration if the stream shall just be forwarded, or distributed 750 based on the different sources. 752 3.17.1. Acting on the source of the packet contents 754 If the "CC" field value of a received packet is 1, it indicates that 755 the text is conveyed from a source indicated in the single member in 756 the CSRC-list, and the receiver MUST act on the source according to 757 its role. If the CC value is 0, the source is indicated in the SSRC 758 field. 760 3.17.2. Detection and indication of possible text loss 762 The RTP sequence numbers of the received packets SHALL be monitored 763 for gaps and packets out of order. If a sequence number gap appears 764 and still exists after some defined short time for jitter resolution, 765 the packets in the gap SHALL be regarded as lost. 767 If it is known that only one source is active in the RTP session, 768 then it is likely that a gap equal to or larger than the agreed 769 number of redundancy generations (including the primary) causes text 770 loss. In that case a t140block SHALL be created with a marker for 771 possible text loss [T140ad1] and assigned to the source and inserted 772 in the reception buffer for that source. 774 If it is known that more than one source is active in the RTP 775 session, then it is not possible in general to evaluate if text was 776 lost when packets were lost. With two active sources and the 777 recommended number of redundancy generations (3), it can take a gap 778 of five consecutive lost packets until any text may be lost, but text 779 loss can also appear if three non-consecutive packets are lost when 780 they contained consecutive data from the same source. A simple 781 method to decide when there is risk for resulting text loss is to 782 evaluate if three or more packets were lost within one second. If 783 this simple method is used, then a t140block SHOULD be created with a 784 marker for possible text loss [T140ad1] and assigned to the SSRC of 785 the transmitter as a general input from the mixer. 787 Implementations MAY apply more refined methods for more reliable 788 detection of if text was lost or not. Any refined method SHALL 789 prefer marking possible loss rather than not marking when it is 790 uncertain if there was loss. 792 3.17.3. Extracting text and handling recovery 794 When applying the following procedures, the effects MUST be 795 considered of possible timestamp wrap around and the RTP session 796 possibly changing SSRC. 798 When a packet is received in an RTP session using the packetization 799 for multi-party aware endpoints, its T140blocks SHALL be extracted in 800 the following way. The description is adapted to the default 801 redundancy case using the original and two redundant generations. 803 The source SHALL be extracted from the CSRC-list if available, 804 otherwise from the SSRC. 806 If the received packet is the first packet received from the source, 807 then all T140blocks in the packet SHALL be retrieved and assigned to 808 a receive buffer for the source beginning with the second generation 809 redundancy, continuing with the first generation redundancy and 810 finally the primary. 812 Note: The normal case is that in the first packet, only the primary 813 data has contents. The redundant data has contents in the first 814 received packet from a source only after initial packet loss. 816 If the packet is not the first packet from a source, then if the 817 second generation redundant data is available, its timestamp SHALL be 818 created by subtracting its timestamp offset from the RTP timestamp. 819 If the resulting timestamp is later than the latest retrieved data 820 from the same source, then the redundant data SHALL be retrieved and 821 appended to the receive buffer. The process SHALL be continued in 822 the same way for the first generation redundant data. After that, 823 the primary data SHALL be retrieved from the packet and appended to 824 the receive buffer for the source. 826 3.17.4. Delete 'BOM' 828 Unicode character 'BOM' is used as a start indication and sometimes 829 used as a filler or keep alive by transmission implementations. 830 These SHALL be deleted after extraction from received packets. 832 3.18. Performance considerations 834 This solution has good performance with low text delays as long as 835 the sum of characters per second during any 10 second interval sent 836 from a number of simultaneously sending participants to a receiving 837 participant does not reach the 'CPS' value. At higher numbers of 838 characters per second sent, a jerkiness is visible in the 839 presentation of text. The solution is therefore suitable for 840 emergency service use, relay service use, and small or well-managed 841 larger multimedia conferences. Only in large unmanaged conferences 842 with a high number of participants there may on very rare occasions 843 appear situations when many participants happen to send text 844 simultaneously, resulting in unpleasantly jerky presentation of text 845 from each sending participant. It should be noted that it is only 846 the number of users sending text within the same moment that causes 847 jerkiness, not the total number of users with RTT capability. 849 3.19. Security for session control and media 851 Security SHOULD be applied when possible regarding the capabilities 852 of the participating devices by use of SIP over TLS by default 853 according to [RFC5630] section 3.1.3 on session control level and by 854 default using DTLS-SRTP [RFC5764] on media level. In applications 855 where legacy endpoints without security may exist, a negotiation 856 SHOULD be performed to decide if security by encryption on media 857 level will be applied. If no other security solution is mandated for 858 the application, then OSRTP [RFC8643] is a suitable method be applied 859 to negotiate SRTP media security with DTLS. Most SDP examples below 860 are for simplicity expressed without the security additions. The 861 principles (but not all details) for applying DTLS-SRTP [RFC5764] 862 security is shown in a couple of the following examples. 864 3.20. SDP offer/answer examples 866 This section shows some examples of SDP for session negotiation of 867 the real-time text media in SIP sessions. Audio is usually provided 868 in the same session, and sometimes also video. The examples only 869 show the part of importance for the real-time text media. The 870 examples relate to the single RTP stream mixing for multi-party aware 871 endpoints and for multi-party unaware endpoints. 873 Note: Multi-party RTT MAY also be provided through other methods, 874 e.g. by a Selective Forwarding Middlebox (SFM). In that case, the 875 SDP of the offer will include something specific for that method, and 876 an answer acknowledging the use of that method would accept it by 877 something specific included in the SDP. The offer may contain also 878 the "rtt-mixer" SDP media attribute for the main RTT media when the 879 offeror has capability for both multi-party methods, while an answer, 880 selecting to use SFM will not include the "rtt-mixer" SDP media 881 attribute. 883 Offer example for "text/red" format and multi-party support: 885 m=text 11000 RTP/AVP 100 98 886 a=rtpmap:98 t140/1000 887 a=rtpmap:100 red/1000 888 a=fmtp:100 98/98/98 889 a=rtt-mixer 891 Answer example from a multi-party capable device 892 m=text 14000 RTP/AVP 100 98 893 a=rtpmap:98 t140/1000 894 a=rtpmap:100 red/1000 895 a=fmtp:100 98/98/98 896 a=rtt-mixer 898 Offer example for "text/red" format including multi-party 899 and security: 900 a=fingerprint: (fingerprint1) 901 m=text 11000 RTP/AVP 100 98 902 a=rtpmap:98 t140/1000 903 a=rtpmap:100 red/1000 904 a=fmtp:100 98/98/98 905 a=rtt-mixer 907 The "fingerprint" is sufficient to offer DTLS-SRTP, with the media 908 line still indicating RTP/AVP. 910 Note: For brevity, the entire value of the SDP fingerprint attribute 911 is not shown in this and the following example. 913 Answer example from a multi-party capable device with security 914 a=fingerprint: (fingerprint2) 915 m=text 16000 RTP/AVP 100 98 916 a=rtpmap:98 t140/1000 917 a=rtpmap:100 red/1000 918 a=fmtp:100 98/98/98 919 a=rtt-mixer 921 With the "fingerprint" the device acknowledges use of SRTP/DTLS. 923 Answer example from a multi-party unaware device that also 924 does not support security: 926 m=text 12000 RTP/AVP 100 98 927 a=rtpmap:98 t140/1000 928 a=rtpmap:100 red/1000 929 a=fmtp:100 98/98/98 931 3.21. Packet sequence example from interleaved transmission 933 This example shows a symbolic flow of packets from a mixer including 934 loss and recovery. The sequence includes interleaved transmission of 935 text from two RTT sources A and B. P indicates primary data. R1 is 936 first redundant generation data and R2 is the second redundant 937 generation data. A1, B1, A2 etc are text chunks (T140blocks) 938 received from the respective sources and sent on to the receiver by 939 the mixer. X indicates dropped packet between the mixer and a 940 receiver. The session is assumed to use original and two redundant 941 generations of RTT. 943 |-----------------------| 944 |Seq no 101, Time=20400 | 945 |CC=1 | 946 |CSRC list A | 947 |R2: A1, Offset=600 | 948 |R1: A2, Offset=300 | 949 |P: A3 | 950 |-----------------------| 952 Assuming that earlier packets ( with text A1 and A2) were received in 953 sequence, text A3 is received from packet 101 and assigned to 954 reception area A. The mixer is now assumed to have received text 955 from source B 100 ms after packet 101 and will send that text. 956 Transmission of A2 and A3 as redundancy is planned for 330 ms after 957 packet 101 if no new text from A is ready to be sent before that. 959 |-----------------------| 960 |Seq no 102, Time=20500 | 961 |CC=1 | 962 |CSRC list B | 963 |R2 Empty, Offset=600 | 964 |R1: Empty, Offset=300 | 965 |P: B1 | 966 |-----------------------| 967 Packet 102 is received. 968 B1 is retrieved from this packet. Redundant transmission of 969 B1 is planned 330 ms after packet 102. 971 X------------------------| 972 X Seq no 103, Timer=20730| 973 X CC=1 | 974 X CSRC list A | 975 X R2: A2, Offset=630 | 976 X R1: A3, Offset=330 | 977 X P: Empty | 978 X------------------------| 979 Packet 103 is assumed to be lost due to network problems. 980 It contains redundancy for A. Sending A3 as second level 981 redundancy is planned for 330 ms after packet 103. 983 X------------------------| 984 X Seq no 104, Timer=20830| 985 X CC=1 | 986 X CSRC list B | 987 X R2: Empty, Offset=600 | 988 X R1: B1, Offset=300 | 989 X P: B2 | 990 X------------------------| 991 Packet 104 contains text from B, including new B2 and 992 redundant B1. It is assumed dropped in network 993 problems. 994 The mixer has A3 redundancy to send but no new text 995 appears from A and therefore the redundancy is sent 996 330 ms after the previous packet with text from A. 998 |------------------------| 999 | Seq no 105, Timer=21060| 1000 | CC=1 | 1001 | CSRC list A | 1002 | R2: A3, Offset=660 | 1003 | R1: Empty, Offset=330 | 1004 | P: Empty | 1005 |------------------------| 1006 Packet 105 is received. 1007 A gap for lost 103, and 104 is detected. 1008 Assume that no other loss was detected the last second. 1009 Then it can be concluded that nothing was totally lost. 1011 R2 is checked. Its original time was 21040-660=20400. 1012 A packet with text from A was received with that 1013 timestamp, so nothing needs to be recovered. 1015 B1 and B2 still needs to be transmitted as redundancy. 1016 This is planned 330 ms after packet 105. That 1017 would be at 21150. 1019 |-----------------------| 1020 |Seq no 106, Timer=21160| 1021 |CC=1 | 1022 |CSRC list B | 1023 | R2: B1, Offset=660 | 1024 | R1: B2, Offset=330 | 1025 | P: Empty | 1026 |-----------------------| 1028 Packet 106 is received. 1030 The second level redundancy in packet 106 is B1 and has timestamp 1031 offset 660 ms. The timestamp of packet 106 minus 660 is 20500 which 1032 is the timestamp of packet 102 THAT was received. So B1 does not 1033 need to be retrieved. The first level redundancy in packet 106 has 1034 offset 330. The timestamp of packet 106 minus 330 is 20830. That is 1035 later than the latest received packet with source B. Therefore B2 is 1036 retrieved and assigned to the input buffer for source B. No primary 1037 is available in packet 106. 1039 After this sequence, A3 and B1 and B2 have been received. In this 1040 case no text was lost. 1042 3.22. Maximum character rate "CPS" 1044 The default maximum rate of reception of "text/t140" real-time text 1045 is in [RFC4103] specified to be 30 characters per second. The value 1046 MAY be modified in the "CPS" parameter of the FMTP attribute in the 1047 media section for the "text/t140" media. A mixer combining real-time 1048 text from a number of sources may occasionally have a higher combined 1049 flow of text coming from the sources. Endpoints SHOULD therefore 1050 specify a suitable higher value for the "CPS" parameter, 1051 corresponding to its real reception capability. A value for "CPS" of 1052 90 SHALL be the default for the "text/t140" stream in the "text/red" 1053 format when multi-party real-time text is negotiated. See [RFC4103] 1054 for the format and use of the "CPS" parameter. The same rules apply 1055 for the multi-party case except for the default value. 1057 4. Presentation level considerations 1059 "Protocol for multimedia application text conversation" [T140] 1060 provides the presentation level requirements for the [RFC4103] 1061 transport. Functions for erasure and other formatting functions and 1062 are specified in [T140] which has the following general statement for 1063 the presentation: 1065 "The display of text from the members of the conversation should be 1066 arranged so that the text from each participant is clearly readable, 1067 and its source and the relative timing of entered text is visualized 1068 in the display. Mechanisms for looking back in the contents from the 1069 current session should be provided. The text should be displayed as 1070 soon as it is received." 1072 Strict application of [T140] is of essence for the interoperability 1073 of real-time text implementations and to fulfill the intention that 1074 the session participants have the same information of the text 1075 contents of the conversation without necessarily having the exact 1076 same layout of the conversation. 1078 [T140] specifies a set of presentation control codes to include in 1079 the stream. Some of them are optional. Implementations MUST be able 1080 to ignore optional control codes that they do not support. 1082 There is no strict "message" concept in real-time text. The Unicode 1083 Line Separator character SHALL be used as a separator allowing a part 1084 of received text to be grouped in presentation. The characters 1085 "CRLF" may be used by other implementations as replacement for Line 1086 Separator. The "CRLF" combination SHALL be erased by just one 1087 erasing action, just as the Line Separator. Presentation functions 1088 are allowed to group text for presentation in smaller groups than the 1089 line separators imply and present such groups with source indication 1090 together with text groups from other sources (see the following 1091 presentation examples). Erasure has no specific limit by any 1092 delimiter in the text stream. 1094 4.1. Presentation by multi-party aware endpoints 1096 A multi-party aware receiving party, presenting real-time text MUST 1097 separate text from different sources and present them in separate 1098 presentation fields. The receiving party MAY separate presentation 1099 of parts of text from a source in readable groups based on other 1100 criteria than line separator and merge these groups in the 1101 presentation area when it benefits the user to most easily find and 1102 read text from the different participants. The criteria MAY e.g. be 1103 a received comma, full stop, or other phrase delimiters, or a long 1104 pause. 1106 When text is received from multiple original sources, the 1107 presentation SHALL provide a view where text is added in multiple 1108 presentation fields. 1110 If the presentation presents text from different sources in one 1111 common area, the presenting endpoint SHOULD insert text from the 1112 local user ended at suitable points merged with received text to 1113 indicate the relative timing for when the text groups were completed. 1114 In this presentation mode, the receiving endpoint SHALL present the 1115 source of the different groups of text. This presentation style is 1116 called the "chat" style here and provides a possibility to follow 1117 text arriving from multiple parties and the approximate relative time 1118 that text is received related to text from the local user. 1120 A view of a three-party RTT call in chat style is shown in this 1121 example . 1123 _________________________________________________ 1124 | |^| 1125 |[Alice] Hi, Alice here. |-| 1126 | | | 1127 |[Bob] Bob as well. | | 1128 | | | 1129 |[Eve] Hi, this is Eve, calling from Paris. | | 1130 | I thought you should be here. | | 1131 | | | 1132 |[Alice] I am coming on Thursday, my | | 1133 | performance is not until Friday morning.| | 1134 | | | 1135 |[Bob] And I on Wednesday evening. | | 1136 | | | 1137 |[Alice] Can we meet on Thursday evening? | | 1138 | | | 1139 |[Eve] Yes, definitely. How about 7pm. | | 1140 | at the entrance of the restaurant | | 1141 | Le Lion Blanc? | | 1142 |[Eve] we can have dinner and then take a walk |-| 1143 |______________________________________________|v| 1144 | But I need to be back to |^| 1145 | the hotel by 11 because I need |-| 1146 | | | 1147 | I wou |-| 1148 |______________________________________________|v| 1149 | of course, I underst | 1150 |________________________________________________| 1152 Figure 3: Example of a three-party RTT call presented in chat style 1153 seen at participant 'Alice's endpoint. 1155 Other presentation styles than the chat style MAY be arranged. 1157 This figure shows how a coordinated column view MAY be presented. 1159 _____________________________________________________________________ 1160 | Bob | Eve | Alice | 1161 |____________________|______________________|_______________________| 1162 | | |I will arrive by TGV. | 1163 |My flight is to Orly| |Convenient to the main | 1164 | |Hi all, can we plan |station. | 1165 | |for the seminar? | | 1166 |Eve, will you do | | | 1167 |your presentation on| | | 1168 |Friday? |Yes, Friday at 10. | | 1169 |Fine, wo | |We need to meet befo | 1170 |___________________________________________________________________| 1171 Figure 4: An example of a coordinated column-view of a three-party 1172 session with entries ordered vertically in approximate time-order. 1174 4.2. Multi-party mixing for multi-party unaware endpoints 1176 When the mixer has indicated RTT multi-party capability in an SDP 1177 negotiation, but the multi-party capability negotiation fails with an 1178 endpoint, then the agreed "text/red" or "text/t140" format SHALL be 1179 used and the mixer SHOULD compose a best-effort presentation of 1180 multi-party real-time text in one stream intended to be presented by 1181 an endpoint with no multi-party awareness, when that is desired in 1182 the actual implementation. The following specifies a procedure which 1183 MAY be applied in that situation. 1185 This presentation format has functional limitations and SHOULD be 1186 used only to enable participation in multi-party calls by legacy 1187 deployed endpoints implementing only RFC 4103 without any multi-party 1188 extensions specified in this document. 1190 The principles and procedures below do not specify any new protocol 1191 elements. They are instead composed from the information in [T140] 1192 and an ambition to provide a best effort presentation on an endpoint 1193 which has functions only for two-party calls. 1195 The mixer mixing for multi-party unaware endpoints SHALL compose a 1196 simulated limited multi-party RTT view suitable for presentation in 1197 one presentation area. The mixer SHALL group text in suitable groups 1198 and prepare for presentation of them by inserting a new line between 1199 them if the transmitted text did not already end with a new line. A 1200 presentable label SHALL be composed and sent for the source initially 1201 in the session and after each source switch. With this procedure the 1202 time for switching from transmission of text from one source to 1203 transmission of text from another source is depending on the actions 1204 of the users. In order to expedite source switch, a user can for 1205 example end its turn with a new line. 1207 4.2.1. Actions by the mixer at reception from the call participants 1209 When text is received by the mixer from the different participants, 1210 the mixer SHALL recover text from redundancy if any packets are lost. 1211 The mark for lost text [T140ad1] SHALL be inserted in the stream if 1212 unrecoverable loss appears. Any Unicode "BOM" characters, possibly 1213 used for keep-alive SHALL be deleted. The time of creation of text 1214 (retrieved from the RTP timestamp) SHALL be stored together with the 1215 received text from each source in queues for transmission to the 1216 recipients in order to be able to evaluate text loss. 1218 4.2.2. Actions by the mixer for transmission to the recipients 1220 The following procedure SHALL be applied for each multi-party unaware 1221 recipient of multi-party text from the mixer. 1223 The text for transmission SHALL be formatted by the mixer for each 1224 receiving user for presentation in one single presentation area. 1225 Text received from a participant SHOULD NOT be included in 1226 transmission to that participant because it is usually presented 1227 locally at transmission time. When there is text available for 1228 transmission from the mixer to a receiving party from more than one 1229 participant, the mixer SHALL switch between transmission of text from 1230 the different sources at suitable points in the transmitted stream. 1232 When switching source, the mixer SHALL insert a line separator if the 1233 already transmitted text did not end with a new line (line separator 1234 or CRLF). A label SHALL be composed from information in the CNAME 1235 and NAME fields in RTCP reports from the participant to have its text 1236 transmitted, or from other session information for that user. The 1237 label SHALL be delimited by suitable characters (e.g. '[ ]') and 1238 transmitted. The CSRC SHALL indicate the selected source. Then text 1239 from that selected participant SHALL be transmitted until a new 1240 suitable point for switching source is reached. 1242 Integrity considerations SHALL be taken when composing the label. 1244 Seeking a suitable point for switching source SHALL be done when 1245 there is older text waiting for transmission from any party than the 1246 age of the last transmitted text. Suitable points for switching are: 1248 * A completed phrase ended by comma 1250 * A completed sentence 1252 * A new line (line separator or CRLF) 1254 * A long pause (e.g. > 10 seconds) in received text from the 1255 currently transmitted source 1257 * If text from one participant has been transmitted with text from 1258 other sources waiting for transmission for a long time (e.g. > 1 1259 minute) and none of the other suitable points for switching has 1260 occurred, a source switch MAY be forced by the mixer at next word 1261 delimiter, and also if even a word delimiter does not occur within 1262 a time (e.g. 15 seconds) after the scan for word delimiter 1263 started. 1265 When switching source, the source which has the oldest text in queue 1266 SHALL be selected to be transmitted. A character display count SHALL 1267 be maintained for the currently transmitted source, starting at zero 1268 after the label is transmitted for the currently transmitted source. 1270 The status SHALL be maintained for the latest control code for Select 1271 Graphic Rendition (SGR) from each source. If there is an SGR code 1272 stored as the status for the current source before the source switch 1273 is done, a reset of SGR SHALL be sent by the sequence SGR 0 [009B 1274 0000 006D] after the new line and before the new label during a 1275 source switch. See SGR below for an explanation. This transmission 1276 does not influence the display count. 1278 If there is an SGR code stored for the new source after the source 1279 switch, that SGR code SHALL be transmitted to the recipient before 1280 the label. This transmission does not influence the display count. 1282 4.2.3. Actions on transmission of text 1284 Text from a source sent to the recipient SHALL increase the display 1285 count by one per transmitted character. 1287 4.2.4. Actions on transmission of control codes 1289 The following control codes specified by T.140 require specific 1290 actions. They SHALL cause specific considerations in the mixer. 1291 Note that the codes presented here are expressed in UCS-16, while 1292 transmission is made in UTF-8 transform of these codes. 1294 BEL 0007 Bell Alert in session, provides for alerting during an 1295 active session. The display count SHALL NOT be altered. 1297 NEW LINE 2028 Line separator. Check and perform a source switch if 1298 appropriate. Increase display count by 1. 1300 CR LF 000D 000A A supported, but not preferred way of requesting a 1301 new line. Check and perform a source switch if appropriate. 1302 Increase display count by 1. 1304 INT ESC 0061 Interrupt (used to initiate mode negotiation 1305 procedure). The display count SHALL NOT be altered. 1307 SGR 009B Ps 006D Select graphic rendition. Ps is rendition 1308 parameters specified in ISO 6429. The display count SHALL NOT be 1309 altered. The SGR code SHOULD be stored for the current source. 1311 SOS 0098 Start of string, used as a general protocol element 1312 introducer, followed by a maximum 256 bytes string and the ST. 1313 The display count SHALL NOT be altered. 1315 ST 009C String terminator, end of SOS string. The display count 1316 SHALL NOT be altered. 1318 ESC 001B Escape - used in control strings. The display count SHALL 1319 NOT be altered for the complete escape code. 1321 Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used 1322 for synchronization and keep-alive, SHALL be deleted from incoming 1323 streams. It SHALL also be sent first after session establishment 1324 to the recipient. The display count SHALL NOT be altered. 1326 Missing text mark (U+FFFD) "Replacement character", represented as a 1327 question mark in a rhombus, or if that is not feasible, replaced 1328 by an apostrophe ', marks place in stream of possible text loss. 1329 This mark SHALL be inserted by the reception procedure in case of 1330 unrecoverable loss of packets. The display count SHALL be 1331 increased by one when sent as for any other character. 1333 SGR If a control code for selecting graphic rendition (SGR), other 1334 than reset of the graphic rendition (SGR 0) is sent to a 1335 recipient, that control code SHALL also be stored as status for 1336 the source in the storage for SGR status. If a reset graphic 1337 rendition (SGR 0) originated from a source is sent, then the SGR 1338 status storage for that source SHALL be cleared. The display 1339 count SHALL NOT be increased. 1341 BS (U+0008) Back Space, intended to erase the last entered character 1342 by a source. Erasure by backspace cannot always be performed as 1343 the erasing party intended. If an erasing action erases all text 1344 up to the end of the leading label after a source switch, then the 1345 mixer MUST NOT transmit more backspaces. Instead it is 1346 RECOMMENDED that a letter "X" is inserted in the text stream for 1347 each backspace as an indication of the intent to erase more. A 1348 new line is usually coded by a Line Separator, but the character 1349 combination "CRLF" MAY be used instead. Erasure of a new line is 1350 in both cases done by just one erasing action (Backspace). If the 1351 display count has a positive value it SHALL be decreased by one 1352 when the BS is sent. If the display count is at zero, it SHALL 1353 NOT not altered. 1355 4.2.5. Packet transmission 1357 A mixer transmitting to a multi-party unaware terminal SHALL send 1358 primary data only from one source per packet. The SSRC SHALL be the 1359 SSRC of the mixer. The CSRC list SHALL contain one member and be the 1360 SSRC of the source of the primary data. 1362 4.2.6. Functional limitations 1364 When a multi-party unaware endpoint presents a conversation in one 1365 display area in a chat style, it inserts source indications for 1366 remote text and local user text as they are merged in completed text 1367 groups. When an endpoint using this layout receives and presents 1368 text mixed for multi-party unaware endpoints, there will be two 1369 levels of source indicators for the received text; one generated by 1370 the mixer and inserted in a label after each source switch, and 1371 another generated by the receiving endpoint and inserted after each 1372 switch between local and remote source in the presentation area. 1373 This will waste display space and look inconsistent to the reader. 1375 New text can be presented only from one source at a time. Switch of 1376 source to be presented takes place at suitable places in the text, 1377 such as end of phrase, end of sentence, line separator and 1378 inactivity. Therefore the time to switch to present waiting text 1379 from other sources may become long and will vary and depend on the 1380 actions of the currently presented source. 1382 Erasure can only be done up to the latest source switch. If a user 1383 tries to erase more text, the erasing actions will be presented as 1384 letter X after the label. 1386 Text loss because of network errors may hit the label between entries 1387 from different parties, causing risk for misunderstanding from which 1388 source a piece of text is. 1390 These facts make it strongly RECOMMENDED to implement multi-party 1391 awareness in RTT endpoints. The use of the mixing method for multi- 1392 party-unaware endpoints should be left for use with endpoints which 1393 are impossible to upgrade to become multi-party aware. 1395 4.2.7. Example views of presentation on multi-party unaware endpoints 1397 The following pictures are examples of the view on a participant's 1398 display for the multi-party-unaware case. 1400 _________________________________________________ 1401 | Conference | Alice | 1402 |________________________|_________________________| 1403 | |I will arrive by TGV. | 1404 |[Bob]:My flight is to |Convenient to the main | 1405 |Orly. |station. | 1406 |[Eve]:Hi all, can we | | 1407 |plan for the seminar. | | 1408 | | | 1409 |[Bob]:Eve, will you do | | 1410 |your presentation on | | 1411 |Friday? | | 1412 |[Eve]:Yes, Friday at 10.| | 1413 |[Bob]: Fine, wo |We need to meet befo | 1414 |________________________|_________________________| 1416 Figure 5: Alice who has a conference-unaware client is receiving the 1417 multi-party real-time text in a single-stream. This figure shows how 1418 a coordinated column view MAY be presented on Alice's device. 1420 _________________________________________________ 1421 | |^| 1422 |[Alice] Hi, Alice here. |-| 1423 | | | 1424 |[mix](Bob) Bob as well. | | 1425 | | | 1426 |(Eve) Hi, this is Eve, calling from Paris | | 1427 | I thought you should be here. | | 1428 | | | 1429 |[Alice] I am coming on Thursday, my | | 1430 | performance is not until Friday morning.| | 1431 | | | 1432 |[mix](Bob) And I on Wednesday evening. | | 1433 | | | 1434 |(Eve) we can have dinner and then walk | | 1435 | | | 1436 |(Eve) But I need to be back to | | 1437 | the hotel by 11 because I need | | 1438 | |-| 1439 |______________________________________________|v| 1440 | of course, I underst | 1441 |________________________________________________| 1443 Figure 6: An example of a view of the multi-party unaware 1444 presentation in chat style. Alice is the local user. 1446 5. Relation to Conference Control 1447 5.1. Use with SIP centralized conferencing framework 1449 The SIP conferencing framework, mainly specified in [RFC4353], 1450 [RFC4579] and [RFC4575] is suitable for coordinating sessions 1451 including multi-party RTT. The RTT stream between the mixer and a 1452 participant is one and the same during the conference. Participants 1453 get announced by notifications when participants are joining or 1454 leaving, and further user information may be provided. The SSRC of 1455 the text to expect from joined users MAY be included in a 1456 notification. The notifications MAY be used both for security 1457 purposes and for translation to a label for presentation to other 1458 users. 1460 5.2. Conference control 1462 In managed conferences, control of the real-time text media SHOULD be 1463 provided in the same way as other for media, e.g. for muting and 1464 unmuting by the direction attributes in SDP [RFC8866]. 1466 Note that floor control functions may be of value for RTT users as 1467 well as for users of other media in a conference. 1469 6. Gateway Considerations 1471 6.1. Gateway considerations with Textphones (e.g. TTYs). 1473 Multi-party RTT sessions may involve gateways of different kinds. 1474 Gateways involved in setting up sessions SHALL correctly reflect the 1475 multi-party capability or unawareness of the combination of the 1476 gateway and the remote endpoint beyond the gateway. 1478 One case that may occur is a gateway to PSTN for communication with 1479 textphones (e.g. TTYs). Textphones are limited devices with no 1480 multi-party awareness, and it SHOULD therefore be suitable for the 1481 gateway to not indicate multi-party awareness for that case. Another 1482 solution is that the gateway indicates multi-party capability towards 1483 the mixer, and includes the multi-party mixer function for multi- 1484 party unaware endpoints itself. This solution makes it possible to 1485 make adaptations for the functional limitations of the textphone 1486 (TTY). 1488 More information on gateways to textphones (TTYs) is found in 1489 [RFC5194] 1491 6.2. Gateway considerations with WebRTC. 1493 Gateway operation to real-time text in WebRTC may also be required. 1494 In WebRTC, RTT is specified in [RFC8865]. 1496 A multi-party bridge may have functionality for communicating by RTT 1497 both in RTP streams with RTT and WebRTC T.140 data channels. Other 1498 configurations may consist of a multi-party bridge with either 1499 technology for RTT transport and a separate gateway for conversion of 1500 the text communication streams between RTP and T.140 data channel. 1502 In WebRTC, it is assumed that for a multi-party session, one T.140 1503 data channel is established for each source from a gateway or bridge 1504 to each participant. Each participant also has a data channel with 1505 two-way connection with the gateway or bridge. 1507 The t140 channel used both ways is for text from the WebRTC user and 1508 from the bridge or gateway itself to the WebRTC user. The label 1509 parameter of this t140 channel is used as NAME field in RTCP to 1510 participants on the RTP side. The other t140 channels are only for 1511 text from other participants to the WebRTC user. 1513 When a new participant has entered the session with RTP transport of 1514 RTT, a new T.140 channel SHOULD be established to WebRTC users with 1515 the label parameter composed from the NAME field in RTCP on the RTP 1516 side. 1518 When a new participant has entered the multi-party session with RTT 1519 transport in a WebRTC T.140 data channel, the new participant SHOULD 1520 be announced by a notification to RTP users. The label parameter 1521 from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP 1522 side, or other available session information. 1524 When a participant on the RTP side disappears, the corresponding 1525 T.140 data channel(s) SHOULD be closed. 1527 When a WebRTC user of T.140 data channels disconnects from the mixer, 1528 the corresponding RTP streams or sources in an RTP-mixed stream 1529 SHOULD be closed. 1531 T.140 data channels MAY be opened and closed by negotiation or 1532 renegotiation of the session or by any other valid means as specified 1533 in section 1 of [RFC8865]. 1535 7. Updates to RFC 4103 1537 This document updates [RFC4103] by introducing an SDP media attribute 1538 "rtt-mixer" for negotiation of multi-party mixing capability with the 1539 [RFC4103] format, and by specifying the rules for packets when multi- 1540 party capability is negotiated and in use. 1542 8. Congestion considerations 1544 The congestion considerations and recommended actions from [RFC4103] 1545 are valid also in multi-party situations. 1547 The first action in case of congestion SHALL be to temporarily 1548 increase the transmission interval up to two seconds. 1550 If the very unlikely situation appears that many participants in a 1551 conference send text simultaneously, a delay will build up for 1552 presentation of text at the receivers because of the limitation in 1553 characters per second("CPS") to be transmitted to the participants. 1554 More delay than 7 seconds can cause confusion in the session. It is 1555 therefore RECOMMENDED that RTP-mixer based mixer discards such text 1556 in excess and inserts a general indication of possible text loss 1557 [T140ad1] in the session. If the main text contributor is indicated 1558 in any way, the mixer MAY avoid deleting text from that participant. 1560 9. Acknowledgements 1562 James Hamlin for format and performance aspects. 1564 10. IANA Considerations 1566 10.1. Registration of the "rtt-mixer" SDP media attribute 1568 [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the 1569 RFC number of this document.] 1571 IANA is asked to register the new SDP attribute "rtt-mixer". 1573 Contact name: IESG 1575 Contact email: iesg@ietf.org 1577 Attribute name: rtt-mixer 1579 Attribute semantics: See RFCXXXX Section 2.3 1581 Attribute value: none 1583 Usage level: media 1585 Purpose: Indicate support by mixer and endpoint of multi-party 1586 mixing for real-time text transmission, using a common RTP-stream 1587 for transmission of text from a number of sources mixed with one 1588 source at a time and the source indicated in a single CSRC-list 1589 member. 1591 Charset Dependent: no 1593 O/A procedure: See RFCXXXX Section 2.3 1595 Mux Category: normal 1597 Reference: RFCXXXX 1599 11. Security Considerations 1601 The RTP-mixer model requires the mixer to be allowed to decrypt, pack 1602 and encrypt secured text from the conference participants. Therefore 1603 the mixer needs to be trusted. This is similar to the situation for 1604 central mixers of audio and video. 1606 The requirement to transfer information about the user in RTCP 1607 reports in SDES, CNAME and NAME fields, and in conference 1608 notifications, for creation of labels may have privacy concerns as 1609 already stated in RFC 3550 [RFC3550], and may be restricted for 1610 privacy reasons. The receiving user will then get a more symbolic 1611 label for the source. 1613 Participants with malicious intentions may appear and e.g. disturb 1614 the multi-party session by a continuous flow of text, or masquerade 1615 as text from other participants. Counteractions should be to require 1616 secure signaling, media and authentication, and to provide higher 1617 level conference functions e.g. for blocking and expelling 1618 participants. 1620 Further security considerations specific for this application are 1621 specified in section Section 3.19. 1623 12. Change history 1625 [RFC Editor: Please remove this section prior to publication.] 1627 12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-14 1629 Changes from comments by Murray Cucherawy during AD review. 1631 Many SHOULD in section 4.2 on multi-party unaware mixing changed to 1632 SHALL, and the whole section instead specified to be optional 1633 depending on the application. 1635 Some SHOULD in section 3 either explained or changed to SHALL. 1637 In order to have explainable conditions behind SHOULDs, the 1638 transmission interval in 3.4 is changed to as soon as text is 1639 available as a main principle. The call participants send with 300 1640 ms interval so that will create realistic load conditions anyway. 1642 12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-13 1644 Changed year to 2021. 1646 Changed reference to draft on RTT in WebRTC to recently published RFC 1647 8865. 1649 Changed label brackets in example from "[]" to "()" to avoid nits 1650 comment. 1652 Changed reference "RFC 4566" to recently published "RFC 8866" 1654 12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-12 1656 Changes according to responses on comments from Brian Rosen in 1657 Avtcore list on 2020-12-05 and -06. 1659 Changes according to responses to comments by Bernard Aboba in 1660 avtcore list 2020-12-06. 1662 Introduction of an optiona RTP multi-stream mixing method for further 1663 study as proposed by Bernard Aboba. 1665 Changes clarifying how to open and close T.140 data channels included 1666 in 6.2 after comments by Lorenzo Miniero. 1668 Changes to satisfy nits check. Some "not" changed to "NOT" in 1669 normative wording combinations. Some lower case normative words 1670 changed to upper case. A normative reference deleted from the 1671 abstract. Two informative documents moved from normative references 1672 to informative references. 1674 12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11 1676 Timestamps and timestamp offsets added to the packet examples in 1677 section 3.23, and the description corrected. 1679 A number of minor corrections added in sections 3.10 - 3.23. 1681 12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10 1683 The packet composition was modified for interleaving packets from 1684 different sources. 1686 The packet reception was modified for the new interleaving method. 1688 The packet sequence examples was adjusted for the new interleaving 1689 method. 1691 Modifications according to responses to Brian Rosen of 2020-11-03 1693 12.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09 1695 Changed name on the SDP media attribute to "rtt-mixer" 1697 Restructure of section 2 for balance between aware and unaware cases. 1699 Moved conference control to own section. 1701 Improved clarification of recovery and loss in the packet sequence 1702 example. 1704 A number of editorial corrections and improvements. 1706 12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 1708 Deleted the method requiring a new packet format "text/rex" because 1709 of the longer standardization and implementation period it needs. 1711 Focus on use of RFC 4103 text/red format with shorter transmission 1712 interval, and source indicated in CSRC. 1714 12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 1716 Added a method based on the "text/red" format and single source per 1717 packet, negotiated by the "rtt-mixer" SDP attribute. 1719 Added reasoning and recommendation about indication of loss. 1721 The highest number of sources in one packet is 15, not 16. Changed. 1723 Added in information on update to RFC 4103 that RFC 4103 explicitly 1724 allows addition of FEC method. The redundancy is a kind of forward 1725 error correction.. 1727 12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 1729 Improved definitions list format. 1731 The format of the media subtype parameters is made to match the 1732 requirements. 1734 The mapping of media subtype parameters to SDP is included. 1736 The "CPS" parameter belongs to the t140 subtype and does not need to 1737 be registered here. 1739 12.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 1741 nomenclature and editorial improvements 1743 "this document" used consistently to refer to this document. 1745 12.11. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 1747 'Redundancy header' renamed to 'data header'. 1749 More clarifications added. 1751 Language and figure number corrections. 1753 12.12. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 1755 Mention possible need to mute and raise hands as for other media. 1756 ---done ---- 1758 Make sure that use in two-party calls is also possible and explained. 1759 - may need more wording - 1761 Clarify the RTT is often used together with other media. --done-- 1763 Tell that text mixing is N-1. A users own text is not received in 1764 the mix. -done- 1766 In 3. correct the interval to: A "text/rex" transmitter SHOULD send 1767 packets distributed in time as long as there is something (new or 1768 redundant T140blocks) to transmit. The maximum transmission interval 1769 SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 1770 receiver as soon as new text to that receiver is available, as long 1771 as the time after the latest sent packet to the same receiver is more 1772 than 150 ms, and also the maximum character rate to the receiver is 1773 not exceeded. The intention is to keep the latency low while keeping 1774 a good protection against text loss in bursty packet loss conditions. 1775 -done- 1777 In 1.3 say that the format is used both ways. -done- 1779 In 13.1 change presentation area to presentation field so that reader 1780 does not think it shall be totally separated. -done- 1781 In Performance and intro, tell the performance in number of 1782 simultaneous sending users and introduced delay 16, 150 vs 1783 requirements 5 vs 500. -done -- 1785 Clarify redundancy level per connection. -done- 1787 Timestamp also for the last data header. To make it possible for all 1788 text to have time offset as for transmission from the source. Make 1789 that header equal to the others. -done- 1791 Mixer always use the CSRC list, even for its own BOM. -done- 1793 Combine all talk about transmission interval (300 ms vs when text has 1794 arrived) in section 3 in one paragraph or close to each other. -done- 1796 Documents the goal of good performance with low delay for 5 1797 simultaneous typers in the introduction. -done- 1799 Describe better that only primary text shall be sent on to receivers. 1800 Redundancy and loss must be resolved by the mixer. -done- 1802 12.13. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 1804 SDP and better description and visibility of security by OSRTP RFC 1805 8634 needed. 1807 The description of gatewaying to WebRTC extended. 1809 The description of the data header in the packet is improved. 1811 12.14. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 1813 2,5,6 More efficient format "text/rex" introduced and attribute 1814 a=rtt-mix deleted. 1816 3. Brief about use of OSRTP for security included- More needed. 1818 4. Brief motivation for the solution and why not rtp-translator is 1819 used added to intro. 1821 7. More limitations for the multi-party unaware mixing method 1822 inserted. 1824 8. Updates to RFC 4102 and 4103 more clearly expressed. 1826 9. Gateway to WebRTC started. More needed. 1828 12.15. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 1829 to draft-ietf-avtcore-multi-party-rtt-mix-00 1831 Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00 1833 Replaced CDATA in IANA registration table with better coding. 1835 Converted to xml2rfc version 3. 1837 12.16. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 1838 to -03 1840 Changed company and e-mail of the author. 1842 Changed title to "RTP-mixer formatting of multi-party Real-time text" 1843 to better match contents. 1845 Check and modification where needed of use of RFC 2119 words SHALL 1846 etc. 1848 More about the CC value in sections on transmitters and receivers so 1849 that 1-to-1 sessions do not use the mixer format. 1851 Enhanced section on presentation for multi-party-unaware endpoints 1853 A paragraph recommending CPS=150 inserted in the performance section. 1855 12.17. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 1856 to -02 1858 In Abstract and 1. Introduction: Introduced wording about regulatory 1859 requirements. 1861 In section 5: The transmission interval is decreased to 100 ms when 1862 there is text from more than one source to transmit. 1864 In section 11 about SDP negotiation, a SHOULD-requirement is 1865 introduced that the mixer should make a mix for multi-party unaware 1866 endpoints if the negotiation is not successful. And a reference to a 1867 later chapter about it. 1869 The presentation considerations chapter 14 is extended with more 1870 information about presentation on multi-party aware endpoints, and a 1871 new section on the multi-party unaware mixing with low functionality 1872 but SHOULD a be implemented in mixers. Presentation examples are 1873 added. 1875 A short chapter 15 on gateway considerations is introduced. 1877 Clarification about the text/t140 format included in chapter 10. 1879 This sentence added to the chapter 10 about use without redundancy. 1880 "The text/red format SHOULD be used unless some other protection 1881 against packet loss is utilized, for example a reliable network or 1882 transport." 1884 Note about deviation from RFC 2198 added in chapter 4. 1886 In chapter 9. "Use with SIP centralized conferencing framework" the 1887 following note is inserted: Note: The CSRC-list in an RTP packet only 1888 includes participants who's text is included in one or more text 1889 blocks. It is not the same as the list of participants in a 1890 conference. With audio and video media, the CSRC-list would often 1891 contain all participants who are not muted whereas text participants 1892 that don't type are completely silent and so don't show up in RTP 1893 packet CSRC-lists. 1895 12.18. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 1896 to -01 1898 Editorial cleanup. 1900 Changed capability indication from fmtp-parameter to SDP attribute 1901 "rtt-mix". 1903 Swapped order of redundancy elements in the example to match reality. 1905 Increased the SDP negotiation section 1907 13. References 1909 13.1. Normative References 1911 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1912 Requirement Levels", BCP 14, RFC 2119, 1913 DOI 10.17487/RFC2119, March 1997, 1914 . 1916 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1917 Jacobson, "RTP: A Transport Protocol for Real-Time 1918 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1919 July 2003, . 1921 [RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type", 1922 RFC 4102, DOI 10.17487/RFC4102, June 2005, 1923 . 1925 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 1926 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 1927 . 1929 [RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session 1930 Initiation Protocol (SIP)", RFC 5630, 1931 DOI 10.17487/RFC5630, October 2009, 1932 . 1934 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1935 Security (DTLS) Extension to Establish Keys for the Secure 1936 Real-time Transport Protocol (SRTP)", RFC 5764, 1937 DOI 10.17487/RFC5764, May 2010, 1938 . 1940 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 1941 Keeping Alive the NAT Mappings Associated with RTP / RTP 1942 Control Protocol (RTCP) Flows", RFC 6263, 1943 DOI 10.17487/RFC6263, June 2011, 1944 . 1946 [RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M. 1947 Thomson, "Session Traversal Utilities for NAT (STUN) Usage 1948 for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675, 1949 October 2015, . 1951 [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 1952 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, 1953 May 2017, . 1955 [RFC8865] Holmberg, C. and G. Hellström, "T.140 Real-Time Text 1956 Conversation over WebRTC Data Channels", RFC 8865, 1957 DOI 10.17487/RFC8865, January 2021, 1958 . 1960 [RFC8866] Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP: 1961 Session Description Protocol", RFC 8866, 1962 DOI 10.17487/RFC8866, January 2021, 1963 . 1965 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 1966 multimedia application text conversation", February 1998, 1967 . 1969 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 1970 Protocol for multimedia application text conversation", 1971 February 2000, 1972 . 1974 13.2. Informative References 1976 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 1977 Session Initiation Protocol (SIP)", RFC 4353, 1978 DOI 10.17487/RFC4353, February 2006, 1979 . 1981 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 1982 Session Initiation Protocol (SIP) Event Package for 1983 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 1984 2006, . 1986 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 1987 (SIP) Call Control - Conferencing for User Agents", 1988 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 1989 . 1991 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 1992 Time Text over IP Using the Session Initiation Protocol 1993 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 1994 . 1996 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 1997 DOI 10.17487/RFC7667, November 2015, 1998 . 2000 [RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. 2001 Stach, "An Opportunistic Approach for Secure Real-time 2002 Transport Protocol (OSRTP)", RFC 8643, 2003 DOI 10.17487/RFC8643, August 2019, 2004 . 2006 Author's Address 2008 Gunnar Hellstrom 2009 Gunnar Hellstrom Accessible Communication 2010 SE-13670 Vendelso 2011 Sweden 2013 Email: gunnar.hellstrom@ghaccess.se