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(See the Legal Provisions document at https://trustee.ietf.org/license-info for more information.) -- The document date (28 April 2021) is 1094 days in the past. Is this intentional? Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Missing Reference: 'Bob' is mentioned on line 1453, but not defined -- Possible downref: Non-RFC (?) normative reference: ref. 'T140' -- Possible downref: Non-RFC (?) normative reference: ref. 'T140ad1' Summary: 0 errors (**), 0 flaws (~~), 4 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) 28 April 2021 5 Intended status: Standards Track 6 Expires: 30 October 2021 8 RTP-mixer formatting of multi-party Real-time text 9 draft-ietf-avtcore-multi-party-rtt-mix-15 11 Abstract 13 Enhancements for RFC 4103 real-time text mixing are provided in this 14 document, suitable for a centralized conference model that enables 15 source identification and rapidly interleaved transmission of text 16 from different sources. The intended use is for real-time text 17 mixers and participant endpoints capable of providing an efficient 18 presentation or other treatment of a multi-party real-time text 19 session. The specified mechanism builds on the standard use of the 20 CSRC list in the RTP packet for source identification. The method 21 makes use of the same "text/t140" and "text/red" formats as for two- 22 party sessions. 24 Solutions using multiple RTP streams in the same RTP session are 25 briefly mentioned, as they could have some benefits over the RTP- 26 mixer model. The possibility to implement the solution in a wide 27 range of existing RTP implementations made the RTP-mixer model be 28 selected to be fully specified in this document. 30 A capability exchange is specified so that it can be verified that a 31 mixer and a participant can handle the multi-party coded real-time 32 text stream using the RTP-mixer method. The capability is indicated 33 by use of an SDP media attribute "rtt-mixer". 35 The document updates RFC 4103 "RTP Payload for Text Conversation". 37 A specification of how a mixer can format text for the case when the 38 endpoint is not multi-party aware is also provided. 40 Status of This Memo 42 This Internet-Draft is submitted in full conformance with the 43 provisions of BCP 78 and BCP 79. 45 Internet-Drafts are working documents of the Internet Engineering 46 Task Force (IETF). Note that other groups may also distribute 47 working documents as Internet-Drafts. The list of current Internet- 48 Drafts is at https://datatracker.ietf.org/drafts/current/. 50 Internet-Drafts are draft documents valid for a maximum of six months 51 and may be updated, replaced, or obsoleted by other documents at any 52 time. It is inappropriate to use Internet-Drafts as reference 53 material or to cite them other than as "work in progress." 55 This Internet-Draft will expire on 30 October 2021. 57 Copyright Notice 59 Copyright (c) 2021 IETF Trust and the persons identified as the 60 document authors. All rights reserved. 62 This document is subject to BCP 78 and the IETF Trust's Legal 63 Provisions Relating to IETF Documents (https://trustee.ietf.org/ 64 license-info) in effect on the date of publication of this document. 65 Please review these documents carefully, as they describe your rights 66 and restrictions with respect to this document. Code Components 67 extracted from this document must include Simplified BSD License text 68 as described in Section 4.e of the Trust Legal Provisions and are 69 provided without warranty as described in the Simplified BSD License. 71 Table of Contents 73 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 74 1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 75 1.2. Selected solution and considered alternatives . . . . . . 6 76 1.3. Intended application . . . . . . . . . . . . . . . . . . 9 77 2. Overview of the two specified solutions and selection of 78 method . . . . . . . . . . . . . . . . . . . . . . . . . 10 79 2.1. The RTP-mixer based solution for multi-party aware 80 endpoints . . . . . . . . . . . . . . . . . . . . . . . . 10 81 2.2. Mixing for multi-party unaware endpoints . . . . . . . . 10 82 2.3. Offer/answer considerations . . . . . . . . . . . . . . . 11 83 2.4. Actions depending on capability negotiation result . . . 11 84 3. Details for the RTP-mixer based multi-party aware mixing 85 method . . . . . . . . . . . . . . . . . . . . . . . . . 12 86 3.1. Use of fields in the RTP packets . . . . . . . . . . . . 12 87 3.2. Initial transmission of a BOM character . . . . . . . . . 12 88 3.3. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 12 89 3.4. Transmission interval . . . . . . . . . . . . . . . . . . 13 90 3.5. Only one source per packet . . . . . . . . . . . . . . . 13 91 3.6. Do not send received text to the originating source . . . 13 92 3.7. Clean incoming text . . . . . . . . . . . . . . . . . . . 13 93 3.8. Redundant transmission principles . . . . . . . . . . . . 13 94 3.9. Interleaving text from different sources . . . . . . . . 14 95 3.10. Text placement in packets . . . . . . . . . . . . . . . . 14 96 3.11. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 15 97 3.12. Creation of the redundancy . . . . . . . . . . . . . . . 15 98 3.13. Timer offset fields . . . . . . . . . . . . . . . . . . . 15 99 3.14. Other RTP header fields . . . . . . . . . . . . . . . . . 16 100 3.15. Pause in transmission . . . . . . . . . . . . . . . . . . 16 101 3.16. RTCP considerations . . . . . . . . . . . . . . . . . . . 16 102 3.17. Reception of multi-party contents . . . . . . . . . . . . 16 103 3.18. Performance considerations . . . . . . . . . . . . . . . 18 104 3.19. Security for session control and media . . . . . . . . . 19 105 3.20. SDP offer/answer examples . . . . . . . . . . . . . . . . 19 106 3.21. Packet sequence example from interleaved transmission . . 21 107 3.22. Maximum character rate "CPS" . . . . . . . . . . . . . . 24 108 4. Presentation level considerations . . . . . . . . . . . . . . 24 109 4.1. Presentation by multi-party aware endpoints . . . . . . . 25 110 4.2. Multi-party mixing for multi-party unaware endpoints . . 27 111 5. Relation to Conference Control . . . . . . . . . . . . . . . 33 112 5.1. Use with SIP centralized conferencing framework . . . . . 33 113 5.2. Conference control . . . . . . . . . . . . . . . . . . . 33 114 6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 33 115 6.1. Gateway considerations with Textphones (e.g. TTYs). . . 33 116 6.2. Gateway considerations with WebRTC. . . . . . . . . . . . 34 117 7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 35 118 8. Congestion considerations . . . . . . . . . . . . . . . . . . 35 119 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 35 120 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 35 121 10.1. Registration of the "rtt-mixer" SDP media attribute . . 35 122 11. Security Considerations . . . . . . . . . . . . . . . . . . . 36 123 12. Change history . . . . . . . . . . . . . . . . . . . . . . . 37 124 12.1. Changes included in 125 draft-ietf-avtcore-multi-party-rtt-mix-15 . . . . . . . 37 126 12.2. Changes included in 127 draft-ietf-avtcore-multi-party-rtt-mix-14 . . . . . . . 37 128 12.3. Changes included in 129 draft-ietf-avtcore-multi-party-rtt-mix-13 . . . . . . . 37 130 12.4. Changes included in 131 draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . . 38 132 12.5. Changes included in 133 draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 38 134 12.6. Changes included in 135 draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 38 136 12.7. Changes included in 137 draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 38 138 12.8. Changes included in 139 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 39 140 12.9. Changes included in 141 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 39 142 12.10. Changes included in 143 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 39 144 12.11. Changes included in 145 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 39 147 12.12. Changes included in 148 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 40 149 12.13. Changes included in 150 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 40 151 12.14. Changes included in 152 draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 41 153 12.15. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 41 154 12.16. Changes from 155 draft-hellstrom-avtcore-multi-party-rtt-source-03 to 156 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 41 157 12.17. Changes from 158 draft-hellstrom-avtcore-multi-party-rtt-source-02 to 159 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 42 160 12.18. Changes from 161 draft-hellstrom-avtcore-multi-party-rtt-source-01 to 162 -02 . . . . . . . . . . . . . . . . . . . . . . . . . . 42 163 12.19. Changes from 164 draft-hellstrom-avtcore-multi-party-rtt-source-00 to 165 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 43 166 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 43 167 13.1. Normative References . . . . . . . . . . . . . . . . . . 43 168 13.2. Informative References . . . . . . . . . . . . . . . . . 44 169 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 45 171 1. Introduction 173 "RTP Payload for Text Conversation" [RFC4103] specifies use of RTP 174 [RFC3550] for transmission of real-time text (RTT) and the "text/ 175 t140" format. It also specifies a redundancy format "text/red" for 176 increased robustness. The "text/red" format is registered in 177 [RFC4102]. 179 Real-time text is usually provided together with audio and sometimes 180 with video in conversational sessions. 182 A requirement related to multi-party sessions from the presentation 183 level standard T.140 [T140] for real-time text is: "The display of 184 text from the members of the conversation should be arranged so that 185 the text from each participant is clearly readable, and its source 186 and the relative timing of entered text is visualized in the 187 display." 189 Another requirement is that the mixing procedure must not introduce 190 delays in the text streams that are experienced to be disturbing the 191 real-time experience of the receiving users. 193 Use of RTT is increasing, and specifically, use in emergency calls is 194 increasing. Emergency call use requires multi-party mixing. RFC 195 4103 "RTP Payload for Text Conversation" mixer implementations can 196 use traditional RTP functions for source identification, but the 197 performance of the mixer when giving turns for the different sources 198 to transmit is limited when using the default transmission 199 characteristics with redundancy. 201 The redundancy scheme of [RFC4103] enables efficient transmission of 202 earlier transmitted redundant text in packets together with new text. 203 However the redundancy header format has no source indicators for the 204 redundant transmissions. The redundant parts in a packet must 205 therefore be from the same source as the new text. The recommended 206 transmission is one new and two redundant generations of text 207 (T140blocks) in each packet and the recommended transmission interval 208 for two-party use is 300 ms. 210 Real-time text mixers for multi-party sessions need to include the 211 source with each transmitted group of text from a conference 212 participant so that the text can be transmitted interleaved with text 213 groups from different sources in the rate they are created. This 214 enables the text groups to be presented by endpoints in suitable 215 grouping with other text from the same source. 217 The presentation can then be arranged so that text from different 218 sources can be presented in real-time and easily read. At the same 219 time it is possible for a reading user to perceive approximately when 220 the text was created in real time by the different parties. The 221 transmission and mixing is intended to be done in a general way so 222 that presentation can be arranged in a layout decided by the 223 endpoint. 225 There are existing implementations of RFC 4103 in endpoints without 226 the updates from this document. These will not be able to receive 227 and present real-time text mixed for multi-party aware endpoints. 229 A negotiation mechanism is therefore needed for verification if the 230 parties are able to handle a common method for multi-party 231 transmission and agreeing on using that method. 233 A fall-back mixing procedure is also needed for cases when the 234 negotiation result indicates that a receiving endpoint is not capable 235 of handling the mixed format. Multi-party unaware endpoints would 236 possibly otherwise present all received multi-party mixed text as if 237 it came from the same source regardless of any accompanying source 238 indication coded in fields in the packet. Or they may have any other 239 undesirable way of acting on the multi-party content. The fall-back 240 method is called the mixing procedure for multi-party unaware 241 endpoints. The fall-back method is naturally not expected to meet 242 all performance requirements placed on the mixing procedure for 243 multi-party aware endpoints. 245 The document updates [RFC4103] by introducing an attribute for 246 indicating capability for the RTP-mixer based multi-party mixing case 247 and rules for source indications and interleaving of text from 248 different sources. 250 1.1. Terminology 252 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 253 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 254 "OPTIONAL" in this document are to be interpreted as described in BCP 255 14 [RFC2119] [RFC8174] when, and only when, they appear in all 256 capitals, as shown above. 258 The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP- 259 mixer, RTP-translator are defined in [RFC3550]. 261 The term "T140block" is defined in [RFC4103] to contain one or more 262 T.140 code elements. 264 "TTY" stands for a text telephone type used in North America. 266 "WebRTC" stands for web based communication specified by W3C and 267 IETF. See [RFC8825]. 269 "DTLS-SRTP" stands for security specified in [RFC5764]. 271 "multi-party aware" stands for an endpoint receiving real-time text 272 from multiple sources through a common conference mixer being able to 273 present the text in real-time separated by source and presented so 274 that a user can get an impression of the approximate relative timing 275 of text from different parties. 277 "multi-party unaware" stands for an endpoint not itself being able to 278 separate text from different sources when received through a common 279 conference mixer. 281 1.2. Selected solution and considered alternatives 283 A number of alternatives were considered when searching an efficient 284 and easily implemented multi-party method for real-time text. This 285 section explains a few of them briefly. 287 Multiple RTP streams, one per participant. 288 One RTP stream per source would be sent in the same RTP session 289 with the "text/red" format. From some points of view, use of 290 multiple RTP streams, one for each source, sent in the same RTP 291 session would be efficient, and would use exactly the same packet 292 format as [RFC4103] and the same payload type. A couple of 293 relevant scenarios using multiple RTP-streams are specified in 294 "RTP Topologies" [RFC7667]. One possibility of special interest 295 is the Selective Forwarding Middlebox (SFM) topology specified in 296 RFC 7667 section 3.7 that could enable end to end encryption. In 297 contrast to audio and video, real-time text is only transmitted 298 when the users actually transmit information. Thus an SFM 299 solution would not need to exclude any party from transmission 300 under normal conditions. In order to allow the mixer to convey 301 the packets with the payload preserved and encrypted, an SFM 302 solution would need to act on some specific characteristics of the 303 "text/red" format. The redundancy headers are part of the 304 payload, so the receiver would need to just assume that the 305 payload type number in the redundancy header is for "text/t140". 306 The characters per second parameter (CPS) would need to act per 307 stream. The relation between the SSRC and the source would need 308 to be conveyed in some specified way, e.g. in the CSRC. Recovery 309 and loss detection would preferably be based on sequence number 310 gap detection. Thus sequence number gaps in the incoming stream 311 to the mixer would need to be reflected in the stream to the 312 participant and no new gaps created by the mixer. However, the 313 RTP implementation in both mixers and endpoints need to support 314 multiple streams in the same RTP session in order to use this 315 mechanism. For best deployment opportunity, it should be possible 316 to upgrade existing endpoint solutions to be multi-party aware 317 with a reasonable effort. There is currently a lack of support 318 for multi-stream RTP in certain implementation technologies. This 319 fact made this solution only briefly mentioned in this document as 320 an option for further study. 322 RTP-mixer based method for multi-party aware endpoints. 323 The "text/red" format in RFC 4103 is sent with shorter 324 transmission interval with the RTP-mixer method and indicating 325 source in CSRC. The "text/red" format with "text/t140" payload in 326 a single RTP stream can be sent when text is available from the 327 call participants instead of at the regular 300 ms. The source is 328 indicated in the CSRC field. Transmission of packets with text 329 from different sources can then be done smoothly while 330 simultaneous transmission occurs as long as it is not limited by 331 the maximum character rate "CPS". With ten participants sending 332 text simultaneously, the switching and transmission performance is 333 good. With more simultaneously sending participants, and 334 receivers with default capacity there will be a noticeable 335 jerkiness and delay in text presentation. The jerkiness will be 336 more expressed the more participants who send text simultaneously. 337 Two seconds jerkiness will be noticeable and slightly unpleasant, 338 but corresponds in time to what typing humans often cause by 339 hesitation or changing position while typing. A benefit of this 340 method is that no new packet format needs to be introduced and 341 implemented. Since simultaneous typing by more than two parties 342 is very rare, this method can be used successfully with good 343 performance. Recovery of text in case of packet loss is based on 344 analysis of timestamps of received redundancy versus earlier 345 received text. Negotiation is based on a new SDP media attribute 346 "rtt-mixer". This method is selected to be the main one specified 347 in this document. 349 Multiple sources per packet. 350 A new "text" media subtype would be specified with up to 15 351 sources in each packet. The mechanism would make use of the RTP 352 mixer model specified in RTP [RFC3550]. Text from up to 15 353 sources can be included in each packet. Packets are normally sent 354 every 300 ms. The mean delay will be 150 ms. The sources are 355 indicated in strict order in the CSRC list of the RTP packets. A 356 new redundancy packet format is specified. This method would 357 result in good performance, but would require standardisation and 358 implementation of new releases in the target technologies that 359 would take more time than desirable to complete. It was therefore 360 not selected to be included in this document. 362 Mixing for multi-party unaware endpoints 363 Presentation of text from multiple parties is prepared by the 364 mixer in one single stream. It is desirable to have a method that 365 does not require any modifications in existing user devices 366 implementing RFC 4103 for RTT without explicit support of multi- 367 party sessions. This is possible by having the mixer insert a new 368 line and a text formatted source label before each switch of text 369 source in the stream. Switch of source can only be done in places 370 in the text where it does not disturb the perception of the 371 contents. Text from only one source can be presented in real time 372 at a time. The delay will therefore be varying. The method also 373 has other limitations, but is included in this document as a 374 fallback method. In calls where parties take turns properly by 375 ending their entries with a new line, the limitations will have 376 limited influence on the user experience. while only two parties 377 send text, these two will see the text in real time with no delay. 378 This method is specified as a fallback method in this document. 380 RTT transport in WebRTC 381 Transport of real-time text in the WebRTC technology is specified 382 to use the WebRTC data channel in [RFC8865]. 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 wants 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 in order to 465 maintain interoperability, if nothing else is specified for the 466 application, format transmitted text to that participant to be 467 suitable to present on a multi-party unaware endpoint as further 468 specified in Section 4.2. 470 2.3. Offer/answer considerations 472 RTP Payload for Text Conversation [RFC4103] specifies use of RTP 473 [RFC3550], and a redundancy format "text/red" for increased 474 robustness of real-time text transmission. This document updates 475 [RFC4103] by introducing a capability negotiation for handling multi- 476 party real-time text, a way to indicate the source of transmitted 477 text, and rules for efficient timing of the transmissions interleaved 478 from 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 is 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 SHALL send packets 566 distributed in time as long as there is something (new or redundant 567 T140blocks) to transmit. The maximum transmission interval SHALL 568 then be 330 ms, when no other limitations cause a longer interval to 569 be temporarily used. It is RECOMMENDED to send the next packet to a 570 receiver as soon as new text to that receiver is available, as long 571 as the maximum character rate ("CPS") to the receiver is not exceeded 572 during any 10 second interval. The intention of these time intervals 573 is to keep the latency low and network load limited while keeping a 574 good protection against text loss in bursty packet loss conditions. 575 The main purpose of the 330 ms interval is for timing of redundant 576 transmission, when no new text from the same source is available. 578 If the "CPS" value is reached, longer transmission intervals SHALL be 579 applied and only part of the text queued for transmission sent at end 580 of each transmission interval, until the transmission rate falls 581 under the "CPS" value again. See also Section 8 583 For a transmitter not acting in a mixer, the transmission interval 584 principles from [RFC4103] apply, and the transmission interval SHALL 585 be 300 ms. 587 3.5. Only one source per packet 589 New text and redundant copies of earlier text from one source SHALL 590 be transmitted in the same packet if available for transmission at 591 the same time. Text from different sources MUST NOT be transmitted 592 in the same packet. 594 3.6. Do not send received text to the originating source 596 Text received by a mixer from a participant SHOULD NOT be included in 597 transmission from the mixer to that participant, because the normal 598 behavior of the endpoint is to present locally produced locally. 600 3.7. Clean incoming text 602 A mixer SHALL handle reception, recovery from packet loss, deletion 603 of superfluous redundancy, marking of possible text loss and deletion 604 of 'BOM' characters from each participant before queueing received 605 text for transmission to receiving participants. 607 3.8. Redundant transmission principles 609 A transmitting party using redundancy SHALL send redundant 610 repetitions of T140blocks already transmitted in earlier packets. 612 The number of redundant generations of T140blocks to include in 613 transmitted packets SHALL be deduced from the SDP negotiation. It 614 SHALL be set to the minimum of the number declared by the two parties 615 negotiating a connection. It is RECOMMENDED to declare and transmit 616 one original and two redundant generations of the T140blocks, because 617 that provides good protection against text loss in case of packet 618 loss, and low overhead. 620 3.9. Interleaving text from different sources 622 When text from more than one source is available for transmission 623 from a mixer, the mixer SHALL let the sources take turns in having 624 their text transmitted. 626 The source with the oldest text received in the mixer or oldest 627 redundant text SHALL be next in turn to get all its available unsent 628 text transmitted. Any redundant repetitions of earlier transmitted 629 text not yet sent the intended number of times SHALL be included as 630 redundant retransmission in the transmission. 632 3.10. Text placement in packets 634 The mixer SHALL compose and transmit an RTP packet to a receiver when 635 one of the following conditions has occurred: 637 * There is unsent text available for transmission to that receiver. 639 * 330 ms has passed since already transmitted text was queued for 640 transmission as redundant text. 642 At time of transmission, the mixer SHALL populate the RTP packet with 643 all T140blocks queued for transmission originating from the source in 644 turn for transmission as long as this is not in conflict with the 645 allowed number of characters per second ("CPS") or the maximum packet 646 size. In this way, the latency of the latest received text is kept 647 low even in moments of simultaneous transmission from many sources. 649 Redundant text SHALL also be included. See Section 3.12 651 The SSRC of the source SHALL be placed as the only member in the 652 CSRC-list. 654 Note: The CSRC-list in an RTP packet only includes the participant 655 whose text is included in text blocks. It is not the same as the 656 total list of participants in a conference. With audio and video 657 media, the CSRC-list would often contain all participants who are not 658 muted whereas text participants that don't type are completely silent 659 and thus are not represented in RTP packet CSRC-lists. 661 3.11. Empty T140blocks 663 If no unsent T140blocks were available for a source at the time of 664 populating a packet, but T140blocks are available which have not yet 665 been sent the full intended number of redundant transmissions, then 666 the primary T140block for that source is composed of an empty 667 T140block, and populated (without taking up any length) in a packet 668 for transmission. The corresponding SSRC SHALL be placed as usual in 669 its place in the CSRC-list. 671 The first packet in the session, the first after a source switch and 672 the first after a pause SHALL be poulated with the available 673 T140blocks for the source in turn to be sent as primary, and empty 674 T140blocks for the agreed number of redundancy generations. 676 3.12. Creation of the redundancy 678 The primary T140block from a source in the latest transmitted packet 679 is saved for populating the first redundant T140block for that source 680 in next transmission of text from that source. The first redundant 681 T140block for that source from the latest transmission is saved for 682 populating the second redundant T140block in next transmission of 683 text from that source. 685 Usually this is the level of redundancy used. If a higher number of 686 redundancy is negotiated, then the procedure SHALL be maintained 687 until all available redundant levels of T140blocks are placed in the 688 packet. If a receiver has negotiated a lower number of "text/red" 689 generations, then that level SHALL be the maximum used by the 690 transmitter. 692 The T140blocks saved for transmission as redundant data are assigned 693 a planned transmission time 330 ms after the current time, but SHOULD 694 be transmitted earlier if new text for the same source gets in turn 695 for transmission before that time. 697 3.13. Timer offset fields 699 The timestamp offset values SHALL be inserted in the redundancy 700 header, with the time offset from the RTP timestamp in the packet 701 when the corresponding T140block was sent as primary. 703 The timestamp offsets are expressed in the same clock tick units as 704 the RTP timestamp. 706 The timestamp offset values for empty T140blocks have no relevance 707 but SHOULD be assigned realistic values. 709 3.14. Other RTP header fields 711 The number of members in the CSRC list ( 0 or 1) SHALL be placed in 712 the "CC" header field. Only mixers place value 1 in the "CC" field. 713 A value of "0" indicates that the source is the transmitting device 714 itself and that the source is indicated by the SSRC field. This 715 value is used by endpoints, and by mixers sending data that it is 716 source of itself. 718 The current time SHALL be inserted in the timestamp. 720 The SSRC of the mixer for the RTT session SHALL be inserted in the 721 SSRC field of the RTP header. 723 The M-bit SHALL be handled as specified in [RFC4103]. 725 3.15. Pause in transmission 727 When there is no new T140block to transmit, and no redundant 728 T140block that has not been retransmitted the intended number of 729 times from any source, the transmission process SHALL be stopped 730 until either new T140blocks arrive, or a keep-alive method calls for 731 transmission of keep-alive packets. 733 3.16. RTCP considerations 735 A mixer SHALL send RTCP reports with SDES, CNAME and NAME information 736 about the sources in the multi-party call. This makes it possible 737 for participants to compose a suitable label for text from each 738 source. 740 Integrity SHALL be considered when composing these fields. They 741 contain name and address information that may be sensitive to 742 transmit in its entirety e.g. to unauthenticated participants. 743 Similar considerations SHALL be taken as for other media. 745 3.17. Reception of multi-party contents 747 The "text/red" receiver included in an endpoint with presentation 748 functions will receive RTP packets in the single stream from the 749 mixer, and SHALL distribute the T140blocks for presentation in 750 presentation areas for each source. Other receiver roles, such as 751 gateways or chained mixers are also feasible, and requires 752 consideration if the stream shall just be forwarded, or distributed 753 based on the different sources. 755 3.17.1. Acting on the source of the packet contents 757 If the "CC" field value of a received packet is 1, it indicates that 758 the text is conveyed from a source indicated in the single member in 759 the CSRC-list, and the receiver MUST act on the source according to 760 its role. If the CC value is 0, the source is indicated in the SSRC 761 field. 763 3.17.2. Detection and indication of possible text loss 765 The RTP sequence numbers of the received packets SHALL be monitored 766 for gaps and packets out of order. If a sequence number gap appears 767 and still exists after some defined short time for jitter resolution, 768 the packets in the gap SHALL be regarded as lost. 770 If it is known that only one source is active in the RTP session, 771 then it is likely that a gap equal to or larger than the agreed 772 number of redundancy generations (including the primary) causes text 773 loss. In that case a t140block SHALL be created with a marker for 774 possible text loss [T140ad1] and assigned to the source and inserted 775 in the reception buffer for that source. 777 If it is known that more than one source is active in the RTP 778 session, then it is not possible in general to evaluate if text was 779 lost when packets were lost. With two active sources and the 780 recommended number of redundancy generations (3), it can take a gap 781 of five consecutive lost packets until any text may be lost, but text 782 loss can also appear if three non-consecutive packets are lost when 783 they contained consecutive data from the same source. A simple 784 method to decide when there is risk for resulting text loss is to 785 evaluate if three or more packets were lost within one second. If 786 this simple method is used, then a t140block SHOULD be created with a 787 marker for possible text loss [T140ad1] and assigned to the SSRC of 788 the transmitter as a general input from the mixer. 790 Implementations MAY apply more refined methods for more reliable 791 detection of if text was lost or not. Any refined method SHALL 792 prefer marking possible loss rather than not marking when it is 793 uncertain if there was loss. 795 3.17.3. Extracting text and handling recovery 797 When applying the following procedures, the effects MUST be 798 considered of possible timestamp wrap around and the RTP session 799 possibly changing SSRC. 801 When a packet is received in an RTP session using the packetization 802 for multi-party aware endpoints, its T140blocks SHALL be extracted in 803 the following way. The description is adapted to the default 804 redundancy case using the original and two redundant generations. 806 The source SHALL be extracted from the CSRC-list if available, 807 otherwise from the SSRC. 809 If the received packet is the first packet received from the source, 810 then all T140blocks in the packet SHALL be retrieved and assigned to 811 a receive buffer for the source beginning with the second generation 812 redundancy, continuing with the first generation redundancy and 813 finally the primary. 815 Note: The normal case is that in the first packet, only the primary 816 data has contents. The redundant data has contents in the first 817 received packet from a source only after initial packet loss. 819 If the packet is not the first packet from a source, then if the 820 second generation redundant data is available, its timestamp SHALL be 821 created by subtracting its timestamp offset from the RTP timestamp. 822 If the resulting timestamp is later than the latest retrieved data 823 from the same source, then the redundant data SHALL be retrieved and 824 appended to the receive buffer. The process SHALL be continued in 825 the same way for the first generation redundant data. After that, 826 the primary data SHALL be retrieved from the packet and appended to 827 the receive buffer for the source. 829 3.17.4. Delete 'BOM' 831 Unicode character 'BOM' is used as a start indication and sometimes 832 used as a filler or keep alive by transmission implementations. 833 These SHALL be deleted after extraction from received packets. 835 3.18. Performance considerations 837 This solution has good performance with low text delays as long as 838 the sum of characters per second during any 10 second interval sent 839 from a number of simultaneously sending participants to a receiving 840 participant does not reach the 'CPS' value. At higher numbers of 841 characters per second sent, a jerkiness is visible in the 842 presentation of text. The solution is therefore suitable for 843 emergency service use, relay service use, and small or well-managed 844 larger multimedia conferences. Only in large unmanaged conferences 845 with a high number of participants there may on very rare occasions 846 appear situations when many participants happen to send text 847 simultaneously, resulting in unpleasantly jerky presentation of text 848 from each sending participant. It should be noted that it is only 849 the number of users sending text within the same moment that causes 850 jerkiness, not the total number of users with RTT capability. 852 3.19. Security for session control and media 854 Security SHOULD be applied when possible regarding the capabilities 855 of the participating devices by use of SIP over TLS by default 856 according to [RFC5630] section 3.1.3 on session control level and by 857 default using DTLS-SRTP [RFC5764] on media level. In applications 858 where legacy endpoints without security may exist, a negotiation 859 SHOULD be performed to decide if security by encryption on media 860 level will be applied. If no other security solution is mandated for 861 the application, then OSRTP [RFC8643] is a suitable method be applied 862 to negotiate SRTP media security with DTLS. Most SDP examples below 863 are for simplicity expressed without the security additions. The 864 principles (but not all details) for applying DTLS-SRTP [RFC5764] 865 security is shown in a couple of the following examples. 867 This document contains two mixing procedures which imply different 868 security levels. The mixing for conference-unaware endpoints has 869 lower security level than the mixing method for conference-aware 870 endpoints, because there may be an opportunity for a malicious mixer 871 or a middleman to masquerade the source labels accompanying the text 872 streams in text format. This is especially true if support of un- 873 encrypted SIP and media is supported because of lack of such support 874 in the target endpoints. However, the mixing for conference-aware 875 endpoints as specified here also requires that the mixer can be 876 trusted. End to end encryption would require further work and could 877 be based on WebRTC as specified in Section 1.2. 879 3.20. SDP offer/answer examples 881 This section shows some examples of SDP for session negotiation of 882 the real-time text media in SIP sessions. Audio is usually provided 883 in the same session, and sometimes also video. The examples only 884 show the part of importance for the real-time text media. The 885 examples relate to the single RTP stream mixing for multi-party aware 886 endpoints and for multi-party unaware endpoints. 888 Note: Multi-party RTT MAY also be provided through other methods, 889 e.g. by a Selective Forwarding Middlebox (SFM). In that case, the 890 SDP of the offer will include something specific for that method, and 891 an answer acknowledging the use of that method would accept it by 892 something specific included in the SDP. The offer may contain also 893 the "rtt-mixer" SDP media attribute for the main RTT media when the 894 offeror has capability for both multi-party methods, while an answer, 895 selecting to use SFM will not include the "rtt-mixer" SDP media 896 attribute. 898 Offer example for "text/red" format and multi-party support: 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 Answer example from a multi-party capable device 907 m=text 14000 RTP/AVP 100 98 908 a=rtpmap:98 t140/1000 909 a=rtpmap:100 red/1000 910 a=fmtp:100 98/98/98 911 a=rtt-mixer 913 Offer example for "text/red" format including multi-party 914 and security: 915 a=fingerprint: (fingerprint1) 916 m=text 11000 RTP/AVP 100 98 917 a=rtpmap:98 t140/1000 918 a=rtpmap:100 red/1000 919 a=fmtp:100 98/98/98 920 a=rtt-mixer 922 The "fingerprint" is sufficient to offer DTLS-SRTP, with the media 923 line still indicating RTP/AVP. 925 Note: For brevity, the entire value of the SDP fingerprint attribute 926 is not shown in this and the following example. 928 Answer example from a multi-party capable device with security 929 a=fingerprint: (fingerprint2) 930 m=text 16000 RTP/AVP 100 98 931 a=rtpmap:98 t140/1000 932 a=rtpmap:100 red/1000 933 a=fmtp:100 98/98/98 934 a=rtt-mixer 936 With the "fingerprint" the device acknowledges use of SRTP/DTLS. 938 Answer example from a multi-party unaware device that also 939 does not support security: 941 m=text 12000 RTP/AVP 100 98 942 a=rtpmap:98 t140/1000 943 a=rtpmap:100 red/1000 944 a=fmtp:100 98/98/98 946 3.21. Packet sequence example from interleaved transmission 948 This example shows a symbolic flow of packets from a mixer including 949 loss and recovery. The sequence includes interleaved transmission of 950 text from two RTT sources A and B. P indicates primary data. R1 is 951 first redundant generation data and R2 is the second redundant 952 generation data. A1, B1, A2 etc are text chunks (T140blocks) 953 received from the respective sources and sent on to the receiver by 954 the mixer. X indicates dropped packet between the mixer and a 955 receiver. The session is assumed to use original and two redundant 956 generations of RTT. 958 |-----------------------| 959 |Seq no 101, Time=20400 | 960 |CC=1 | 961 |CSRC list A | 962 |R2: A1, Offset=600 | 963 |R1: A2, Offset=300 | 964 |P: A3 | 965 |-----------------------| 967 Assuming that earlier packets ( with text A1 and A2) were received in 968 sequence, text A3 is received from packet 101 and assigned to 969 reception area A. The mixer is now assumed to have received text 970 from source B 100 ms after packet 101 and will send that text. 971 Transmission of A2 and A3 as redundancy is planned for 330 ms after 972 packet 101 if no new text from A is ready to be sent before that. 974 |-----------------------| 975 |Seq no 102, Time=20500 | 976 |CC=1 | 977 |CSRC list B | 978 |R2 Empty, Offset=600 | 979 |R1: Empty, Offset=300 | 980 |P: B1 | 981 |-----------------------| 982 Packet 102 is received. 983 B1 is retrieved from this packet. Redundant transmission of 984 B1 is planned 330 ms after packet 102. 986 X------------------------| 987 X Seq no 103, Timer=20730| 988 X CC=1 | 989 X CSRC list A | 990 X R2: A2, Offset=630 | 991 X R1: A3, Offset=330 | 992 X P: Empty | 993 X------------------------| 994 Packet 103 is assumed to be lost due to network problems. 995 It contains redundancy for A. Sending A3 as second level 996 redundancy is planned for 330 ms after packet 103. 998 X------------------------| 999 X Seq no 104, Timer=20830| 1000 X CC=1 | 1001 X CSRC list B | 1002 X R2: Empty, Offset=600 | 1003 X R1: B1, Offset=300 | 1004 X P: B2 | 1005 X------------------------| 1006 Packet 104 contains text from B, including new B2 and 1007 redundant B1. It is assumed dropped in network 1008 problems. 1009 The mixer has A3 redundancy to send but no new text 1010 appears from A and therefore the redundancy is sent 1011 330 ms after the previous packet with text from A. 1013 |------------------------| 1014 | Seq no 105, Timer=21060| 1015 | CC=1 | 1016 | CSRC list A | 1017 | R2: A3, Offset=660 | 1018 | R1: Empty, Offset=330 | 1019 | P: Empty | 1020 |------------------------| 1021 Packet 105 is received. 1022 A gap for lost 103, and 104 is detected. 1023 Assume that no other loss was detected the last second. 1024 Then it can be concluded that nothing was totally lost. 1026 R2 is checked. Its original time was 21040-660=20400. 1027 A packet with text from A was received with that 1028 timestamp, so nothing needs to be recovered. 1030 B1 and B2 still needs to be transmitted as redundancy. 1031 This is planned 330 ms after packet 105. That 1032 would be at 21150. 1034 |-----------------------| 1035 |Seq no 106, Timer=21160| 1036 |CC=1 | 1037 |CSRC list B | 1038 | R2: B1, Offset=660 | 1039 | R1: B2, Offset=330 | 1040 | P: Empty | 1041 |-----------------------| 1043 Packet 106 is received. 1045 The second level redundancy in packet 106 is B1 and has timestamp 1046 offset 660 ms. The timestamp of packet 106 minus 660 is 20500 which 1047 is the timestamp of packet 102 THAT was received. So B1 does not 1048 need to be retrieved. The first level redundancy in packet 106 has 1049 offset 330. The timestamp of packet 106 minus 330 is 20830. That is 1050 later than the latest received packet with source B. Therefore B2 is 1051 retrieved and assigned to the input buffer for source B. No primary 1052 is available in packet 106. 1054 After this sequence, A3 and B1 and B2 have been received. In this 1055 case no text was lost. 1057 3.22. Maximum character rate "CPS" 1059 The default maximum rate of reception of "text/t140" real-time text 1060 is in [RFC4103] specified to be 30 characters per second. The value 1061 MAY be modified in the "CPS" parameter of the FMTP attribute in the 1062 media section for the "text/t140" media. A mixer combining real-time 1063 text from a number of sources may occasionally have a higher combined 1064 flow of text coming from the sources. Endpoints SHOULD therefore 1065 specify a suitable higher value for the "CPS" parameter, 1066 corresponding to its real reception capability. A value for "CPS" of 1067 90 SHALL be the default for the "text/t140" stream in the "text/red" 1068 format when multi-party real-time text is negotiated. See [RFC4103] 1069 for the format and use of the "CPS" parameter. The same rules apply 1070 for the multi-party case except for the default value. 1072 4. Presentation level considerations 1074 "Protocol for multimedia application text conversation" [T140] 1075 provides the presentation level requirements for the [RFC4103] 1076 transport. Functions for erasure and other formatting functions and 1077 are specified in [T140] which has the following general statement for 1078 the presentation: 1080 "The display of text from the members of the conversation should be 1081 arranged so that the text from each participant is clearly readable, 1082 and its source and the relative timing of entered text is visualized 1083 in the display. Mechanisms for looking back in the contents from the 1084 current session should be provided. The text should be displayed as 1085 soon as it is received." 1087 Strict application of [T140] is of essence for the interoperability 1088 of real-time text implementations and to fulfill the intention that 1089 the session participants have the same information of the text 1090 contents of the conversation without necessarily having the exact 1091 same layout of the conversation. 1093 [T140] specifies a set of presentation control codes to include in 1094 the stream. Some of them are optional. Implementations MUST be able 1095 to ignore optional control codes that they do not support. 1097 There is no strict "message" concept in real-time text. The Unicode 1098 Line Separator character SHALL be used as a separator allowing a part 1099 of received text to be grouped in presentation. The characters 1100 "CRLF" may be used by other implementations as replacement for Line 1101 Separator. The "CRLF" combination SHALL be erased by just one 1102 erasing action, just as the Line Separator. Presentation functions 1103 are allowed to group text for presentation in smaller groups than the 1104 line separators imply and present such groups with source indication 1105 together with text groups from other sources (see the following 1106 presentation examples). Erasure has no specific limit by any 1107 delimiter in the text stream. 1109 4.1. Presentation by multi-party aware endpoints 1111 A multi-party aware receiving party, presenting real-time text MUST 1112 separate text from different sources and present them in separate 1113 presentation fields. The receiving party MAY separate presentation 1114 of parts of text from a source in readable groups based on other 1115 criteria than line separator and merge these groups in the 1116 presentation area when it benefits the user to most easily find and 1117 read text from the different participants. The criteria MAY e.g. be 1118 a received comma, full stop, or other phrase delimiters, or a long 1119 pause. 1121 When text is received from multiple original sources, the 1122 presentation SHALL provide a view where text is added in multiple 1123 presentation fields. 1125 If the presentation presents text from different sources in one 1126 common area, the presenting endpoint SHOULD insert text from the 1127 local user ended at suitable points merged with received text to 1128 indicate the relative timing for when the text groups were completed. 1129 In this presentation mode, the receiving endpoint SHALL present the 1130 source of the different groups of text. This presentation style is 1131 called the "chat" style here and provides a possibility to follow 1132 text arriving from multiple parties and the approximate relative time 1133 that text is received related to text from the local user. 1135 A view of a three-party RTT call in chat style is shown in this 1136 example . 1138 _________________________________________________ 1139 | |^| 1140 |[Alice] Hi, Alice here. |-| 1141 | | | 1142 |[Bob] Bob as well. | | 1143 | | | 1144 |[Eve] Hi, this is Eve, calling from Paris. | | 1145 | I thought you should be here. | | 1146 | | | 1147 |[Alice] I am coming on Thursday, my | | 1148 | performance is not until Friday morning.| | 1149 | | | 1150 |[Bob] And I on Wednesday evening. | | 1151 | | | 1152 |[Alice] Can we meet on Thursday evening? | | 1153 | | | 1154 |[Eve] Yes, definitely. How about 7pm. | | 1155 | at the entrance of the restaurant | | 1156 | Le Lion Blanc? | | 1157 |[Eve] we can have dinner and then take a walk |-| 1158 |______________________________________________|v| 1159 | But I need to be back to |^| 1160 | the hotel by 11 because I need |-| 1161 | | | 1162 | I wou |-| 1163 |______________________________________________|v| 1164 | of course, I underst | 1165 |________________________________________________| 1167 Figure 3: Example of a three-party RTT call presented in chat style 1168 seen at participant 'Alice's endpoint. 1170 Other presentation styles than the chat style MAY be arranged. 1172 This figure shows how a coordinated column view MAY be presented. 1174 _____________________________________________________________________ 1175 | Bob | Eve | Alice | 1176 |____________________|______________________|_______________________| 1177 | | |I will arrive by TGV. | 1178 |My flight is to Orly| |Convenient to the main | 1179 | |Hi all, can we plan |station. | 1180 | |for the seminar? | | 1181 |Eve, will you do | | | 1182 |your presentation on| | | 1183 |Friday? |Yes, Friday at 10. | | 1184 |Fine, wo | |We need to meet befo | 1185 |___________________________________________________________________| 1186 Figure 4: An example of a coordinated column-view of a three-party 1187 session with entries ordered vertically in approximate time-order. 1189 4.2. Multi-party mixing for multi-party unaware endpoints 1191 When the mixer has indicated RTT multi-party capability in an SDP 1192 negotiation, but the multi-party capability negotiation fails with an 1193 endpoint, then the agreed "text/red" or "text/t140" format SHALL be 1194 used and the mixer SHOULD compose a best-effort presentation of 1195 multi-party real-time text in one stream intended to be presented by 1196 an endpoint with no multi-party awareness, when that is desired in 1197 the actual implementation. The following specifies a procedure which 1198 MAY be applied in that situation. 1200 This presentation format has functional limitations and SHOULD be 1201 used only to enable participation in multi-party calls by legacy 1202 deployed endpoints implementing only RFC 4103 without any multi-party 1203 extensions specified in this document. 1205 The principles and procedures below do not specify any new protocol 1206 elements. They are instead composed from the information in [T140] 1207 and an ambition to provide a best effort presentation on an endpoint 1208 which has functions only for two-party calls. 1210 The mixer mixing for multi-party unaware endpoints SHALL compose a 1211 simulated limited multi-party RTT view suitable for presentation in 1212 one presentation area. The mixer SHALL group text in suitable groups 1213 and prepare for presentation of them by inserting a new line between 1214 them if the transmitted text did not already end with a new line. A 1215 presentable label SHALL be composed and sent for the source initially 1216 in the session and after each source switch. With this procedure the 1217 time for switching from transmission of text from one source to 1218 transmission of text from another source is depending on the actions 1219 of the users. In order to expedite source switch, a user can for 1220 example end its turn with a new line. 1222 4.2.1. Actions by the mixer at reception from the call participants 1224 When text is received by the mixer from the different participants, 1225 the mixer SHALL recover text from redundancy if any packets are lost. 1226 The mark for lost text [T140ad1] SHALL be inserted in the stream if 1227 unrecoverable loss appears. Any Unicode "BOM" characters, possibly 1228 used for keep-alive SHALL be deleted. The time of creation of text 1229 (retrieved from the RTP timestamp) SHALL be stored together with the 1230 received text from each source in queues for transmission to the 1231 recipients in order to be able to evaluate text loss. 1233 4.2.2. Actions by the mixer for transmission to the recipients 1235 The following procedure SHALL be applied for each multi-party unaware 1236 recipient of multi-party text from the mixer. 1238 The text for transmission SHALL be formatted by the mixer for each 1239 receiving user for presentation in one single presentation area. 1240 Text received from a participant SHOULD NOT be included in 1241 transmission to that participant because it is usually presented 1242 locally at transmission time. When there is text available for 1243 transmission from the mixer to a receiving party from more than one 1244 participant, the mixer SHALL switch between transmission of text from 1245 the different sources at suitable points in the transmitted stream. 1247 When switching source, the mixer SHALL insert a line separator if the 1248 already transmitted text did not end with a new line (line separator 1249 or CRLF). A label SHALL be composed from information in the CNAME 1250 and NAME fields in RTCP reports from the participant to have its text 1251 transmitted, or from other session information for that user. The 1252 label SHALL be delimited by suitable characters (e.g. '[ ]') and 1253 transmitted. The CSRC SHALL indicate the selected source. Then text 1254 from that selected participant SHALL be transmitted until a new 1255 suitable point for switching source is reached. 1257 Integrity considerations SHALL be taken when composing the label. 1259 Seeking a suitable point for switching source SHALL be done when 1260 there is older text waiting for transmission from any party than the 1261 age of the last transmitted text. Suitable points for switching are: 1263 * A completed phrase ended by comma 1265 * A completed sentence 1267 * A new line (line separator or CRLF) 1269 * A long pause (e.g. > 10 seconds) in received text from the 1270 currently transmitted source 1272 * If text from one participant has been transmitted with text from 1273 other sources waiting for transmission for a long time (e.g. > 1 1274 minute) and none of the other suitable points for switching has 1275 occurred, a source switch MAY be forced by the mixer at next word 1276 delimiter, and also if even a word delimiter does not occur within 1277 a time (e.g. 15 seconds) after the scan for word delimiter 1278 started. 1280 When switching source, the source which has the oldest text in queue 1281 SHALL be selected to be transmitted. A character display count SHALL 1282 be maintained for the currently transmitted source, starting at zero 1283 after the label is transmitted for the currently transmitted source. 1285 The status SHALL be maintained for the latest control code for Select 1286 Graphic Rendition (SGR) from each source. If there is an SGR code 1287 stored as the status for the current source before the source switch 1288 is done, a reset of SGR SHALL be sent by the sequence SGR 0 [009B 1289 0000 006D] after the new line and before the new label during a 1290 source switch. See SGR below for an explanation. This transmission 1291 does not influence the display count. 1293 If there is an SGR code stored for the new source after the source 1294 switch, that SGR code SHALL be transmitted to the recipient before 1295 the label. This transmission does not influence the display count. 1297 4.2.3. Actions on transmission of text 1299 Text from a source sent to the recipient SHALL increase the display 1300 count by one per transmitted character. 1302 4.2.4. Actions on transmission of control codes 1304 The following control codes specified by T.140 require specific 1305 actions. They SHALL cause specific considerations in the mixer. 1306 Note that the codes presented here are expressed in UCS-16, while 1307 transmission is made in UTF-8 transform of these codes. 1309 BEL 0007 Bell Alert in session, provides for alerting during an 1310 active session. The display count SHALL NOT be altered. 1312 NEW LINE 2028 Line separator. Check and perform a source switch if 1313 appropriate. Increase display count by 1. 1315 CR LF 000D 000A A supported, but not preferred way of requesting a 1316 new line. Check and perform a source switch if appropriate. 1317 Increase display count by 1. 1319 INT ESC 0061 Interrupt (used to initiate mode negotiation 1320 procedure). The display count SHALL NOT be altered. 1322 SGR 009B Ps 006D Select graphic rendition. Ps is rendition 1323 parameters specified in ISO 6429. The display count SHALL NOT be 1324 altered. The SGR code SHOULD be stored for the current source. 1326 SOS 0098 Start of string, used as a general protocol element 1327 introducer, followed by a maximum 256 bytes string and the ST. 1328 The display count SHALL NOT be altered. 1330 ST 009C String terminator, end of SOS string. The display count 1331 SHALL NOT be altered. 1333 ESC 001B Escape - used in control strings. The display count SHALL 1334 NOT be altered for the complete escape code. 1336 Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used 1337 for synchronization and keep-alive, SHALL be deleted from incoming 1338 streams. It SHALL also be sent first after session establishment 1339 to the recipient. The display count SHALL NOT be altered. 1341 Missing text mark (U+FFFD) "Replacement character", represented as a 1342 question mark in a rhombus, or if that is not feasible, replaced 1343 by an apostrophe ', marks place in stream of possible text loss. 1344 This mark SHALL be inserted by the reception procedure in case of 1345 unrecoverable loss of packets. The display count SHALL be 1346 increased by one when sent as for any other character. 1348 SGR If a control code for selecting graphic rendition (SGR), other 1349 than reset of the graphic rendition (SGR 0) is sent to a 1350 recipient, that control code SHALL also be stored as status for 1351 the source in the storage for SGR status. If a reset graphic 1352 rendition (SGR 0) originated from a source is sent, then the SGR 1353 status storage for that source SHALL be cleared. The display 1354 count SHALL NOT be increased. 1356 BS (U+0008) Back Space, intended to erase the last entered character 1357 by a source. Erasure by backspace cannot always be performed as 1358 the erasing party intended. If an erasing action erases all text 1359 up to the end of the leading label after a source switch, then the 1360 mixer MUST NOT transmit more backspaces. Instead it is 1361 RECOMMENDED that a letter "X" is inserted in the text stream for 1362 each backspace as an indication of the intent to erase more. A 1363 new line is usually coded by a Line Separator, but the character 1364 combination "CRLF" MAY be used instead. Erasure of a new line is 1365 in both cases done by just one erasing action (Backspace). If the 1366 display count has a positive value it SHALL be decreased by one 1367 when the BS is sent. If the display count is at zero, it SHALL 1368 NOT not altered. 1370 4.2.5. Packet transmission 1372 A mixer transmitting to a multi-party unaware terminal SHALL send 1373 primary data only from one source per packet. The SSRC SHALL be the 1374 SSRC of the mixer. The CSRC list SHALL contain one member and be the 1375 SSRC of the source of the primary data. 1377 4.2.6. Functional limitations 1379 When a multi-party unaware endpoint presents a conversation in one 1380 display area in a chat style, it inserts source indications for 1381 remote text and local user text as they are merged in completed text 1382 groups. When an endpoint using this layout receives and presents 1383 text mixed for multi-party unaware endpoints, there will be two 1384 levels of source indicators for the received text; one generated by 1385 the mixer and inserted in a label after each source switch, and 1386 another generated by the receiving endpoint and inserted after each 1387 switch between local and remote source in the presentation area. 1388 This will waste display space and look inconsistent to the reader. 1390 New text can be presented only from one source at a time. Switch of 1391 source to be presented takes place at suitable places in the text, 1392 such as end of phrase, end of sentence, line separator and 1393 inactivity. Therefore the time to switch to present waiting text 1394 from other sources may become long and will vary and depend on the 1395 actions of the currently presented source. 1397 Erasure can only be done up to the latest source switch. If a user 1398 tries to erase more text, the erasing actions will be presented as 1399 letter X after the label. 1401 Text loss because of network errors may hit the label between entries 1402 from different parties, causing risk for misunderstanding from which 1403 source a piece of text is. 1405 These facts make it strongly RECOMMENDED to implement multi-party 1406 awareness in RTT endpoints. The use of the mixing method for multi- 1407 party-unaware endpoints should be left for use with endpoints which 1408 are impossible to upgrade to become multi-party aware. 1410 4.2.7. Example views of presentation on multi-party unaware endpoints 1412 The following pictures are examples of the view on a participant's 1413 display for the multi-party-unaware case. 1415 _________________________________________________ 1416 | Conference | Alice | 1417 |________________________|_________________________| 1418 | |I will arrive by TGV. | 1419 |[Bob]:My flight is to |Convenient to the main | 1420 |Orly. |station. | 1421 |[Eve]:Hi all, can we | | 1422 |plan for the seminar. | | 1423 | | | 1424 |[Bob]:Eve, will you do | | 1425 |your presentation on | | 1426 |Friday? | | 1427 |[Eve]:Yes, Friday at 10.| | 1428 |[Bob]: Fine, wo |We need to meet befo | 1429 |________________________|_________________________| 1431 Figure 5: Alice who has a conference-unaware client is receiving the 1432 multi-party real-time text in a single-stream. 1434 This figure shows how a coordinated column view MAY be presented on 1435 Alice's device in a view with two-columns. The mixer inserts labels 1436 to show how the sources alternate in the column with received text. 1437 The mixer alternates between the sources at suitable points in the 1438 text exchange so that text entries from each party can be 1439 conveniently read. 1441 _________________________________________________ 1442 | |^| 1443 |(Alice) Hi, Alice here. |-| 1444 | | | 1445 |(mix)[Bob)] Bob as well. | | 1446 | | | 1447 |[Eve] Hi, this is Eve, calling from Paris | | 1448 | I thought you should be here. | | 1449 | | | 1450 |(Alice) I am coming on Thursday, my | | 1451 | performance is not until Friday morning.| | 1452 | | | 1453 |(mix)[Bob] And I on Wednesday evening. | | 1454 | | | 1455 |[Eve] we can have dinner and then walk | | 1456 | | | 1457 |[Eve] But I need to be back to | | 1458 | the hotel by 11 because I need | | 1459 | |-| 1460 |______________________________________________|v| 1461 | of course, I underst | 1462 |________________________________________________| 1464 Figure 6: An example of a view of the multi-party unaware 1465 presentation in chat style. Alice is the local user. 1467 In this view, there is a tradition in receiving applications to 1468 include a label showing the source of the text, here shown with 1469 parenthesis "()". The mixer also inserts source labels for the 1470 multi-party call participants, here shown with brackets "[]". 1472 5. Relation to Conference Control 1474 5.1. Use with SIP centralized conferencing framework 1476 The SIP conferencing framework, mainly specified in [RFC4353], 1477 [RFC4579] and [RFC4575] is suitable for coordinating sessions 1478 including multi-party RTT. The RTT stream between the mixer and a 1479 participant is one and the same during the conference. Participants 1480 get announced by notifications when participants are joining or 1481 leaving, and further user information may be provided. The SSRC of 1482 the text to expect from joined users MAY be included in a 1483 notification. The notifications MAY be used both for security 1484 purposes and for translation to a label for presentation to other 1485 users. 1487 5.2. Conference control 1489 In managed conferences, control of the real-time text media SHOULD be 1490 provided in the same way as other for media, e.g. for muting and 1491 unmuting by the direction attributes in SDP [RFC8866]. 1493 Note that floor control functions may be of value for RTT users as 1494 well as for users of other media in a conference. 1496 6. Gateway Considerations 1498 6.1. Gateway considerations with Textphones (e.g. TTYs). 1500 Multi-party RTT sessions may involve gateways of different kinds. 1501 Gateways involved in setting up sessions SHALL correctly reflect the 1502 multi-party capability or unawareness of the combination of the 1503 gateway and the remote endpoint beyond the gateway. 1505 One case that may occur is a gateway to PSTN for communication with 1506 textphones (e.g. TTYs). Textphones are limited devices with no 1507 multi-party awareness, and it SHOULD therefore be suitable for the 1508 gateway to not indicate multi-party awareness for that case. Another 1509 solution is that the gateway indicates multi-party capability towards 1510 the mixer, and includes the multi-party mixer function for multi- 1511 party unaware endpoints itself. This solution makes it possible to 1512 make adaptations for the functional limitations of the textphone 1513 (TTY). 1515 More information on gateways to textphones (TTYs) is found in 1516 [RFC5194] 1518 6.2. Gateway considerations with WebRTC. 1520 Gateway operation to real-time text in WebRTC may also be required. 1521 In WebRTC, RTT is specified in [RFC8865]. 1523 A multi-party bridge may have functionality for communicating by RTT 1524 both in RTP streams with RTT and WebRTC T.140 data channels. Other 1525 configurations may consist of a multi-party bridge with either 1526 technology for RTT transport and a separate gateway for conversion of 1527 the text communication streams between RTP and T.140 data channel. 1529 In WebRTC, it is assumed that for a multi-party session, one T.140 1530 data channel is established for each source from a gateway or bridge 1531 to each participant. Each participant also has a data channel with 1532 two-way connection with the gateway or bridge. 1534 The t140 channel used both ways is for text from the WebRTC user and 1535 from the bridge or gateway itself to the WebRTC user. The label 1536 parameter of this t140 channel is used as NAME field in RTCP to 1537 participants on the RTP side. The other t140 channels are only for 1538 text from other participants to the WebRTC user. 1540 When a new participant has entered the session with RTP transport of 1541 RTT, a new T.140 channel SHOULD be established to WebRTC users with 1542 the label parameter composed from the NAME field in RTCP on the RTP 1543 side. 1545 When a new participant has entered the multi-party session with RTT 1546 transport in a WebRTC T.140 data channel, the new participant SHOULD 1547 be announced by a notification to RTP users. The label parameter 1548 from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP 1549 side, or other available session information. 1551 When a participant on the RTP side disappears, the corresponding 1552 T.140 data channel(s) SHOULD be closed. 1554 When a WebRTC user of T.140 data channels disconnects from the mixer, 1555 the corresponding RTP streams or sources in an RTP-mixed stream 1556 SHOULD be closed. 1558 T.140 data channels MAY be opened and closed by negotiation or 1559 renegotiation of the session or by any other valid means as specified 1560 in section 1 of [RFC8865]. 1562 7. Updates to RFC 4103 1564 This document updates [RFC4103] by introducing an SDP media attribute 1565 "rtt-mixer" for negotiation of multi-party mixing capability with the 1566 [RFC4103] format, and by specifying the rules for packets when multi- 1567 party capability is negotiated and in use. 1569 8. Congestion considerations 1571 The congestion considerations and recommended actions from [RFC4103] 1572 are valid also in multi-party situations. 1574 The first action in case of congestion SHALL be to temporarily 1575 increase the transmission interval up to two seconds. 1577 If the very unlikely situation appears that many participants in a 1578 conference send text simultaneously for a long period, a delay may 1579 build up for presentation of text at the receivers if the limitation 1580 in characters per second("CPS") to be transmitted to the participants 1581 is exceeded. More delay than 7 seconds can cause confusion in the 1582 session. It is therefore RECOMMENDED that an RTP-mixer based mixer 1583 discards such text in excess and inserts a general indication of 1584 possible text loss [T140ad1] in the session. If the main text 1585 contributor is indicated in any way, the mixer MAY avoid deleting 1586 text from that participant. It should however be noted that human 1587 creation of text normally contains pauses, when the transmission can 1588 catch up, so that the transmission overload situations are expected 1589 to be very rare. 1591 9. Acknowledgements 1593 James Hamlin for format and performance aspects. 1595 10. IANA Considerations 1597 10.1. Registration of the "rtt-mixer" SDP media attribute 1599 [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the 1600 RFC number of this document.] 1601 IANA is asked to register the new SDP attribute "rtt-mixer". 1603 Contact name: IESG 1605 Contact email: iesg@ietf.org 1607 Attribute name: rtt-mixer 1609 Attribute semantics: See RFCXXXX Section 2.3 1611 Attribute value: none 1613 Usage level: media 1615 Purpose: Indicate support by mixer and endpoint of multi-party 1616 mixing for real-time text transmission, using a common RTP-stream 1617 for transmission of text from a number of sources mixed with one 1618 source at a time and the source indicated in a single CSRC-list 1619 member. 1621 Charset Dependent: no 1623 O/A procedure: See RFCXXXX Section 2.3 1625 Mux Category: normal 1627 Reference: RFCXXXX 1629 11. Security Considerations 1631 The RTP-mixer model requires the mixer to be allowed to decrypt, pack 1632 and encrypt secured text from the conference participants. Therefore 1633 the mixer needs to be trusted. This is similar to the situation for 1634 central mixers of audio and video. 1636 The requirement to transfer information about the user in RTCP 1637 reports in SDES, CNAME and NAME fields, and in conference 1638 notifications, for creation of labels may have privacy concerns as 1639 already stated in RFC 3550 [RFC3550], and may be restricted for 1640 privacy reasons. The receiving user will then get a more symbolic 1641 label for the source. 1643 Participants with malicious intentions may appear and e.g. disturb 1644 the multi-party session by a continuous flow of text, or masquerade 1645 as text from other participants. Counteractions should be to require 1646 secure signaling, media and authentication, and to provide higher 1647 level conference functions e.g. for blocking and expelling 1648 participants. 1650 Further security considerations specific for this application are 1651 specified in section Section 3.19. 1653 12. Change history 1655 [RFC Editor: Please remove this section prior to publication.] 1657 12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-15 1659 Actions on review comments from Jurgen Schonwalder: 1661 A bit more about congestion situations and that they are expected to 1662 be very rare. 1664 Explanation of differences in security between the conference-aware 1665 and the conference-unaware case added in security section. 1667 Presentation examples with suource labels made less confusing, and 1668 explained. 1670 Reference to T.140 inserted at first mentioning of T.140. 1672 Reference to RFC 8825 inserted to explain WebRTC 1674 Nit in wording in terminology section adjusted. 1676 12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-14 1678 Changes from comments by Murray Cucherawy during AD review. 1680 Many SHOULD in section 4.2 on multi-party unaware mixing changed to 1681 SHALL, and the whole section instead specified to be optional 1682 depending on the application. 1684 Some SHOULD in section 3 either explained or changed to SHALL. 1686 In order to have explainable conditions behind SHOULDs, the 1687 transmission interval in 3.4 is changed to as soon as text is 1688 available as a main principle. The call participants send with 300 1689 ms interval so that will create realistic load conditions anyway. 1691 12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-13 1693 Changed year to 2021. 1695 Changed reference to draft on RTT in WebRTC to recently published RFC 1696 8865. 1698 Changed label brackets in example from "[]" to "()" to avoid nits 1699 comment. 1701 Changed reference "RFC 4566" to recently published "RFC 8866" 1703 12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-12 1705 Changes according to responses on comments from Brian Rosen in 1706 Avtcore list on 2020-12-05 and -06. 1708 Changes according to responses to comments by Bernard Aboba in 1709 avtcore list 2020-12-06. 1711 Introduction of an optiona RTP multi-stream mixing method for further 1712 study as proposed by Bernard Aboba. 1714 Changes clarifying how to open and close T.140 data channels included 1715 in 6.2 after comments by Lorenzo Miniero. 1717 Changes to satisfy nits check. Some "not" changed to "NOT" in 1718 normative wording combinations. Some lower case normative words 1719 changed to upper case. A normative reference deleted from the 1720 abstract. Two informative documents moved from normative references 1721 to informative references. 1723 12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11 1725 Timestamps and timestamp offsets added to the packet examples in 1726 section 3.23, and the description corrected. 1728 A number of minor corrections added in sections 3.10 - 3.23. 1730 12.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10 1732 The packet composition was modified for interleaving packets from 1733 different sources. 1735 The packet reception was modified for the new interleaving method. 1737 The packet sequence examples was adjusted for the new interleaving 1738 method. 1740 Modifications according to responses to Brian Rosen of 2020-11-03 1742 12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09 1744 Changed name on the SDP media attribute to "rtt-mixer" 1745 Restructure of section 2 for balance between aware and unaware cases. 1747 Moved conference control to own section. 1749 Improved clarification of recovery and loss in the packet sequence 1750 example. 1752 A number of editorial corrections and improvements. 1754 12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 1756 Deleted the method requiring a new packet format "text/rex" because 1757 of the longer standardization and implementation period it needs. 1759 Focus on use of RFC 4103 text/red format with shorter transmission 1760 interval, and source indicated in CSRC. 1762 12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 1764 Added a method based on the "text/red" format and single source per 1765 packet, negotiated by the "rtt-mixer" SDP attribute. 1767 Added reasoning and recommendation about indication of loss. 1769 The highest number of sources in one packet is 15, not 16. Changed. 1771 Added in information on update to RFC 4103 that RFC 4103 explicitly 1772 allows addition of FEC method. The redundancy is a kind of forward 1773 error correction.. 1775 12.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 1777 Improved definitions list format. 1779 The format of the media subtype parameters is made to match the 1780 requirements. 1782 The mapping of media subtype parameters to SDP is included. 1784 The "CPS" parameter belongs to the t140 subtype and does not need to 1785 be registered here. 1787 12.11. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 1789 nomenclature and editorial improvements 1791 "this document" used consistently to refer to this document. 1793 12.12. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 1795 'Redundancy header' renamed to 'data header'. 1797 More clarifications added. 1799 Language and figure number corrections. 1801 12.13. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 1803 Mention possible need to mute and raise hands as for other media. 1804 ---done ---- 1806 Make sure that use in two-party calls is also possible and explained. 1807 - may need more wording - 1809 Clarify the RTT is often used together with other media. --done-- 1811 Tell that text mixing is N-1. A users own text is not received in 1812 the mix. -done- 1814 In 3. correct the interval to: A "text/rex" transmitter SHOULD send 1815 packets distributed in time as long as there is something (new or 1816 redundant T140blocks) to transmit. The maximum transmission interval 1817 SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 1818 receiver as soon as new text to that receiver is available, as long 1819 as the time after the latest sent packet to the same receiver is more 1820 than 150 ms, and also the maximum character rate to the receiver is 1821 not exceeded. The intention is to keep the latency low while keeping 1822 a good protection against text loss in bursty packet loss conditions. 1823 -done- 1825 In 1.3 say that the format is used both ways. -done- 1827 In 13.1 change presentation area to presentation field so that reader 1828 does not think it shall be totally separated. -done- 1830 In Performance and intro, tell the performance in number of 1831 simultaneous sending users and introduced delay 16, 150 vs 1832 requirements 5 vs 500. -done -- 1834 Clarify redundancy level per connection. -done- 1836 Timestamp also for the last data header. To make it possible for all 1837 text to have time offset as for transmission from the source. Make 1838 that header equal to the others. -done- 1840 Mixer always use the CSRC list, even for its own BOM. -done- 1841 Combine all talk about transmission interval (300 ms vs when text has 1842 arrived) in section 3 in one paragraph or close to each other. -done- 1844 Documents the goal of good performance with low delay for 5 1845 simultaneous typers in the introduction. -done- 1847 Describe better that only primary text shall be sent on to receivers. 1848 Redundancy and loss must be resolved by the mixer. -done- 1850 12.14. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 1852 SDP and better description and visibility of security by OSRTP RFC 1853 8634 needed. 1855 The description of gatewaying to WebRTC extended. 1857 The description of the data header in the packet is improved. 1859 12.15. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 1861 2,5,6 More efficient format "text/rex" introduced and attribute 1862 a=rtt-mix deleted. 1864 3. Brief about use of OSRTP for security included- More needed. 1866 4. Brief motivation for the solution and why not rtp-translator is 1867 used added to intro. 1869 7. More limitations for the multi-party unaware mixing method 1870 inserted. 1872 8. Updates to RFC 4102 and 4103 more clearly expressed. 1874 9. Gateway to WebRTC started. More needed. 1876 12.16. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 1877 to draft-ietf-avtcore-multi-party-rtt-mix-00 1879 Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00 1881 Replaced CDATA in IANA registration table with better coding. 1883 Converted to xml2rfc version 3. 1885 12.17. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 1886 to -03 1888 Changed company and e-mail of the author. 1890 Changed title to "RTP-mixer formatting of multi-party Real-time text" 1891 to better match contents. 1893 Check and modification where needed of use of RFC 2119 words SHALL 1894 etc. 1896 More about the CC value in sections on transmitters and receivers so 1897 that 1-to-1 sessions do not use the mixer format. 1899 Enhanced section on presentation for multi-party-unaware endpoints 1901 A paragraph recommending CPS=150 inserted in the performance section. 1903 12.18. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 1904 to -02 1906 In Abstract and 1. Introduction: Introduced wording about regulatory 1907 requirements. 1909 In section 5: The transmission interval is decreased to 100 ms when 1910 there is text from more than one source to transmit. 1912 In section 11 about SDP negotiation, a SHOULD-requirement is 1913 introduced that the mixer should make a mix for multi-party unaware 1914 endpoints if the negotiation is not successful. And a reference to a 1915 later chapter about it. 1917 The presentation considerations chapter 14 is extended with more 1918 information about presentation on multi-party aware endpoints, and a 1919 new section on the multi-party unaware mixing with low functionality 1920 but SHOULD a be implemented in mixers. Presentation examples are 1921 added. 1923 A short chapter 15 on gateway considerations is introduced. 1925 Clarification about the text/t140 format included in chapter 10. 1927 This sentence added to the chapter 10 about use without redundancy. 1928 "The text/red format SHOULD be used unless some other protection 1929 against packet loss is utilized, for example a reliable network or 1930 transport." 1932 Note about deviation from RFC 2198 added in chapter 4. 1934 In chapter 9. "Use with SIP centralized conferencing framework" the 1935 following note is inserted: Note: The CSRC-list in an RTP packet only 1936 includes participants who's text is included in one or more text 1937 blocks. It is not the same as the list of participants in a 1938 conference. With audio and video media, the CSRC-list would often 1939 contain all participants who are not muted whereas text participants 1940 that don't type are completely silent and so don't show up in RTP 1941 packet CSRC-lists. 1943 12.19. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 1944 to -01 1946 Editorial cleanup. 1948 Changed capability indication from fmtp-parameter to SDP attribute 1949 "rtt-mix". 1951 Swapped order of redundancy elements in the example to match reality. 1953 Increased the SDP negotiation section 1955 13. References 1957 13.1. Normative References 1959 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1960 Requirement Levels", BCP 14, RFC 2119, 1961 DOI 10.17487/RFC2119, March 1997, 1962 . 1964 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1965 Jacobson, "RTP: A Transport Protocol for Real-Time 1966 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1967 July 2003, . 1969 [RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type", 1970 RFC 4102, DOI 10.17487/RFC4102, June 2005, 1971 . 1973 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 1974 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 1975 . 1977 [RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session 1978 Initiation Protocol (SIP)", RFC 5630, 1979 DOI 10.17487/RFC5630, October 2009, 1980 . 1982 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1983 Security (DTLS) Extension to Establish Keys for the Secure 1984 Real-time Transport Protocol (SRTP)", RFC 5764, 1985 DOI 10.17487/RFC5764, May 2010, 1986 . 1988 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 1989 Keeping Alive the NAT Mappings Associated with RTP / RTP 1990 Control Protocol (RTCP) Flows", RFC 6263, 1991 DOI 10.17487/RFC6263, June 2011, 1992 . 1994 [RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M. 1995 Thomson, "Session Traversal Utilities for NAT (STUN) Usage 1996 for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675, 1997 October 2015, . 1999 [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2000 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, 2001 May 2017, . 2003 [RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for 2004 Browser-Based Applications", RFC 8825, 2005 DOI 10.17487/RFC8825, January 2021, 2006 . 2008 [RFC8865] Holmberg, C. and G. Hellström, "T.140 Real-Time Text 2009 Conversation over WebRTC Data Channels", RFC 8865, 2010 DOI 10.17487/RFC8865, January 2021, 2011 . 2013 [RFC8866] Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP: 2014 Session Description Protocol", RFC 8866, 2015 DOI 10.17487/RFC8866, January 2021, 2016 . 2018 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 2019 multimedia application text conversation", February 1998, 2020 . 2022 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 2023 Protocol for multimedia application text conversation", 2024 February 2000, 2025 . 2027 13.2. Informative References 2029 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 2030 Session Initiation Protocol (SIP)", RFC 4353, 2031 DOI 10.17487/RFC4353, February 2006, 2032 . 2034 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 2035 Session Initiation Protocol (SIP) Event Package for 2036 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 2037 2006, . 2039 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 2040 (SIP) Call Control - Conferencing for User Agents", 2041 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 2042 . 2044 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 2045 Time Text over IP Using the Session Initiation Protocol 2046 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 2047 . 2049 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 2050 DOI 10.17487/RFC7667, November 2015, 2051 . 2053 [RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. 2054 Stach, "An Opportunistic Approach for Secure Real-time 2055 Transport Protocol (OSRTP)", RFC 8643, 2056 DOI 10.17487/RFC8643, August 2019, 2057 . 2059 Author's Address 2061 Gunnar Hellstrom 2062 Gunnar Hellstrom Accessible Communication 2063 SE-13670 Vendelso 2064 Sweden 2066 Email: gunnar.hellstrom@ghaccess.se