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Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Missing Reference: 'Bob' is mentioned on line 1427, but not defined ** Obsolete normative reference: RFC 4566 (Obsoleted by RFC 8866) -- Possible downref: Non-RFC (?) normative reference: ref. 'T140' -- Possible downref: Non-RFC (?) normative reference: ref. 'T140ad1' Summary: 1 error (**), 0 flaws (~~), 2 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) 15 December 2020 5 Intended status: Standards Track 6 Expires: 18 June 2021 8 RTP-mixer formatting of multi-party Real-time text 9 draft-ietf-avtcore-multi-party-rtt-mix-12 11 Abstract 13 Real-time text mixers for multi-party sessions need to identify the 14 source of each transmitted group of text so that the text can be 15 presented by endpoints in suitable grouping with other text from the 16 same source, while new text from other sources is also presented in 17 readable grouping as received interleaved in real-time. 19 Use of RTT is increasing, and specifically, use in emergency calls is 20 increasing. Emergency call use requires multi-party mixing. RFC 21 4103 "RTP Payload for Text Conversation" mixer implementations can 22 use traditional RTP functions for source identification, but the 23 performance of the mixer when giving turns for the different sources 24 to transmit is limited when using the default transmission 25 characteristics with redundancy. 27 Enhancements for RFC 4103 real-time text mixing are provided in this 28 document, suitable for a centralized conference model that enables 29 source identification and rapidly interleaved transmission of text 30 from different sources. The intended use is for real-time text 31 mixers and participant endpoints capable of providing an efficient 32 presentation or other treatment of a multi-party real-time text 33 session. The specified mechanism builds on the standard use of the 34 CSRC list in the RTP packet for source identification. The method 35 makes use of the same "text/t140" and "text/red" formats as for two- 36 party sessions. 38 Solutions using multiple RTP streams in the same RTP session are 39 briefly mentioned, as they could have some benefits over the RTP- 40 mixer model. The possibility to implement the solution in a wide 41 range of existing RTP implementations made the RTP-mixer model be 42 selected to be fully specified in this document. 44 A capability exchange is specified so that it can be verified that a 45 mixer and a participant can handle the multi-party coded real-time 46 text stream using the RTP-mixer method. The capability is indicated 47 by use of an SDP media attribute "rtt-mixer". 49 The document updates RFC 4103 "RTP Payload for Text Conversation". 51 A specification of how a mixer can format text for the case when the 52 endpoint is not multi-party aware is also provided. 54 Status of This Memo 56 This Internet-Draft is submitted in full conformance with the 57 provisions of BCP 78 and BCP 79. 59 Internet-Drafts are working documents of the Internet Engineering 60 Task Force (IETF). Note that other groups may also distribute 61 working documents as Internet-Drafts. The list of current Internet- 62 Drafts is at https://datatracker.ietf.org/drafts/current/. 64 Internet-Drafts are draft documents valid for a maximum of six months 65 and may be updated, replaced, or obsoleted by other documents at any 66 time. It is inappropriate to use Internet-Drafts as reference 67 material or to cite them other than as "work in progress." 69 This Internet-Draft will expire on 18 June 2021. 71 Copyright Notice 73 Copyright (c) 2020 IETF Trust and the persons identified as the 74 document authors. All rights reserved. 76 This document is subject to BCP 78 and the IETF Trust's Legal 77 Provisions Relating to IETF Documents (https://trustee.ietf.org/ 78 license-info) in effect on the date of publication of this document. 79 Please review these documents carefully, as they describe your rights 80 and restrictions with respect to this document. Code Components 81 extracted from this document must include Simplified BSD License text 82 as described in Section 4.e of the Trust Legal Provisions and are 83 provided without warranty as described in the Simplified BSD License. 85 Table of Contents 87 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 88 1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 89 1.2. Selected solution and considered alternatives . . . . . . 6 90 1.3. Intended application . . . . . . . . . . . . . . . . . . 9 91 2. Overview of the two specified solutions and selection of 92 method . . . . . . . . . . . . . . . . . . . . . . . . . 10 93 2.1. The RTP-mixer based solution for multi-party aware 94 endpoints . . . . . . . . . . . . . . . . . . . . . . . . 10 95 2.2. Mixing for multi-party unaware endpoints . . . . . . . . 10 96 2.3. Offer/answer considerations . . . . . . . . . . . . . . . 11 97 2.4. Actions depending on capability negotiation result . . . 11 98 3. Details for the RTP-mixer based multi-party aware mixing 99 method . . . . . . . . . . . . . . . . . . . . . . . . . 12 100 3.1. Use of fields in the RTP packets . . . . . . . . . . . . 12 101 3.2. Initial transmission of a BOM character . . . . . . . . . 12 102 3.3. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 12 103 3.4. Transmission interval . . . . . . . . . . . . . . . . . . 13 104 3.5. Only one source per packet . . . . . . . . . . . . . . . 13 105 3.6. Do not send received text to the originating source . . . 13 106 3.7. Clean incoming text . . . . . . . . . . . . . . . . . . . 13 107 3.8. Redundant transmission principles . . . . . . . . . . . . 13 108 3.9. Interleaving text from different sources . . . . . . . . 14 109 3.10. Text placement in packets . . . . . . . . . . . . . . . . 14 110 3.11. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 15 111 3.12. Creation of the redundancy . . . . . . . . . . . . . . . 15 112 3.13. Timer offset fields . . . . . . . . . . . . . . . . . . . 15 113 3.14. Other RTP header fields . . . . . . . . . . . . . . . . . 16 114 3.15. Pause in transmission . . . . . . . . . . . . . . . . . . 16 115 3.16. RTCP considerations . . . . . . . . . . . . . . . . . . . 16 116 3.17. Reception of multi-party contents . . . . . . . . . . . . 16 117 3.18. Performance considerations . . . . . . . . . . . . . . . 18 118 3.19. Security for session control and media . . . . . . . . . 19 119 3.20. SDP offer/answer examples . . . . . . . . . . . . . . . . 19 120 3.21. Packet sequence example from interleaved transmission . . 20 121 3.22. Maximum character rate "CPS" . . . . . . . . . . . . . . 23 122 4. Presentation level considerations . . . . . . . . . . . . . . 23 123 4.1. Presentation by multi-party aware endpoints . . . . . . . 24 124 4.2. Multi-party mixing for multi-party unaware endpoints . . 26 125 5. Relation to Conference Control . . . . . . . . . . . . . . . 31 126 5.1. Use with SIP centralized conferencing framework . . . . . 32 127 5.2. Conference control . . . . . . . . . . . . . . . . . . . 32 128 6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 32 129 6.1. Gateway considerations with Textphones (e.g. TTYs). . . 32 130 6.2. Gateway considerations with WebRTC. . . . . . . . . . . . 33 131 7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 34 132 8. Congestion considerations . . . . . . . . . . . . . . . . . . 34 133 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 34 134 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 34 135 10.1. Registration of the "rtt-mixer" sdp media attribute . . 34 136 11. Security Considerations . . . . . . . . . . . . . . . . . . . 35 137 12. Change history . . . . . . . . . . . . . . . . . . . . . . . 35 138 12.1. Changes included in 139 draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . . 35 140 12.2. Changes included in 141 draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 36 142 12.3. Changes included in 143 draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 36 145 12.4. Changes included in 146 draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 36 147 12.5. Changes included in 148 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 36 149 12.6. Changes included in 150 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 37 151 12.7. Changes included in 152 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 37 153 12.8. Changes included in 154 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 37 155 12.9. Changes included in 156 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 37 157 12.10. Changes included in 158 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 37 159 12.11. Changes included in 160 draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 38 161 12.12. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 39 162 12.13. Changes from 163 draft-hellstrom-avtcore-multi-party-rtt-source-03 to 164 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 39 165 12.14. Changes from 166 draft-hellstrom-avtcore-multi-party-rtt-source-02 to 167 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 39 168 12.15. Changes from 169 draft-hellstrom-avtcore-multi-party-rtt-source-01 to 170 -02 . . . . . . . . . . . . . . . . . . . . . . . . . . 40 171 12.16. Changes from 172 draft-hellstrom-avtcore-multi-party-rtt-source-00 to 173 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 40 174 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 41 175 13.1. Normative References . . . . . . . . . . . . . . . . . . 41 176 13.2. Informative References . . . . . . . . . . . . . . . . . 42 177 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 43 179 1. Introduction 181 "RTP Payload for Text Conversation" [RFC4103] specifies use of RFC 182 3550 RTP [RFC3550] for transmission of real-time text (RTT) and the 183 "text/t140" format. It also specifies a redundancy format "text/red" 184 for increased robustness. RFC 4102 [RFC4102] registers the "text/ 185 red" format. 187 Real-time text is usually provided together with audio and sometimes 188 with video in conversational sessions. 190 A requirement related to multi-party sessions from the presentation 191 level standard T.140 for real-time text is: "The display of text from 192 the members of the conversation should be arranged so that the text 193 from each participant is clearly readable, and its source and the 194 relative timing of entered text is visualized in the display." 196 Another requirement is that the mixing procedure must not introduce 197 delays in the text streams that are experienced to be disturbing the 198 real-time experience of the receiving users. 200 The redundancy scheme of RFC 4103 [RFC4103] enables efficient 201 transmission of earlier transmitted redundant text in packets 202 together with new text. However the redundancy header format has no 203 source indicators for the redundant transmissions. The redundant 204 parts in a packet must therefore be from the same source as the new 205 text. The recommended transmission is one new and two redundant 206 generations of text (T140blocks) in each packet and the recommended 207 transmission interval for two-party use is 300 ms. 209 Real-time text mixers for multi-party sessions need to include the 210 source with each transmitted group of text from a conference 211 participant so that the text can be transmitted interleaved with text 212 groups from different sources in the rate they are created. This 213 enables the text groups to be presented by endpoints in suitable 214 grouping with other text from the same source. 216 The presentation can then be arranged so that text from different 217 sources can be presented in real-time and easily read. At the same 218 time it is possible for a reading user to perceive approximately when 219 the text was created in real time by the different parties. The 220 transmission and mixing is intended to be done in a general way so 221 that presentation can be arranged in a layout decided by the 222 endpoint. 224 There are existing implementations of RFC 4103 in endpoints without 225 the updates from this document. These will not be able to receive 226 and present real-time text mixed for multi-party aware endpoints. 228 A negotiation mechanism is therefore needed for verification if the 229 parties are able to handle a common method for multi-party 230 transmission and agreeing on using that method. 232 A fall-back mixing procedure is also needed for cases when the 233 negotiation result indicates that a receiving endpoint is not capable 234 of handling the mixed format. Multi-party unaware endpoints would 235 possibly otherwise present all received multi-party mixed text as if 236 it came from the same source regardless of any accompanying source 237 indication coded in fields in the packet. Or they may have any other 238 undesirable way of acting on the multi-party content. The fall-back 239 method is called the mixing procedure for multi-party unaware 240 endpoints. The fall-back method is naturally not expected to meet 241 all performance requirements placed on the mixing procedure for 242 multi-party aware endpoints. 244 The document updates RFC 4103[RFC4103] by introducing an attribute 245 for indicating capability for the RTP-mixer based multi-party mixing 246 case and rules for source indications and interleaving of text from 247 different sources. 249 1.1. Terminology 251 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 252 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 253 document are to be interpreted as described in [RFC2119]. 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 RFC 4103 [RFC4103] to contain one 259 or more 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 RFC 5764 [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 with 100 ms packet intervals 324 instead of the regular 300 ms. The source is indicated in the 325 CSRC field. Transmission of packets with text from different 326 sources can then be done every 100 ms while simultaneous 327 transmission occurs. With five participants sending text 328 simultaneously, the switching and transmission performance is 329 good. With more simultaneously sending participants, there will 330 be a noticeable jerkiness in text presentation. The jerkiness 331 will be more expressed the more participants who send text 332 simultaneously. With ten sending participants, the jerkiness will 333 be about one second. Text sent from a source at the end of the 334 period its text is sent by the mixer will have close to zero extra 335 delay. Recent text will be presented with no or low delay. The 336 one second jerkiness will be noticeable and slightly unpleasant, 337 but corresponds in time to what typing humans often cause by 338 hesitation or changing position while typing. A benefit of this 339 method is that no new packet format needs to be introduced and 340 implemented. Since simultaneous typing by more than two parties 341 is very rare, this method can be used successfully with good 342 performance. Recovery of text in case of packet loss is based on 343 analysis of timestamps of received redundancy versus earlier 344 received text. Negotiation is based on a new sdp media attribute 345 "rtt-mixer". This method is selected to be the main one specified 346 in this document. 348 Multiple sources per packet. 349 A new "text" media subtype would be specified with up to 15 350 sources in each packet. The mechanism would make use of the RTP 351 mixer model specified in RFC3550[RFC3550]. Text from up to 15 352 sources can be included in each packet. Packets are normally sent 353 every 300 ms. The mean delay will be 150 ms. The sources are 354 indicated in strict order in the CSRC list of the RTP packets. A 355 new redundancy packet format is specified. This method would 356 result in good performance, but would require standardisation and 357 implementation of new releases in the target technologies that 358 would take more time than desirable to complete. It was therefore 359 not selected to be included in this document. 361 Mixing for multi-party unaware endpoints 362 Presentation of text from multiple parties is prepared by the 363 mixer in one single stream. It is desirable to have a method that 364 does not require any modifications in existing user devices 365 implementing RFC 4103 for RTT without explicit support of multi- 366 party sessions. This is possible by having the mixer insert a new 367 line and a text formatted source label before each switch of text 368 source in the stream. Switch of source can only be done in places 369 in the text where it does not disturb the perception of the 370 contents. Text from only one source can be presented in real time 371 at a time. The delay will therefore be varying. The method also 372 has other limitations, but is included in this document as a 373 fallback method. In calls where parties take turns properly by 374 ending their entries with a new line, the limitations will have 375 limited influence on the user experience. while only two parties 376 send text, these two will see the text in real time with no delay. 377 This method is specified as a fallback method in this document. 379 RTT transport in WebRTC 380 Transport of real-time text in the WebRTC technology is specified 381 to use the WebRTC data channel in 382 [I-D.ietf-mmusic-t140-usage-data-channel]. That specification 383 contains a section briefly describing its use in multi-party 384 sessions. The focus of this document is RTP transport. 385 Therefore, even if the WebRTC transport provides good multi-party 386 performance, it is just mentioned in this document in relation to 387 providing gateways with multi-party capabilities between RTP and 388 WebRTC technologies. 390 1.3. Intended application 392 The method for multi-party real-time text specified in this document 393 is primarily intended for use in transmission between mixers and 394 endpoints in centralised mixing configurations. It is also 395 applicable between mixers. An often mentioned application is for 396 emergency service calls with real-time text and voice, where a 397 calltaker want to make an attended handover of a call to another 398 agent, and stay observing the session. Multimedia conference 399 sessions with support for participants to contribute in text is 400 another application. Conferences with central support for speech-to- 401 text conversion is yet another mentioned application. 403 In all these applications, normally only one participant at a time 404 will send long text utterances. In some cases, one other participant 405 will occasionally contribute with a longer comment simultaneously. 406 That may also happen in some rare cases when text is interpreted to 407 text in another language in a conference. Apart from these cases, 408 other participants are only expected to contribute with very brief 409 utterings while others are sending text. 411 Users expect that the text they send is presented in real-time in a 412 readable way to the other participants even if they send 413 simultaneously with other users and even when they make brief edit 414 operations of their text by backspacing and correcting their text. 416 Text is supposed to be human generated, by some text input means, 417 such as typing on a keyboard or using speech-to-text technology. 418 Occasional small cut-and-paste operations may appear even if that is 419 not the initial purpose of real-time text. 421 The real-time characteristics of real-time text is essential for the 422 participants to be able to contribute to a conversation. If the text 423 is too much delayed from typing a letter to its presentation, then, 424 in some conference situations, the opportunity to comment will be 425 gone and someone else will grab the turn. A delay of more than one 426 second in such situations is an obstacle for good conversation. 428 2. Overview of the two specified solutions and selection of method 430 This section contains a brief introduction of the two methods 431 specified in this document. 433 2.1. The RTP-mixer based solution for multi-party aware endpoints 435 This method specifies negotiated use of the RFC 4103 format for 436 multi-party transmission in a single RTP stream. The main purpose of 437 this document is to specify a method for true multi-party real-time 438 text mixing for multi-party aware endpoints that can be widely 439 deployed. The RTP-mixer based method makes use of the current format 440 for real-time text in [RFC4103]. It is an update of RFC 4103 by a 441 clarification on one way to use it in the multi-party situation. 442 That is done by completing a negotiation for this kind of multi-party 443 capability and by interleaving packets from different sources. The 444 source is indicated in the CSRC element in the RTP packets. Specific 445 considerations are made to be able to recover text after packet loss. 447 The detailed procedures for the RTP-mixer based multi-party aware 448 case are specified in Section 3. 450 Please use [RFC4103] as reference when reading the specification. 452 2.2. Mixing for multi-party unaware endpoints 454 A method is also specified in this document for cases when the 455 endpoint participating in a multi-party call does not itself 456 implement any solution, or not the same, as the mixer. The method 457 requires the mixer to insert text dividers and readable labels and 458 only send text from one source at a time until a suitable point 459 appears for source change. This solution is a fallback method with 460 functional limitations. It acts on the presentation level. 462 A party acting as a mixer, which has not negotiated any method for 463 true multi-party RTT handling, but negotiated a "text/red" or "text/ 464 t140" format in a session with a participant SHOULD, if nothing else 465 is specified for the application, format transmitted text to that 466 participant to be suitable to present on a multi-party unaware 467 endpoint as further specified in Section 4.2. 469 2.3. Offer/answer considerations 471 RFC 4103[RFC4103] specifies use of RFC 3550 RTP[RFC3550], and a 472 redundancy format "text/red" for increased robustness of real-time 473 text transmission. This document updates RFC 4103[RFC4103] by 474 introducing a capability negotiation for handling multi-party real- 475 time text, a way to indicate the source of transmitted text, and 476 rules for efficient timing of the transmissions interleaved from 477 different sources. 479 The capability negotiation for the "RTP-mixer based multi-party 480 method" is based on use of the sdp media attribute "rtt-mixer". 482 Both parties SHALL indicate their capability in a session setup or 483 modification, and evaluate the capability of the counterpart. 485 The syntax is as follows: 486 "a=rtt-mixer" 488 If any other method for RTP-based multi-party real-time text gets 489 specified, it is assumed that it will be recognized by some specific 490 SDP feature exchange. 492 It is possible to both indicate capability for the RTP-mixer based 493 method and another method. An answer MUST NOT accept more than one 494 method. 496 2.4. Actions depending on capability negotiation result 498 A transmitting party SHALL send text according to the RTP-mixer based 499 multi-party method only when the negotiation for that method was 500 successful and when it conveys text for another source. In all other 501 cases, the packets SHALL be populated and interpreted as for a two- 502 party session. 504 A party which has negotiated the "rtt-mixer" sdp media attribute MUST 505 populate the CSRC-list and format the packets according to Section 3 506 if it acts as an rtp-mixer and sends multi-party text. 508 A party which has negotiated the "rtt-mixer" sdp media attribute MUST 509 interpret the contents of the "CC" field, the CSRC-list and the 510 packets according to Section 3 in received RTP packets in the 511 corresponding RTP stream. 513 A party which has not successfully completed the negotiation of the 514 "rtt-mixer" sdp media attribute MUST NOT transmit packets interleaved 515 from different sources in the same RTP stream as specified in 516 Section 3. If the party is a mixer and did declare the "rtt-mixer" 517 sdp media attribute, it SHOULD perform the procedure for multi-party 518 unaware endpoints. If the party is not a mixer, it SHOULD transmit 519 according to [RFC4103]. 521 3. Details for the RTP-mixer based multi-party aware mixing method 523 3.1. Use of fields in the RTP packets 525 The CC field SHALL show the number of members in the CSRC list, which 526 SHALL be one (1) in transmissions from a mixer when conveying text 527 from other sources in a multi-party session, and otherwise 0. 529 When text is conveyed by a mixer during a multi-party session, a CSRC 530 list SHALL be included in the packet. The single member in the CSRC- 531 list SHALL contain the SSRC of the source of the T140blocks in the 532 packet. 534 When redundancy is used, the RECOMMENDED level of redundancy is to 535 use one primary and two redundant generations of T140blocks. In some 536 cases, a primary or redundant T140block is empty, but is still 537 represented by a member in the redundancy header. 539 From other aspects, the contents of the RTP packets are equal to what 540 is specified in [RFC4103]. 542 3.2. Initial transmission of a BOM character 544 As soon as a participant is known to participate in a session with 545 another entity and being available for text reception, a Unicode BOM 546 character SHALL be sent to it by the other entity according to the 547 procedures in this section. If the transmitter is a mixer, then the 548 source of this character SHALL be indicated to be the mixer itself. 550 Note that the BOM character SHALL be transmitted with the same 551 redundancy procedures as any other text. 553 3.3. Keep-alive 555 After that, the transmitter SHALL send keep-alive traffic to the 556 receiver(s) at regular intervals when no other traffic has occurred 557 during that interval, if that is decided for the actual connection. 558 It is RECOMMENDED to use the keep-alive solution from [RFC6263]. The 559 consent check of [RFC7675] is a possible alternative if it is used 560 anyway for other reasons. 562 3.4. Transmission interval 564 A "text/red" or "text/t140" transmitter in a mixer SHOULD send 565 packets distributed in time as long as there is something (new or 566 redundant T140blocks) to transmit. The maximum transmission interval 567 SHOULD then be 330 ms. It is RECOMMENDED to send the next packet to 568 a receiver as soon as new text to that receiver is available, as long 569 as the time after the latest sent packet to the same receiver is more 570 than or equal to 100 ms, and also the maximum character rate ("CPS") 571 to the receiver is not exceeded. The intention is to keep the 572 latency low and network load limited while keeping a good protection 573 against text loss in bursty packet loss conditions. 575 For a transmitter not acting in a mixer, the same transmission 576 interval principles apply, but the maximum transmission interval 577 SHOULD be 300 ms. 579 3.5. Only one source per packet 581 New text and redundant copies of earlier text from one source SHALL 582 be transmitted in the same packet if available for transmission at 583 the same time. Text from different sources MUST NOT be transmitted 584 in the same packet. 586 3.6. Do not send received text to the originating source 588 Text received by a mixer from a participant SHOULD NOT be included in 589 transmission from the mixer to that participant. 591 3.7. Clean incoming text 593 A mixer SHALL handle reception, recovery from packet loss, deletion 594 of superfluous redundancy, marking of possible text loss and deletion 595 of 'BOM' characters from each participant before queueing received 596 text for transmission to receiving participants. 598 3.8. Redundant transmission principles 600 A transmitting party using redundancy SHALL send redundant 601 repetitions of T140blocks already transmitted in earlier packets. 603 The number of redundant generations of T140blocks to include in 604 transmitted packets SHALL be deduced from the SDP negotiation. It 605 SHOULD be set to the minimum of the number declared by the two 606 parties negotiating a connection. It is RECOMMENDED to declare and 607 transmit one original and two redundant generations of the 608 T140blocks. 610 3.9. Interleaving text from different sources 612 When text from more than one source is available for transmission 613 from a mixer, the mixer SHALL let the sources take turns in having 614 their text transmitted. 616 The source with the oldest text received in the mixer or oldest 617 redundant text SHOULD be next in turn to get all its available unsent 618 text transmitted. The age of redundant text SHOULD then be 619 considered to be 330 ms after its previous transmission. 621 3.10. Text placement in packets 623 The mixer SHOULD compose and transmit an RTP packet to a receiver 624 when one of the following conditions has occurred: 626 * 100 ms has passed since the latest transmission to that receiver, 627 and there is unsent text available for transmission. 629 * New text has arrived and more than 100 ms has passed since latest 630 transmission to that receiver. 632 * 330 ms has passed since already transmitted text was queued for 633 transmission as redundant text, and more than 100 ms has passed 634 since the latest transmission to that receiver, and the redundant 635 text is still not sent. 637 At time of transmission, the mixer SHALL populate the RTP packet with 638 all T140blocks queued for transmission originating from the source in 639 turn for transmission as long as this is not in conflict with the 640 allowed number of characters per second ("CPS") or the maximum packet 641 size. In this way, the latency of the latest received text is kept 642 low even in moments of simultaneous transmission from many sources. 644 Redundant text SHALL also be included. See Section 3.12 646 The SSRC of the source SHALL be placed as the only member in the 647 CSRC-list. 649 Note: The CSRC-list in an RTP packet only includes the participant 650 whose text is included in text blocks. It is not the same as the 651 total list of participants in a conference. With audio and video 652 media, the CSRC-list would often contain all participants who are not 653 muted whereas text participants that don't type are completely silent 654 and thus are not represented in RTP packet CSRC-lists. 656 3.11. Empty T140blocks 658 If no unsent T140blocks were available for a source at the time of 659 populating a packet, but T140blocks are available which have not yet 660 been sent the full intended number of redundant transmissions, then 661 the primary T140block for that source is composed of an empty 662 T140block, and populated (without taking up any length) in a packet 663 for transmission. The corresponding SSRC SHALL be placed as usual in 664 its place in the CSRC-list. 666 The first packet in the session, the first after a source switch and 667 the first after a pause SHALL be poulated with the available 668 T140blocks for the source in turn to be sent as primary, and empty 669 T140blocks for the agreed number of redundancy generations. 671 3.12. Creation of the redundancy 673 The primary T140block from a source in the latest transmitted packet 674 is saved for populating the first redundant T140block for that source 675 in next transmission of text from that source. The first redundant 676 T140block for that source from the latest transmission is saved for 677 populating the second redundant T140block in next transmission of 678 text from that source. 680 Usually this is the level of redundancy used. If a higher number of 681 redundancy is negotiated, then the procedure SHALL be maintained 682 until all available redundant levels of T140blocks are placed in the 683 packet. If a receiver has negotiated a lower number of "text/red" 684 generations, then that level SHOULD be the maximum used by the 685 transmitter. 687 The T140blocks saved for transmission as redundant data are assigned 688 a planned transmission time 330 ms after the current time, but SHOULD 689 be transmitted earlier if new text for the same source gets in turn 690 for transmission before that time. 692 3.13. Timer offset fields 694 The timestamp offset values are inserted in the redundancy header, 695 with the time offset from the RTP timestamp in the packet when the 696 corresponding T140block was sent as primary. 698 The timestamp offsets are expressed in the same clock tick units as 699 the RTP timestamp. 701 The timestamp offset values for empty T140blocks have no relevance 702 but SHOULD be assigned realistic values. 704 3.14. Other RTP header fields 706 The number of members in the CSRC list ( 0 or 1) SHALL be placed in 707 the "CC" header field. Only mixers place value 1 in the "CC" field. 708 A value of "0" indicates that the source is the transmitting device 709 itself and that the source is indicated by the SSRC field. This 710 value is used by endpoints, and by mixers sending data that it is 711 source of itself. 713 The current time SHALL be inserted in the timestamp. 715 The SSRC of the mixer for the RTT session SHALL be inserted in the 716 SSRC field of the RTP header. 718 The M-bit SHALL be handled as specified in [RFC4103]. 720 3.15. Pause in transmission 722 When there is no new T140block to transmit, and no redundant 723 T140block that has not been retransmitted the intended number of 724 times from any source, the transmission process can stop until either 725 new T140blocks arrive, or a keep-alive method calls for transmission 726 of keep-alive packets. 728 3.16. RTCP considerations 730 A mixer SHALL send RTCP reports with SDES, CNAME and NAME information 731 about the sources in the multi-party call. This makes it possible 732 for participants to compose a suitable label for text from each 733 source. 735 Integrity SHALL be considered when composing these fields. They 736 contain name and address information that may be sensitive to 737 transmit in its entirety e.g. to unauthenticated participants. 738 Similar considerations SHOULD be taken as for other media. 740 3.17. Reception of multi-party contents 742 The "text/red" receiver included in an endpoint with presentation 743 functions will receive RTP packets in the single stream from the 744 mixer, and SHALL distribute the T140blocks for presentation in 745 presentation areas for each source. Other receiver roles, such as 746 gateways or chained mixers are also feasible, and requires 747 consideration if the stream shall just be forwarded, or distributed 748 based on the different sources. 750 3.17.1. Acting on the source of the packet contents 752 If the "CC" field value of a received packet is 1, it indicates that 753 the text is conveyed from a source indicated in the single member in 754 the CSRC-list, and the receiver MUST act on the source according to 755 its role. If the CC value is 0, the source is indicated in the SSRC 756 field. 758 3.17.2. Detection and indication of possible text loss 760 The RTP sequence numbers of the received packets SHALL be monitored 761 for gaps and packets out of order. If a sequence number gap appears 762 and still exists after some defined short time for jitter resolution, 763 the packets in the gap SHALL be regarded as lost. 765 If it is known that only one source is active in the RTP session, 766 then it is likely that a gap equal to or larger than the agreed 767 number of redundancy generations (including the primary) causes text 768 loss. In that case a t140block SHALL be created with a marker for 769 possible text loss [T140ad1] and assigned to the source and inserted 770 in the reception buffer for that source. 772 If it is known that more than one source is active in the RTP 773 session, then it is not possible in general to evaluate if text was 774 lost when packets were lost. With two active sources and the 775 recommended number of redundancy generations (3), it can take a gap 776 of five consecutive lost packets until any text may be lost, but text 777 loss can also appear if three non-consecutive packets are lost when 778 they contained consecutive data from the same source. A simple 779 method to decide when there is risk for resulting text loss is to 780 evaluate if three or more packets were lost within one second. Then 781 a t140block SHOULD be created with a marker for possible text loss 782 [T140ad1] and assigned to the SSRC of the transmitter as a general 783 input from the mixer. 785 Implementations MAY apply more refined methods for more reliable 786 detection of if text was lost or not. Any refined method SHOULD 787 prefer marking possible loss rather than not marking when it is 788 uncertain if there was loss. 790 3.17.3. Extracting text and handling recovery 792 When applying the following procedures, the effects MUST be 793 considered of possible timestamp wrap around and the RTP session 794 possibly changing SSRC. 796 When a packet is received in an RTP session using the packetization 797 for multi-party aware endpoints, its T140blocks SHALL be extracted in 798 the following way. The description is adapted to the default 799 redundancy case using the original and two redundant generations. 801 The source SHALL be extracted from the CSRC-list if available, 802 otherwise from the SSRC. 804 If the received packet is the first packet received from the source, 805 then all T140blocks in the packet SHALL be retrieved and assigned to 806 a receive buffer for the source beginning with the second generation 807 redundancy, continuing with the first generation redundancy and 808 finally the primary. 810 Note: The normal case is that in the first packet, only the primary 811 data has contents. The redundant data has contents in the first 812 received packet from a source only after initial packet loss. 814 If the packet is not the first packet from a source, then if the 815 second generation redundant data is available, its timestamp SHALL be 816 created by subtracting its timestamp offset from the RTP timestamp. 817 If the resulting timestamp is later than the latest retrieved data 818 from the same source, then the redundant data SHALL be retrieved and 819 appended to the receive buffer. The process SHALL be continued in 820 the same way for the first generation redundant data. After that, 821 the primary data SHALL be retrieved from the packet and appended to 822 the receive buffer for the source. 824 3.17.4. Delete 'BOM' 826 Unicode character 'BOM' is used as a start indication and sometimes 827 used as a filler or keep alive by transmission implementations. 828 These SHALL be deleted after extraction from received packets. 830 3.18. Performance considerations 832 This solution has good performance for up to five participants 833 simultaneously sending text. At higher numbers of participants 834 simultaneously sending text, a jerkiness is visible in the 835 presentation of text. With ten participants simultaneously 836 transmitting text, the jerkiness is about one second. Even so, the 837 transmission of text catches up, so there is limited delay of new 838 text. The solution is therefore suitable for emergency service use, 839 relay service use, and small or well-managed larger multimedia 840 conferences. Only in large unmanaged conferences with a high number 841 of participants there may on very rare occasions appear situations 842 when many participants happen to send text simultaneously, resulting 843 in unpleasantly jerky presentation of text from each sending 844 participant. It should be noted that it is only the number of users 845 sending text within the same moment that causes jerkiness, not the 846 total number of users with RTT capability. 848 3.19. Security for session control and media 850 Security SHOULD be applied by use of SIP over TLS by default 851 according to [RFC5630] section 3.1.3 on session control level and by 852 default using DTLS-SRTP [RFC5764] on media level. In applications 853 where legacy endpoints without security may exist, a negotiation 854 SHOULD be performed to decide if security by encryption on media 855 level will be applied. If no other security solution is mandated for 856 the application, then OSRTP [RFC8643] is a suitable method be applied 857 to negotiate SRTP media security with DTLS. Most SDP examples below 858 are for simplicity expressed without the security additions. The 859 principles (but not all details) for applying DTLS-SRTP [RFC5764] 860 security is shown in a couple of the following examples. 862 3.20. SDP offer/answer examples 864 This sections shows some examples of SDP for session negotiation of 865 the real-time text media in SIP sessions. Audio is usually provided 866 in the same session, and sometimes also video. The examples only 867 show the part of importance for the real-time text media. The 868 examples relate to the single RTP stream mixing for multi-party aware 869 endpoints and for multi-party unaware endpoints. 871 Note: Multi-party RTT may also be provided through other methods, 872 e.g. by a Selective Forwarding Middlebox (SFM). In that case, the 873 SDP of the offer will include something specific for that method, and 874 an answer acknowledging the use of that method would accept it by 875 something specific included in the SDP. The offer may contain also 876 the "rtt-mixer" sdp media attribute for the main RTT media when the 877 offeror has capability for both multi-party methods, while an answer, 878 selecting to use SFM will not include the "rtt-mixer" sdp media 879 attribute. 881 Offer example for "text/red" format and multi-party support: 883 m=text 11000 RTP/AVP 100 98 884 a=rtpmap:98 t140/1000 885 a=rtpmap:100 red/1000 886 a=fmtp:100 98/98/98 887 a=rtt-mixer 889 Answer example from a multi-party capable device 890 m=text 14000 RTP/AVP 100 98 891 a=rtpmap:98 t140/1000 892 a=rtpmap:100 red/1000 893 a=fmtp:100 98/98/98 894 a=rtt-mixer 896 Offer example for "text/red" format including multi-party 897 and security: 898 a=fingerprint: (fingerprint1) 899 m=text 11000 RTP/AVP 100 98 900 a=rtpmap:98 t140/1000 901 a=rtpmap:100 red/1000 902 a=fmtp:100 98/98/98 903 a=rtt-mixer 905 The "fingerprint" is sufficient to offer DTLS-SRTP, with the media 906 line still indicating RTP/AVP. 908 Note: For brevity, the entire value of the SDP fingerprint attribute 909 is not shown in this and the following example. 911 Answer example from a multi-party capable device with security 912 a=fingerprint: (fingerprint2) 913 m=text 16000 RTP/AVP 100 98 914 a=rtpmap:98 t140/1000 915 a=rtpmap:100 red/1000 916 a=fmtp:100 98/98/98 917 a=rtt-mixer 919 With the "fingerprint" the device acknowledges use of SRTP/DTLS. 921 Answer example from a multi-party unaware device that also 922 does not support security: 924 m=text 12000 RTP/AVP 100 98 925 a=rtpmap:98 t140/1000 926 a=rtpmap:100 red/1000 927 a=fmtp:100 98/98/98 929 3.21. Packet sequence example from interleaved transmission 931 This example shows a symbolic flow of packets from a mixer including 932 loss and recovery. The sequence includes interleaved transmission of 933 text from two RTT sources A and B. P indicates primary data. R1 is 934 first redundant generation data and R2 is the second redundant 935 generation data. A1, B1, A2 etc are text chunks (T140blocks) 936 received from the respective sources and sent on to the receiver by 937 the mixer. X indicates dropped packet between the mixer and a 938 receiver. The session is assumed to use original and two redundant 939 generations of RTT. 941 |-----------------------| 942 |Seq no 101, Time=20400 | 943 |CC=1 | 944 |CSRC list A | 945 |R2: A1, Offset=600 | 946 |R1: A2, Offset=300 | 947 |P: A3 | 948 |-----------------------| 950 Assuming that earlier packets ( with text A1 and A2) were received in 951 sequence, text A3 is received from packet 101 and assigned to 952 reception area A. The mixer is now assumed to have received text 953 from source B and will send that text 100 ms after packet 101. 954 Transmission of A2 and A3 as redundancy is planned for 330 ms after 955 packet 101 if no new text from A is ready to be sent before that. 957 |-----------------------| 958 |Seq no 102, Time=20500 | 959 |CC=1 | 960 |CSRC list B | 961 |R2 Empty, Offset=600 | 962 |R1: Empty, Offset=300 | 963 |P: B1 | 964 |-----------------------| 965 Packet 102 is received. 966 B1 is retrieved from this packet. Redundant transmission of 967 B1 is planned 330 ms after packet 102. 969 X------------------------| 970 X Seq no 103, Timer=20730| 971 X CC=1 | 972 X CSRC list A | 973 X R2: A2, Offset=630 | 974 X R1: A3, Offset=330 | 975 X P: Empty | 976 X------------------------| 977 Packet 103 is assumed to be lost due to network problems. 978 It contains redundancy for A. Sending A3 as second level 979 redundancy is planned for 330 ms after packet 104. 981 X------------------------| 982 X Seq no 104, Timer=20820| 983 X CC=1 | 984 X CSRC list B | 985 X R2: Empty, Offset=600 | 986 X R1: B1, Offset=300 | 987 X P: B2 | 988 X------------------------| 989 Packet 104 contains text from B, including new B2 and 990 redundant B1. It is assumed dropped in network 991 problems. 992 The mixer has A3 redundancy to send but no new text 993 appears from A and therefore the redundancy is sent 994 330 ms after the previous packet with text from A. 996 |------------------------| 997 | Seq no 105, Timer=21060| 998 | CC=1 | 999 | CSRC list A | 1000 | R2: A3, Offset=660 | 1001 | R1: Empty, Offset=330 | 1002 | P: Empty | 1003 |------------------------| 1004 Packet 105 is received. 1005 A gap for lost 103, and 104 is detected. 1006 Assume that no other loss was detected the last second. 1007 Then it can be concluded that nothing was totally lost. 1009 R2 is checked. Its original time was 21040-660=20400. 1010 A packet with text from A was received with that 1011 timestamp, so nothing needs to be recovered. 1013 B1 and B2 still needs to be transmitted as redundancy. 1014 This is planned 330 ms after packet 105. But that 1015 would be at 21150 which is only 90 ms after the 1016 latest packet. It is instead transmitted at 1017 time 21160. 1019 |-----------------------| 1020 |Seq no 106, Timer=21160| 1021 |CC=1 | 1022 |CSRC list B | 1023 | R2: B1, Offset=660 | 1024 | R1: B2, Offset=340 | 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 101 THAT was received. So B1 does not 1033 need to be retrieved. The first level redundancy in packet 106 has 1034 offset 340. The timestamp of packet 106 minus 340 is 20820. 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 RFC 4103 [RFC4103] specified to be 30 characters per second. 1046 The value MAY be modified in the CPS parameter of the FMTP attribute 1047 in the media section for the "text/t140" media. A mixer combining 1048 real-time text from a number of sources may occasionally have a 1049 higher combined flow of text coming from the sources. Endpoints 1050 SHOULD therefore specify a suitable higher value for the CPS 1051 parameter, corresponding to its real reception capability. A value 1052 for "CPS" of 90 SHALL be the default for the "text/t140" stream in 1053 the "text/red" format when multi-party real-time text is negotiated. 1054 See RFC 4103 [RFC4103] for the format and use of the CPS parameter. 1055 The same rules apply for the multi-party case except for the default 1056 value. 1058 4. Presentation level considerations 1060 ITU-T T.140 [T140] provides the presentation level requirements for 1061 the RFC 4103 [RFC4103] transport. T.140 [T140] has functions for 1062 erasure and other formatting functions and has the following general 1063 statement for 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." 1071 Strict application of T.140 [T140] is of essence for the 1072 interoperability of real-time text implementations and to fulfill the 1073 intention that the session participants have the same information of 1074 the text contents of the conversation without necessarily having the 1075 exact same layout of the conversation. 1077 T.140 [T140] specifies a set of presentation control codes to include 1078 in the stream. Some of them are optional. Implementations MUST be 1079 able to ignore optional control codes that they do not support. 1081 There is no strict "message" concept in real-time text. The Unicode 1082 Line Separator character SHALL be used as a separator allowing a part 1083 of received text to be grouped in presentation. The characters 1084 "CRLF" may be used by other implementations as replacement for Line 1085 Separator. The "CRLF" combination SHALL be erased by just one 1086 erasing action, just as the Line Separator. Presentation functions 1087 are allowed to group text for presentation in smaller groups than the 1088 line separators imply and present such groups with source indication 1089 together with text groups from other sources (see the following 1090 presentation examples). Erasure has no specific limit by any 1091 delimiter in the text stream. 1093 4.1. Presentation by multi-party aware endpoints 1095 A multi-party aware receiving party, presenting real-time text MUST 1096 separate text from different sources and present them in separate 1097 presentation fields. The receiving party MAY separate presentation 1098 of parts of text from a source in readable groups based on other 1099 criteria than line separator and merge these groups in the 1100 presentation area when it benefits the user to most easily find and 1101 read text from the different participants. The criteria MAY e.g. be 1102 a received comma, full stop, or other phrase delimiters, or a long 1103 pause. 1105 When text is received from multiple original sources, the 1106 presentation SHOULD provide a view where text is added in multiple 1107 presentation fields. 1109 If the presentation presents text from different sources in one 1110 common area, the presenting endpoint SHOULD insert text from the 1111 local user ended at suitable points merged with received text to 1112 indicate the relative timing for when the text groups were completed. 1113 In this presentation mode, the receiving endpoint SHALL present the 1114 source of the different groups of text. This presentation style is 1115 called the "chat" style here. 1117 A view of a three-party RTT call in chat style is shown in this 1118 example . 1120 _________________________________________________ 1121 | |^| 1122 |[Alice] Hi, Alice here. |-| 1123 | | | 1124 |[Bob] Bob as well. | | 1125 | | | 1126 |[Eve] Hi, this is Eve, calling from Paris. | | 1127 | I thought you should be here. | | 1128 | | | 1129 |[Alice] I am coming on Thursday, my | | 1130 | performance is not until Friday morning.| | 1131 | | | 1132 |[Bob] And I on Wednesday evening. | | 1133 | | | 1134 |[Alice] Can we meet on Thursday evening? | | 1135 | | | 1136 |[Eve] Yes, definitely. How about 7pm. | | 1137 | at the entrance of the restaurant | | 1138 | Le Lion Blanc? | | 1139 |[Eve] we can have dinner and then take a walk |-| 1140 |______________________________________________|v| 1141 | But I need to be back to |^| 1142 | the hotel by 11 because I need |-| 1143 | | | 1144 | I wou |-| 1145 |______________________________________________|v| 1146 | of course, I underst | 1147 |________________________________________________| 1149 Figure 3: Example of a three-party RTT call presented in chat style 1150 seen at participant 'Alice's endpoint. 1152 Other presentation styles than the chat style may be arranged. 1154 This figure shows how a coordinated column view MAY be presented. 1156 _____________________________________________________________________ 1157 | Bob | Eve | Alice | 1158 |____________________|______________________|_______________________| 1159 | | |I will arrive by TGV. | 1160 |My flight is to Orly| |Convenient to the main | 1161 | |Hi all, can we plan |station. | 1162 | |for the seminar? | | 1163 |Eve, will you do | | | 1164 |your presentation on| | | 1165 |Friday? |Yes, Friday at 10. | | 1166 |Fine, wo | |We need to meet befo | 1167 |___________________________________________________________________| 1168 Figure 4: An example of a coordinated column-view of a three-party 1169 session with entries ordered vertically in approximate time-order. 1171 4.2. Multi-party mixing for multi-party unaware endpoints 1173 When the mixer has indicated RTT multi-party capability in an SDP 1174 negotiation, but the multi-party capability negotiation fails with an 1175 endpoint, then the agreed "text/red" or "text/t140" format SHALL be 1176 used and the mixer SHOULD compose a best-effort presentation of 1177 multi-party real-time text in one stream intended to be presented by 1178 an endpoint with no multi-party awareness. 1180 This presentation format has functional limitations and SHOULD be 1181 used only to enable participation in multi-party calls by legacy 1182 deployed endpoints implementing only RFC 4103 without any multi-party 1183 extensions specified in this document. 1185 The principles and procedures below do not specify any new protocol 1186 elements. They are instead composed from the information in ITU-T 1187 T.140 [T140] and an ambition to provide a best effort presentation on 1188 an endpoint which has functions only for two-party calls. 1190 The mixer mixing for multi-party unaware endpoints SHALL compose a 1191 simulated limited multi-party RTT view suitable for presentation in 1192 one presentation area. The mixer SHALL group text in suitable groups 1193 and prepare for presentation of them by inserting a new line between 1194 them if the transmitted text did not already end with a new line. A 1195 presentable label SHOULD be composed and sent for the source 1196 initially in the session and after each source switch. With this 1197 procedure the time for switching from transmission of text from one 1198 source to transmission of text from another source is depending on 1199 the actions of the users. In order to expedite source switch, a user 1200 can for example end its turn with a new line. 1202 4.2.1. Actions by the mixer at reception from the call participants 1204 When text is received by the mixer from the different participants, 1205 the mixer SHALL recover text from redundancy if any packets are lost. 1206 The mark for lost text [T140ad1] SHOULD be inserted in the stream if 1207 unrecoverable loss appears. Any Unicode "BOM" characters, possibly 1208 used for keep-alive SHALL be deleted. The time of creation of text 1209 (retrieved from the RTP timestamp) SHOULD be stored together with the 1210 received text from each source in queues for transmission to the 1211 recipients. 1213 4.2.2. Actions by the mixer for transmission to the recipients 1215 The following procedure SHOULD be applied for each multi-party 1216 unaware recipient of multi-party text from the mixer. 1218 The text for transmission SHOULD be formatted by the mixer for each 1219 receiving user for presentation in one single presentation area. 1220 Text received from a participant SHOULD NOT be included in 1221 transmission to that participant. When there is text available for 1222 transmission from the mixer to a receiving party from more than one 1223 participant, the mixer SHOULD switch between transmission of text 1224 from the different sources at suitable points in the transmitted 1225 stream. 1227 When switching source, the mixer SHOULD insert a line separator if 1228 the already transmitted text did not end with a new line (line 1229 separator or CRLF). A label SHOULD be composed from information in 1230 the CNAME and NAME fields in RTCP reports from the participant to 1231 have its text transmitted, or from other session information for that 1232 user. The label SHOULD be delimited by suitable characters (e.g. '[ 1233 ]') and transmitted. The CSRC SHOULD indicate the selected source. 1234 Then text from that selected participant SHOULD be transmitted until 1235 a new suitable point for switching source is reached. 1237 Integrity considerations SHALL be taken when composing the label. 1239 Seeking a suitable point for switching source SHOULD be done when 1240 there is older text waiting for transmission from any party than the 1241 age of the last transmitted text. Suitable points for switching are: 1243 * A completed phrase ended by comma 1245 * A completed sentence 1247 * A new line (line separator or CRLF) 1249 * A long pause (e.g. > 10 seconds) in received text from the 1250 currently transmitted source 1252 * If text from one participant has been transmitted with text from 1253 other sources waiting for transmission for a long time (e.g. > 1 1254 minute) and none of the other suitable points for switching has 1255 occurred, a source switch MAY be forced by the mixer at next word 1256 delimiter, and also if even a word delimiter does not occur within 1257 a time (e.g. 15 seconds) after the scan for word delimiter 1258 started. 1260 When switching source, the source which has the oldest text in queue 1261 SHOULD be selected to be transmitted. A character display count 1262 SHOULD be maintained for the currently transmitted source, starting 1263 at zero after the label is transmitted for the currently transmitted 1264 source. 1266 The status SHOULD be maintained for the latest control code for 1267 Select Graphic Rendition (SGR) from each source. If there is an SGR 1268 code stored as the status for the current source before the source 1269 switch is done, a reset of SGR SHOULD be sent by the sequence SGR 0 1270 [009B 0000 006D] after the new line and before the new label during a 1271 source switch. See SGR below for an explanation. This transmission 1272 does not influence the display count. 1274 If there is an SGR code stored for the new source after the source 1275 switch, that SGR code SHOULD be transmitted to the recipient before 1276 the label. This transmission does not influence the display count. 1278 4.2.3. Actions on transmission of text 1280 Text from a source sent to the recipient SHOULD increase the display 1281 count by one per transmitted character. 1283 4.2.4. Actions on transmission of control codes 1285 The following control codes specified by T.140 require specific 1286 actions. They SHOULD cause specific considerations in the mixer. 1287 Note that the codes presented here are expressed in UCS-16, while 1288 transmission is made in UTF-8 transform of these codes. 1290 BEL 0007 Bell Alert in session, provides for alerting during an 1291 active session. The display count SHOULD NOT be altered. 1293 NEW LINE 2028 Line separator. Check and perform a source switch if 1294 appropriate. Increase display count by 1. 1296 CR LF 000D 000A A supported, but not preferred way of requesting a 1297 new line. Check and perform a source switch if appropriate. 1298 Increase display count by 1. 1300 INT ESC 0061 Interrupt (used to initiate mode negotiation 1301 procedure). The display count SHOULD NOT be altered. 1303 SGR 009B Ps 006D Select graphic rendition. Ps is rendition 1304 parameters specified in ISO 6429. The display count SHOULD NOT be 1305 altered. The SGR code SHOULD be stored for the current source. 1307 SOS 0098 Start of string, used as a general protocol element 1308 introducer, followed by a maximum 256 bytes string and the ST. 1309 The display count SHOULD NOT be altered. 1311 ST 009C String terminator, end of SOS string. The display count 1312 SHOULD NOT be altered. 1314 ESC 001B Escape - used in control strings. The display count SHOULD 1315 NOT be altered for the complete escape code. 1317 Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used 1318 for synchronization and keep-alive. SHOULD be deleted from 1319 incoming streams. Shall be sent first after session establishment 1320 to the recipient. The display count SHOULD NOT be altered. 1322 Missing text mark (U+FFFD) "Replacement character", represented as a 1323 question mark in a rhombus, or if that is not feasible, replaced 1324 by an apostrophe ', marks place in stream of possible text loss. 1325 SHOULD be inserted by the reception procedure in case of 1326 unrecoverable loss of packets. The display count SHOULD be 1327 increased by one when sent as for any other character. 1329 SGR If a control code for selecting graphic rendition (SGR), other 1330 than reset of the graphic rendition (SGR 0) is sent to a 1331 recipient, that control code SHOULD also be stored as status for 1332 the source in the storage for SGR status. If a reset graphic 1333 rendition (SGR 0) originated from a source is sent, then the SGR 1334 status storage for that source SHOULD be cleared. The display 1335 count SHOULD NOT be increased. 1337 BS (U+0008) Back Space, intended to erase the last entered character 1338 by a source. Erasure by backspace cannot always be performed as 1339 the erasing party intended. If an erasing action erases all text 1340 up to the end of the leading label after a source switch, then the 1341 mixer MUST NOT transmit more backspaces. Instead it is 1342 RECOMMENDED that a letter "X" is inserted in the text stream for 1343 each backspace as an indication of the intent to erase more. A 1344 new line is usually coded by a Line Separator, but the character 1345 combination "CRLF" MAY be used instead. Erasure of a new line is 1346 in both cases done by just one erasing action (Backspace). If the 1347 display count has a positive value it is decreased by one when the 1348 BS is sent. If the display count is at zero, it is not altered. 1350 4.2.5. Packet transmission 1352 A mixer transmitting to a multi-party unaware terminal SHOULD send 1353 primary data only from one source per packet. The SSRC SHOULD be the 1354 SSRC of the mixer. The CSRC list SHOULD contain one member and be 1355 the SSRC of the source of the primary data. 1357 4.2.6. Functional limitations 1359 When a multi-party unaware endpoint presents a conversation in one 1360 display area in a chat style, it inserts source indications for 1361 remote text and local user text as they are merged in completed text 1362 groups. When an endpoint using this layout receives and presents 1363 text mixed for multi-party unaware endpoints, there will be two 1364 levels of source indicators for the received text; one generated by 1365 the mixer and inserted in a label after each source switch, and 1366 another generated by the receiving endpoint and inserted after each 1367 switch between local and remote source in the presentation area. 1368 This will waste display space and look inconsistent to the reader. 1370 New text can be presented only from one source at a time. Switch of 1371 source to be presented takes place at suitable places in the text, 1372 such as end of phrase, end of sentence, line separator and 1373 inactivity. Therefore the time to switch to present waiting text 1374 from other sources may become long and will vary and depend on the 1375 actions of the currently presented source. 1377 Erasure can only be done up to the latest source switch. If a user 1378 tries to erase more text, the erasing actions will be presented as 1379 letter X after the label. 1381 Text loss because of network errors may hit the label between entries 1382 from different parties, causing risk for misunderstanding from which 1383 source a piece of text is. 1385 These facts makes it strongly RECOMMENDED to implement multi-party 1386 awareness in RTT endpoints. The use of the mixing method for multi- 1387 party-unaware endpoints should be left for use with endpoints which 1388 are impossible to upgrade to become multi-party aware. 1390 4.2.7. Example views of presentation on multi-party unaware endpoints 1392 The following pictures are examples of the view on a participant's 1393 display for the multi-party-unaware case. 1395 _________________________________________________ 1396 | Conference | Alice | 1397 |________________________|_________________________| 1398 | |I will arrive by TGV. | 1399 |[Bob]:My flight is to |Convenient to the main | 1400 |Orly. |station. | 1401 |[Eve]:Hi all, can we | | 1402 |plan for the seminar. | | 1403 | | | 1404 |[Bob]:Eve, will you do | | 1405 |your presentation on | | 1406 |Friday? | | 1407 |[Eve]:Yes, Friday at 10.| | 1408 |[Bob]: Fine, wo |We need to meet befo | 1409 |________________________|_________________________| 1411 Figure 5: Alice who has a conference-unaware client is receiving the 1412 multi-party real-time text in a single-stream. This figure shows how 1413 a coordinated column view MAY be presented on Alice's device. 1415 _________________________________________________ 1416 | |^| 1417 |[Alice] Hi, Alice here. |-| 1418 | | | 1419 |[mix][Bob] Bob as well. | | 1420 | | | 1421 |[Eve] Hi, this is Eve, calling from Paris | | 1422 | I thought you should be here. | | 1423 | | | 1424 |[Alice] I am coming on Thursday, my | | 1425 | performance is not until Friday morning.| | 1426 | | | 1427 |[mix][Bob] And I on Wednesday evening. | | 1428 | | | 1429 |[Eve] we can have dinner and then walk | | 1430 | | | 1431 |[Eve] But I need to be back to | | 1432 | the hotel by 11 because I need | | 1433 | |-| 1434 |______________________________________________|v| 1435 | of course, I underst | 1436 |________________________________________________| 1438 Figure 6: An example of a view of the multi-party unaware 1439 presentation in chat style. Alice is the local user. 1441 5. Relation to Conference Control 1442 5.1. Use with SIP centralized conferencing framework 1444 The SIP conferencing framework, mainly specified in RFC 1445 4353[RFC4353], RFC 4579[RFC4579] and RFC 4575[RFC4575] is suitable 1446 for coordinating sessions including multi-party RTT. The RTT stream 1447 between the mixer and a participant is one and the same during the 1448 conference. Participants get announced by notifications when 1449 participants are joining or leaving, and further user information may 1450 be provided. The SSRC of the text to expect from joined users MAY be 1451 included in a notification. The notifications MAY be used both for 1452 security purposes and for translation to a label for presentation to 1453 other users. 1455 5.2. Conference control 1457 In managed conferences, control of the real-time text media SHOULD be 1458 provided in the same way as other for media, e.g. for muting and 1459 unmuting by the direction attributes in SDP [RFC4566]. 1461 Note that floor control functions may be of value for RTT users as 1462 well as for users of other media in a conference. 1464 6. Gateway Considerations 1466 6.1. Gateway considerations with Textphones (e.g. TTYs). 1468 Multi-party RTT sessions may involve gateways of different kinds. 1469 Gateways involved in setting up sessions SHALL correctly reflect the 1470 multi-party capability or unawareness of the combination of the 1471 gateway and the remote endpoint beyond the gateway. 1473 One case that may occur is a gateway to PSTN for communication with 1474 textphones (e.g. TTYs). Textphones are limited devices with no 1475 multi-party awareness, and it SHOULD therefore be suitable for the 1476 gateway to not indicate multi-party awareness for that case. Another 1477 solution is that the gateway indicates multi-party capability towards 1478 the mixer, and includes the multi-party mixer function for multi- 1479 party unaware endpoints itself. This solution makes it possible to 1480 make adaptations for the functional limitations of the textphone 1481 (TTY). 1483 More information on gateways to textphones (TTYs) is found in RFC 1484 5194[RFC5194] 1486 6.2. Gateway considerations with WebRTC. 1488 Gateway operation to real-time text in WebRTC may also be required. 1489 In WebRTC, RTT is specified in 1490 [I-D.ietf-mmusic-t140-usage-data-channel]. 1492 A multi-party bridge may have functionality for communicating by RTT 1493 both in RTP streams with RTT and WebRTC T.140 data channels. Other 1494 configurations may consist of a multi-party bridge with either 1495 technology for RTT transport and a separate gateway for conversion of 1496 the text communication streams between RTP and T.140 data channel. 1498 In WebRTC, it is assumed that for a multi-party session, one T.140 1499 data channel is established for each source from a gateway or bridge 1500 to each participant. Each participant also has a data channel with 1501 two-way connection with the gateway or bridge. 1503 The t140 channel used both ways is for text from the WebRTC user and 1504 from the bridge or gateway itself to the WebRTC user. The label 1505 parameter of this t140 channel is used as NAME field in RTCP to 1506 participants on the RTP side. The other t140 channels are only for 1507 text from other participants to the WebRTC user. 1509 When a new participant has entered the session with RTP transport of 1510 RTT, a new T.140 channel SHOULD be established to WebRTC users with 1511 the label parameter composed from the NAME field in RTCP on the RTP 1512 side. 1514 When a new participant has entered the multi-party session with RTT 1515 transport in a WebRTC T.140 data channel, the new participant SHOULD 1516 be announced by a notification to RTP users. The label parameter 1517 from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP 1518 side, or other available session information. 1520 When a participant on the RTP side disappears, the corresponding 1521 T.140 data channel(s) SHOULD be closed. 1523 When a WebRTC user of T.140 data channels disconnects from the mixer, 1524 the corresponding RTP streams or sources in an RTP-mixed stream 1525 SHOULD be closed. 1527 T.140 data channels MAY be opened and closed by negotiation or 1528 renegotiation of the session or by any other valid means as specified 1529 in section 1 of [I-D.ietf-mmusic-t140-usage-data-channel]. 1531 7. Updates to RFC 4103 1533 This document updates RFC 4103[RFC4103] by introducing an sdp media 1534 attribute "rtt-mixer" for negotiation of multi-party mixing 1535 capability with the [RFC4103] format, and by specifying the rules for 1536 packets when multi-party capability is negotiated and in use. 1538 8. Congestion considerations 1540 The congestion considerations and recommended actions from RFC 4103 1541 [RFC4103] are valid also in multi-party situations. 1543 The first action in case of congestion SHOULD be to temporarily 1544 increase the transmission interval up to two seconds. 1546 If the very unlikely situation appears that more than 70 participants 1547 in a conference send text simultaneously, it will take more than 7 1548 seconds between presentation of text from each of these participants. 1549 More time than that can cause confusion in the session. It is 1550 therefore RECOMMENDED that RTP-mixer based mixer discards such text 1551 in excess and inserts a general indication of possible text loss 1552 [T140ad1] in the session. If the main text contributor is indicated 1553 in any way, the mixer MAY avoid deleting text from that participant. 1555 9. Acknowledgements 1557 James Hamlin for format and performance aspects. 1559 10. IANA Considerations 1561 10.1. Registration of the "rtt-mixer" sdp media attribute 1563 [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the 1564 RFC number of this document.] 1566 IANA is asked to register the new sdp attribute "rtt-mixer". 1568 Contact name: IESG 1570 Contact email: iesg@ietf.org 1572 Attribute name: rtt-mixer 1574 Attribute semantics: See RFCXXXX Section 2.3 1576 Attribute value: none 1578 Usage level: media 1579 Purpose: Indicate support by mixer and endpoint of multi-party 1580 mixing for real-time text transmission, using a common RTP-stream 1581 for transmission of text from a number of sources mixed with one 1582 source at a time and the source indicated in a single CSRC-list 1583 member. 1585 Charset Dependent: no 1587 O/A procedure: See RFCXXXX Section 2.3 1589 Mux Category: normal 1591 Reference: RFCXXXX 1593 11. Security Considerations 1595 The RTP-mixer model requires the mixer to be allowed to decrypt, pack 1596 and encrypt secured text from the conference participants. Therefore 1597 the mixer needs to be trusted. This is similar to the situation for 1598 central mixers of audio and video. 1600 The requirement to transfer information about the user in RTCP 1601 reports in SDES, CNAME and NAME fields, and in conference 1602 notifications, for creation of labels may have privacy concerns as 1603 already stated in RFC 3550 [RFC3550], and may be restricted of 1604 privacy reasons. The receiving user will then get a more symbolic 1605 label for the source. 1607 Participants with malicious intentions may appear and e.g. disturb 1608 the multi-party session by a continuous flow of text, or masquerading 1609 as text from other participants. Counteractions should be to require 1610 secure signaling, media and authentication, and to provide higher 1611 level conference functions e.g. for blocking and expelling 1612 participants. 1614 12. Change history 1616 12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-12 1618 Changes according to responses on comments from Brian Rosen in 1619 Avtcore list on 2020-12-05 and -06. 1621 Changes according to responses to comments by Bernard Aboba in 1622 avtcore list 2020-12-06. 1624 Introduction of an optiona RTP multi-stream mixing method for further 1625 study as proposed by Bernard Aboba. 1627 Changes clarifying how to open and close T.140 data channels included 1628 in 6.2 after comments by Lorenzo Miniero. 1630 Changes to satisfy nits check. Some "not" changed to "NOT" in 1631 normative wording combinations. Some lower case normative words 1632 changed to upper case. A normative reference deleted from the 1633 abstract. Two informative documents moved from normative references 1634 to informative references. 1636 12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11 1638 Timestamps and timestamp offsets added to the packet examples in 1639 section 3.23, and the description corrected. 1641 A number of minor corrections added in sections 3.10 - 3.23. 1643 12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10 1645 The packet composition was modified for interleaving packets from 1646 different sources. 1648 The packet reception was modified for the new interleaving method. 1650 The packet sequence examples was adjusted for the new interleaving 1651 method. 1653 Modifications according to responses to Brian Rosen of 2020-11-03 1655 12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09 1657 Changed name on the SDP media attribute to "rtt-mixer" 1659 Restructure of section 2 for balance between aware and unaware cases. 1661 Moved conference control to own section. 1663 Improved clarification of recovery and loss in the packet sequence 1664 example. 1666 A number of editorial corrections and improvements. 1668 12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 1670 Deleted the method requiring a new packet format "text/rex" because 1671 of the longer standardization and implementation period it needs. 1673 Focus on use of RFC 4103 text/red format with shorter transmission 1674 interval, and source indicated in CSRC. 1676 12.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 1678 Added a method based on the "text/red" format and single source per 1679 packet, negotiated by the "rtt-mixer" sdp attribute. 1681 Added reasoning and recommendation about indication of loss. 1683 The highest number of sources in one packet is 15, not 16. Changed. 1685 Added in information on update to RFC 4103 that RFC 4103 explicitly 1686 allows addition of FEC method. The redundancy is a kind of forward 1687 error correction.. 1689 12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 1691 Improved definitions list format. 1693 The format of the media subtype parameters is made to match the 1694 requirements. 1696 The mapping of media subtype parameters to sdp is included. 1698 The CPS parameter belongs to the t140 subtype and does not need to be 1699 registered here. 1701 12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 1703 nomenclature and editorial improvements 1705 "this document" used consistently to refer to this document. 1707 12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 1709 'Redundancy header' renamed to 'data header'. 1711 More clarifications added. 1713 Language and figure number corrections. 1715 12.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 1717 Mention possible need to mute and raise hands as for other media. 1718 ---done ---- 1720 Make sure that use in two-party calls is also possible and explained. 1721 - may need more wording - 1723 Clarify the RTT is often used together with other media. --done-- 1724 Tell that text mixing is N-1. A users own text is not received in 1725 the mix. -done- 1727 In 3. correct the interval to: A "text/rex" transmitter SHOULD send 1728 packets distributed in time as long as there is something (new or 1729 redundant T140blocks) to transmit. The maximum transmission interval 1730 SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 1731 receiver as soon as new text to that receiver is available, as long 1732 as the time after the latest sent packet to the same receiver is more 1733 than 150 ms, and also the maximum character rate to the receiver is 1734 not exceeded. The intention is to keep the latency low while keeping 1735 a good protection against text loss in bursty packet loss conditions. 1736 -done- 1738 In 1.3 say that the format is used both ways. -done- 1740 In 13.1 change presentation area to presentation field so that reader 1741 does not think it shall be totally separated. -done- 1743 In Performance and intro, tell the performance in number of 1744 simultaneous sending users and introduced delay 16, 150 vs 1745 requirements 5 vs 500. -done -- 1747 Clarify redundancy level per connection. -done- 1749 Timestamp also for the last data header. To make it possible for all 1750 text to have time offset as for transmission from the source. Make 1751 that header equal to the others. -done- 1753 Mixer always use the CSRC list, even for its own BOM. -done- 1755 Combine all talk about transmission interval (300 ms vs when text has 1756 arrived) in section 3 in one paragraph or close to each other. -done- 1758 Documents the goal of good performance with low delay for 5 1759 simultaneous typers in the introduction. -done- 1761 Describe better that only primary text shall be sent on to receivers. 1762 Redundancy and loss must be resolved by the mixer. -done- 1764 12.11. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 1766 SDP and better description and visibility of security by OSRTP RFC 1767 8634 needed. 1769 The description of gatewaying to WebRTC extended. 1771 The description of the data header in the packet is improved. 1773 12.12. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 1775 2,5,6 More efficient format "text/rex" introduced and attribute 1776 a=rtt-mix deleted. 1778 3. Brief about use of OSRTP for security included- More needed. 1780 4. Brief motivation for the solution and why not rtp-translator is 1781 used added to intro. 1783 7. More limitations for the multi-party unaware mixing method 1784 inserted. 1786 8. Updates to RFC 4102 and 4103 more clearly expressed. 1788 9. Gateway to WebRTC started. More needed. 1790 12.13. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 1791 to draft-ietf-avtcore-multi-party-rtt-mix-00 1793 Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00 1795 Replaced CDATA in IANA registration table with better coding. 1797 Converted to xml2rfc version 3. 1799 12.14. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 1800 to -03 1802 Changed company and e-mail of the author. 1804 Changed title to "RTP-mixer formatting of multi-party Real-time text" 1805 to better match contents. 1807 Check and modification where needed of use of RFC 2119 words SHALL 1808 etc. 1810 More about the CC value in sections on transmitters and receivers so 1811 that 1-to-1 sessions do not use the mixer format. 1813 Enhanced section on presentation for multi-party-unaware endpoints 1815 A paragraph recommending CPS=150 inserted in the performance section. 1817 12.15. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 1818 to -02 1820 In Abstract and 1. Introduction: Introduced wording about regulatory 1821 requirements. 1823 In section 5: The transmission interval is decreased to 100 ms when 1824 there is text from more than one source to transmit. 1826 In section 11 about SDP negotiation, a SHOULD-requirement is 1827 introduced that the mixer should make a mix for multi-party unaware 1828 endpoints if the negotiation is not successful. And a reference to a 1829 later chapter about it. 1831 The presentation considerations chapter 14 is extended with more 1832 information about presentation on multi-party aware endpoints, and a 1833 new section on the multi-party unaware mixing with low functionality 1834 but SHOULD a be implemented in mixers. Presentation examples are 1835 added. 1837 A short chapter 15 on gateway considerations is introduced. 1839 Clarification about the text/t140 format included in chapter 10. 1841 This sentence added to the chapter 10 about use without redundancy. 1842 "The text/red format SHOULD be used unless some other protection 1843 against packet loss is utilized, for example a reliable network or 1844 transport." 1846 Note about deviation from RFC 2198 added in chapter 4. 1848 In chapter 9. "Use with SIP centralized conferencing framework" the 1849 following note is inserted: Note: The CSRC-list in an RTP packet only 1850 includes participants who's text is included in one or more text 1851 blocks. It is not the same as the list of participants in a 1852 conference. With audio and video media, the CSRC-list would often 1853 contain all participants who are not muted whereas text participants 1854 that don't type are completely silent and so don't show up in RTP 1855 packet CSRC-lists. 1857 12.16. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 1858 to -01 1860 Editorial cleanup. 1862 Changed capability indication from fmtp-parameter to SDP attribute 1863 "rtt-mix". 1865 Swapped order of redundancy elements in the example to match reality. 1867 Increased the SDP negotiation section 1869 13. References 1871 13.1. Normative References 1873 [I-D.ietf-mmusic-t140-usage-data-channel] 1874 Holmberg, C. and G. Hellstrom, "T.140 Real-time Text 1875 Conversation over WebRTC Data Channels", Work in Progress, 1876 Internet-Draft, draft-ietf-mmusic-t140-usage-data-channel- 1877 14, 10 April 2020, . 1880 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1881 Requirement Levels", BCP 14, RFC 2119, 1882 DOI 10.17487/RFC2119, March 1997, 1883 . 1885 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1886 Jacobson, "RTP: A Transport Protocol for Real-Time 1887 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1888 July 2003, . 1890 [RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type", 1891 RFC 4102, DOI 10.17487/RFC4102, June 2005, 1892 . 1894 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 1895 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 1896 . 1898 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1899 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 1900 July 2006, . 1902 [RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session 1903 Initiation Protocol (SIP)", RFC 5630, 1904 DOI 10.17487/RFC5630, October 2009, 1905 . 1907 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1908 Security (DTLS) Extension to Establish Keys for the Secure 1909 Real-time Transport Protocol (SRTP)", RFC 5764, 1910 DOI 10.17487/RFC5764, May 2010, 1911 . 1913 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 1914 Keeping Alive the NAT Mappings Associated with RTP / RTP 1915 Control Protocol (RTCP) Flows", RFC 6263, 1916 DOI 10.17487/RFC6263, June 2011, 1917 . 1919 [RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M. 1920 Thomson, "Session Traversal Utilities for NAT (STUN) Usage 1921 for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675, 1922 October 2015, . 1924 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 1925 multimedia application text conversation", February 1998, 1926 . 1928 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 1929 Protocol for multimedia application text conversation", 1930 February 2000, 1931 . 1933 13.2. Informative References 1935 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 1936 Session Initiation Protocol (SIP)", RFC 4353, 1937 DOI 10.17487/RFC4353, February 2006, 1938 . 1940 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 1941 Session Initiation Protocol (SIP) Event Package for 1942 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 1943 2006, . 1945 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 1946 (SIP) Call Control - Conferencing for User Agents", 1947 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 1948 . 1950 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 1951 Time Text over IP Using the Session Initiation Protocol 1952 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 1953 . 1955 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 1956 DOI 10.17487/RFC7667, November 2015, 1957 . 1959 [RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. 1960 Stach, "An Opportunistic Approach for Secure Real-time 1961 Transport Protocol (OSRTP)", RFC 8643, 1962 DOI 10.17487/RFC8643, August 2019, 1963 . 1965 Author's Address 1967 Gunnar Hellstrom 1968 Gunnar Hellstrom Accessible Communication 1969 SE-13670 Vendelso 1970 Sweden 1972 Email: gunnar.hellstrom@ghaccess.se