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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) -- Possible downref: Non-RFC (?) normative reference: ref. 'T140' -- Possible downref: Non-RFC (?) normative reference: ref. 'T140ad1' Summary: 0 errors (**), 0 flaws (~~), 1 warning (==), 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) 10 February 2021 5 Intended status: Standards Track 6 Expires: 14 August 2021 8 RTP-mixer formatting of multi-party Real-time text 9 draft-ietf-avtcore-multi-party-rtt-mix-13 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 14 August 2021. 71 Copyright Notice 73 Copyright (c) 2021 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. . . . . . . . . . . . 32 131 7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 33 132 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-13 . . . . . . . 35 140 12.2. Changes included in 141 draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . . 35 142 12.3. Changes included in 143 draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 36 145 12.4. Changes included in 146 draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 36 147 12.5. Changes included in 148 draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 36 149 12.6. Changes included in 150 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 37 151 12.7. Changes included in 152 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 37 153 12.8. Changes included in 154 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 37 155 12.9. Changes included in 156 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 37 157 12.10. Changes included in 158 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 37 159 12.11. Changes included in 160 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 38 161 12.12. Changes included in 162 draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 39 163 12.13. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 39 164 12.14. Changes from 165 draft-hellstrom-avtcore-multi-party-rtt-source-03 to 166 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 39 167 12.15. Changes from 168 draft-hellstrom-avtcore-multi-party-rtt-source-02 to 169 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 39 170 12.16. Changes from 171 draft-hellstrom-avtcore-multi-party-rtt-source-01 to 172 -02 . . . . . . . . . . . . . . . . . . . . . . . . . . 40 173 12.17. Changes from 174 draft-hellstrom-avtcore-multi-party-rtt-source-00 to 175 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 41 176 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 41 177 13.1. Normative References . . . . . . . . . . . . . . . . . . 41 178 13.2. Informative References . . . . . . . . . . . . . . . . . 42 179 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 43 181 1. Introduction 183 "RTP Payload for Text Conversation" [RFC4103] specifies use of RFC 184 3550 RTP [RFC3550] for transmission of real-time text (RTT) and the 185 "text/t140" format. It also specifies a redundancy format "text/red" 186 for increased robustness. RFC 4102 [RFC4102] registers the "text/ 187 red" format. 189 Real-time text is usually provided together with audio and sometimes 190 with video in conversational sessions. 192 A requirement related to multi-party sessions from the presentation 193 level standard T.140 for real-time text is: "The display of text from 194 the members of the conversation should be arranged so that the text 195 from each participant is clearly readable, and its source and the 196 relative timing of entered text is visualized in the display." 198 Another requirement is that the mixing procedure must not introduce 199 delays in the text streams that are experienced to be disturbing the 200 real-time experience of the receiving users. 202 The redundancy scheme of RFC 4103 [RFC4103] enables efficient 203 transmission of earlier transmitted redundant text in packets 204 together with new text. However the redundancy header format has no 205 source indicators for the redundant transmissions. The redundant 206 parts in a packet must therefore be from the same source as the new 207 text. The recommended transmission is one new and two redundant 208 generations of text (T140blocks) in each packet and the recommended 209 transmission interval for two-party use is 300 ms. 211 Real-time text mixers for multi-party sessions need to include the 212 source with each transmitted group of text from a conference 213 participant so that the text can be transmitted interleaved with text 214 groups from different sources in the rate they are created. This 215 enables the text groups to be presented by endpoints in suitable 216 grouping with other text from the same source. 218 The presentation can then be arranged so that text from different 219 sources can be presented in real-time and easily read. At the same 220 time it is possible for a reading user to perceive approximately when 221 the text was created in real time by the different parties. The 222 transmission and mixing is intended to be done in a general way so 223 that presentation can be arranged in a layout decided by the 224 endpoint. 226 There are existing implementations of RFC 4103 in endpoints without 227 the updates from this document. These will not be able to receive 228 and present real-time text mixed for multi-party aware endpoints. 230 A negotiation mechanism is therefore needed for verification if the 231 parties are able to handle a common method for multi-party 232 transmission and agreeing on using that method. 234 A fall-back mixing procedure is also needed for cases when the 235 negotiation result indicates that a receiving endpoint is not capable 236 of handling the mixed format. Multi-party unaware endpoints would 237 possibly otherwise present all received multi-party mixed text as if 238 it came from the same source regardless of any accompanying source 239 indication coded in fields in the packet. Or they may have any other 240 undesirable way of acting on the multi-party content. The fall-back 241 method is called the mixing procedure for multi-party unaware 242 endpoints. The fall-back method is naturally not expected to meet 243 all performance requirements placed on the mixing procedure for 244 multi-party aware endpoints. 246 The document updates RFC 4103[RFC4103] by introducing an attribute 247 for indicating capability for the RTP-mixer based multi-party mixing 248 case and rules for source indications and interleaving of text from 249 different sources. 251 1.1. Terminology 253 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 254 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 255 document are to be interpreted as described in [RFC2119]. 257 The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP- 258 mixer, RTP-translator as defined in [RFC3550] 260 The term "T140block" is defined in RFC 4103 [RFC4103] to contain one 261 or more T.140 code elements. 263 "TTY" stands for a text telephone type used in North America. 265 "WebRTC" stands for web based communication specified by W3C and 266 IETF. 268 "DTLS-SRTP" stands for security specified in RFC 5764 [RFC5764]. 270 "multi-party aware" stands for an endpoint receiving real-time text 271 from multiple sources through a common conference mixer being able to 272 present the text in real-time separated by source and presented so 273 that a user can get an impression of the approximate relative timing 274 of text from different parties. 276 "multi-party unaware" stands for an endpoint not itself being able to 277 separate text from different sources when received through a common 278 conference mixer. 280 1.2. Selected solution and considered alternatives 282 A number of alternatives were considered when searching an efficient 283 and easily implemented multi-party method for real-time text. This 284 section explains a few of them briefly. 286 Multiple RTP streams, one per participant. 287 One RTP stream per source would be sent in the same RTP session 288 with the "text/red" format. From some points of view, use of 289 multiple RTP streams, one for each source, sent in the same RTP 290 session would be efficient, and would use exactly the same packet 291 format as [RFC4103] and the same payload type. A couple of 292 relevant scenarios using multiple RTP-streams are specified in 293 "RTP Topologies" [RFC7667]. One possibility of special interest 294 is the Selective Forwarding Middlebox (SFM) topology specified in 295 RFC 7667 section 3.7 that could enable end to end encryption. In 296 contrast to audio and video, real-time text is only transmitted 297 when the users actually transmit information. Thus an SFM 298 solution would not need to exclude any party from transmission 299 under normal conditions. In order to allow the mixer to convey 300 the packets with the payload preserved and encrypted, an SFM 301 solution would need to act on some specific characteristics of the 302 "text/red" format. The redundancy headers are part of the 303 payload, so the receiver would need to just assume that the 304 payload type number in the redundancy header is for "text/t140". 305 The characters per second parameter (CPS) would need to act per 306 stream. The relation between the SSRC and the source would need 307 to be conveyed in some specified way, e.g. in the CSRC. Recovery 308 and loss detection would preferably be based on sequence number 309 gap detection. Thus sequence number gaps in the incoming stream 310 to the mixer would need to be reflected in the stream to the 311 participant and no new gaps created by the mixer. However, the 312 RTP implementation in both mixers and endpoints need to support 313 multiple streams in the same RTP session in order to use this 314 mechanism. For best deployment opportunity, it should be possible 315 to upgrade existing endpoint solutions to be multi-party aware 316 with a reasonable effort. There is currently a lack of support 317 for multi-stream RTP in certain implementation technologies. This 318 fact made this solution only briefly mentioned in this document as 319 an option for further study. 321 RTP-mixer based method for multi-party aware endpoints. 322 The "text/red" format in RFC 4103 is sent with shorter 323 transmission interval with the RTP-mixer method and indicating 324 source in CSRC. The "text/red" format with "text/t140" payload in 325 a single RTP stream can be sent with 100 ms packet intervals 326 instead of the regular 300 ms. The source is indicated in the 327 CSRC field. Transmission of packets with text from different 328 sources can then be done every 100 ms while simultaneous 329 transmission occurs. With five participants sending text 330 simultaneously, the switching and transmission performance is 331 good. With more simultaneously sending participants, there will 332 be a noticeable jerkiness in text presentation. The jerkiness 333 will be more expressed the more participants who send text 334 simultaneously. With ten sending participants, the jerkiness will 335 be about one second. Text sent from a source at the end of the 336 period its text is sent by the mixer will have close to zero extra 337 delay. Recent text will be presented with no or low delay. The 338 one second jerkiness will be noticeable and slightly unpleasant, 339 but corresponds in time to what typing humans often cause by 340 hesitation or changing position while typing. A benefit of this 341 method is that no new packet format needs to be introduced and 342 implemented. Since simultaneous typing by more than two parties 343 is very rare, this method can be used successfully with good 344 performance. Recovery of text in case of packet loss is based on 345 analysis of timestamps of received redundancy versus earlier 346 received text. Negotiation is based on a new sdp media attribute 347 "rtt-mixer". This method is selected to be the main one specified 348 in this document. 350 Multiple sources per packet. 351 A new "text" media subtype would be specified with up to 15 352 sources in each packet. The mechanism would make use of the RTP 353 mixer model specified in RFC3550[RFC3550]. Text from up to 15 354 sources can be included in each packet. Packets are normally sent 355 every 300 ms. The mean delay will be 150 ms. The sources are 356 indicated in strict order in the CSRC list of the RTP packets. A 357 new redundancy packet format is specified. This method would 358 result in good performance, but would require standardisation and 359 implementation of new releases in the target technologies that 360 would take more time than desirable to complete. It was therefore 361 not selected to be included in this document. 363 Mixing for multi-party unaware endpoints 364 Presentation of text from multiple parties is prepared by the 365 mixer in one single stream. It is desirable to have a method that 366 does not require any modifications in existing user devices 367 implementing RFC 4103 for RTT without explicit support of multi- 368 party sessions. This is possible by having the mixer insert a new 369 line and a text formatted source label before each switch of text 370 source in the stream. Switch of source can only be done in places 371 in the text where it does not disturb the perception of the 372 contents. Text from only one source can be presented in real time 373 at a time. The delay will therefore be varying. The method also 374 has other limitations, but is included in this document as a 375 fallback method. In calls where parties take turns properly by 376 ending their entries with a new line, the limitations will have 377 limited influence on the user experience. while only two parties 378 send text, these two will see the text in real time with no delay. 379 This method is specified as a fallback method in this document. 381 RTT transport in WebRTC 382 Transport of real-time text in the WebRTC technology is specified 383 to use the WebRTC data channel in [RFC8865]. That specification 384 contains a section briefly describing its use in multi-party 385 sessions. The focus of this document is RTP transport. 386 Therefore, even if the WebRTC transport provides good multi-party 387 performance, it is just mentioned in this document in relation to 388 providing gateways with multi-party capabilities between RTP and 389 WebRTC technologies. 391 1.3. Intended application 393 The method for multi-party real-time text specified in this document 394 is primarily intended for use in transmission between mixers and 395 endpoints in centralised mixing configurations. It is also 396 applicable between mixers. An often mentioned application is for 397 emergency service calls with real-time text and voice, where a 398 calltaker want to make an attended handover of a call to another 399 agent, and stay observing the session. Multimedia conference 400 sessions with support for participants to contribute in text is 401 another application. Conferences with central support for speech-to- 402 text conversion is yet another mentioned application. 404 In all these applications, normally only one participant at a time 405 will send long text utterances. In some cases, one other participant 406 will occasionally contribute with a longer comment simultaneously. 407 That may also happen in some rare cases when text is interpreted to 408 text in another language in a conference. Apart from these cases, 409 other participants are only expected to contribute with very brief 410 utterings while others are sending text. 412 Users expect that the text they send is presented in real-time in a 413 readable way to the other participants even if they send 414 simultaneously with other users and even when they make brief edit 415 operations of their text by backspacing and correcting their text. 417 Text is supposed to be human generated, by some text input means, 418 such as typing on a keyboard or using speech-to-text technology. 419 Occasional small cut-and-paste operations may appear even if that is 420 not the initial purpose of real-time text. 422 The real-time characteristics of real-time text is essential for the 423 participants to be able to contribute to a conversation. If the text 424 is too much delayed from typing a letter to its presentation, then, 425 in some conference situations, the opportunity to comment will be 426 gone and someone else will grab the turn. A delay of more than one 427 second in such situations is an obstacle for good conversation. 429 2. Overview of the two specified solutions and selection of method 431 This section contains a brief introduction of the two methods 432 specified in this document. 434 2.1. The RTP-mixer based solution for multi-party aware endpoints 436 This method specifies negotiated use of the RFC 4103 format for 437 multi-party transmission in a single RTP stream. The main purpose of 438 this document is to specify a method for true multi-party real-time 439 text mixing for multi-party aware endpoints that can be widely 440 deployed. The RTP-mixer based method makes use of the current format 441 for real-time text in [RFC4103]. It is an update of RFC 4103 by a 442 clarification on one way to use it in the multi-party situation. 443 That is done by completing a negotiation for this kind of multi-party 444 capability and by interleaving packets from different sources. The 445 source is indicated in the CSRC element in the RTP packets. Specific 446 considerations are made to be able to recover text after packet loss. 448 The detailed procedures for the RTP-mixer based multi-party aware 449 case are specified in Section 3. 451 Please use [RFC4103] as reference when reading the specification. 453 2.2. Mixing for multi-party unaware endpoints 455 A method is also specified in this document for cases when the 456 endpoint participating in a multi-party call does not itself 457 implement any solution, or not the same, as the mixer. The method 458 requires the mixer to insert text dividers and readable labels and 459 only send text from one source at a time until a suitable point 460 appears for source change. This solution is a fallback method with 461 functional limitations. It acts on the presentation level. 463 A party acting as a mixer, which has not negotiated any method for 464 true multi-party RTT handling, but negotiated a "text/red" or "text/ 465 t140" format in a session with a participant SHOULD, if nothing else 466 is specified for the application, format transmitted text to that 467 participant to be suitable to present on a multi-party unaware 468 endpoint as further specified in Section 4.2. 470 2.3. Offer/answer considerations 472 RFC 4103[RFC4103] specifies use of RFC 3550 RTP[RFC3550], and a 473 redundancy format "text/red" for increased robustness of real-time 474 text transmission. This document updates RFC 4103[RFC4103] by 475 introducing a capability negotiation for handling multi-party real- 476 time text, a way to indicate the source of transmitted text, and 477 rules for efficient timing of the transmissions interleaved from 478 different sources. 480 The capability negotiation for the "RTP-mixer based multi-party 481 method" is based on use of the sdp media attribute "rtt-mixer". 483 Both parties SHALL indicate their capability in a session setup or 484 modification, and evaluate the capability of the counterpart. 486 The syntax is as follows: 487 "a=rtt-mixer" 489 If any other method for RTP-based multi-party real-time text gets 490 specified, it is assumed that it will be recognized by some specific 491 SDP feature exchange. 493 It is possible to both indicate capability for the RTP-mixer based 494 method and another method. An answer MUST NOT accept more than one 495 method. 497 2.4. Actions depending on capability negotiation result 499 A transmitting party SHALL send text according to the RTP-mixer based 500 multi-party method only when the negotiation for that method was 501 successful and when it conveys text for another source. In all other 502 cases, the packets SHALL be populated and interpreted as for a two- 503 party session. 505 A party which has negotiated the "rtt-mixer" sdp media attribute MUST 506 populate the CSRC-list and format the packets according to Section 3 507 if it acts as an rtp-mixer and sends multi-party text. 509 A party which has negotiated the "rtt-mixer" sdp media attribute MUST 510 interpret the contents of the "CC" field, the CSRC-list and the 511 packets according to Section 3 in received RTP packets in the 512 corresponding RTP stream. 514 A party which has not successfully completed the negotiation of the 515 "rtt-mixer" sdp media attribute MUST NOT transmit packets interleaved 516 from different sources in the same RTP stream as specified in 517 Section 3. If the party is a mixer and did declare the "rtt-mixer" 518 sdp media attribute, it SHOULD perform the procedure for multi-party 519 unaware endpoints. If the party is not a mixer, it SHOULD transmit 520 according to [RFC4103]. 522 3. Details for the RTP-mixer based multi-party aware mixing method 524 3.1. Use of fields in the RTP packets 526 The CC field SHALL show the number of members in the CSRC list, which 527 SHALL be one (1) in transmissions from a mixer when conveying text 528 from other sources in a multi-party session, and otherwise 0. 530 When text is conveyed by a mixer during a multi-party session, a CSRC 531 list SHALL be included in the packet. The single member in the CSRC- 532 list SHALL contain the SSRC of the source of the T140blocks in the 533 packet. 535 When redundancy is used, the RECOMMENDED level of redundancy is to 536 use one primary and two redundant generations of T140blocks. In some 537 cases, a primary or redundant T140block is empty, but is still 538 represented by a member in the redundancy header. 540 From other aspects, the contents of the RTP packets are equal to what 541 is specified in [RFC4103]. 543 3.2. Initial transmission of a BOM character 545 As soon as a participant is known to participate in a session with 546 another entity and being available for text reception, a Unicode BOM 547 character SHALL be sent to it by the other entity according to the 548 procedures in this section. If the transmitter is a mixer, then the 549 source of this character SHALL be indicated to be the mixer itself. 551 Note that the BOM character SHALL be transmitted with the same 552 redundancy procedures as any other text. 554 3.3. Keep-alive 556 After that, the transmitter SHALL send keep-alive traffic to the 557 receiver(s) at regular intervals when no other traffic has occurred 558 during that interval, if that is decided for the actual connection. 559 It is RECOMMENDED to use the keep-alive solution from [RFC6263]. The 560 consent check of [RFC7675] is a possible alternative if it is used 561 anyway for other reasons. 563 3.4. Transmission interval 565 A "text/red" or "text/t140" transmitter in a mixer SHOULD send 566 packets distributed in time as long as there is something (new or 567 redundant T140blocks) to transmit. The maximum transmission interval 568 SHOULD then be 330 ms. It is RECOMMENDED to send the next packet to 569 a receiver as soon as new text to that receiver is available, as long 570 as the time after the latest sent packet to the same receiver is more 571 than or equal to 100 ms, and also the maximum character rate ("CPS") 572 to the receiver is not exceeded. The intention is to keep the 573 latency low and network load limited while keeping a good protection 574 against text loss in bursty packet loss conditions. 576 For a transmitter not acting in a mixer, the same transmission 577 interval principles apply, but the maximum transmission interval 578 SHOULD be 300 ms. 580 3.5. Only one source per packet 582 New text and redundant copies of earlier text from one source SHALL 583 be transmitted in the same packet if available for transmission at 584 the same time. Text from different sources MUST NOT be transmitted 585 in the same packet. 587 3.6. Do not send received text to the originating source 589 Text received by a mixer from a participant SHOULD NOT be included in 590 transmission from the mixer to that participant. 592 3.7. Clean incoming text 594 A mixer SHALL handle reception, recovery from packet loss, deletion 595 of superfluous redundancy, marking of possible text loss and deletion 596 of 'BOM' characters from each participant before queueing received 597 text for transmission to receiving participants. 599 3.8. Redundant transmission principles 601 A transmitting party using redundancy SHALL send redundant 602 repetitions of T140blocks already transmitted in earlier packets. 604 The number of redundant generations of T140blocks to include in 605 transmitted packets SHALL be deduced from the SDP negotiation. It 606 SHOULD be set to the minimum of the number declared by the two 607 parties negotiating a connection. It is RECOMMENDED to declare and 608 transmit one original and two redundant generations of the 609 T140blocks. 611 3.9. Interleaving text from different sources 613 When text from more than one source is available for transmission 614 from a mixer, the mixer SHALL let the sources take turns in having 615 their text transmitted. 617 The source with the oldest text received in the mixer or oldest 618 redundant text SHOULD be next in turn to get all its available unsent 619 text transmitted. The age of redundant text SHOULD then be 620 considered to be 330 ms after its previous transmission. 622 3.10. Text placement in packets 624 The mixer SHOULD compose and transmit an RTP packet to a receiver 625 when one of the following conditions has occurred: 627 * 100 ms has passed since the latest transmission to that receiver, 628 and there is unsent text available for transmission. 630 * New text has arrived and more than 100 ms has passed since latest 631 transmission to that receiver. 633 * 330 ms has passed since already transmitted text was queued for 634 transmission as redundant text, and more than 100 ms has passed 635 since the latest transmission to that receiver, and the redundant 636 text is still not sent. 638 At time of transmission, the mixer SHALL populate the RTP packet with 639 all T140blocks queued for transmission originating from the source in 640 turn for transmission as long as this is not in conflict with the 641 allowed number of characters per second ("CPS") or the maximum packet 642 size. In this way, the latency of the latest received text is kept 643 low even in moments of simultaneous transmission from many sources. 645 Redundant text SHALL also be included. See Section 3.12 647 The SSRC of the source SHALL be placed as the only member in the 648 CSRC-list. 650 Note: The CSRC-list in an RTP packet only includes the participant 651 whose text is included in text blocks. It is not the same as the 652 total list of participants in a conference. With audio and video 653 media, the CSRC-list would often contain all participants who are not 654 muted whereas text participants that don't type are completely silent 655 and thus are not represented in RTP packet CSRC-lists. 657 3.11. Empty T140blocks 659 If no unsent T140blocks were available for a source at the time of 660 populating a packet, but T140blocks are available which have not yet 661 been sent the full intended number of redundant transmissions, then 662 the primary T140block for that source is composed of an empty 663 T140block, and populated (without taking up any length) in a packet 664 for transmission. The corresponding SSRC SHALL be placed as usual in 665 its place in the CSRC-list. 667 The first packet in the session, the first after a source switch and 668 the first after a pause SHALL be poulated with the available 669 T140blocks for the source in turn to be sent as primary, and empty 670 T140blocks for the agreed number of redundancy generations. 672 3.12. Creation of the redundancy 674 The primary T140block from a source in the latest transmitted packet 675 is saved for populating the first redundant T140block for that source 676 in next transmission of text from that source. The first redundant 677 T140block for that source from the latest transmission is saved for 678 populating the second redundant T140block in next transmission of 679 text from that source. 681 Usually this is the level of redundancy used. If a higher number of 682 redundancy is negotiated, then the procedure SHALL be maintained 683 until all available redundant levels of T140blocks are placed in the 684 packet. If a receiver has negotiated a lower number of "text/red" 685 generations, then that level SHOULD be the maximum used by the 686 transmitter. 688 The T140blocks saved for transmission as redundant data are assigned 689 a planned transmission time 330 ms after the current time, but SHOULD 690 be transmitted earlier if new text for the same source gets in turn 691 for transmission before that time. 693 3.13. Timer offset fields 695 The timestamp offset values are inserted in the redundancy header, 696 with the time offset from the RTP timestamp in the packet when the 697 corresponding T140block was sent as primary. 699 The timestamp offsets are expressed in the same clock tick units as 700 the RTP timestamp. 702 The timestamp offset values for empty T140blocks have no relevance 703 but SHOULD be assigned realistic values. 705 3.14. Other RTP header fields 707 The number of members in the CSRC list ( 0 or 1) SHALL be placed in 708 the "CC" header field. Only mixers place value 1 in the "CC" field. 709 A value of "0" indicates that the source is the transmitting device 710 itself and that the source is indicated by the SSRC field. This 711 value is used by endpoints, and by mixers sending data that it is 712 source of itself. 714 The current time SHALL be inserted in the timestamp. 716 The SSRC of the mixer for the RTT session SHALL be inserted in the 717 SSRC field of the RTP header. 719 The M-bit SHALL be handled as specified in [RFC4103]. 721 3.15. Pause in transmission 723 When there is no new T140block to transmit, and no redundant 724 T140block that has not been retransmitted the intended number of 725 times from any source, the transmission process can stop until either 726 new T140blocks arrive, or a keep-alive method calls for transmission 727 of keep-alive packets. 729 3.16. RTCP considerations 731 A mixer SHALL send RTCP reports with SDES, CNAME and NAME information 732 about the sources in the multi-party call. This makes it possible 733 for participants to compose a suitable label for text from each 734 source. 736 Integrity SHALL be considered when composing these fields. They 737 contain name and address information that may be sensitive to 738 transmit in its entirety e.g. to unauthenticated participants. 739 Similar considerations SHOULD be taken as for other media. 741 3.17. Reception of multi-party contents 743 The "text/red" receiver included in an endpoint with presentation 744 functions will receive RTP packets in the single stream from the 745 mixer, and SHALL distribute the T140blocks for presentation in 746 presentation areas for each source. Other receiver roles, such as 747 gateways or chained mixers are also feasible, and requires 748 consideration if the stream shall just be forwarded, or distributed 749 based on the different sources. 751 3.17.1. Acting on the source of the packet contents 753 If the "CC" field value of a received packet is 1, it indicates that 754 the text is conveyed from a source indicated in the single member in 755 the CSRC-list, and the receiver MUST act on the source according to 756 its role. If the CC value is 0, the source is indicated in the SSRC 757 field. 759 3.17.2. Detection and indication of possible text loss 761 The RTP sequence numbers of the received packets SHALL be monitored 762 for gaps and packets out of order. If a sequence number gap appears 763 and still exists after some defined short time for jitter resolution, 764 the packets in the gap SHALL be regarded as lost. 766 If it is known that only one source is active in the RTP session, 767 then it is likely that a gap equal to or larger than the agreed 768 number of redundancy generations (including the primary) causes text 769 loss. In that case a t140block SHALL be created with a marker for 770 possible text loss [T140ad1] and assigned to the source and inserted 771 in the reception buffer for that source. 773 If it is known that more than one source is active in the RTP 774 session, then it is not possible in general to evaluate if text was 775 lost when packets were lost. With two active sources and the 776 recommended number of redundancy generations (3), it can take a gap 777 of five consecutive lost packets until any text may be lost, but text 778 loss can also appear if three non-consecutive packets are lost when 779 they contained consecutive data from the same source. A simple 780 method to decide when there is risk for resulting text loss is to 781 evaluate if three or more packets were lost within one second. Then 782 a t140block SHOULD be created with a marker for possible text loss 783 [T140ad1] and assigned to the SSRC of the transmitter as a general 784 input from the mixer. 786 Implementations MAY apply more refined methods for more reliable 787 detection of if text was lost or not. Any refined method SHOULD 788 prefer marking possible loss rather than not marking when it is 789 uncertain if there was loss. 791 3.17.3. Extracting text and handling recovery 793 When applying the following procedures, the effects MUST be 794 considered of possible timestamp wrap around and the RTP session 795 possibly changing SSRC. 797 When a packet is received in an RTP session using the packetization 798 for multi-party aware endpoints, its T140blocks SHALL be extracted in 799 the following way. The description is adapted to the default 800 redundancy case using the original and two redundant generations. 802 The source SHALL be extracted from the CSRC-list if available, 803 otherwise from the SSRC. 805 If the received packet is the first packet received from the source, 806 then all T140blocks in the packet SHALL be retrieved and assigned to 807 a receive buffer for the source beginning with the second generation 808 redundancy, continuing with the first generation redundancy and 809 finally the primary. 811 Note: The normal case is that in the first packet, only the primary 812 data has contents. The redundant data has contents in the first 813 received packet from a source only after initial packet loss. 815 If the packet is not the first packet from a source, then if the 816 second generation redundant data is available, its timestamp SHALL be 817 created by subtracting its timestamp offset from the RTP timestamp. 818 If the resulting timestamp is later than the latest retrieved data 819 from the same source, then the redundant data SHALL be retrieved and 820 appended to the receive buffer. The process SHALL be continued in 821 the same way for the first generation redundant data. After that, 822 the primary data SHALL be retrieved from the packet and appended to 823 the receive buffer for the source. 825 3.17.4. Delete 'BOM' 827 Unicode character 'BOM' is used as a start indication and sometimes 828 used as a filler or keep alive by transmission implementations. 829 These SHALL be deleted after extraction from received packets. 831 3.18. Performance considerations 833 This solution has good performance for up to five participants 834 simultaneously sending text. At higher numbers of participants 835 simultaneously sending text, a jerkiness is visible in the 836 presentation of text. With ten participants simultaneously 837 transmitting text, the jerkiness is about one second. Even so, the 838 transmission of text catches up, so there is limited delay of new 839 text. The solution is therefore suitable for emergency service use, 840 relay service use, and small or well-managed larger multimedia 841 conferences. Only in large unmanaged conferences with a high number 842 of participants there may on very rare occasions appear situations 843 when many participants happen to send text simultaneously, resulting 844 in unpleasantly jerky presentation of text from each sending 845 participant. It should be noted that it is only the number of users 846 sending text within the same moment that causes jerkiness, not the 847 total number of users with RTT capability. 849 3.19. Security for session control and media 851 Security SHOULD be applied by use of SIP over TLS by default 852 according to [RFC5630] section 3.1.3 on session control level and by 853 default using DTLS-SRTP [RFC5764] on media level. In applications 854 where legacy endpoints without security may exist, a negotiation 855 SHOULD be performed to decide if security by encryption on media 856 level will be applied. If no other security solution is mandated for 857 the application, then OSRTP [RFC8643] is a suitable method be applied 858 to negotiate SRTP media security with DTLS. Most SDP examples below 859 are for simplicity expressed without the security additions. The 860 principles (but not all details) for applying DTLS-SRTP [RFC5764] 861 security is shown in a couple of the following examples. 863 3.20. SDP offer/answer examples 865 This sections shows some examples of SDP for session negotiation of 866 the real-time text media in SIP sessions. Audio is usually provided 867 in the same session, and sometimes also video. The examples only 868 show the part of importance for the real-time text media. The 869 examples relate to the single RTP stream mixing for multi-party aware 870 endpoints and for multi-party unaware endpoints. 872 Note: Multi-party RTT may also be provided through other methods, 873 e.g. by a Selective Forwarding Middlebox (SFM). In that case, the 874 SDP of the offer will include something specific for that method, and 875 an answer acknowledging the use of that method would accept it by 876 something specific included in the SDP. The offer may contain also 877 the "rtt-mixer" sdp media attribute for the main RTT media when the 878 offeror has capability for both multi-party methods, while an answer, 879 selecting to use SFM will not include the "rtt-mixer" sdp media 880 attribute. 882 Offer example for "text/red" format and multi-party support: 884 m=text 11000 RTP/AVP 100 98 885 a=rtpmap:98 t140/1000 886 a=rtpmap:100 red/1000 887 a=fmtp:100 98/98/98 888 a=rtt-mixer 890 Answer example from a multi-party capable device 891 m=text 14000 RTP/AVP 100 98 892 a=rtpmap:98 t140/1000 893 a=rtpmap:100 red/1000 894 a=fmtp:100 98/98/98 895 a=rtt-mixer 897 Offer example for "text/red" format including multi-party 898 and security: 899 a=fingerprint: (fingerprint1) 900 m=text 11000 RTP/AVP 100 98 901 a=rtpmap:98 t140/1000 902 a=rtpmap:100 red/1000 903 a=fmtp:100 98/98/98 904 a=rtt-mixer 906 The "fingerprint" is sufficient to offer DTLS-SRTP, with the media 907 line still indicating RTP/AVP. 909 Note: For brevity, the entire value of the SDP fingerprint attribute 910 is not shown in this and the following example. 912 Answer example from a multi-party capable device with security 913 a=fingerprint: (fingerprint2) 914 m=text 16000 RTP/AVP 100 98 915 a=rtpmap:98 t140/1000 916 a=rtpmap:100 red/1000 917 a=fmtp:100 98/98/98 918 a=rtt-mixer 920 With the "fingerprint" the device acknowledges use of SRTP/DTLS. 922 Answer example from a multi-party unaware device that also 923 does not support security: 925 m=text 12000 RTP/AVP 100 98 926 a=rtpmap:98 t140/1000 927 a=rtpmap:100 red/1000 928 a=fmtp:100 98/98/98 930 3.21. Packet sequence example from interleaved transmission 932 This example shows a symbolic flow of packets from a mixer including 933 loss and recovery. The sequence includes interleaved transmission of 934 text from two RTT sources A and B. P indicates primary data. R1 is 935 first redundant generation data and R2 is the second redundant 936 generation data. A1, B1, A2 etc are text chunks (T140blocks) 937 received from the respective sources and sent on to the receiver by 938 the mixer. X indicates dropped packet between the mixer and a 939 receiver. The session is assumed to use original and two redundant 940 generations of RTT. 942 |-----------------------| 943 |Seq no 101, Time=20400 | 944 |CC=1 | 945 |CSRC list A | 946 |R2: A1, Offset=600 | 947 |R1: A2, Offset=300 | 948 |P: A3 | 949 |-----------------------| 951 Assuming that earlier packets ( with text A1 and A2) were received in 952 sequence, text A3 is received from packet 101 and assigned to 953 reception area A. The mixer is now assumed to have received text 954 from source B and will send that text 100 ms after packet 101. 955 Transmission of A2 and A3 as redundancy is planned for 330 ms after 956 packet 101 if no new text from A is ready to be sent before that. 958 |-----------------------| 959 |Seq no 102, Time=20500 | 960 |CC=1 | 961 |CSRC list B | 962 |R2 Empty, Offset=600 | 963 |R1: Empty, Offset=300 | 964 |P: B1 | 965 |-----------------------| 966 Packet 102 is received. 967 B1 is retrieved from this packet. Redundant transmission of 968 B1 is planned 330 ms after packet 102. 970 X------------------------| 971 X Seq no 103, Timer=20730| 972 X CC=1 | 973 X CSRC list A | 974 X R2: A2, Offset=630 | 975 X R1: A3, Offset=330 | 976 X P: Empty | 977 X------------------------| 978 Packet 103 is assumed to be lost due to network problems. 979 It contains redundancy for A. Sending A3 as second level 980 redundancy is planned for 330 ms after packet 104. 982 X------------------------| 983 X Seq no 104, Timer=20820| 984 X CC=1 | 985 X CSRC list B | 986 X R2: Empty, Offset=600 | 987 X R1: B1, Offset=300 | 988 X P: B2 | 989 X------------------------| 990 Packet 104 contains text from B, including new B2 and 991 redundant B1. It is assumed dropped in network 992 problems. 993 The mixer has A3 redundancy to send but no new text 994 appears from A and therefore the redundancy is sent 995 330 ms after the previous packet with text from A. 997 |------------------------| 998 | Seq no 105, Timer=21060| 999 | CC=1 | 1000 | CSRC list A | 1001 | R2: A3, Offset=660 | 1002 | R1: Empty, Offset=330 | 1003 | P: Empty | 1004 |------------------------| 1005 Packet 105 is received. 1006 A gap for lost 103, and 104 is detected. 1007 Assume that no other loss was detected the last second. 1008 Then it can be concluded that nothing was totally lost. 1010 R2 is checked. Its original time was 21040-660=20400. 1011 A packet with text from A was received with that 1012 timestamp, so nothing needs to be recovered. 1014 B1 and B2 still needs to be transmitted as redundancy. 1015 This is planned 330 ms after packet 105. But that 1016 would be at 21150 which is only 90 ms after the 1017 latest packet. It is instead transmitted at 1018 time 21160. 1020 |-----------------------| 1021 |Seq no 106, Timer=21160| 1022 |CC=1 | 1023 |CSRC list B | 1024 | R2: B1, Offset=660 | 1025 | R1: B2, Offset=340 | 1026 | P: Empty | 1027 |-----------------------| 1029 Packet 106 is received. 1031 The second level redundancy in packet 106 is B1 and has timestamp 1032 offset 660 ms. The timestamp of packet 106 minus 660 is 20500 which 1033 is the timestamp of packet 101 THAT was received. So B1 does not 1034 need to be retrieved. The first level redundancy in packet 106 has 1035 offset 340. The timestamp of packet 106 minus 340 is 20820. That is 1036 later than the latest received packet with source B. Therefore B2 is 1037 retrieved and assigned to the input buffer for source B. No primary 1038 is available in packet 106 1040 After this sequence, A3 and B1 and B2 have been received. In this 1041 case no text was lost. 1043 3.22. Maximum character rate "CPS" 1045 The default maximum rate of reception of "text/t140" real-time text 1046 is in RFC 4103 [RFC4103] specified to be 30 characters per second. 1047 The value MAY be modified in the CPS parameter of the FMTP attribute 1048 in the media section for the "text/t140" media. A mixer combining 1049 real-time text from a number of sources may occasionally have a 1050 higher combined flow of text coming from the sources. Endpoints 1051 SHOULD therefore specify a suitable higher value for the CPS 1052 parameter, corresponding to its real reception capability. A value 1053 for "CPS" of 90 SHALL be the default for the "text/t140" stream in 1054 the "text/red" format when multi-party real-time text is negotiated. 1055 See RFC 4103 [RFC4103] for the format and use of the CPS parameter. 1056 The same rules apply for the multi-party case except for the default 1057 value. 1059 4. Presentation level considerations 1061 ITU-T T.140 [T140] provides the presentation level requirements for 1062 the RFC 4103 [RFC4103] transport. T.140 [T140] has functions for 1063 erasure and other formatting functions and has the following general 1064 statement for the presentation: 1066 "The display of text from the members of the conversation should be 1067 arranged so that the text from each participant is clearly readable, 1068 and its source and the relative timing of entered text is visualized 1069 in the display. Mechanisms for looking back in the contents from the 1070 current session should be provided. The text should be displayed as 1071 soon as it is received." 1072 Strict application of T.140 [T140] is of essence for the 1073 interoperability of real-time text implementations and to fulfill the 1074 intention that the session participants have the same information of 1075 the text contents of the conversation without necessarily having the 1076 exact same layout of the conversation. 1078 T.140 [T140] specifies a set of presentation control codes to include 1079 in the stream. Some of them are optional. Implementations MUST be 1080 able 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 SHOULD 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. 1118 A view of a three-party RTT call in chat style is shown in this 1119 example . 1121 _________________________________________________ 1122 | |^| 1123 |[Alice] Hi, Alice here. |-| 1124 | | | 1125 |[Bob] Bob as well. | | 1126 | | | 1127 |[Eve] Hi, this is Eve, calling from Paris. | | 1128 | I thought you should be here. | | 1129 | | | 1130 |[Alice] I am coming on Thursday, my | | 1131 | performance is not until Friday morning.| | 1132 | | | 1133 |[Bob] And I on Wednesday evening. | | 1134 | | | 1135 |[Alice] Can we meet on Thursday evening? | | 1136 | | | 1137 |[Eve] Yes, definitely. How about 7pm. | | 1138 | at the entrance of the restaurant | | 1139 | Le Lion Blanc? | | 1140 |[Eve] we can have dinner and then take a walk |-| 1141 |______________________________________________|v| 1142 | But I need to be back to |^| 1143 | the hotel by 11 because I need |-| 1144 | | | 1145 | I wou |-| 1146 |______________________________________________|v| 1147 | of course, I underst | 1148 |________________________________________________| 1150 Figure 3: Example of a three-party RTT call presented in chat style 1151 seen at participant 'Alice's endpoint. 1153 Other presentation styles than the chat style may be arranged. 1155 This figure shows how a coordinated column view MAY be presented. 1157 _____________________________________________________________________ 1158 | Bob | Eve | Alice | 1159 |____________________|______________________|_______________________| 1160 | | |I will arrive by TGV. | 1161 |My flight is to Orly| |Convenient to the main | 1162 | |Hi all, can we plan |station. | 1163 | |for the seminar? | | 1164 |Eve, will you do | | | 1165 |your presentation on| | | 1166 |Friday? |Yes, Friday at 10. | | 1167 |Fine, wo | |We need to meet befo | 1168 |___________________________________________________________________| 1169 Figure 4: An example of a coordinated column-view of a three-party 1170 session with entries ordered vertically in approximate time-order. 1172 4.2. Multi-party mixing for multi-party unaware endpoints 1174 When the mixer has indicated RTT multi-party capability in an SDP 1175 negotiation, but the multi-party capability negotiation fails with an 1176 endpoint, then the agreed "text/red" or "text/t140" format SHALL be 1177 used and the mixer SHOULD compose a best-effort presentation of 1178 multi-party real-time text in one stream intended to be presented by 1179 an endpoint with no multi-party awareness. 1181 This presentation format has functional limitations and SHOULD be 1182 used only to enable participation in multi-party calls by legacy 1183 deployed endpoints implementing only RFC 4103 without any multi-party 1184 extensions specified in this document. 1186 The principles and procedures below do not specify any new protocol 1187 elements. They are instead composed from the information in ITU-T 1188 T.140 [T140] and an ambition to provide a best effort presentation on 1189 an endpoint which has functions only for two-party calls. 1191 The mixer mixing for multi-party unaware endpoints SHALL compose a 1192 simulated limited multi-party RTT view suitable for presentation in 1193 one presentation area. The mixer SHALL group text in suitable groups 1194 and prepare for presentation of them by inserting a new line between 1195 them if the transmitted text did not already end with a new line. A 1196 presentable label SHOULD be composed and sent for the source 1197 initially in the session and after each source switch. With this 1198 procedure the time for switching from transmission of text from one 1199 source to transmission of text from another source is depending on 1200 the actions of the users. In order to expedite source switch, a user 1201 can for example end its turn with a new line. 1203 4.2.1. Actions by the mixer at reception from the call participants 1205 When text is received by the mixer from the different participants, 1206 the mixer SHALL recover text from redundancy if any packets are lost. 1207 The mark for lost text [T140ad1] SHOULD be inserted in the stream if 1208 unrecoverable loss appears. Any Unicode "BOM" characters, possibly 1209 used for keep-alive SHALL be deleted. The time of creation of text 1210 (retrieved from the RTP timestamp) SHOULD be stored together with the 1211 received text from each source in queues for transmission to the 1212 recipients. 1214 4.2.2. Actions by the mixer for transmission to the recipients 1216 The following procedure SHOULD be applied for each multi-party 1217 unaware recipient of multi-party text from the mixer. 1219 The text for transmission SHOULD be formatted by the mixer for each 1220 receiving user for presentation in one single presentation area. 1221 Text received from a participant SHOULD NOT be included in 1222 transmission to that participant. When there is text available for 1223 transmission from the mixer to a receiving party from more than one 1224 participant, the mixer SHOULD switch between transmission of text 1225 from the different sources at suitable points in the transmitted 1226 stream. 1228 When switching source, the mixer SHOULD insert a line separator if 1229 the already transmitted text did not end with a new line (line 1230 separator or CRLF). A label SHOULD be composed from information in 1231 the CNAME and NAME fields in RTCP reports from the participant to 1232 have its text transmitted, or from other session information for that 1233 user. The label SHOULD be delimited by suitable characters (e.g. '[ 1234 ]') and transmitted. The CSRC SHOULD indicate the selected source. 1235 Then text from that selected participant SHOULD be transmitted until 1236 a new suitable point for switching source is reached. 1238 Integrity considerations SHALL be taken when composing the label. 1240 Seeking a suitable point for switching source SHOULD be done when 1241 there is older text waiting for transmission from any party than the 1242 age of the last transmitted text. Suitable points for switching are: 1244 * A completed phrase ended by comma 1246 * A completed sentence 1248 * A new line (line separator or CRLF) 1250 * A long pause (e.g. > 10 seconds) in received text from the 1251 currently transmitted source 1253 * If text from one participant has been transmitted with text from 1254 other sources waiting for transmission for a long time (e.g. > 1 1255 minute) and none of the other suitable points for switching has 1256 occurred, a source switch MAY be forced by the mixer at next word 1257 delimiter, and also if even a word delimiter does not occur within 1258 a time (e.g. 15 seconds) after the scan for word delimiter 1259 started. 1261 When switching source, the source which has the oldest text in queue 1262 SHOULD be selected to be transmitted. A character display count 1263 SHOULD be maintained for the currently transmitted source, starting 1264 at zero after the label is transmitted for the currently transmitted 1265 source. 1267 The status SHOULD be maintained for the latest control code for 1268 Select Graphic Rendition (SGR) from each source. If there is an SGR 1269 code stored as the status for the current source before the source 1270 switch is done, a reset of SGR SHOULD be sent by the sequence SGR 0 1271 [009B 0000 006D] after the new line and before the new label during a 1272 source switch. See SGR below for an explanation. This transmission 1273 does not influence the display count. 1275 If there is an SGR code stored for the new source after the source 1276 switch, that SGR code SHOULD be transmitted to the recipient before 1277 the label. This transmission does not influence the display count. 1279 4.2.3. Actions on transmission of text 1281 Text from a source sent to the recipient SHOULD increase the display 1282 count by one per transmitted character. 1284 4.2.4. Actions on transmission of control codes 1286 The following control codes specified by T.140 require specific 1287 actions. They SHOULD cause specific considerations in the mixer. 1288 Note that the codes presented here are expressed in UCS-16, while 1289 transmission is made in UTF-8 transform of these codes. 1291 BEL 0007 Bell Alert in session, provides for alerting during an 1292 active session. The display count SHOULD NOT be altered. 1294 NEW LINE 2028 Line separator. Check and perform a source switch if 1295 appropriate. Increase display count by 1. 1297 CR LF 000D 000A A supported, but not preferred way of requesting a 1298 new line. Check and perform a source switch if appropriate. 1299 Increase display count by 1. 1301 INT ESC 0061 Interrupt (used to initiate mode negotiation 1302 procedure). The display count SHOULD NOT be altered. 1304 SGR 009B Ps 006D Select graphic rendition. Ps is rendition 1305 parameters specified in ISO 6429. The display count SHOULD NOT be 1306 altered. The SGR code SHOULD be stored for the current source. 1308 SOS 0098 Start of string, used as a general protocol element 1309 introducer, followed by a maximum 256 bytes string and the ST. 1310 The display count SHOULD NOT be altered. 1312 ST 009C String terminator, end of SOS string. The display count 1313 SHOULD NOT be altered. 1315 ESC 001B Escape - used in control strings. The display count SHOULD 1316 NOT be altered for the complete escape code. 1318 Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used 1319 for synchronization and keep-alive. SHOULD be deleted from 1320 incoming streams. Shall be sent first after session establishment 1321 to the recipient. The display count SHOULD NOT be altered. 1323 Missing text mark (U+FFFD) "Replacement character", represented as a 1324 question mark in a rhombus, or if that is not feasible, replaced 1325 by an apostrophe ', marks place in stream of possible text loss. 1326 SHOULD be inserted by the reception procedure in case of 1327 unrecoverable loss of packets. The display count SHOULD be 1328 increased by one when sent as for any other character. 1330 SGR If a control code for selecting graphic rendition (SGR), other 1331 than reset of the graphic rendition (SGR 0) is sent to a 1332 recipient, that control code SHOULD also be stored as status for 1333 the source in the storage for SGR status. If a reset graphic 1334 rendition (SGR 0) originated from a source is sent, then the SGR 1335 status storage for that source SHOULD be cleared. The display 1336 count SHOULD NOT be increased. 1338 BS (U+0008) Back Space, intended to erase the last entered character 1339 by a source. Erasure by backspace cannot always be performed as 1340 the erasing party intended. If an erasing action erases all text 1341 up to the end of the leading label after a source switch, then the 1342 mixer MUST NOT transmit more backspaces. Instead it is 1343 RECOMMENDED that a letter "X" is inserted in the text stream for 1344 each backspace as an indication of the intent to erase more. A 1345 new line is usually coded by a Line Separator, but the character 1346 combination "CRLF" MAY be used instead. Erasure of a new line is 1347 in both cases done by just one erasing action (Backspace). If the 1348 display count has a positive value it is decreased by one when the 1349 BS is sent. If the display count is at zero, it is not altered. 1351 4.2.5. Packet transmission 1353 A mixer transmitting to a multi-party unaware terminal SHOULD send 1354 primary data only from one source per packet. The SSRC SHOULD be the 1355 SSRC of the mixer. The CSRC list SHOULD contain one member and be 1356 the SSRC of the source of the primary data. 1358 4.2.6. Functional limitations 1360 When a multi-party unaware endpoint presents a conversation in one 1361 display area in a chat style, it inserts source indications for 1362 remote text and local user text as they are merged in completed text 1363 groups. When an endpoint using this layout receives and presents 1364 text mixed for multi-party unaware endpoints, there will be two 1365 levels of source indicators for the received text; one generated by 1366 the mixer and inserted in a label after each source switch, and 1367 another generated by the receiving endpoint and inserted after each 1368 switch between local and remote source in the presentation area. 1369 This will waste display space and look inconsistent to the reader. 1371 New text can be presented only from one source at a time. Switch of 1372 source to be presented takes place at suitable places in the text, 1373 such as end of phrase, end of sentence, line separator and 1374 inactivity. Therefore the time to switch to present waiting text 1375 from other sources may become long and will vary and depend on the 1376 actions of the currently presented source. 1378 Erasure can only be done up to the latest source switch. If a user 1379 tries to erase more text, the erasing actions will be presented as 1380 letter X after the label. 1382 Text loss because of network errors may hit the label between entries 1383 from different parties, causing risk for misunderstanding from which 1384 source a piece of text is. 1386 These facts makes it strongly RECOMMENDED to implement multi-party 1387 awareness in RTT endpoints. The use of the mixing method for multi- 1388 party-unaware endpoints should be left for use with endpoints which 1389 are impossible to upgrade to become multi-party aware. 1391 4.2.7. Example views of presentation on multi-party unaware endpoints 1393 The following pictures are examples of the view on a participant's 1394 display for the multi-party-unaware case. 1396 _________________________________________________ 1397 | Conference | Alice | 1398 |________________________|_________________________| 1399 | |I will arrive by TGV. | 1400 |[Bob]:My flight is to |Convenient to the main | 1401 |Orly. |station. | 1402 |[Eve]:Hi all, can we | | 1403 |plan for the seminar. | | 1404 | | | 1405 |[Bob]:Eve, will you do | | 1406 |your presentation on | | 1407 |Friday? | | 1408 |[Eve]:Yes, Friday at 10.| | 1409 |[Bob]: Fine, wo |We need to meet befo | 1410 |________________________|_________________________| 1412 Figure 5: Alice who has a conference-unaware client is receiving the 1413 multi-party real-time text in a single-stream. This figure shows how 1414 a coordinated column view MAY be presented on Alice's device. 1416 _________________________________________________ 1417 | |^| 1418 |[Alice] Hi, Alice here. |-| 1419 | | | 1420 |[mix](Bob) Bob as well. | | 1421 | | | 1422 |(Eve) Hi, this is Eve, calling from Paris | | 1423 | I thought you should be here. | | 1424 | | | 1425 |[Alice] I am coming on Thursday, my | | 1426 | performance is not until Friday morning.| | 1427 | | | 1428 |[mix](Bob) And I on Wednesday evening. | | 1429 | | | 1430 |(Eve) we can have dinner and then walk | | 1431 | | | 1432 |(Eve) But I need to be back to | | 1433 | the hotel by 11 because I need | | 1434 | |-| 1435 |______________________________________________|v| 1436 | of course, I underst | 1437 |________________________________________________| 1439 Figure 6: An example of a view of the multi-party unaware 1440 presentation in chat style. Alice is the local user. 1442 5. Relation to Conference Control 1443 5.1. Use with SIP centralized conferencing framework 1445 The SIP conferencing framework, mainly specified in RFC 1446 4353[RFC4353], RFC 4579[RFC4579] and RFC 4575[RFC4575] is suitable 1447 for coordinating sessions including multi-party RTT. The RTT stream 1448 between the mixer and a participant is one and the same during the 1449 conference. Participants get announced by notifications when 1450 participants are joining or leaving, and further user information may 1451 be provided. The SSRC of the text to expect from joined users MAY be 1452 included in a notification. The notifications MAY be used both for 1453 security purposes and for translation to a label for presentation to 1454 other users. 1456 5.2. Conference control 1458 In managed conferences, control of the real-time text media SHOULD be 1459 provided in the same way as other for media, e.g. for muting and 1460 unmuting by the direction attributes in SDP [RFC8866]. 1462 Note that floor control functions may be of value for RTT users as 1463 well as for users of other media in a conference. 1465 6. Gateway Considerations 1467 6.1. Gateway considerations with Textphones (e.g. TTYs). 1469 Multi-party RTT sessions may involve gateways of different kinds. 1470 Gateways involved in setting up sessions SHALL correctly reflect the 1471 multi-party capability or unawareness of the combination of the 1472 gateway and the remote endpoint beyond the gateway. 1474 One case that may occur is a gateway to PSTN for communication with 1475 textphones (e.g. TTYs). Textphones are limited devices with no 1476 multi-party awareness, and it SHOULD therefore be suitable for the 1477 gateway to not indicate multi-party awareness for that case. Another 1478 solution is that the gateway indicates multi-party capability towards 1479 the mixer, and includes the multi-party mixer function for multi- 1480 party unaware endpoints itself. This solution makes it possible to 1481 make adaptations for the functional limitations of the textphone 1482 (TTY). 1484 More information on gateways to textphones (TTYs) is found in RFC 1485 5194[RFC5194] 1487 6.2. Gateway considerations with WebRTC. 1489 Gateway operation to real-time text in WebRTC may also be required. 1490 In WebRTC, RTT is specified in [RFC8865]. 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 [RFC8865]. 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 1580 Purpose: Indicate support by mixer and endpoint of multi-party 1581 mixing for real-time text transmission, using a common RTP-stream 1582 for transmission of text from a number of sources mixed with one 1583 source at a time and the source indicated in a single CSRC-list 1584 member. 1586 Charset Dependent: no 1588 O/A procedure: See RFCXXXX Section 2.3 1590 Mux Category: normal 1592 Reference: RFCXXXX 1594 11. Security Considerations 1596 The RTP-mixer model requires the mixer to be allowed to decrypt, pack 1597 and encrypt secured text from the conference participants. Therefore 1598 the mixer needs to be trusted. This is similar to the situation for 1599 central mixers of audio and video. 1601 The requirement to transfer information about the user in RTCP 1602 reports in SDES, CNAME and NAME fields, and in conference 1603 notifications, for creation of labels may have privacy concerns as 1604 already stated in RFC 3550 [RFC3550], and may be restricted of 1605 privacy reasons. The receiving user will then get a more symbolic 1606 label for the source. 1608 Participants with malicious intentions may appear and e.g. disturb 1609 the multi-party session by a continuous flow of text, or masquerading 1610 as text from other participants. Counteractions should be to require 1611 secure signaling, media and authentication, and to provide higher 1612 level conference functions e.g. for blocking and expelling 1613 participants. 1615 12. Change history 1617 12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-13 1619 Changed year to 2021. 1621 Changed reference to draft on RTT in WebRTC to recently published RFC 1622 8865. 1624 Changed label brackets in example from "[]" to "()" to avoid nits 1625 comment. 1627 Changed reference "RFC 4566" to recently published "RFC 8866" 1629 12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-12 1631 Changes according to responses on comments from Brian Rosen in 1632 Avtcore list on 2020-12-05 and -06. 1634 Changes according to responses to comments by Bernard Aboba in 1635 avtcore list 2020-12-06. 1637 Introduction of an optiona RTP multi-stream mixing method for further 1638 study as proposed by Bernard Aboba. 1640 Changes clarifying how to open and close T.140 data channels included 1641 in 6.2 after comments by Lorenzo Miniero. 1643 Changes to satisfy nits check. Some "not" changed to "NOT" in 1644 normative wording combinations. Some lower case normative words 1645 changed to upper case. A normative reference deleted from the 1646 abstract. Two informative documents moved from normative references 1647 to informative references. 1649 12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11 1651 Timestamps and timestamp offsets added to the packet examples in 1652 section 3.23, and the description corrected. 1654 A number of minor corrections added in sections 3.10 - 3.23. 1656 12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10 1658 The packet composition was modified for interleaving packets from 1659 different sources. 1661 The packet reception was modified for the new interleaving method. 1663 The packet sequence examples was adjusted for the new interleaving 1664 method. 1666 Modifications according to responses to Brian Rosen of 2020-11-03 1668 12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09 1670 Changed name on the SDP media attribute to "rtt-mixer" 1672 Restructure of section 2 for balance between aware and unaware cases. 1674 Moved conference control to own section. 1676 Improved clarification of recovery and loss in the packet sequence 1677 example. 1679 A number of editorial corrections and improvements. 1681 12.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 1683 Deleted the method requiring a new packet format "text/rex" because 1684 of the longer standardization and implementation period it needs. 1686 Focus on use of RFC 4103 text/red format with shorter transmission 1687 interval, and source indicated in CSRC. 1689 12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 1691 Added a method based on the "text/red" format and single source per 1692 packet, negotiated by the "rtt-mixer" sdp attribute. 1694 Added reasoning and recommendation about indication of loss. 1696 The highest number of sources in one packet is 15, not 16. Changed. 1698 Added in information on update to RFC 4103 that RFC 4103 explicitly 1699 allows addition of FEC method. The redundancy is a kind of forward 1700 error correction.. 1702 12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 1704 Improved definitions list format. 1706 The format of the media subtype parameters is made to match the 1707 requirements. 1709 The mapping of media subtype parameters to sdp is included. 1711 The CPS parameter belongs to the t140 subtype and does not need to be 1712 registered here. 1714 12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 1716 nomenclature and editorial improvements 1718 "this document" used consistently to refer to this document. 1720 12.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 1722 'Redundancy header' renamed to 'data header'. 1724 More clarifications added. 1726 Language and figure number corrections. 1728 12.11. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 1730 Mention possible need to mute and raise hands as for other media. 1731 ---done ---- 1733 Make sure that use in two-party calls is also possible and explained. 1734 - may need more wording - 1736 Clarify the RTT is often used together with other media. --done-- 1738 Tell that text mixing is N-1. A users own text is not received in 1739 the mix. -done- 1741 In 3. correct the interval to: A "text/rex" transmitter SHOULD send 1742 packets distributed in time as long as there is something (new or 1743 redundant T140blocks) to transmit. The maximum transmission interval 1744 SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 1745 receiver as soon as new text to that receiver is available, as long 1746 as the time after the latest sent packet to the same receiver is more 1747 than 150 ms, and also the maximum character rate to the receiver is 1748 not exceeded. The intention is to keep the latency low while keeping 1749 a good protection against text loss in bursty packet loss conditions. 1750 -done- 1752 In 1.3 say that the format is used both ways. -done- 1754 In 13.1 change presentation area to presentation field so that reader 1755 does not think it shall be totally separated. -done- 1757 In Performance and intro, tell the performance in number of 1758 simultaneous sending users and introduced delay 16, 150 vs 1759 requirements 5 vs 500. -done -- 1761 Clarify redundancy level per connection. -done- 1763 Timestamp also for the last data header. To make it possible for all 1764 text to have time offset as for transmission from the source. Make 1765 that header equal to the others. -done- 1767 Mixer always use the CSRC list, even for its own BOM. -done- 1769 Combine all talk about transmission interval (300 ms vs when text has 1770 arrived) in section 3 in one paragraph or close to each other. -done- 1772 Documents the goal of good performance with low delay for 5 1773 simultaneous typers in the introduction. -done- 1774 Describe better that only primary text shall be sent on to receivers. 1775 Redundancy and loss must be resolved by the mixer. -done- 1777 12.12. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 1779 SDP and better description and visibility of security by OSRTP RFC 1780 8634 needed. 1782 The description of gatewaying to WebRTC extended. 1784 The description of the data header in the packet is improved. 1786 12.13. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 1788 2,5,6 More efficient format "text/rex" introduced and attribute 1789 a=rtt-mix deleted. 1791 3. Brief about use of OSRTP for security included- More needed. 1793 4. Brief motivation for the solution and why not rtp-translator is 1794 used added to intro. 1796 7. More limitations for the multi-party unaware mixing method 1797 inserted. 1799 8. Updates to RFC 4102 and 4103 more clearly expressed. 1801 9. Gateway to WebRTC started. More needed. 1803 12.14. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 1804 to draft-ietf-avtcore-multi-party-rtt-mix-00 1806 Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00 1808 Replaced CDATA in IANA registration table with better coding. 1810 Converted to xml2rfc version 3. 1812 12.15. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 1813 to -03 1815 Changed company and e-mail of the author. 1817 Changed title to "RTP-mixer formatting of multi-party Real-time text" 1818 to better match contents. 1820 Check and modification where needed of use of RFC 2119 words SHALL 1821 etc. 1823 More about the CC value in sections on transmitters and receivers so 1824 that 1-to-1 sessions do not use the mixer format. 1826 Enhanced section on presentation for multi-party-unaware endpoints 1828 A paragraph recommending CPS=150 inserted in the performance section. 1830 12.16. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 1831 to -02 1833 In Abstract and 1. Introduction: Introduced wording about regulatory 1834 requirements. 1836 In section 5: The transmission interval is decreased to 100 ms when 1837 there is text from more than one source to transmit. 1839 In section 11 about SDP negotiation, a SHOULD-requirement is 1840 introduced that the mixer should make a mix for multi-party unaware 1841 endpoints if the negotiation is not successful. And a reference to a 1842 later chapter about it. 1844 The presentation considerations chapter 14 is extended with more 1845 information about presentation on multi-party aware endpoints, and a 1846 new section on the multi-party unaware mixing with low functionality 1847 but SHOULD a be implemented in mixers. Presentation examples are 1848 added. 1850 A short chapter 15 on gateway considerations is introduced. 1852 Clarification about the text/t140 format included in chapter 10. 1854 This sentence added to the chapter 10 about use without redundancy. 1855 "The text/red format SHOULD be used unless some other protection 1856 against packet loss is utilized, for example a reliable network or 1857 transport." 1859 Note about deviation from RFC 2198 added in chapter 4. 1861 In chapter 9. "Use with SIP centralized conferencing framework" the 1862 following note is inserted: Note: The CSRC-list in an RTP packet only 1863 includes participants who's text is included in one or more text 1864 blocks. It is not the same as the list of participants in a 1865 conference. With audio and video media, the CSRC-list would often 1866 contain all participants who are not muted whereas text participants 1867 that don't type are completely silent and so don't show up in RTP 1868 packet CSRC-lists. 1870 12.17. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 1871 to -01 1873 Editorial cleanup. 1875 Changed capability indication from fmtp-parameter to SDP attribute 1876 "rtt-mix". 1878 Swapped order of redundancy elements in the example to match reality. 1880 Increased the SDP negotiation section 1882 13. References 1884 13.1. Normative References 1886 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1887 Requirement Levels", BCP 14, RFC 2119, 1888 DOI 10.17487/RFC2119, March 1997, 1889 . 1891 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1892 Jacobson, "RTP: A Transport Protocol for Real-Time 1893 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1894 July 2003, . 1896 [RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type", 1897 RFC 4102, DOI 10.17487/RFC4102, June 2005, 1898 . 1900 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 1901 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 1902 . 1904 [RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session 1905 Initiation Protocol (SIP)", RFC 5630, 1906 DOI 10.17487/RFC5630, October 2009, 1907 . 1909 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1910 Security (DTLS) Extension to Establish Keys for the Secure 1911 Real-time Transport Protocol (SRTP)", RFC 5764, 1912 DOI 10.17487/RFC5764, May 2010, 1913 . 1915 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 1916 Keeping Alive the NAT Mappings Associated with RTP / RTP 1917 Control Protocol (RTCP) Flows", RFC 6263, 1918 DOI 10.17487/RFC6263, June 2011, 1919 . 1921 [RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M. 1922 Thomson, "Session Traversal Utilities for NAT (STUN) Usage 1923 for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675, 1924 October 2015, . 1926 [RFC8865] Holmberg, C. and G. Hellstrom, "T.140 Real-Time Text 1927 Conversation over WebRTC Data Channels", RFC 8865, 1928 DOI 10.17487/RFC8865, January 2021, 1929 . 1931 [RFC8866] Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP: 1932 Session Description Protocol", RFC 8866, 1933 DOI 10.17487/RFC8866, January 2021, 1934 . 1936 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 1937 multimedia application text conversation", February 1998, 1938 . 1940 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 1941 Protocol for multimedia application text conversation", 1942 February 2000, 1943 . 1945 13.2. Informative References 1947 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 1948 Session Initiation Protocol (SIP)", RFC 4353, 1949 DOI 10.17487/RFC4353, February 2006, 1950 . 1952 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 1953 Session Initiation Protocol (SIP) Event Package for 1954 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 1955 2006, . 1957 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 1958 (SIP) Call Control - Conferencing for User Agents", 1959 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 1960 . 1962 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 1963 Time Text over IP Using the Session Initiation Protocol 1964 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 1965 . 1967 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 1968 DOI 10.17487/RFC7667, November 2015, 1969 . 1971 [RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. 1972 Stach, "An Opportunistic Approach for Secure Real-time 1973 Transport Protocol (OSRTP)", RFC 8643, 1974 DOI 10.17487/RFC8643, August 2019, 1975 . 1977 Author's Address 1979 Gunnar Hellstrom 1980 Gunnar Hellstrom Accessible Communication 1981 SE-13670 Vendelso 1982 Sweden 1984 Email: gunnar.hellstrom@ghaccess.se