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(See the Legal Provisions document at https://trustee.ietf.org/license-info for more information.) -- The document date (26 May 2021) is 1038 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: 'CC' is mentioned on line 273, but not defined == Missing Reference: 'Bob' is mentioned on line 1566, but not defined -- Possible downref: Non-RFC (?) normative reference: ref. 'T140' -- Possible downref: Non-RFC (?) normative reference: ref. 'T140ad1' Summary: 0 errors (**), 0 flaws (~~), 5 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) 26 May 2021 5 Intended status: Standards Track 6 Expires: 27 November 2021 8 RTP-mixer formatting of multiparty Real-time text 9 draft-ietf-avtcore-multi-party-rtt-mix-20 11 Abstract 13 This document provides enhancements for RFC 4103 real-time text 14 mixing 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 multiparty real-time text 19 session. The specified mechanism builds on the standard use of the 20 Contributing Source (CSRC) list in the Realtime Protocol (RTP) packet 21 for source identification. The method makes use of the same "text/ 22 t140" and "text/red" formats as for two-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 multiparty-coded real-time 32 text stream using the RTP-mixer method. The capability is indicated 33 by use of an RFC 8866 Session Description Protocol (SDP) media 34 attribute "rtt-mixer". 36 The document updates RFC 4103 "RTP Payload for Text Conversation". 38 A specification of how a mixer can format text for the case when the 39 endpoint is not multiparty-aware is also provided. 41 Status of This Memo 43 This Internet-Draft is submitted in full conformance with the 44 provisions of BCP 78 and BCP 79. 46 Internet-Drafts are working documents of the Internet Engineering 47 Task Force (IETF). Note that other groups may also distribute 48 working documents as Internet-Drafts. The list of current Internet- 49 Drafts is at https://datatracker.ietf.org/drafts/current/. 51 Internet-Drafts are draft documents valid for a maximum of six months 52 and may be updated, replaced, or obsoleted by other documents at any 53 time. It is inappropriate to use Internet-Drafts as reference 54 material or to cite them other than as "work in progress." 56 This Internet-Draft will expire on 27 November 2021. 58 Copyright Notice 60 Copyright (c) 2021 IETF Trust and the persons identified as the 61 document authors. All rights reserved. 63 This document is subject to BCP 78 and the IETF Trust's Legal 64 Provisions Relating to IETF Documents (https://trustee.ietf.org/ 65 license-info) in effect on the date of publication of this document. 66 Please review these documents carefully, as they describe your rights 67 and restrictions with respect to this document. Code Components 68 extracted from this document must include Simplified BSD License text 69 as described in Section 4.e of the Trust Legal Provisions and are 70 provided without warranty as described in the Simplified BSD License. 72 Table of Contents 74 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 75 1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 76 1.2. Selected solution and considered alternatives . . . . . . 7 77 1.3. Intended application . . . . . . . . . . . . . . . . . . 9 78 2. Overview of the two specified solutions and selection of 79 method . . . . . . . . . . . . . . . . . . . . . . . . . 10 80 2.1. The RTP-mixer-based solution for multiparty-aware 81 endpoints . . . . . . . . . . . . . . . . . . . . . . . . 10 82 2.2. Mixing for multiparty-unaware endpoints . . . . . . . . . 11 83 2.3. Offer/answer considerations . . . . . . . . . . . . . . . 11 84 2.4. Actions depending on capability negotiation result . . . 13 85 3. Details for the RTP-mixer-based mixing method for 86 multiparty-aware endpoints . . . . . . . . . . . . . . . 13 87 3.1. Use of fields in the RTP packets . . . . . . . . . . . . 13 88 3.2. Initial transmission of a BOM character . . . . . . . . . 14 89 3.3. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 14 90 3.4. Transmission interval . . . . . . . . . . . . . . . . . . 14 91 3.5. Only one source per packet . . . . . . . . . . . . . . . 15 92 3.6. Do not send received text to the originating source . . . 15 93 3.7. Clean incoming text . . . . . . . . . . . . . . . . . . . 16 94 3.8. Redundant transmission principles . . . . . . . . . . . . 16 95 3.9. Text placement in packets . . . . . . . . . . . . . . . . 16 96 3.10. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 17 97 3.11. Creation of the redundancy . . . . . . . . . . . . . . . 17 98 3.12. Timer offset fields . . . . . . . . . . . . . . . . . . . 18 99 3.13. Other RTP header fields . . . . . . . . . . . . . . . . . 18 100 3.14. Pause in transmission . . . . . . . . . . . . . . . . . . 18 101 3.15. RTCP considerations . . . . . . . . . . . . . . . . . . . 19 102 3.16. Reception of multiparty contents . . . . . . . . . . . . 19 103 3.17. Performance considerations . . . . . . . . . . . . . . . 21 104 3.18. Security for session control and media . . . . . . . . . 21 105 3.19. SDP offer/answer examples . . . . . . . . . . . . . . . . 22 106 3.20. Packet sequence example from interleaved transmission . . 23 107 3.21. Maximum character rate "cps" . . . . . . . . . . . . . . 26 108 4. Presentation level considerations . . . . . . . . . . . . . . 26 109 4.1. Presentation by multiparty-aware endpoints . . . . . . . 27 110 4.2. Multiparty mixing for multiparty-unaware endpoints . . . 29 111 5. Relation to Conference Control . . . . . . . . . . . . . . . 35 112 5.1. Use with SIP centralized conferencing framework . . . . . 36 113 5.2. Conference control . . . . . . . . . . . . . . . . . . . 36 114 6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 36 115 6.1. Gateway considerations with Textphones . . . . . . . . . 36 116 6.2. Gateway considerations with WebRTC . . . . . . . . . . . 36 117 7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 37 118 8. Congestion considerations . . . . . . . . . . . . . . . . . . 38 119 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 38 120 9.1. Registration of the "rtt-mixer" SDP media attribute . . . 38 121 10. Security Considerations . . . . . . . . . . . . . . . . . . . 39 122 11. Change history . . . . . . . . . . . . . . . . . . . . . . . 40 123 11.1. Changes included in 124 draft-ietf-avtcore-multi-party-rtt-mix-20 . . . . . . . 40 125 11.2. Changes included in 126 draft-ietf-avtcore-multi-party-rtt-mix-19 . . . . . . . 40 127 11.3. Changes included in 128 draft-ietf-avtcore-multi-party-rtt-mix-18 . . . . . . . 40 129 11.4. Changes included in 130 draft-ietf-avtcore-multi-party-rtt-mix-17 . . . . . . . 40 131 11.5. Changes included in 132 draft-ietf-avtcore-multi-party-rtt-mix-16 . . . . . . . 40 133 11.6. Changes included in 134 draft-ietf-avtcore-multi-party-rtt-mix-15 . . . . . . . 41 135 11.7. Changes included in 136 draft-ietf-avtcore-multi-party-rtt-mix-14 . . . . . . . 41 137 11.8. Changes included in 138 draft-ietf-avtcore-multi-party-rtt-mix-13 . . . . . . . 41 139 11.9. Changes included in 140 draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . . 42 142 11.10. Changes included in 143 draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 42 144 11.11. Changes included in 145 draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 42 146 11.12. Changes included in 147 draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 42 148 11.13. Changes included in 149 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 43 150 11.14. Changes included in 151 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 43 152 11.15. Changes included in 153 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 43 154 11.16. Changes included in 155 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 43 156 11.17. Changes included in 157 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 43 158 11.18. Changes included in 159 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 44 160 11.19. Changes included in 161 draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 45 162 11.20. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 45 163 11.21. Changes from 164 draft-hellstrom-avtcore-multi-party-rtt-source-03 to 165 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 45 166 11.22. Changes from 167 draft-hellstrom-avtcore-multi-party-rtt-source-02 to 168 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 45 169 11.23. Changes from 170 draft-hellstrom-avtcore-multi-party-rtt-source-01 to 171 -02 . . . . . . . . . . . . . . . . . . . . . . . . . . 46 172 11.24. Changes from 173 draft-hellstrom-avtcore-multi-party-rtt-source-00 to 174 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 47 175 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 47 176 12.1. Normative References . . . . . . . . . . . . . . . . . . 47 177 12.2. Informative References . . . . . . . . . . . . . . . . . 48 178 Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 49 179 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 49 181 1. Introduction 183 "RTP Payload for Text Conversation" [RFC4103] specifies use of the 184 Real-Time Transport Protocol (RTP) [RFC3550] for transmission of 185 real-time text (RTT) and the "text/t140" format. It also specifies a 186 redundancy format "text/red" for increased robustness. The "text/ 187 red" format is registered in [RFC4102]. 189 Real-time text is usually provided together with audio and sometimes 190 with video in conversational sessions. 192 A requirement related to multiparty sessions from the presentation 193 level standard T.140 [T140] for real-time text is: "The display of 194 text from the members of the conversation should be arranged so that 195 the text from each participant is clearly readable, and its source 196 and the relative timing of entered text is visualized in the 197 display." 199 Another requirement is that the mixing procedure must not introduce 200 delays in the text streams that are experienced to be disturbing the 201 real-time experience of the receiving users. 203 Use of RTT is increasing, and specifically, use in emergency calls is 204 increasing. Emergency call use requires multiparty mixing because it 205 is common that one agent needs to transfer the call to another 206 specialized agent but is obliged to stay on the call at least to 207 verify that the transfer was successful. Mixer implementations for 208 RFC 4103 "RTP Payload for Text Conversation" can use traditional RFC 209 3550 RTP functions for mixing and source identification, but the 210 performance of the mixer when giving turns for the different sources 211 to transmit is limited when using the default transmission 212 characteristics with redundancy. 214 The redundancy scheme of [RFC4103] enables efficient transmission of 215 earlier transmitted redundant text in packets together with new text. 216 However, the redundancy header format has no source indicators for 217 the redundant transmissions. The redundant parts in a packet must 218 therefore be from the same source as the new text. The recommended 219 transmission is one new and two redundant generations of text 220 (T140blocks) in each packet and the recommended transmission interval 221 for two-party use is 300 ms. 223 Real-time text mixers for multiparty sessions need to include the 224 source with each transmitted group of text from a conference 225 participant so that the text can be transmitted interleaved with text 226 groups from different sources at the rate they are created. This 227 enables the text groups to be presented by endpoints in suitable 228 grouping with other text from the same source. 230 The presentation can then be arranged so that text from different 231 sources can be presented in real-time and easily read. At the same 232 time it is possible for a reading user to perceive approximately when 233 the text was created in real time by the different parties. The 234 transmission and mixing is intended to be done in a general way, so 235 that presentation can be arranged in a layout decided by the 236 endpoint. 238 There are existing implementations of RFC 4103 in endpoints without 239 the updates from this document. These will not be able to receive 240 and present real-time text mixed for multiparty-aware endpoints. 242 A negotiation mechanism is therefore needed for verification if the 243 parties are able to handle a common method for multiparty 244 transmission and agreeing on using that method. 246 A fallback mixing procedure is also needed for cases when the 247 negotiation result indicates that a receiving endpoint is not capable 248 of handling the mixed format. Multiparty-unaware endpoints would 249 possibly otherwise present all received multiparty mixed text as if 250 it came from the same source regardless of any accompanying source 251 indication coded in fields in the packet. Or they may have other 252 undesirable ways of acting on the multiparty content. The fallback 253 method is called the mixing procedure for multiparty-unaware 254 endpoints. The fallback method is naturally not expected to meet all 255 performance requirements placed on the mixing procedure for 256 multiparty-aware endpoints. 258 The document updates [RFC4103] by introducing an attribute for 259 declaring support of the RTP-mixer-based multiparty mixing case and 260 rules for source indications and interleaving of text from different 261 sources. 263 1.1. Terminology 265 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 266 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 267 "OPTIONAL" in this document are to be interpreted as described in BCP 268 14 [RFC2119] [RFC8174] when, and only when, they appear in all 269 capitals, as shown above. 271 The terms Source Description (SDES), Canonical name (CNAME), Name 272 (NAME), Synchronization Source (SSRC), Contributing Source (CSRC), 273 CSRC list, CSRC count [CC], Real-Time control protocol (RTCP), RTP- 274 mixer, RTP-translator are defined in [RFC3550]. 276 The term "T140block" is defined in [RFC4103] to contain one or more 277 T.140 code elements. 279 "TTY" stands for a textphone type used in North America. 281 Web based real-time communication (WebRTC) is specified by the World 282 Wide Web Consortium (W3C) and IETF. See [RFC8825]. 284 "DTLS-SRTP" is a Datagram Transport Layer Security (DTLS) extension 285 for use with Secure Real-Time Transport Protocol/Secure Real-Time 286 Control Protocol (SRTP/SRTCP) specified in [RFC5764]. 288 "multiparty-aware" describes an endpoint receiving real-time text 289 from multiple sources through a common conference mixer being able to 290 present the text in real-time, separated by source, and presented so 291 that a user can get an impression of the approximate relative timing 292 of text from different parties. 294 "multiparty-unaware" describes an endpoint not itself being able to 295 separate text from different sources when received through a common 296 conference mixer. 298 1.2. Selected solution and considered alternatives 300 A number of alternatives were considered when searching an efficient 301 and easily implemented multiparty method for real-time text. This 302 section explains a few of them briefly. 304 Multiple RTP streams, one per participant 305 One RTP stream per source would be sent in the same RTP session 306 with the "text/red" format. From some points of view, use of 307 multiple RTP streams, one for each source, sent in the same RTP 308 session would be efficient, and would use exactly the same packet 309 format as [RFC4103] and the same payload type. A couple of 310 relevant scenarios using multiple RTP-streams are specified in 311 "RTP Topologies" [RFC7667]. One possibility of special interest 312 is the Selective Forwarding Middlebox (SFM) topology specified in 313 RFC 7667 section 3.7 that could enable end-to-end encryption. In 314 contrast to audio and video, real-time text is only transmitted 315 when the users actually transmit information. Thus, an SFM 316 solution would not need to exclude any party from transmission 317 under normal conditions. In order to allow the mixer to convey 318 the packets with the payload preserved and encrypted, an SFM 319 solution would need to act on some specific characteristics of the 320 "text/red" format. The redundancy headers are part of the 321 payload, so the receiver would need to just assume that the 322 payload type number in the redundancy header is for "text/t140". 323 The characters per second parameter (cps) would need to act per 324 stream. The relation between the SSRC and the source would need 325 to be conveyed in some specified way, e.g., in the CSRC. Recovery 326 and loss detection would preferably be based on sequence number 327 gap detection. Thus, sequence number gaps in the incoming stream 328 to the mixer would need to be reflected in the stream to the 329 participant, with no new gaps created by the mixer. However, the 330 RTP implementation in both mixers and endpoints need to support 331 multiple streams in the same RTP session in order to use this 332 mechanism. For best deployment opportunity, it should be possible 333 to upgrade existing endpoint solutions to be multiparty-aware with 334 a reasonable effort. There is currently a lack of support for 335 multi-stream RTP in certain implementations. This fact led to 336 this solution being only briefly mentioned in this document as an 337 option for further study. 339 RTP-mixer-based method for multiparty-aware endpoints 340 The "text/red" format in RFC 4103 is sent with a shorter 341 transmission interval with the RTP-mixer method and indicating the 342 source in the CSRC field. The "text/red" format with a "text/ 343 t140" payload in a single RTP stream can be sent when text is 344 available from the call participants instead of at the regular 300 345 ms. Transmission of packets with text from different sources can 346 then be done smoothly while simultaneous transmission occurs as 347 long as it is not limited by the maximum character rate "cps". 348 With ten participants sending text simultaneously, the switching 349 and transmission performance is good. With more simultaneously 350 sending participants, and with receivers having the default 351 capacity there will be a noticeable jerkiness and delay in text 352 presentation. The jerkiness will be more expressed the more 353 participants who send text simultaneously. Two seconds jerkiness 354 will be noticeable and slightly unpleasant, but it corresponds in 355 time to what typing humans often cause by hesitation or changing 356 position while typing. A benefit of this method is that no new 357 packet format needs to be introduced and implemented. Since 358 simultaneous typing by more than two parties is expected to be 359 very rare as described in Section 1.3, this method can be used 360 successfully with good performance. Recovery of text in case of 361 packet loss is based on analysis of timestamps of received 362 redundancy versus earlier received text. Negotiation is based on 363 a new SDP media attribute "rtt-mixer". This method is selected to 364 be the main one specified in this document. 366 Multiple sources per packet 367 A new "text" media subtype would be specified with up to 15 368 sources in each packet. The mechanism would make use of the RTP 369 mixer model specified in RTP [RFC3550]. The sources are indicated 370 in strict order in the CSRC list of the RTP packets. The CSRC 371 list can have up to 15 members. Therefore, text from up to 15 372 sources can be included in each packet. Packets are normally sent 373 with 300 ms intervals. The mean delay will be 150 ms. A new 374 redundancy packet format is specified. This method would result 375 in good performance, but would require standardization and 376 implementation of new releases in the target technologies that 377 would take more time than desirable to complete. It was therefore 378 not selected to be included in this document. 380 Mixing for multiparty-unaware endpoints 381 Presentation of text from multiple parties is prepared by the 382 mixer in one single stream. It is desirable to have a method that 383 does not require any modifications in existing user devices 384 implementing RFC 4103 for RTT without explicit support of 385 multiparty sessions. This is possible by having the mixer insert 386 a new line and a text formatted source label before each switch of 387 text source in the stream. Switch of source can only be done in 388 places in the text where it does not disturb the perception of the 389 contents. Text from only one source can be presented in real time 390 at a time. The delay will therefore vary. The method also has 391 other limitations, but is included in this document as a fallback 392 method. In calls where parties take turns properly by ending 393 their entries with a new line, the limitations will have limited 394 influence on the user experience. when only two parties send text, 395 these two will see the text in real time with no delay. This 396 method is specified as a fallback method in this document. 398 RTT transport in WebRTC 399 Transport of real-time text in the WebRTC technology is specified 400 to use the WebRTC data channel in [RFC8865]. That specification 401 contains a section briefly describing its use in multiparty 402 sessions. The focus of this document is RTP transport. 403 Therefore, even if the WebRTC transport provides good multiparty 404 performance, it is just mentioned in this document in relation to 405 providing gateways with multiparty capabilities between RTP and 406 WebRTC technologies. 408 1.3. Intended application 410 The method for multiparty real-time text specified in this document 411 is primarily intended for use in transmission between mixers and 412 endpoints in centralized mixing configurations. It is also 413 applicable between mixers. An often mentioned application is for 414 emergency service calls with real-time text and voice, where a call 415 taker wants to make an attended handover of a call to another agent, 416 and stay to observe the session. Multimedia conference sessions with 417 support for participants to contribute in text is another 418 application. Conferences with central support for speech-to-text 419 conversion is yet another mentioned application. 421 In all these applications, normally only one participant at a time 422 will send long text utterances. In some cases, one other participant 423 will occasionally contribute with a longer comment simultaneously. 424 That may also happen in some rare cases when text is interpreted to 425 text in another language in a conference. Apart from these cases, 426 other participants are only expected to contribute with very brief 427 utterings while others are sending text. 429 Users expect that the text they send is presented in real-time in a 430 readable way to the other participants even if they send 431 simultaneously with other users and even when they make brief edit 432 operations of their text by backspacing and correcting their text. 434 Text is supposed to be human generated, by some text input means, 435 such as typing on a keyboard or using speech-to-text technology. 436 Occasional small cut-and-paste operations may appear even if that is 437 not the initial purpose of real-time text. 439 The real-time characteristics of real-time text is essential for the 440 participants to be able to contribute to a conversation. If the text 441 is too much delayed from typing a letter to its presentation, then, 442 in some conference situations, the opportunity to comment will be 443 gone and someone else will grab the turn. A delay of more than one 444 second in such situations is an obstacle for good conversation. 446 2. Overview of the two specified solutions and selection of method 448 This section contains a brief introduction of the two methods 449 specified in this document. 451 2.1. The RTP-mixer-based solution for multiparty-aware endpoints 453 This method specifies negotiated use of the RFC 4103 format for 454 multiparty transmission in a single RTP stream. The main purpose of 455 this document is to specify a method for true multiparty real-time 456 text mixing for multiparty-aware endpoints that can be widely 457 deployed. The RTP-mixer-based method makes use of the current format 458 for real-time text in [RFC4103]. It is an update of RFC 4103 by a 459 clarification on one way to use it in the multiparty situation. That 460 is done by completing a negotiation for this kind of multiparty 461 capability and by interleaving packets from different sources. The 462 source is indicated in the CSRC element in the RTP packets. Specific 463 considerations are made to be able to recover text after packet loss. 465 The detailed procedures for the RTP-mixer-based multiparty-aware case 466 are specified in Section 3. 468 Please use [RFC4103] as reference when reading the specification. 470 2.2. Mixing for multiparty-unaware endpoints 472 A method is also specified in this document for cases when the 473 endpoint participating in a multiparty call does not itself implement 474 any solution, or not the same, as the mixer. The method requires the 475 mixer to insert text dividers and readable labels and only send text 476 from one source at a time until a suitable point appears for source 477 change. This solution is a fallback method with functional 478 limitations. It acts on the presentation level. 480 A mixer SHOULD by default format and transmit text to a call 481 participant to be suitable to present on a multiparty-unaware 482 endpoint which has not negotiated any method for true multiparty RTT 483 handling, but negotiated a "text/red" or "text/t140" format in a 484 session. This SHOULD be done if nothing else is specified for the 485 application in order to maintain interoperability. Section 4.2 486 specifies how this mixing is done. 488 2.3. Offer/answer considerations 490 RTP Payload for Text Conversation [RFC4103] specifies use of RTP 491 [RFC3550], and a redundancy format "text/red" for increased 492 robustness of real-time text transmission. This document updates 493 [RFC4103] by introducing a capability negotiation for handling 494 multiparty real-time text, a way to indicate the source of 495 transmitted text, and rules for efficient timing of the transmissions 496 interleaved from different sources. 498 The capability negotiation for the "RTP-mixer-based multiparty 499 method" is based on use of the SDP media attribute "rtt-mixer". 501 The syntax is as follows: 502 "a=rtt-mixer" 504 If any other method for RTP-based multiparty real-time text gets 505 specified by additional work, it is assumed that it will be 506 recognized by some specific SDP feature exchange. 508 2.3.1. Initial offer 510 A party intending to set up a session and being willing to use the 511 RTP-mixer-based method of this specification for sending or receiving 512 or both sending and receiving real-time text SHALL include the "rtt- 513 mixer" SDP attribute in the corresponding "text" media section in the 514 initial offer. 516 The party MAY indicate capability for both the RTP-mixer-based method 517 of this specification and other methods. 519 When the offeror has sent the offer including the "rtt-mixer" 520 attribute, it MUST be prepared to receive and handle real-time text 521 formatted according to both the method for multiparty-aware parties 522 specified in Section 3 in this specification and two-party formatted 523 real-time text. 525 2.3.2. Answering the offer 527 A party receiving an offer containing the "rtt-mixer" SDP attribute 528 and being willing to use the RTP-mixer-based method of this 529 specification for sending or receiving or both sending and receiving 530 SHALL include the "rtt-mixer" SDP attribute in the corresponding 531 "text" media section in the answer. 533 If the offer did not contain the "rtt-mixer" attribute, the answer 534 MUST NOT contain the "rtt-mixer" attribute. 536 Even when the "rtt-mixer" attribute is successfully negotiated, the 537 parties MAY send and receive two-party coded real-time text. 539 An answer MUST NOT include acceptance of more than one method for 540 multiparty real-time text in the same RTP session. 542 When the answer including acceptance is transmitted, the answerer 543 MUST be prepared to act on received text in the negotiated session 544 according to the method for multiparty-aware parties specified in 545 Section 3 of this specification. Reception of text for a two-party 546 session SHALL also be supported. 548 2.3.3. Offeror processing the answer 550 When the answer is processed by the offeror, it MUST act as specified 551 in Section 2.4 553 2.3.4. Modifying a session 555 A session MAY be modified at any time by any party offering a 556 modified SDP with or without the "rtt-mixer" SDP attribute expressing 557 a desired change in the support of multiparty real-time text. 559 If the modified offer adds indication of support for multiparty real- 560 time text by including the "rtt-mixer" SDP attribute, the procedures 561 specified in the previous subsections SHALL be applied. 563 If the modified offer deletes indication of support for multiparty 564 real-time text by excluding the "rtt-mixer" SDP attribute, the answer 565 MUST NOT contain the "rtt-mixer" attribute. After processing this 566 SDP exchange, the parties MUST NOT send real-time text formatted for 567 multiparty-aware parties according to this specification. 569 2.4. Actions depending on capability negotiation result 571 A transmitting party SHALL send text according to the RTP-mixer-based 572 multiparty method only when the negotiation for that method was 573 successful and when it conveys text for another source. In all other 574 cases, the packets SHALL be populated and interpreted as for a two- 575 party session. 577 A party which has negotiated the "rtt-mixer" SDP media attribute MUST 578 populate the CSRC-list, and format the packets according to Section 3 579 if it acts as an rtp-mixer and sends multiparty text. 581 A party which has negotiated the "rtt-mixer" SDP media attribute MUST 582 interpret the contents of the "CC" field, the CSRC-list and the 583 packets according to Section 3 in received RTP packets in the 584 corresponding RTP stream. 586 A party which has not successfully completed the negotiation of the 587 "rtt-mixer" SDP media attribute MUST NOT transmit packets interleaved 588 from different sources in the same RTP stream as specified in 589 Section 3. If the party is a mixer and did declare the "rtt-mixer" 590 SDP media attribute, it SHOULD perform the procedure for multiparty- 591 unaware endpoints. If the party is not a mixer, it SHOULD transmit 592 as in a two-party session according to [RFC4103]. 594 3. Details for the RTP-mixer-based mixing method for multiparty-aware 595 endpoints 597 3.1. Use of fields in the RTP packets 599 The CC field SHALL show the number of members in the CSRC list, which 600 SHALL be one (1) in transmissions from a mixer when conveying text 601 from other sources in a multiparty session, and otherwise 0. 603 When text is conveyed by a mixer during a multiparty session, a CSRC 604 list SHALL be included in the packet. The single member in the CSRC- 605 list SHALL contain the SSRC of the source of the T140blocks in the 606 packet. 608 When redundancy is used, the RECOMMENDED level of redundancy is to 609 use one primary and two redundant generations of T140blocks. In some 610 cases, a primary or redundant T140block is empty, but is still 611 represented by a member in the redundancy header. 613 In other regards, the contents of the RTP packets are equal to what 614 is specified in [RFC4103]. 616 3.2. Initial transmission of a BOM character 618 As soon as a participant is known to participate in a session with 619 another entity and is available for text reception, a Unicode Byte- 620 Order Mark (BOM) character SHALL be sent to it by the other entity 621 according to the procedures in this section. This is useful in many 622 configurations to open ports and firewalls and setting up the 623 connection between the application and the network. If the 624 transmitter is a mixer, then the source of this character SHALL be 625 indicated to be the mixer itself. 627 Note that the BOM character SHALL be transmitted with the same 628 redundancy procedures as any other text. 630 3.3. Keep-alive 632 After that, the transmitter SHALL send keep-alive traffic to the 633 receiver(s) at regular intervals when no other traffic has occurred 634 during that interval, if that is decided for the actual connection. 635 It is RECOMMENDED to use the keep-alive solution from [RFC6263]. The 636 consent check of [RFC7675] is a possible alternative if it is used 637 anyway for other reasons. 639 3.4. Transmission interval 641 A "text/red" or "text/t140" transmitter in a mixer SHALL send packets 642 distributed in time as long as there is something (new or redundant 643 T140blocks) to transmit. The maximum transmission interval between 644 text transmissions from the same source SHALL then be 330 ms, when no 645 other limitations cause a longer interval to be temporarily used. It 646 is RECOMMENDED to send the next packet to a receiver as soon as new 647 text to that receiver is available, as long as the mean character 648 rate of new text to the receiver calculated over the last 10 one- 649 second intervals does not exceed the "cps" value of the receiver. 650 The intention is to keep the latency low and network load limited 651 while keeping good protection against text loss in bursty packet loss 652 conditions. The main purpose of the 330 ms interval is for timing of 653 redundant transmission, when no new text from the same source is 654 available. 656 The reason for the value 330 ms is that many sources of text will 657 transmit new text with 300 ms intervals during periods of continuous 658 user typing, and then reception in the mixer of such new text will 659 cause a combined transmission of the new text and the unsent 660 redundancy from the previous transmission. Only when the user stops 661 typing, the 330 ms interval will be applied to send the redundancy. 663 If the Characters Per Second (cps) value is reached, a longer 664 transmission interval SHALL be applied for text from all sources as 665 specified in [RFC4103] and only as much of the text queued for 666 transmission SHALL be sent at the end of each transmission interval 667 as can be allowed without exceeding the "cps" value. Division of 668 text for partial transmission MUST then be made at T140block borders. 669 When the transmission rate falls under the "cps" value again, the 670 transmission intervals SHALL be returned to 330 ms and transmission 671 of new text SHALL return to be made as soon as new text is available. 673 NOTE: that extending the transmission intervals during high load 674 periods does not change the number of characters to be conveyed. It 675 just evens out the load in time and reduces the number of packets per 676 second. With human created conversational text, the sending user 677 will eventually take a pause letting transmission catch up. 679 See also Section 8. 681 For a transmitter not acting as a mixer, the transmission interval 682 principles from [RFC4103] apply, and the normal transmission interval 683 SHALL be 300 ms. 685 3.5. Only one source per packet 687 New text and redundant copies of earlier text from one source SHALL 688 be transmitted in the same packet if available for transmission at 689 the same time. Text from different sources MUST NOT be transmitted 690 in the same packet. 692 3.6. Do not send received text to the originating source 694 Text received by a mixer from a participant SHOULD NOT be included in 695 transmission from the mixer to that participant, because the normal 696 behavior of the endpoint is to present locally-produced text locally. 698 3.7. Clean incoming text 700 A mixer SHALL handle reception, recovery from packet loss, deletion 701 of superfluous redundancy, marking of possible text loss and deletion 702 of 'BOM' characters from each participant before queueing received 703 text for transmission to receiving participants as specified in 704 [RFC4103] for single-party sources and Section 3.16 for multiparty 705 sources (chained mixers). 707 3.8. Redundant transmission principles 709 A transmitting party using redundancy SHALL send redundant 710 repetitions of T140blocks already transmitted in earlier packets. 712 The number of redundant generations of T140blocks to include in 713 transmitted packets SHALL be deduced from the SDP negotiation. It 714 SHALL be set to the minimum of the number declared by the two parties 715 negotiating a connection. It is RECOMMENDED to declare and transmit 716 one original and two redundant generations of the T140blocks because 717 that provides good protection against text loss in case of packet 718 loss, and low overhead. 720 3.9. Text placement in packets 722 The mixer SHALL compose and transmit an RTP packet to a receiver when 723 one or more of the following conditions have occurred: 725 * The transmission interval is the normal 330 ms and there is newly 726 received unsent text available for transmission to that receiver. 728 * The current transmission interval has passed and is longer than 729 the normal 330 ms and there is newly received unsent text 730 available for transmission to that receiver. 732 * The current transmission interval ( normally 330 ms) has passed 733 since already transmitted text was queued for transmission as 734 redundant text. 736 The principles from [RFC4103] apply for populating the header, the 737 redundancy header and the data in the packet with specifics specified 738 here and in the following sections. 740 At the time of transmission, the mixer SHALL populate the RTP packet 741 with all T140blocks queued for transmission originating from the 742 source in turn for transmission as long as this is not in conflict 743 with the allowed number of characters per second ("cps") or the 744 maximum packet size. In this way, the latency of the latest received 745 text is kept low even in moments of simultaneous transmission from 746 many sources. 748 Redundant text SHALL also be included, and the assessment of how much 749 new text can be included within the maximum packet size MUST take 750 into account that the redundancy has priority to be transmitted in 751 its entirety. See Section 3.4 753 The SSRC of the source SHALL be placed as the only member in the 754 CSRC-list. 756 Note: The CSRC-list in an RTP packet only includes the participant 757 whose text is included in text blocks. It is not the same as the 758 total list of participants in a conference. With audio and video 759 media, the CSRC-list would often contain all participants who are not 760 muted whereas text participants that don't type are completely silent 761 and thus are not represented in RTP packet CSRC-lists. 763 3.10. Empty T140blocks 765 If no unsent T140blocks were available for a source at the time of 766 populating a packet, but T140blocks are available which have not yet 767 been sent the full intended number of redundant transmissions, then 768 the primary T140block for that source is composed of an empty 769 T140block, and populated (without taking up any length) in a packet 770 for transmission. The corresponding SSRC SHALL be placed as usual in 771 its place in the CSRC-list. 773 The first packet in the session, the first after a source switch, and 774 the first after a pause SHALL be populated with the available 775 T140blocks for the source in turn to be sent as primary, and empty 776 T140blocks for the agreed number of redundancy generations. 778 3.11. Creation of the redundancy 780 The primary T140block from a source in the latest transmitted packet 781 is saved for populating the first redundant T140block for that source 782 in the next transmission of text from that source. The first 783 redundant T140block for that source from the latest transmission is 784 saved for populating the second redundant T140block in the next 785 transmission of text from that source. 787 Usually this is the level of redundancy used. If a higher level of 788 redundancy is negotiated, then the procedure SHALL be maintained 789 until all available redundant levels of T140blocks are placed in the 790 packet. If a receiver has negotiated a lower number of "text/red" 791 generations, then that level SHALL be the maximum used by the 792 transmitter. 794 The T140blocks saved for transmission as redundant data are assigned 795 a planned transmission time 330 ms after the current time, but SHOULD 796 be transmitted earlier if new text for the same source gets in turn 797 for transmission before that time. 799 3.12. Timer offset fields 801 The timestamp offset values SHALL be inserted in the redundancy 802 header, with the time offset from the RTP timestamp in the packet 803 when the corresponding T140block was sent as primary. 805 The timestamp offsets are expressed in the same clock tick units as 806 the RTP timestamp. 808 The timestamp offset values for empty T140blocks have no relevance 809 but SHOULD be assigned realistic values. 811 3.13. Other RTP header fields 813 The number of members in the CSRC list (0 or 1) SHALL be placed in 814 the "CC" header field. Only mixers place value 1 in the "CC" field. 815 A value of "0" indicates that the source is the transmitting device 816 itself and that the source is indicated by the SSRC field. This 817 value is used by endpoints, and by mixers sending self-sourced data. 819 The current time SHALL be inserted in the timestamp. 821 The SSRC header field SHALL contain the SSRC of the RTP session where 822 the packet will be transmitted. 824 The M-bit SHALL be handled as specified in [RFC4103]. 826 3.14. Pause in transmission 828 When there is no new T140block to transmit, and no redundant 829 T140block that has not been retransmitted the intended number of 830 times from any source, the transmission process SHALL be stopped 831 until either new T140blocks arrive, or a keep-alive method calls for 832 transmission of keep-alive packets. 834 3.15. RTCP considerations 836 A mixer SHALL send RTCP reports with SDES, CNAME, and NAME 837 information about the sources in the multiparty call. This makes it 838 possible for participants to compose a suitable label for text from 839 each source. 841 Privacy considerations SHALL be taken when composing these fields. 842 They contain name and address information that may be sensitive to 843 transmit in its entirety, e.g., to unauthenticated participants. 845 3.16. Reception of multiparty contents 847 The "text/red" receiver included in an endpoint with presentation 848 functions will receive RTP packets in the single stream from the 849 mixer, and SHALL distribute the T140blocks for presentation in 850 presentation areas for each source. Other receiver roles, such as 851 gateways or chained mixers, are also feasible. They require 852 considerations if the stream shall just be forwarded, or distributed 853 based on the different sources. 855 3.16.1. Acting on the source of the packet contents 857 If the "CC" field value of a received packet is 1, it indicates that 858 the text is conveyed from a source indicated in the single member in 859 the CSRC-list, and the receiver MUST act on the source according to 860 its role. If the CC value is 0, the source is indicated in the SSRC 861 field. 863 3.16.2. Detection and indication of possible text loss 865 The receiver SHALL monitor the RTP sequence numbers of the received 866 packets for gaps and packets out of order. If a sequence number gap 867 appears and still exists after some defined short time for jitter and 868 reordering resolution, the packets in the gap SHALL be regarded as 869 lost. 871 If it is known that only one source is active in the RTP session, 872 then it is likely that a gap equal to or larger than the agreed 873 number of redundancy generations (including the primary) causes text 874 loss. In that case, the receiver SHALL create a t140block with a 875 marker for possible text loss [T140ad1] and associate it with the 876 source and insert it in the reception buffer for that source. 878 If it is known that more than one source is active in the RTP 879 session, then it is not possible in general to evaluate if text was 880 lost when packets were lost. With two active sources and the 881 recommended number of redundancy generations (3), it can take a gap 882 of five consecutive lost packets until any text may be lost, but text 883 loss can also appear if three non-consecutive packets are lost when 884 they contained consecutive data from the same source. A simple 885 method to decide when there is risk for resulting text loss is to 886 evaluate if three or more packets were lost within one second. If 887 this simple method is used, then a t140block SHOULD be created with a 888 marker for possible text loss [T140ad1] and associated with the SSRC 889 of the RTP session as a general input from the mixer. 891 Implementations MAY apply more refined methods for more reliable 892 detection of whether text was lost or not. Any refined method SHOULD 893 prefer marking possible loss rather than not marking when it is 894 uncertain if there was loss. 896 3.16.3. Extracting text and handling recovery 898 When applying the following procedures, the effects MUST be 899 considered of possible timestamp wrap around and the RTP session 900 possibly changing SSRC. 902 When a packet is received in an RTP session using the packetization 903 for multiparty-aware endpoints, its T140blocks SHALL be extracted in 904 the following way. 906 The source SHALL be extracted from the CSRC-list if available, 907 otherwise from the SSRC. 909 If the received packet is the first packet received from the source, 910 then all T140blocks in the packet SHALL be retrieved and assigned to 911 a receive buffer for the source beginning with the oldest available 912 redundant generation, continuing with the younger redundant 913 generations in age order and finally the primary. 915 Note: The normal case is that in the first packet, only the primary 916 data has contents. The redundant data has contents in the first 917 received packet from a source only after initial packet loss. 919 If the packet is not the first packet from a source, then if 920 redundant data is available, the process SHALL start with the oldest 921 generation. The timestamp of that redundant data SHALL be created by 922 subtracting its timestamp offset from the RTP timestamp. If the 923 resulting timestamp is later than the latest retrieved data from the 924 same source, then the redundant data SHALL be retrieved and appended 925 to the receive buffer. The process SHALL be continued in the same 926 way for all younger generations of redundant data. After that, the 927 timestamp of the packet SHALL be compared with the timestamp of the 928 latest retrieved data from the same source and if it is later, then 929 the primary data SHALL be retrieved from the packet and appended to 930 the receive buffer for the source. 932 3.16.4. Delete 'BOM' 934 Unicode character 'BOM' is used as a start indication and sometimes 935 used as a filler or keep alive by transmission implementations. 936 These SHALL be deleted after extraction from received packets. 938 3.17. Performance considerations 940 This solution has good performance with low text delays, as long as 941 the mean number of characters per second sent during any 10-second 942 interval from a number of simultaneously sending participants to a 943 receiving participant, does not reach the "cps" value. At higher 944 numbers of sent characters per second, a jerkiness is visible in the 945 presentation of text. The solution is therefore suitable for 946 emergency service use, relay service use, and small or well-managed 947 larger multimedia conferences. Only in large unmanaged conferences 948 with a high number of participants there may on very rare occasions 949 appear situations when many participants happen to send text 950 simultaneously. In such circumstances, the result may be 951 unpleasantly jerky presentation of text from each sending 952 participant. It should be noted that it is only the number of users 953 sending text within the same moment that causes jerkiness, not the 954 total number of users with RTT capability. 956 3.18. Security for session control and media 958 Security mechanisms to provide confidentiality and integrity 959 protection and peer authentication SHOULD be applied when possible 960 regarding the capabilities of the participating devices by use of SIP 961 over TLS by default according to [RFC5630] section 3.1.3 on the 962 session control level and by default using DTLS-SRTP [RFC5764] on the 963 media level. In applications where legacy endpoints without security 964 are allowed, a negotiation SHOULD be performed to decide if 965 encryption on the media level will be applied. If no other security 966 solution is mandated for the application, then OSRTP [RFC8643] is a 967 suitable method to be applied to negotiate SRTP media security with 968 DTLS. Most SDP examples below are for simplicity expressed without 969 the security additions. The principles (but not all details) for 970 applying DTLS-SRTP [RFC5764] security are shown in a couple of the 971 following examples. 973 Further general security considerations are covered in Section 10. 975 End-to-end encryption would require further work and could be based 976 on WebRTC as specified in Section 1.2 or on double encryption as 977 specified in [RFC8723]. 979 3.19. SDP offer/answer examples 981 This section shows some examples of SDP for session negotiation of 982 the real-time text media in SIP sessions. Audio is usually provided 983 in the same session, and sometimes also video. The examples only 984 show the part of importance for the real-time text media. The 985 examples relate to the single RTP stream mixing for multiparty-aware 986 endpoints and for multiparty-unaware endpoints. 988 Note: Multiparty RTT MAY also be provided through other methods, 989 e.g., by a Selective Forwarding Middlebox (SFM). In that case, the 990 SDP of the offer will include something specific for that method, 991 e.g., an SDP attribute or another media format. An answer selecting 992 the use of that method would accept it by a corresponding 993 acknowledgement included in the SDP. The offer may contain also the 994 "rtt-mixer" SDP media attribute for the main RTT media when the 995 offeror has capability for both multiparty methods, while an answer, 996 selecting to use SFM will not include the "rtt-mixer" SDP media 997 attribute. 999 Offer example for "text/red" format and multiparty support: 1001 m=text 11000 RTP/AVP 100 98 1002 a=rtpmap:98 t140/1000 1003 a=rtpmap:100 red/1000 1004 a=fmtp:100 98/98/98 1005 a=rtt-mixer 1007 Answer example from a multiparty-aware device 1008 m=text 14000 RTP/AVP 100 98 1009 a=rtpmap:98 t140/1000 1010 a=rtpmap:100 red/1000 1011 a=fmtp:100 98/98/98 1012 a=rtt-mixer 1014 Offer example for "text/red" format including multiparty 1015 and security: 1016 a=fingerprint: (fingerprint1) 1017 m=text 11000 RTP/AVP 100 98 1018 a=rtpmap:98 t140/1000 1019 a=rtpmap:100 red/1000 1020 a=fmtp:100 98/98/98 1021 a=rtt-mixer 1023 The "fingerprint" is sufficient to offer DTLS-SRTP, with the media 1024 line still indicating RTP/AVP. 1026 Note: For brevity, the entire value of the SDP fingerprint attribute 1027 is not shown in this and the following example. 1029 Answer example from a multiparty-aware device with security 1030 a=fingerprint: (fingerprint2) 1031 m=text 16000 RTP/AVP 100 98 1032 a=rtpmap:98 t140/1000 1033 a=rtpmap:100 red/1000 1034 a=fmtp:100 98/98/98 1035 a=rtt-mixer 1037 With the "fingerprint" the device acknowledges use of SRTP/DTLS. 1039 Answer example from a multiparty-unaware device that also 1040 does not support security: 1042 m=text 12000 RTP/AVP 100 98 1043 a=rtpmap:98 t140/1000 1044 a=rtpmap:100 red/1000 1045 a=fmtp:100 98/98/98 1047 3.20. Packet sequence example from interleaved transmission 1049 This example shows a symbolic flow of packets from a mixer including 1050 loss and recovery. The sequence includes interleaved transmission of 1051 text from two RTT sources A and B. P indicates primary data. R1 is 1052 first redundant generation data and R2 is the second redundant 1053 generation data. A1, B1, A2 etc. are text chunks (T140blocks) 1054 received from the respective sources and sent on to the receiver by 1055 the mixer. X indicates a dropped packet between the mixer and a 1056 receiver. The session is assumed to use original and two redundant 1057 generations of RTT. 1059 |-----------------------| 1060 |Seq no 101, Time=20400 | 1061 |CC=1 | 1062 |CSRC list A | 1063 |R2: A1, Offset=600 | 1064 |R1: A2, Offset=300 | 1065 |P: A3 | 1066 |-----------------------| 1068 Assuming that earlier packets (with text A1 and A2) were received in 1069 sequence, text A3 is received from packet 101 and assigned to 1070 reception buffer A. The mixer is now assumed to have received 1071 initial text from source B 100 ms after packet 101 and will send that 1072 text. Transmission of A2 and A3 as redundancy is planned for 330 ms 1073 after packet 101 if no new text from A is ready to be sent before 1074 that. 1076 |-----------------------| 1077 |Seq no 102, Time=20500 | 1078 |CC=1 | 1079 |CSRC list B | 1080 |R2 Empty, Offset=600 | 1081 |R1: Empty, Offset=300 | 1082 |P: B1 | 1083 |-----------------------| 1084 Packet 102 is received. 1085 B1 is retrieved from this packet. Redundant transmission of 1086 B1 is planned 330 ms after packet 102. 1088 X------------------------| 1089 X Seq no 103, Timer=20730| 1090 X CC=1 | 1091 X CSRC list A | 1092 X R2: A2, Offset=630 | 1093 X R1: A3, Offset=330 | 1094 X P: Empty | 1095 X------------------------| 1096 Packet 103 is assumed to be lost due to network problems. 1097 It contains redundancy for A. Sending A3 as second level 1098 redundancy is planned for 330 ms after packet 103. 1100 X------------------------| 1101 X Seq no 104, Timer=20800| 1102 X CC=1 | 1103 X CSRC list B | 1104 X R2: Empty, Offset=600 | 1105 X R1: B1, Offset=300 | 1106 X P: B2 | 1107 X------------------------| 1108 Packet 104 contains text from B, including new B2 and 1109 redundant B1. It is assumed dropped due to network 1110 problems. 1111 The mixer has A3 redundancy to send, but no new text 1112 appears from A and therefore the redundancy is sent 1113 330 ms after the previous packet with text from A. 1115 |------------------------| 1116 | Seq no 105, Timer=21060| 1117 | CC=1 | 1118 | CSRC list A | 1119 | R2: A3, Offset=660 | 1120 | R1: Empty, Offset=330 | 1121 | P: Empty | 1122 |------------------------| 1123 Packet 105 is received. 1124 A gap for lost packets 103 and 104 is detected. 1125 Assume that no other loss was detected during the last second. 1126 Then it can be concluded that nothing was totally lost. 1128 R2 is checked. Its original time was 21060-660=20400. 1129 A packet with text from A was received with that 1130 timestamp, so nothing needs to be recovered. 1132 B1 and B2 still need to be transmitted as redundancy. 1133 This is planned 330 ms after packet 104. That 1134 would be at 21130. 1136 |-----------------------| 1137 |Seq no 106, Timer=21130| 1138 |CC=1 | 1139 |CSRC list B | 1140 | R2: B1, Offset=630 | 1141 | R1: B2, Offset=330 | 1142 | P: Empty | 1143 |-----------------------| 1145 Packet 106 is received. 1147 The second level redundancy in packet 106 is B1 and has timestamp 1148 offset 630 ms. The timestamp of packet 106 minus 630 is 20500 which 1149 is the timestamp of packet 102 that was received. So B1 does not 1150 need to be retrieved. The first level redundancy in packet 106 has 1151 offset 330. The timestamp of packet 106 minus 330 is 20800. That is 1152 later than the latest received packet with source B. Therefore B2 is 1153 retrieved and assigned to the input buffer for source B. No primary 1154 is available in packet 106. 1156 After this sequence, A3 and B1 and B2 have been received. In this 1157 case no text was lost. 1159 3.21. Maximum character rate "cps" 1161 The default maximum rate of reception of "text/t140" real-time text 1162 is in [RFC4103] specified to be 30 characters per second. The actual 1163 rate is calculated without regard to any redundant text transmission 1164 and is in the multiparty case evaluated for all sources contributing 1165 to transmission to a receiver. The value MAY be modified in the 1166 "cps" parameter of the FMTP attribute in the media section for the 1167 "text/t140" media. A mixer combining real-time text from a number of 1168 sources may occasionally have a higher combined flow of text coming 1169 from the sources. Endpoints SHOULD therefore specify a suitable 1170 higher value for the "cps" parameter, corresponding to its real 1171 reception capability. A value for "cps" of 90 SHALL be the default 1172 for the "text/t140" stream in the "text/red" format when multiparty 1173 real-time text is negotiated. See [RFC4103] for the format and use 1174 of the "cps" parameter. The same rules apply for the multiparty case 1175 except for the default value. 1177 4. Presentation level considerations 1179 "Protocol for multimedia application text conversation" [T140] 1180 provides the presentation level requirements for the [RFC4103] 1181 transport. Functions for erasure and other formatting functions are 1182 specified in [T140] which has the following general statement for the 1183 presentation: 1185 "The display of text from the members of the conversation should be 1186 arranged so that the text from each participant is clearly readable, 1187 and its source and the relative timing of entered text is visualized 1188 in the display. Mechanisms for looking back in the contents from the 1189 current session should be provided. The text should be displayed as 1190 soon as it is received." 1191 Strict application of [T140] is of essence for the interoperability 1192 of real-time text implementations and to fulfill the intention that 1193 the session participants have the same information conveyed in the 1194 text contents of the conversation without necessarily having the 1195 exact same layout of the conversation. 1197 [T140] specifies a set of presentation control codes to include in 1198 the stream. Some of them are optional. Implementations MUST ignore 1199 optional control codes that they do not support. 1201 There is no strict "message" concept in real-time text. The Unicode 1202 Line Separator character SHALL be used as a separator allowing a part 1203 of received text to be grouped in presentation. The characters 1204 "CRLF" may be used by other implementations as a replacement for Line 1205 Separator. The "CRLF" combination SHALL be erased by just one 1206 erasing action, the same as the Line Separator. Presentation 1207 functions are allowed to group text for presentation in smaller 1208 groups than the line separators imply and present such groups with 1209 source indication together with text groups from other sources (see 1210 the following presentation examples). Erasure has no specific limit 1211 by any delimiter in the text stream. 1213 4.1. Presentation by multiparty-aware endpoints 1215 A multiparty-aware receiving party, presenting real-time text MUST 1216 separate text from different sources and present them in separate 1217 presentation fields. The receiving party MAY separate presentation 1218 of parts of text from a source in readable groups based on other 1219 criteria than line separator and merge these groups in the 1220 presentation area when it benefits the user to most easily find and 1221 read text from the different participants. The criteria MAY e.g., be 1222 a received comma, full stop, or other phrase delimiters, or a long 1223 pause. 1225 When text is received from multiple original sources, the 1226 presentation SHALL provide a view where text is added in multiple 1227 presentation fields. 1229 If the presentation presents text from different sources in one 1230 common area, the presenting endpoint SHOULD insert text from the 1231 local user ended at suitable points merged with received text to 1232 indicate the relative timing for when the text groups were completed. 1233 In this presentation mode, the receiving endpoint SHALL present the 1234 source of the different groups of text. This presentation style is 1235 called the "chat" style here and provides a possibility to follow 1236 text arriving from multiple parties and the approximate relative time 1237 that text is received related to text from the local user. 1239 A view of a three-party RTT call in chat style is shown in this 1240 example . 1242 _________________________________________________ 1243 | |^| 1244 |[Alice] Hi, Alice here. |-| 1245 | | | 1246 |[Bob] Bob as well. | | 1247 | | | 1248 |[Eve] Hi, this is Eve, calling from Paris. | | 1249 | I thought you should be here. | | 1250 | | | 1251 |[Alice] I am coming on Thursday, my | | 1252 | performance is not until Friday morning.| | 1253 | | | 1254 |[Bob] And I on Wednesday evening. | | 1255 | | | 1256 |[Alice] Can we meet on Thursday evening? | | 1257 | | | 1258 |[Eve] Yes, definitely. How about 7pm. | | 1259 | at the entrance of the restaurant | | 1260 | Le Lion Blanc? | | 1261 |[Eve] we can have dinner and then take a walk |-| 1262 |______________________________________________|v| 1263 | But I need to be back to |^| 1264 | the hotel by 11 because I need |-| 1265 | | | 1266 | I wou |-| 1267 |______________________________________________|v| 1268 | of course, I underst | 1269 |________________________________________________| 1271 Figure 3: Example of a three-party RTT call presented in chat style 1272 seen at participant 'Alice's endpoint. 1274 Other presentation styles than the chat style MAY be arranged. 1276 This figure shows how a coordinated column view MAY be presented. 1278 _____________________________________________________________________ 1279 | Bob | Eve | Alice | 1280 |____________________|______________________|_______________________| 1281 | | |I will arrive by TGV. | 1282 |My flight is to Orly| |Convenient to the main | 1283 | |Hi all, can we plan |station. | 1284 | |for the seminar? | | 1285 |Eve, will you do | | | 1286 |your presentation on| | | 1287 |Friday? |Yes, Friday at 10. | | 1288 |Fine, wo | |We need to meet befo | 1289 |___________________________________________________________________| 1291 Figure 4: An example of a coordinated column-view of a three-party 1292 session with entries ordered vertically in approximate time-order. 1294 4.2. Multiparty mixing for multiparty-unaware endpoints 1296 When the mixer has indicated RTT multiparty capability in an SDP 1297 negotiation, but the multiparty capability negotiation fails with an 1298 endpoint, then the agreed "text/red" or "text/t140" format SHALL be 1299 used and the mixer SHOULD compose a best-effort presentation of 1300 multiparty real-time text in one stream intended to be presented by 1301 an endpoint with no multiparty awareness, when that is desired in the 1302 actual implementation. The following specifies a procedure which MAY 1303 be applied in that situation. 1305 This presentation format has functional limitations and SHOULD be 1306 used only to enable participation in multiparty calls by legacy 1307 deployed endpoints implementing only RFC 4103 without any multiparty 1308 extensions specified in this document. 1310 The principles and procedures below do not specify any new protocol 1311 elements. They are instead composed of information from [T140] and 1312 an ambition to provide a best-effort presentation on an endpoint 1313 which has functions originally intended only for two-party calls. 1315 The mixer mixing for multiparty-unaware endpoints SHALL compose a 1316 simulated, limited multiparty RTT view suitable for presentation in 1317 one presentation area. The mixer SHALL group text in suitable groups 1318 and prepare for presentation of them by inserting a line separator 1319 between them if the transmitted text did not already end with a new 1320 line (line separator or CRLF). A presentable label SHALL be composed 1321 and sent for the source initially in the session and after each 1322 source switch. With this procedure the time for switching from 1323 transmission of text from one source to transmission of text from 1324 another source depends on the actions of the users. In order to 1325 expedite source switching, a user can, for example, end its turn with 1326 a new line. 1328 4.2.1. Actions by the mixer at reception from the call participants 1330 When text is received by the mixer from the different participants, 1331 the mixer SHALL recover text from redundancy if any packets are lost. 1332 The mark for lost text [T140ad1] SHALL be inserted in the stream if 1333 unrecoverable loss appears. Any Unicode "BOM" characters, possibly 1334 used for keep-alive, SHALL be deleted. The time of creation of text 1335 (retrieved from the RTP timestamp) SHALL be stored together with the 1336 received text from each source in queues for transmission to the 1337 recipients in order to be able to evaluate text loss. 1339 4.2.2. Actions by the mixer for transmission to the recipients 1341 The following procedure SHALL be applied for each multiparty-unaware 1342 recipient of multiparty text from the mixer. 1344 The text for transmission SHALL be formatted by the mixer for each 1345 receiving user for presentation in one single presentation area. 1346 Text received from a participant SHOULD NOT be included in 1347 transmission to that participant because it is usually presented 1348 locally at transmission time. When there is text available for 1349 transmission from the mixer to a receiving party from more than one 1350 participant, the mixer SHALL switch between transmission of text from 1351 the different sources at suitable points in the transmitted stream. 1353 When switching source, the mixer SHALL insert a line separator if the 1354 already transmitted text did not end with a new line (line separator 1355 or CRLF). A label SHALL be composed of information in the CNAME and 1356 NAME fields in RTCP reports from the participant to have its text 1357 transmitted, or from other session information for that user. The 1358 label SHALL be delimited by suitable characters (e.g., '[ ]') and 1359 transmitted. The CSRC SHALL indicate the selected source. Then text 1360 from that selected participant SHALL be transmitted until a new 1361 suitable point for switching source is reached. 1363 Information available to the mixer for composing the label may 1364 contain sensitive personal information that SHOULD NOT be revealed in 1365 sessions not securely authenticated and confidentiality protected. 1366 Privacy considerations regarding how much personal information is 1367 included in the label SHOULD therefore be taken when composing the 1368 label. 1370 Seeking a suitable point for switching source SHALL be done when 1371 there is older text waiting for transmission from any party than the 1372 age of the last transmitted text. Suitable points for switching are: 1374 * A completed phrase ended by comma 1376 * A completed sentence 1378 * A new line (line separator or CRLF) 1380 * A long pause (e.g., > 10 seconds) in received text from the 1381 currently transmitted source 1383 * If text from one participant has been transmitted with text from 1384 other sources waiting for transmission for a long time (e.g., > 1 1385 minute) and none of the other suitable points for switching has 1386 occurred, a source switch MAY be forced by the mixer at the next 1387 word delimiter, and also even if a word delimiter does not occur 1388 within a time (e.g., 15 seconds) after the scan for a word 1389 delimiter started. 1391 When switching source, the source which has the oldest text in queue 1392 SHALL be selected to be transmitted. A character display count SHALL 1393 be maintained for the currently transmitted source, starting at zero 1394 after the label is transmitted for the currently transmitted source. 1396 The status SHALL be maintained for the latest control code for Select 1397 Graphic Rendition (SGR) from each source. If there is an SGR code 1398 stored as the status for the current source before the source switch 1399 is done, a reset of SGR SHALL be sent by the sequence SGR 0 [009B 1400 0000 006D] after the new line and before the new label during a 1401 source switch. See SGR below for an explanation. This transmission 1402 does not influence the display count. 1404 If there is an SGR code stored for the new source after the source 1405 switch, that SGR code SHALL be transmitted to the recipient before 1406 the label. This transmission does not influence the display count. 1408 4.2.3. Actions on transmission of text 1410 Text from a source sent to the recipient SHALL increase the display 1411 count by one per transmitted character. 1413 4.2.4. Actions on transmission of control codes 1415 The following control codes specified by T.140 require specific 1416 actions. They SHALL cause specific considerations in the mixer. 1417 Note that the codes presented here are expressed in UCS-16, while 1418 transmission is made in the UTF-8 encoding of these codes. 1420 BEL 0007 Bell Alert in session. Provides for alerting during an 1421 active session. The display count SHALL NOT be altered. 1423 NEW LINE 2028 Line separator. Check and perform a source switch if 1424 appropriate. Increase the display count by 1. 1426 CR LF 000D 000A A supported but not preferred way of requesting a 1427 new line. Check and perform a source switch if appropriate. 1428 Increase the display count by 1. 1430 INT ESC 0061 Interrupt (used to initiate the mode negotiation 1431 procedure). The display count SHALL NOT be altered. 1433 SGR 009B Ps 006D Select graphic rendition. Ps is the rendition 1434 parameters specified in ISO 6429. The display count SHALL NOT be 1435 altered. The SGR code SHOULD be stored for the current source. 1437 SOS 0098 Start of string, used as a general protocol element 1438 introducer, followed by a maximum 256-byte string and the ST. The 1439 display count SHALL NOT be altered. 1441 ST 009C String terminator, end of SOS string. The display count 1442 SHALL NOT be altered. 1444 ESC 001B Escape - used in control strings. The display count SHALL 1445 NOT be altered for the complete escape code. 1447 Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used 1448 for synchronization and keep-alive. It SHALL be deleted from 1449 incoming streams. It SHALL also be sent first after session 1450 establishment to the recipient. The display count SHALL NOT be 1451 altered. 1453 Missing text mark (U+FFFD) "Replacement character", represented as a 1454 question mark in a rhombus, or if that is not feasible, replaced 1455 by an apostrophe '. It marks the place in the stream of possible 1456 text loss. This mark SHALL be inserted by the reception procedure 1457 in case of unrecoverable loss of packets. The display count SHALL 1458 be increased by one when sent as for any other character. 1460 SGR If a control code for selecting graphic rendition (SGR) other 1461 than reset of the graphic rendition (SGR 0) is sent to a 1462 recipient, that control code SHALL also be stored as the status 1463 for the source in the storage for SGR status. If a reset graphic 1464 rendition (SGR 0) originating from a source is sent, then the SGR 1465 status storage for that source SHALL be cleared. The display 1466 count SHALL NOT be increased. 1468 BS (U+0008) Back Space, intended to erase the last entered character 1469 by a source. Erasure by backspace cannot always be performed as 1470 the erasing party intended. If an erasing action erases all text 1471 up to the end of the leading label after a source switch, then the 1472 mixer MUST NOT transmit more backspaces. Instead, it is 1473 RECOMMENDED that a letter "X" is inserted in the text stream for 1474 each backspace as an indication of the intent to erase more. A 1475 new line is usually coded by a Line Separator, but the character 1476 combination "CRLF" MAY be used instead. Erasure of a new line is 1477 in both cases done by just one erasing action (Backspace). If the 1478 display count has a positive value it SHALL be decreased by one 1479 when the BS is sent. If the display count is at zero, it SHALL 1480 NOT be altered. 1482 4.2.5. Packet transmission 1484 A mixer transmitting to a multiparty-unaware terminal SHALL send 1485 primary data only from one source per packet. The SSRC SHALL be the 1486 SSRC of the mixer. The CSRC list SHALL contain one member and be the 1487 SSRC of the source of the primary data. 1489 4.2.6. Functional limitations 1491 When a multiparty-unaware endpoint presents a conversation in one 1492 display area in a chat style, it inserts source indications for 1493 remote text and local user text as they are merged in completed text 1494 groups. When an endpoint using this layout receives and presents 1495 text mixed for multiparty-unaware endpoints, there will be two levels 1496 of source indicators for the received text; one generated by the 1497 mixer and inserted in a label after each source switch, and another 1498 generated by the receiving endpoint and inserted after each switch 1499 between local and remote source in the presentation area. This will 1500 waste display space and look inconsistent to the reader. 1502 New text can be presented only from one source at a time. Switch of 1503 source to be presented takes place at suitable places in the text, 1504 such as end of phrase, end of sentence, line separator and 1505 inactivity. Therefore, the time to switch to present waiting text 1506 from other sources may become long and will vary and depend on the 1507 actions of the currently presented source. 1509 Erasure can only be done up to the latest source switch. If a user 1510 tries to erase more text, the erasing actions will be presented as 1511 letter X after the label. 1513 Text loss because of network errors may hit the label between entries 1514 from different parties, causing risk for misunderstanding from which 1515 source a piece of text is. 1517 These facts make it strongly RECOMMENDED implementing multiparty 1518 awareness in RTT endpoints. The use of the mixing method for 1519 multiparty-unaware endpoints should be left for use with endpoints 1520 which are impossible to upgrade to become multiparty-aware. 1522 4.2.7. Example views of presentation on multiparty-unaware endpoints 1524 The following pictures are examples of the view on a participant's 1525 display for the multiparty-unaware case. 1527 _________________________________________________ 1528 | Conference | Alice | 1529 |________________________|_________________________| 1530 | |I will arrive by TGV. | 1531 |[Bob]:My flight is to |Convenient to the main | 1532 |Orly. |station. | 1533 |[Eve]:Hi all, can we | | 1534 |plan for the seminar. | | 1535 | | | 1536 |[Bob]:Eve, will you do | | 1537 |your presentation on | | 1538 |Friday? | | 1539 |[Eve]:Yes, Friday at 10.| | 1540 |[Bob]: Fine, wo |We need to meet befo | 1541 |________________________|_________________________| 1543 Figure 5: Alice who has a conference-unaware client is receiving the 1544 multiparty real-time text in a single-stream. 1546 This figure shows how a coordinated column view MAY be presented on 1547 Alice's device in a view with two-columns. The mixer inserts labels 1548 to show how the sources alternate in the column with received text. 1550 The mixer alternates between the sources at suitable points in the 1551 text exchange so that text entries from each party can be 1552 conveniently read. 1554 _________________________________________________ 1555 | |^| 1556 |(Alice) Hi, Alice here. |-| 1557 | | | 1558 |(mix)[Bob)] Bob as well. | | 1559 | | | 1560 |[Eve] Hi, this is Eve, calling from Paris | | 1561 | I thought you should be here. | | 1562 | | | 1563 |(Alice) I am coming on Thursday, my | | 1564 | performance is not until Friday morning.| | 1565 | | | 1566 |(mix)[Bob] And I on Wednesday evening. | | 1567 | | | 1568 |[Eve] we can have dinner and then walk | | 1569 | | | 1570 |[Eve] But I need to be back to | | 1571 | the hotel by 11 because I need | | 1572 | |-| 1573 |______________________________________________|v| 1574 | of course, I underst | 1575 |________________________________________________| 1577 Figure 6: An example of a view of the multiparty-unaware presentation 1578 in chat style. Alice is the local user. 1580 In this view, there is a tradition in receiving applications to 1581 include a label showing the source of the text, here shown with 1582 parenthesis "()". The mixer also inserts source labels for the 1583 multiparty call participants, here shown with brackets "[]". 1585 5. Relation to Conference Control 1586 5.1. Use with SIP centralized conferencing framework 1588 The Session Initiation Protocol (SIP) conferencing framework, mainly 1589 specified in [RFC4353], [RFC4579] and [RFC4575] is suitable for 1590 coordinating sessions including multiparty RTT. The RTT stream 1591 between the mixer and a participant is one and the same during the 1592 conference. Participants get announced by notifications when 1593 participants are joining or leaving, and further user information may 1594 be provided. The SSRC of the text to expect from joined users MAY be 1595 included in a notification. The notifications MAY be used both for 1596 security purposes and for translation to a label for presentation to 1597 other users. 1599 5.2. Conference control 1601 In managed conferences, control of the real-time text media SHOULD be 1602 provided in the same way as other for media, e.g., for muting and 1603 unmuting by the direction attributes in SDP [RFC8866]. 1605 Note that floor control functions may be of value for RTT users as 1606 well as for users of other media in a conference. 1608 6. Gateway Considerations 1610 6.1. Gateway considerations with Textphones 1612 multiparty RTT sessions may involve gateways of different kinds. 1613 Gateways involved in setting up sessions SHALL correctly reflect the 1614 multiparty capability or unawareness of the combination of the 1615 gateway and the remote endpoint beyond the gateway. 1617 One case that may occur is a gateway to Public Switched Telephone 1618 Network (PSTN) for communication with textphones (e.g., TTYs). 1619 Textphones are limited devices with no multiparty awareness, and it 1620 SHOULD therefore be suitable for the gateway to not indicate 1621 multiparty awareness for that case. Another solution is that the 1622 gateway indicates multiparty capability towards the mixer, and 1623 includes the multiparty mixer function for multiparty-unaware 1624 endpoints itself. This solution makes it possible to adapt to the 1625 functional limitations of the textphone. 1627 More information on gateways to textphones is found in [RFC5194] 1629 6.2. Gateway considerations with WebRTC 1631 Gateway operation to real-time text in WebRTC may also be required. 1632 In WebRTC, RTT is specified in [RFC8865]. 1634 A multiparty bridge may have functionality for communicating by RTT 1635 both in RTP streams with RTT and WebRTC T.140 data channels. Other 1636 configurations may consist of a multiparty bridge with either 1637 technology for RTT transport and a separate gateway for conversion of 1638 the text communication streams between RTP and T.140 data channel. 1640 In WebRTC, it is assumed that for a multiparty session, one T.140 1641 data channel is established for each source from a gateway or bridge 1642 to each participant. Each participant also has a data channel with a 1643 two-way connection with the gateway or bridge. 1645 The T.140 data channel used both ways is for text from the WebRTC 1646 user and from the bridge or gateway itself to the WebRTC user. The 1647 label parameter of this T.140 data channel is used as the NAME field 1648 in RTCP to participants on the RTP side. The other T.140 data 1649 channels are only for text from other participants to the WebRTC 1650 user. 1652 When a new participant has entered the session with RTP transport of 1653 RTT, a new T.140 channel SHOULD be established to WebRTC users with 1654 the label parameter composed of information from the NAME field in 1655 RTCP on the RTP side. 1657 When a new participant has entered the multiparty session with RTT 1658 transport in a WebRTC T.140 data channel, the new participant SHOULD 1659 be announced by a notification to RTP users. The label parameter 1660 from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP 1661 side, or other available session information. 1663 When a participant on the RTP side is disconnected from the 1664 multiparty session, the corresponding T.140 data channel(s) SHOULD be 1665 closed. 1667 When a WebRTC user of T.140 data channels disconnects from the mixer, 1668 the corresponding RTP streams or sources in an RTP-mixed stream 1669 SHOULD be closed. 1671 T.140 data channels MAY be opened and closed by negotiation or 1672 renegotiation of the session or by any other valid means as specified 1673 in section 1 of [RFC8865]. 1675 7. Updates to RFC 4103 1677 This document updates [RFC4103] by introducing an SDP media attribute 1678 "rtt-mixer" for negotiation of multiparty-mixing capability with the 1679 [RFC4103] format, and by specifying the rules for packets when 1680 multiparty capability is negotiated and in use. 1682 8. Congestion considerations 1684 The congestion considerations and recommended actions from [RFC4103] 1685 are also valid in multiparty situations. 1687 The time values SHALL then be applied per source of text sent to a 1688 receiver. 1690 If the very unlikely situation appears that many participants in a 1691 conference send text simultaneously for a long period, a delay may 1692 build up for presentation of text at the receivers if the limitation 1693 in characters per second ("cps") to be transmitted to the 1694 participants is exceeded. More delay than 7 seconds can cause 1695 confusion in the session. It is therefore RECOMMENDED that an RTP- 1696 mixer-based mixer discards such text causing excessive delays and 1697 inserts a general indication of possible text loss [T140ad1] in the 1698 session. If the main text contributor is indicated in any way, the 1699 mixer MAY avoid deleting text from that participant. It should 1700 however be noted that human creation of text normally contains 1701 pauses, when the transmission can catch up, so that the transmission 1702 overload situations are expected to be very rare. 1704 9. IANA Considerations 1706 9.1. Registration of the "rtt-mixer" SDP media attribute 1708 [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the 1709 RFC number of this document.] 1711 IANA is asked to register the new SDP attribute "rtt-mixer". 1713 Contact name: IESG 1715 Contact email: iesg@ietf.org 1717 Attribute name: rtt-mixer 1719 Attribute semantics: See RFCXXXX Section 2.3 1721 Attribute value: none 1723 Usage level: media 1725 Purpose: Indicate support by mixer and endpoint of multiparty mixing 1726 for real-time text transmission, using a common RTP-stream for 1727 transmission of text from a number of sources mixed with one 1728 source at a time and the source indicated in a single CSRC-list 1729 member. 1731 Charset Dependent: no 1733 O/A procedure: See RFCXXXX Section 2.3 1735 Mux Category: normal 1737 Reference: RFCXXXX 1739 10. Security Considerations 1741 The RTP-mixer model requires the mixer to be allowed to decrypt, 1742 pack, and encrypt secured text from the conference participants. 1743 Therefore, the mixer needs to be trusted to maintain confidentiality 1744 and integrity of the RTT data. This situation is similar to the 1745 situation for handling audio and video media in centralized mixers. 1747 The requirement to transfer information about the user in RTCP 1748 reports in SDES, CNAME, and NAME fields, and in conference 1749 notifications, may have privacy concerns as already stated in RFC 1750 3550 [RFC3550], and may be restricted for privacy reasons. When used 1751 for creation of readable labels in the presentation, the receiving 1752 user will then get a more symbolic label for the source. 1754 The services available through the RTT mixer may have special 1755 interest for deaf and hard-of-hearing persons. Some users may want 1756 to refrain from revealing such characteristics broadly in 1757 conferences. The design of the conference systems where the mixer is 1758 included MAY need to be made with confidentiality of such 1759 characteristics in mind. 1761 Participants with malicious intentions may appear and e.g., disturb 1762 the multiparty session by emitting a continuous flow of text. They 1763 may also send text that appears to originate from other participants. 1764 Counteractions should be to require secure signaling, media and 1765 authentication, and to provide higher-layer conference functions 1766 e.g., for blocking, muting, and expelling participants. 1768 Participants with malicious intentions may also try to disturb the 1769 presentation by sending incomplete or malformed control codes. 1770 Handling of text from the different sources by the receivers MUST 1771 therefore be well separated so that the effects of such actions only 1772 affect text from the source causing the action. 1774 Care should be taken that if use of the mixer is allowed for users 1775 both with and without security procedures, opens for possible attacks 1776 by both unauthenticated call participants and even eavesdropping and 1777 manipulating of content non-participants. 1779 As already stated in Section 3.18, security in media SHOULD be 1780 applied by using DTLS-SRTP [RFC5764] on the media level. 1782 Further security considerations specific for this application are 1783 specified in Section 3.18. 1785 11. Change history 1787 [RFC Editor: Please remove this section prior to publication.] 1789 11.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-20 1791 Inclusion of edits as respone to a comment by Benjamin Kaduk in 1792 section 3.16.3 to make the recovery procedure generic. 1794 Added persons to the acknowledgements and moved acknowledgements to 1795 last in the document. 1797 11.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-19 1799 Edits because of comments in a review by Francesca Palombini. 1801 Edits because of comments from Benjamin Kaduk. 1803 Proposed to not change anything because of Robert Wilton's comments. 1805 Two added sentences in the security section to meet comments by Roman 1806 Danyliw. 1808 11.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-18 1810 Edits of nits as proposed in a review by Lars Eggert. 1812 Edits as response to review by Martin Duke. 1814 11.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-17 1816 Actions on Gen-ART review comments. 1818 Actions on SecDir review comments. 1820 11.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-16 1822 Improvements in the offer/answer considerations section by adding 1823 subsections for each phase in the negotiation as requested by IANA 1824 expert review. 1826 11.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-15 1828 Actions on review comments from Jurgen Schonwalder: 1830 A bit more about congestion situations and that they are expected to 1831 be very rare. 1833 Explanation of differences in security between the conference-aware 1834 and the conference-unaware case added in security section. 1836 Presentation examples with suource labels made less confusing, and 1837 explained. 1839 Reference to T.140 inserted at first mentioning of T.140. 1841 Reference to RFC 8825 inserted to explain WebRTC 1843 Nit in wording in terminology section adjusted. 1845 11.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-14 1847 Changes from comments by Murray Kucherawy during AD review. 1849 Many SHOULD in section 4.2 on multiparty-unaware mixing changed to 1850 SHALL, and the whole section instead specified to be optional 1851 depending on the application. 1853 Some SHOULD in section 3 either explained or changed to SHALL. 1855 In order to have explainable conditions behind SHOULDs, the 1856 transmission interval in 3.4 is changed to as soon as text is 1857 available as a main principle. The call participants send with 300 1858 ms interval so that will create realistic load conditions anyway. 1860 11.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-13 1862 Changed year to 2021. 1864 Changed reference to draft on RTT in WebRTC to recently published RFC 1865 8865. 1867 Changed label brackets in example from "[]" to "()" to avoid nits 1868 comment. 1870 Changed reference "RFC 4566" to recently published "RFC 8866" 1872 11.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-12 1874 Changes according to responses on comments from Brian Rosen in 1875 Avtcore list on 2020-12-05 and -06. 1877 Changes according to responses to comments by Bernard Aboba in 1878 avtcore list 2020-12-06. 1880 Introduction of an optiona RTP multi-stream mixing method for further 1881 study as proposed by Bernard Aboba. 1883 Changes clarifying how to open and close T.140 data channels included 1884 in 6.2 after comments by Lorenzo Miniero. 1886 Changes to satisfy nits check. Some "not" changed to "NOT" in 1887 normative wording combinations. Some lower case normative words 1888 changed to upper case. A normative reference deleted from the 1889 abstract. Two informative documents moved from normative references 1890 to informative references. 1892 11.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11 1894 Timestamps and timestamp offsets added to the packet examples in 1895 section 3.23, and the description corrected. 1897 A number of minor corrections added in sections 3.10 - 3.23. 1899 11.11. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10 1901 The packet composition was modified for interleaving packets from 1902 different sources. 1904 The packet reception was modified for the new interleaving method. 1906 The packet sequence examples was adjusted for the new interleaving 1907 method. 1909 Modifications according to responses to Brian Rosen of 2020-11-03 1911 11.12. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09 1913 Changed name on the SDP media attribute to "rtt-mixer" 1915 Restructure of section 2 for balance between aware and unaware cases. 1917 Moved conference control to own section. 1919 Improved clarification of recovery and loss in the packet sequence 1920 example. 1922 A number of editorial corrections and improvements. 1924 11.13. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 1926 Deleted the method requiring a new packet format "text/rex" because 1927 of the longer standardization and implementation period it needs. 1929 Focus on use of RFC 4103 text/red format with shorter transmission 1930 interval, and source indicated in CSRC. 1932 11.14. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 1934 Added a method based on the "text/red" format and single source per 1935 packet, negotiated by the "rtt-mixer" SDP attribute. 1937 Added reasoning and recommendation about indication of loss. 1939 The highest number of sources in one packet is 15, not 16. Changed. 1941 Added in information on update to RFC 4103 that RFC 4103 explicitly 1942 allows addition of FEC method. The redundancy is a kind of forward 1943 error correction. 1945 11.15. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 1947 Improved definitions list format. 1949 The format of the media subtype parameters is made to match the 1950 requirements. 1952 The mapping of media subtype parameters to SDP is included. 1954 The "cps" parameter belongs to the t140 subtype and does not need to 1955 be registered here. 1957 11.16. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 1959 nomenclature and editorial improvements 1961 "this document" used consistently to refer to this document. 1963 11.17. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 1965 'Redundancy header' renamed to 'data header'. 1967 More clarifications added. 1969 Language and figure number corrections. 1971 11.18. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 1973 Mention possible need to mute and raise hands as for other media. 1974 ---done ---- 1976 Make sure that use in two-party calls is also possible and explained. 1977 - may need more wording - 1979 Clarify the RTT is often used together with other media. --done-- 1981 Tell that text mixing is N-1. A users own text is not received in 1982 the mix. -done- 1984 In 3. correct the interval to: A "text/rex" transmitter SHOULD send 1985 packets distributed in time as long as there is something (new or 1986 redundant T140blocks) to transmit. The maximum transmission interval 1987 SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 1988 receiver as soon as new text to that receiver is available, as long 1989 as the time after the latest sent packet to the same receiver is more 1990 than 150 ms, and also the maximum character rate to the receiver is 1991 not exceeded. The intention is to keep the latency low while keeping 1992 a good protection against text loss in bursty packet loss conditions. 1993 -done- 1995 In 1.3 say that the format is used both ways. -done- 1997 In 13.1 change presentation area to presentation field so that reader 1998 does not think it shall be totally separated. -done- 2000 In Performance and intro, tell the performance in number of 2001 simultaneous sending users and introduced delay 16, 150 vs 2002 requirements 5 vs 500. -done -- 2004 Clarify redundancy level per connection. -done- 2006 Timestamp also for the last data header. To make it possible for all 2007 text to have time offset as for transmission from the source. Make 2008 that header equal to the others. -done- 2010 Mixer always use the CSRC list, even for its own BOM. -done- 2012 Combine all talk about transmission interval (300 ms vs when text has 2013 arrived) in section 3 in one paragraph or close to each other. -done- 2014 Documents the goal of good performance with low delay for 5 2015 simultaneous typers in the introduction. -done- 2017 Describe better that only primary text shall be sent on to receivers. 2018 Redundancy and loss must be resolved by the mixer. -done- 2020 11.19. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 2022 SDP and better description and visibility of security by OSRTP RFC 2023 8634 needed. 2025 The description of gatewaying to WebRTC extended. 2027 The description of the data header in the packet is improved. 2029 11.20. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 2031 2,5,6 More efficient format "text/rex" introduced and attribute 2032 a=rtt-mix deleted. 2034 3. Brief about use of OSRTP for security included- More needed. 2036 4. Brief motivation for the solution and why not rtp-translator is 2037 used added to intro. 2039 7. More limitations for the multiparty-unaware mixing method 2040 inserted. 2042 8. Updates to RFC 4102 and 4103 more clearly expressed. 2044 9. Gateway to WebRTC started. More needed. 2046 11.21. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 2047 to draft-ietf-avtcore-multi-party-rtt-mix-00 2049 Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00 2051 Replaced CDATA in IANA registration table with better coding. 2053 Converted to xml2rfc version 3. 2055 11.22. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 2056 to -03 2058 Changed company and e-mail of the author. 2060 Changed title to "RTP-mixer formatting of multi-party Real-time text" 2061 to better match contents. 2063 Check and modification where needed of use of RFC 2119 words SHALL 2064 etc. 2066 More about the CC value in sections on transmitters and receivers so 2067 that 1-to-1 sessions do not use the mixer format. 2069 Enhanced section on presentation for multiparty-unaware endpoints 2071 A paragraph recommending cps=150 inserted in the performance section. 2073 11.23. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 2074 to -02 2076 In Abstract and 1. Introduction: Introduced wording about regulatory 2077 requirements. 2079 In section 5: The transmission interval is decreased to 100 ms when 2080 there is text from more than one source to transmit. 2082 In section 11 about SDP negotiation, a SHOULD-requirement is 2083 introduced that the mixer should make a mix for multiparty-unaware 2084 endpoints if the negotiation is not successful. And a reference to a 2085 later chapter about it. 2087 The presentation considerations chapter 14 is extended with more 2088 information about presentation on multiparty-aware endpoints, and a 2089 new section on the multiparty-unaware mixing with low functionality 2090 but SHOULD be implemented in mixers. Presentation examples are 2091 added. 2093 A short chapter 15 on gateway considerations is introduced. 2095 Clarification about the text/t140 format included in chapter 10. 2097 This sentence added to the chapter 10 about use without redundancy. 2098 "The text/red format SHOULD be used unless some other protection 2099 against packet loss is utilized, for example a reliable network or 2100 transport." 2102 Note about deviation from RFC 2198 added in chapter 4. 2104 In chapter 9. "Use with SIP centralized conferencing framework" the 2105 following note is inserted: Note: The CSRC-list in an RTP packet only 2106 includes participants whose text is included in one or more text 2107 blocks. It is not the same as the list of participants in a 2108 conference. With audio and video media, the CSRC-list would often 2109 contain all participants who are not muted whereas text participants 2110 that don't type are completely silent and so don't show up in RTP 2111 packet CSRC-lists. 2113 11.24. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 2114 to -01 2116 Editorial cleanup. 2118 Changed capability indication from fmtp-parameter to SDP attribute 2119 "rtt-mix". 2121 Swapped order of redundancy elements in the example to match reality. 2123 Increased the SDP negotiation section 2125 12. References 2127 12.1. Normative References 2129 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 2130 Requirement Levels", BCP 14, RFC 2119, 2131 DOI 10.17487/RFC2119, March 1997, 2132 . 2134 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 2135 Jacobson, "RTP: A Transport Protocol for Real-Time 2136 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 2137 July 2003, . 2139 [RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type", 2140 RFC 4102, DOI 10.17487/RFC4102, June 2005, 2141 . 2143 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 2144 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 2145 . 2147 [RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session 2148 Initiation Protocol (SIP)", RFC 5630, 2149 DOI 10.17487/RFC5630, October 2009, 2150 . 2152 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 2153 Security (DTLS) Extension to Establish Keys for the Secure 2154 Real-time Transport Protocol (SRTP)", RFC 5764, 2155 DOI 10.17487/RFC5764, May 2010, 2156 . 2158 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 2159 Keeping Alive the NAT Mappings Associated with RTP / RTP 2160 Control Protocol (RTCP) Flows", RFC 6263, 2161 DOI 10.17487/RFC6263, June 2011, 2162 . 2164 [RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M. 2165 Thomson, "Session Traversal Utilities for NAT (STUN) Usage 2166 for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675, 2167 October 2015, . 2169 [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2170 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, 2171 May 2017, . 2173 [RFC8865] Holmberg, C. and G. Hellström, "T.140 Real-Time Text 2174 Conversation over WebRTC Data Channels", RFC 8865, 2175 DOI 10.17487/RFC8865, January 2021, 2176 . 2178 [RFC8866] Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP: 2179 Session Description Protocol", RFC 8866, 2180 DOI 10.17487/RFC8866, January 2021, 2181 . 2183 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 2184 multimedia application text conversation", February 1998, 2185 . 2187 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 2188 Protocol for multimedia application text conversation", 2189 February 2000, 2190 . 2192 12.2. Informative References 2194 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 2195 Session Initiation Protocol (SIP)", RFC 4353, 2196 DOI 10.17487/RFC4353, February 2006, 2197 . 2199 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 2200 Session Initiation Protocol (SIP) Event Package for 2201 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 2202 2006, . 2204 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 2205 (SIP) Call Control - Conferencing for User Agents", 2206 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 2207 . 2209 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 2210 Time Text over IP Using the Session Initiation Protocol 2211 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 2212 . 2214 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 2215 DOI 10.17487/RFC7667, November 2015, 2216 . 2218 [RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. 2219 Stach, "An Opportunistic Approach for Secure Real-time 2220 Transport Protocol (OSRTP)", RFC 8643, 2221 DOI 10.17487/RFC8643, August 2019, 2222 . 2224 [RFC8723] Jennings, C., Jones, P., Barnes, R., and A.B. Roach, 2225 "Double Encryption Procedures for the Secure Real-Time 2226 Transport Protocol (SRTP)", RFC 8723, 2227 DOI 10.17487/RFC8723, April 2020, 2228 . 2230 [RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for 2231 Browser-Based Applications", RFC 8825, 2232 DOI 10.17487/RFC8825, January 2021, 2233 . 2235 Acknowledgements 2237 The author want to thank the following persons for support, reviews 2238 and valuable comments: Bernard Aboba, Amanda Baber, Roman Danyliw, 2239 Spencer Dawkins, Martin Duke, Lars Eggert, James Hamlin, Benjamin 2240 Kaduk, Murray Kucherawy, Paul Kyziwat, Jonathan Lennox, Lorenzo 2241 Miniero, Dan Mongrain, Francesca Palombini, Colin Perkins, Brian 2242 Rosen, Juergen Schoenwaelder, Rich Salz, Robert Wilton, Dale Worley, 2243 Peter Yee and Yong Xin. 2245 Author's Address 2246 Gunnar Hellstrom 2247 Gunnar Hellstrom Accessible Communication 2248 SE-13670 Vendelso 2249 Sweden 2251 Email: gunnar.hellstrom@ghaccess.se