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(See the Legal Provisions document at https://trustee.ietf.org/license-info for more information.) -- The document date (25 May 2021) is 1066 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 271, but not defined == Missing Reference: 'Bob' is mentioned on line 1564, 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) 25 May 2021 5 Intended status: Standards Track 6 Expires: 26 November 2021 8 RTP-mixer formatting of multiparty Real-time text 9 draft-ietf-avtcore-multi-party-rtt-mix-19 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 26 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. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 38 120 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 38 121 10.1. Registration of the "rtt-mixer" SDP media attribute . . 38 122 11. Security Considerations . . . . . . . . . . . . . . . . . . . 39 123 12. Change history . . . . . . . . . . . . . . . . . . . . . . . 40 124 12.1. Changes included in 125 draft-ietf-avtcore-multi-party-rtt-mix-19 . . . . . . . 40 126 12.2. Changes included in 127 draft-ietf-avtcore-multi-party-rtt-mix-18 . . . . . . . 40 128 12.3. Changes included in 129 draft-ietf-avtcore-multi-party-rtt-mix-17 . . . . . . . 40 130 12.4. Changes included in 131 draft-ietf-avtcore-multi-party-rtt-mix-16 . . . . . . . 40 132 12.5. Changes included in 133 draft-ietf-avtcore-multi-party-rtt-mix-15 . . . . . . . 40 134 12.6. Changes included in 135 draft-ietf-avtcore-multi-party-rtt-mix-14 . . . . . . . 41 136 12.7. Changes included in 137 draft-ietf-avtcore-multi-party-rtt-mix-13 . . . . . . . 41 138 12.8. Changes included in 139 draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . . 41 140 12.9. Changes included in 141 draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 42 143 12.10. Changes included in 144 draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 42 145 12.11. Changes included in 146 draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 42 147 12.12. Changes included in 148 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 43 149 12.13. Changes included in 150 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 43 151 12.14. Changes included in 152 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 43 153 12.15. Changes included in 154 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 43 155 12.16. Changes included in 156 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 43 157 12.17. Changes included in 158 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 44 159 12.18. Changes included in 160 draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 45 161 12.19. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 45 162 12.20. Changes from 163 draft-hellstrom-avtcore-multi-party-rtt-source-03 to 164 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 45 165 12.21. Changes from 166 draft-hellstrom-avtcore-multi-party-rtt-source-02 to 167 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 45 168 12.22. Changes from 169 draft-hellstrom-avtcore-multi-party-rtt-source-01 to 170 -02 . . . . . . . . . . . . . . . . . . . . . . . . . . 46 171 12.23. Changes from 172 draft-hellstrom-avtcore-multi-party-rtt-source-00 to 173 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 47 174 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 47 175 13.1. Normative References . . . . . . . . . . . . . . . . . . 47 176 13.2. Informative References . . . . . . . . . . . . . . . . . 48 177 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 49 179 1. Introduction 181 "RTP Payload for Text Conversation" [RFC4103] specifies use of the 182 Real-Time Transport Protocol (RTP) [RFC3550] for transmission of 183 real-time text (RTT) and the "text/t140" format. It also specifies a 184 redundancy format "text/red" for increased robustness. The "text/ 185 red" format is registered in [RFC4102]. 187 Real-time text is usually provided together with audio and sometimes 188 with video in conversational sessions. 190 A requirement related to multiparty sessions from the presentation 191 level standard T.140 [T140] for real-time text is: "The display of 192 text from the members of the conversation should be arranged so that 193 the text from each participant is clearly readable, and its source 194 and the relative timing of entered text is visualized in the 195 display." 197 Another requirement is that the mixing procedure must not introduce 198 delays in the text streams that are experienced to be disturbing the 199 real-time experience of the receiving users. 201 Use of RTT is increasing, and specifically, use in emergency calls is 202 increasing. Emergency call use requires multiparty mixing because it 203 is common that one agent needs to transfer the call to another 204 specialized agent but is obliged to stay on the call at least to 205 verify that the transfer was successful. Mixer implementations for 206 RFC 4103 "RTP Payload for Text Conversation" can use traditional RFC 207 3550 RTP functions for mixing and source identification, but the 208 performance of the mixer when giving turns for the different sources 209 to transmit is limited when using the default transmission 210 characteristics with redundancy. 212 The redundancy scheme of [RFC4103] enables efficient transmission of 213 earlier transmitted redundant text in packets together with new text. 214 However, the redundancy header format has no source indicators for 215 the redundant transmissions. The redundant parts in a packet must 216 therefore be from the same source as the new text. The recommended 217 transmission is one new and two redundant generations of text 218 (T140blocks) in each packet and the recommended transmission interval 219 for two-party use is 300 ms. 221 Real-time text mixers for multiparty sessions need to include the 222 source with each transmitted group of text from a conference 223 participant so that the text can be transmitted interleaved with text 224 groups from different sources at the rate they are created. This 225 enables the text groups to be presented by endpoints in suitable 226 grouping with other text from the same source. 228 The presentation can then be arranged so that text from different 229 sources can be presented in real-time and easily read. At the same 230 time it is possible for a reading user to perceive approximately when 231 the text was created in real time by the different parties. The 232 transmission and mixing is intended to be done in a general way, so 233 that presentation can be arranged in a layout decided by the 234 endpoint. 236 There are existing implementations of RFC 4103 in endpoints without 237 the updates from this document. These will not be able to receive 238 and present real-time text mixed for multiparty-aware endpoints. 240 A negotiation mechanism is therefore needed for verification if the 241 parties are able to handle a common method for multiparty 242 transmission and agreeing on using that method. 244 A fallback mixing procedure is also needed for cases when the 245 negotiation result indicates that a receiving endpoint is not capable 246 of handling the mixed format. Multiparty-unaware endpoints would 247 possibly otherwise present all received multiparty mixed text as if 248 it came from the same source regardless of any accompanying source 249 indication coded in fields in the packet. Or they may have other 250 undesirable ways of acting on the multiparty content. The fallback 251 method is called the mixing procedure for multiparty-unaware 252 endpoints. The fallback method is naturally not expected to meet all 253 performance requirements placed on the mixing procedure for 254 multiparty-aware endpoints. 256 The document updates [RFC4103] by introducing an attribute for 257 declaring support of the RTP-mixer-based multiparty mixing case and 258 rules for source indications and interleaving of text from different 259 sources. 261 1.1. Terminology 263 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 264 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 265 "OPTIONAL" in this document are to be interpreted as described in BCP 266 14 [RFC2119] [RFC8174] when, and only when, they appear in all 267 capitals, as shown above. 269 The terms Source Description (SDES), Canonical name (CNAME), Name 270 (NAME), Synchronization Source (SSRC), Contributing Source (CSRC), 271 CSRC list, CSRC count [CC], Real-Time control protocol (RTCP), RTP- 272 mixer, RTP-translator are defined in [RFC3550]. 274 The term "T140block" is defined in [RFC4103] to contain one or more 275 T.140 code elements. 277 "TTY" stands for a textphone type used in North America. 279 Web based real-time communication (WebRTC) is specified by the World 280 Wide Web Consortium (W3C) and IETF. See [RFC8825]. 282 "DTLS-SRTP" is a Datagram Transport Layer Security (DTLS) extension 283 for use with Secure Real-Time Transport Protocol/Secure Real-Time 284 Control Protocol (SRTP/SRTCP) specified in [RFC5764]. 286 "multiparty-aware" describes an endpoint receiving real-time text 287 from multiple sources through a common conference mixer being able to 288 present the text in real-time, separated by source, and presented so 289 that a user can get an impression of the approximate relative timing 290 of text from different parties. 292 "multiparty-unaware" describes an endpoint not itself being able to 293 separate text from different sources when received through a common 294 conference mixer. 296 1.2. Selected solution and considered alternatives 298 A number of alternatives were considered when searching an efficient 299 and easily implemented multiparty method for real-time text. This 300 section explains a few of them briefly. 302 Multiple RTP streams, one per participant 303 One RTP stream per source would be sent in the same RTP session 304 with the "text/red" format. From some points of view, use of 305 multiple RTP streams, one for each source, sent in the same RTP 306 session would be efficient, and would use exactly the same packet 307 format as [RFC4103] and the same payload type. A couple of 308 relevant scenarios using multiple RTP-streams are specified in 309 "RTP Topologies" [RFC7667]. One possibility of special interest 310 is the Selective Forwarding Middlebox (SFM) topology specified in 311 RFC 7667 section 3.7 that could enable end-to-end encryption. In 312 contrast to audio and video, real-time text is only transmitted 313 when the users actually transmit information. Thus, an SFM 314 solution would not need to exclude any party from transmission 315 under normal conditions. In order to allow the mixer to convey 316 the packets with the payload preserved and encrypted, an SFM 317 solution would need to act on some specific characteristics of the 318 "text/red" format. The redundancy headers are part of the 319 payload, so the receiver would need to just assume that the 320 payload type number in the redundancy header is for "text/t140". 321 The characters per second parameter (cps) would need to act per 322 stream. The relation between the SSRC and the source would need 323 to be conveyed in some specified way, e.g., in the CSRC. Recovery 324 and loss detection would preferably be based on sequence number 325 gap detection. Thus, sequence number gaps in the incoming stream 326 to the mixer would need to be reflected in the stream to the 327 participant, with no new gaps created by the mixer. However, the 328 RTP implementation in both mixers and endpoints need to support 329 multiple streams in the same RTP session in order to use this 330 mechanism. For best deployment opportunity, it should be possible 331 to upgrade existing endpoint solutions to be multiparty-aware with 332 a reasonable effort. There is currently a lack of support for 333 multi-stream RTP in certain implementations. This fact led to 334 this solution being only briefly mentioned in this document as an 335 option for further study. 337 RTP-mixer-based method for multiparty-aware endpoints 338 The "text/red" format in RFC 4103 is sent with a shorter 339 transmission interval with the RTP-mixer method and indicating the 340 source in the CSRC field. The "text/red" format with a "text/ 341 t140" payload in a single RTP stream can be sent when text is 342 available from the call participants instead of at the regular 300 343 ms. Transmission of packets with text from different sources can 344 then be done smoothly while simultaneous transmission occurs as 345 long as it is not limited by the maximum character rate "cps". 346 With ten participants sending text simultaneously, the switching 347 and transmission performance is good. With more simultaneously 348 sending participants, and with receivers having the default 349 capacity there will be a noticeable jerkiness and delay in text 350 presentation. The jerkiness will be more expressed the more 351 participants who send text simultaneously. Two seconds jerkiness 352 will be noticeable and slightly unpleasant, but it corresponds in 353 time to what typing humans often cause by hesitation or changing 354 position while typing. A benefit of this method is that no new 355 packet format needs to be introduced and implemented. Since 356 simultaneous typing by more than two parties is expected to be 357 very rare as described in Section 1.3, this method can be used 358 successfully with good performance. Recovery of text in case of 359 packet loss is based on analysis of timestamps of received 360 redundancy versus earlier received text. Negotiation is based on 361 a new SDP media attribute "rtt-mixer". This method is selected to 362 be the main one specified in this document. 364 Multiple sources per packet 365 A new "text" media subtype would be specified with up to 15 366 sources in each packet. The mechanism would make use of the RTP 367 mixer model specified in RTP [RFC3550]. The sources are indicated 368 in strict order in the CSRC list of the RTP packets. The CSRC 369 list can have up to 15 members. Therefore, text from up to 15 370 sources can be included in each packet. Packets are normally sent 371 with 300 ms intervals. The mean delay will be 150 ms. A new 372 redundancy packet format is specified. This method would result 373 in good performance, but would require standardization and 374 implementation of new releases in the target technologies that 375 would take more time than desirable to complete. It was therefore 376 not selected to be included in this document. 378 Mixing for multiparty-unaware endpoints 379 Presentation of text from multiple parties is prepared by the 380 mixer in one single stream. It is desirable to have a method that 381 does not require any modifications in existing user devices 382 implementing RFC 4103 for RTT without explicit support of 383 multiparty sessions. This is possible by having the mixer insert 384 a new line and a text formatted source label before each switch of 385 text source in the stream. Switch of source can only be done in 386 places in the text where it does not disturb the perception of the 387 contents. Text from only one source can be presented in real time 388 at a time. The delay will therefore vary. The method also has 389 other limitations, but is included in this document as a fallback 390 method. In calls where parties take turns properly by ending 391 their entries with a new line, the limitations will have limited 392 influence on the user experience. when only two parties send text, 393 these two will see the text in real time with no delay. This 394 method is specified as a fallback method in this document. 396 RTT transport in WebRTC 397 Transport of real-time text in the WebRTC technology is specified 398 to use the WebRTC data channel in [RFC8865]. That specification 399 contains a section briefly describing its use in multiparty 400 sessions. The focus of this document is RTP transport. 401 Therefore, even if the WebRTC transport provides good multiparty 402 performance, it is just mentioned in this document in relation to 403 providing gateways with multiparty capabilities between RTP and 404 WebRTC technologies. 406 1.3. Intended application 408 The method for multiparty real-time text specified in this document 409 is primarily intended for use in transmission between mixers and 410 endpoints in centralized mixing configurations. It is also 411 applicable between mixers. An often mentioned application is for 412 emergency service calls with real-time text and voice, where a call 413 taker wants to make an attended handover of a call to another agent, 414 and stay to observe the session. Multimedia conference sessions with 415 support for participants to contribute in text is another 416 application. Conferences with central support for speech-to-text 417 conversion is yet another mentioned application. 419 In all these applications, normally only one participant at a time 420 will send long text utterances. In some cases, one other participant 421 will occasionally contribute with a longer comment simultaneously. 422 That may also happen in some rare cases when text is interpreted to 423 text in another language in a conference. Apart from these cases, 424 other participants are only expected to contribute with very brief 425 utterings while others are sending text. 427 Users expect that the text they send is presented in real-time in a 428 readable way to the other participants even if they send 429 simultaneously with other users and even when they make brief edit 430 operations of their text by backspacing and correcting their text. 432 Text is supposed to be human generated, by some text input means, 433 such as typing on a keyboard or using speech-to-text technology. 434 Occasional small cut-and-paste operations may appear even if that is 435 not the initial purpose of real-time text. 437 The real-time characteristics of real-time text is essential for the 438 participants to be able to contribute to a conversation. If the text 439 is too much delayed from typing a letter to its presentation, then, 440 in some conference situations, the opportunity to comment will be 441 gone and someone else will grab the turn. A delay of more than one 442 second in such situations is an obstacle for good conversation. 444 2. Overview of the two specified solutions and selection of method 446 This section contains a brief introduction of the two methods 447 specified in this document. 449 2.1. The RTP-mixer-based solution for multiparty-aware endpoints 451 This method specifies negotiated use of the RFC 4103 format for 452 multiparty transmission in a single RTP stream. The main purpose of 453 this document is to specify a method for true multiparty real-time 454 text mixing for multiparty-aware endpoints that can be widely 455 deployed. The RTP-mixer-based method makes use of the current format 456 for real-time text in [RFC4103]. It is an update of RFC 4103 by a 457 clarification on one way to use it in the multiparty situation. That 458 is done by completing a negotiation for this kind of multiparty 459 capability and by interleaving packets from different sources. The 460 source is indicated in the CSRC element in the RTP packets. Specific 461 considerations are made to be able to recover text after packet loss. 463 The detailed procedures for the RTP-mixer-based multiparty-aware case 464 are specified in Section 3. 466 Please use [RFC4103] as reference when reading the specification. 468 2.2. Mixing for multiparty-unaware endpoints 470 A method is also specified in this document for cases when the 471 endpoint participating in a multiparty call does not itself implement 472 any solution, or not the same, as the mixer. The method requires the 473 mixer to insert text dividers and readable labels and only send text 474 from one source at a time until a suitable point appears for source 475 change. This solution is a fallback method with functional 476 limitations. It acts on the presentation level. 478 A mixer SHOULD by default format and transmit text to a call 479 participant to be suitable to present on a multiparty-unaware 480 endpoint which has not negotiated any method for true multiparty RTT 481 handling, but negotiated a "text/red" or "text/t140" format in a 482 session. This SHOULD be done if nothing else is specified for the 483 application in order to maintain interoperability. Section 4.2 484 specifies how this mixing is done. 486 2.3. Offer/answer considerations 488 RTP Payload for Text Conversation [RFC4103] specifies use of RTP 489 [RFC3550], and a redundancy format "text/red" for increased 490 robustness of real-time text transmission. This document updates 491 [RFC4103] by introducing a capability negotiation for handling 492 multiparty real-time text, a way to indicate the source of 493 transmitted text, and rules for efficient timing of the transmissions 494 interleaved from different sources. 496 The capability negotiation for the "RTP-mixer-based multiparty 497 method" is based on use of the SDP media attribute "rtt-mixer". 499 The syntax is as follows: 500 "a=rtt-mixer" 502 If any other method for RTP-based multiparty real-time text gets 503 specified by additional work, it is assumed that it will be 504 recognized by some specific SDP feature exchange. 506 2.3.1. Initial offer 508 A party intending to set up a session and being willing to use the 509 RTP-mixer-based method of this specification for sending or receiving 510 or both sending and receiving real-time text SHALL include the "rtt- 511 mixer" SDP attribute in the corresponding "text" media section in the 512 initial offer. 514 The party MAY indicate capability for both the RTP-mixer-based method 515 of this specification and other methods. 517 When the offeror has sent the offer including the "rtt-mixer" 518 attribute, it MUST be prepared to receive and handle real-time text 519 formatted according to both the method for multiparty-aware parties 520 specified in Section 3 in this specification and two-party formatted 521 real-time text. 523 2.3.2. Answering the offer 525 A party receiving an offer containing the "rtt-mixer" SDP attribute 526 and being willing to use the RTP-mixer-based method of this 527 specification for sending or receiving or both sending and receiving 528 SHALL include the "rtt-mixer" SDP attribute in the corresponding 529 "text" media section in the answer. 531 If the offer did not contain the "rtt-mixer" attribute, the answer 532 MUST NOT contain the "rtt-mixer" attribute. 534 Even when the "rtt-mixer" attribute is successfully negotiated, the 535 parties MAY send and receive two-party coded real-time text. 537 An answer MUST NOT include acceptance of more than one method for 538 multiparty real-time text in the same RTP session. 540 When the answer including acceptance is transmitted, the answerer 541 MUST be prepared to act on received text in the negotiated session 542 according to the method for multiparty-aware parties specified in 543 Section 3 of this specification. Reception of text for a two-party 544 session SHALL also be supported. 546 2.3.3. Offeror processing the answer 548 When the answer is processed by the offeror, it MUST act as specified 549 in Section 2.4 551 2.3.4. Modifying a session 553 A session MAY be modified at any time by any party offering a 554 modified SDP with or without the "rtt-mixer" SDP attribute expressing 555 a desired change in the support of multiparty real-time text. 557 If the modified offer adds indication of support for multiparty real- 558 time text by including the "rtt-mixer" SDP attribute, the procedures 559 specified in the previous subsections SHALL be applied. 561 If the modified offer deletes indication of support for multiparty 562 real-time text by excluding the "rtt-mixer" SDP attribute, the answer 563 MUST NOT contain the "rtt-mixer" attribute. After processing this 564 SDP exchange, the parties MUST NOT send real-time text formatted for 565 multiparty-aware parties according to this specification. 567 2.4. Actions depending on capability negotiation result 569 A transmitting party SHALL send text according to the RTP-mixer-based 570 multiparty method only when the negotiation for that method was 571 successful and when it conveys text for another source. In all other 572 cases, the packets SHALL be populated and interpreted as for a two- 573 party session. 575 A party which has negotiated the "rtt-mixer" SDP media attribute MUST 576 populate the CSRC-list, and format the packets according to Section 3 577 if it acts as an rtp-mixer and sends multiparty text. 579 A party which has negotiated the "rtt-mixer" SDP media attribute MUST 580 interpret the contents of the "CC" field, the CSRC-list and the 581 packets according to Section 3 in received RTP packets in the 582 corresponding RTP stream. 584 A party which has not successfully completed the negotiation of the 585 "rtt-mixer" SDP media attribute MUST NOT transmit packets interleaved 586 from different sources in the same RTP stream as specified in 587 Section 3. If the party is a mixer and did declare the "rtt-mixer" 588 SDP media attribute, it SHOULD perform the procedure for multiparty- 589 unaware endpoints. If the party is not a mixer, it SHOULD transmit 590 as in a two-party session according to [RFC4103]. 592 3. Details for the RTP-mixer-based mixing method for multiparty-aware 593 endpoints 595 3.1. Use of fields in the RTP packets 597 The CC field SHALL show the number of members in the CSRC list, which 598 SHALL be one (1) in transmissions from a mixer when conveying text 599 from other sources in a multiparty session, and otherwise 0. 601 When text is conveyed by a mixer during a multiparty session, a CSRC 602 list SHALL be included in the packet. The single member in the CSRC- 603 list SHALL contain the SSRC of the source of the T140blocks in the 604 packet. 606 When redundancy is used, the RECOMMENDED level of redundancy is to 607 use one primary and two redundant generations of T140blocks. In some 608 cases, a primary or redundant T140block is empty, but is still 609 represented by a member in the redundancy header. 611 In other regards, the contents of the RTP packets are equal to what 612 is specified in [RFC4103]. 614 3.2. Initial transmission of a BOM character 616 As soon as a participant is known to participate in a session with 617 another entity and is available for text reception, a Unicode Byte- 618 Order Mark (BOM) character SHALL be sent to it by the other entity 619 according to the procedures in this section. This is useful in many 620 configurations to open ports and firewalls and setting up the 621 connection between the application and the network. If the 622 transmitter is a mixer, then the source of this character SHALL be 623 indicated to be the mixer itself. 625 Note that the BOM character SHALL be transmitted with the same 626 redundancy procedures as any other text. 628 3.3. Keep-alive 630 After that, the transmitter SHALL send keep-alive traffic to the 631 receiver(s) at regular intervals when no other traffic has occurred 632 during that interval, if that is decided for the actual connection. 633 It is RECOMMENDED to use the keep-alive solution from [RFC6263]. The 634 consent check of [RFC7675] is a possible alternative if it is used 635 anyway for other reasons. 637 3.4. Transmission interval 639 A "text/red" or "text/t140" transmitter in a mixer SHALL send packets 640 distributed in time as long as there is something (new or redundant 641 T140blocks) to transmit. The maximum transmission interval between 642 text transmissions from the same source SHALL then be 330 ms, when no 643 other limitations cause a longer interval to be temporarily used. It 644 is RECOMMENDED to send the next packet to a receiver as soon as new 645 text to that receiver is available, as long as the mean character 646 rate of new text to the receiver calculated over the last 10 one- 647 second intervals does not exceed the "cps" value of the receiver. 648 The intention is to keep the latency low and network load limited 649 while keeping good protection against text loss in bursty packet loss 650 conditions. The main purpose of the 330 ms interval is for timing of 651 redundant transmission, when no new text from the same source is 652 available. 654 The reason for the value 330 ms is that many sources of text will 655 transmit new text with 300 ms intervals during periods of continuous 656 user typing, and then reception in the mixer of such new text will 657 cause a combined transmission of the new text and the unsent 658 redundancy from the previous transmission. Only when the user stops 659 typing, the 330 ms interval will be applied to send the redundancy. 661 If the Characters Per Second (cps) value is reached, a longer 662 transmission interval SHALL be applied for text from all sources as 663 specified in [RFC4103] and only as much of the text queued for 664 transmission SHALL be sent at the end of each transmission interval 665 as can be allowed without exceeding the "cps" value. Division of 666 text for partial transmission MUST then be made at T140block borders. 667 When the transmission rate falls under the "cps" value again, the 668 transmission intervals SHALL be returned to 330 ms and transmission 669 of new text SHALL return to be made as soon as new text is available. 671 NOTE: that extending the transmission intervals during high load 672 periods does not change the number of characters to be conveyed. It 673 just evens out the load in time and reduces the number of packets per 674 second. With human created conversational text, the sending user 675 will eventually take a pause letting transmission catch up. 677 See also Section 8. 679 For a transmitter not acting as a mixer, the transmission interval 680 principles from [RFC4103] apply, and the normal transmission interval 681 SHALL be 300 ms. 683 3.5. Only one source per packet 685 New text and redundant copies of earlier text from one source SHALL 686 be transmitted in the same packet if available for transmission at 687 the same time. Text from different sources MUST NOT be transmitted 688 in the same packet. 690 3.6. Do not send received text to the originating source 692 Text received by a mixer from a participant SHOULD NOT be included in 693 transmission from the mixer to that participant, because the normal 694 behavior of the endpoint is to present locally-produced text locally. 696 3.7. Clean incoming text 698 A mixer SHALL handle reception, recovery from packet loss, deletion 699 of superfluous redundancy, marking of possible text loss and deletion 700 of 'BOM' characters from each participant before queueing received 701 text for transmission to receiving participants as specified in 702 [RFC4103] for single-party sources and Section 3.16 for multiparty 703 sources (chained mixers). 705 3.8. Redundant transmission principles 707 A transmitting party using redundancy SHALL send redundant 708 repetitions of T140blocks already transmitted in earlier packets. 710 The number of redundant generations of T140blocks to include in 711 transmitted packets SHALL be deduced from the SDP negotiation. It 712 SHALL be set to the minimum of the number declared by the two parties 713 negotiating a connection. It is RECOMMENDED to declare and transmit 714 one original and two redundant generations of the T140blocks because 715 that provides good protection against text loss in case of packet 716 loss, and low overhead. 718 3.9. Text placement in packets 720 The mixer SHALL compose and transmit an RTP packet to a receiver when 721 one or more of the following conditions have occurred: 723 * The transmission interval is the normal 330 ms and there is newly 724 received unsent text available for transmission to that receiver. 726 * The current transmission interval has passed and is longer than 727 the normal 330 ms and there is newly received unsent text 728 available for transmission to that receiver. 730 * The current transmission interval ( normally 330 ms) has passed 731 since already transmitted text was queued for transmission as 732 redundant text. 734 The principles from [RFC4103] apply for populating the header, the 735 redundancy header and the data in the packet with specifics specified 736 here and in the following sections. 738 At the time of transmission, the mixer SHALL populate the RTP packet 739 with all T140blocks queued for transmission originating from the 740 source in turn for transmission as long as this is not in conflict 741 with the allowed number of characters per second ("cps") or the 742 maximum packet size. In this way, the latency of the latest received 743 text is kept low even in moments of simultaneous transmission from 744 many sources. 746 Redundant text SHALL also be included, and the assessment of how much 747 new text can be included within the maximum packet size MUST take 748 into account that the redundancy has priority to be transmitted in 749 its entirety. See Section 3.4 751 The SSRC of the source SHALL be placed as the only member in the 752 CSRC-list. 754 Note: The CSRC-list in an RTP packet only includes the participant 755 whose text is included in text blocks. It is not the same as the 756 total list of participants in a conference. With audio and video 757 media, the CSRC-list would often contain all participants who are not 758 muted whereas text participants that don't type are completely silent 759 and thus are not represented in RTP packet CSRC-lists. 761 3.10. Empty T140blocks 763 If no unsent T140blocks were available for a source at the time of 764 populating a packet, but T140blocks are available which have not yet 765 been sent the full intended number of redundant transmissions, then 766 the primary T140block for that source is composed of an empty 767 T140block, and populated (without taking up any length) in a packet 768 for transmission. The corresponding SSRC SHALL be placed as usual in 769 its place in the CSRC-list. 771 The first packet in the session, the first after a source switch, and 772 the first after a pause SHALL be populated with the available 773 T140blocks for the source in turn to be sent as primary, and empty 774 T140blocks for the agreed number of redundancy generations. 776 3.11. Creation of the redundancy 778 The primary T140block from a source in the latest transmitted packet 779 is saved for populating the first redundant T140block for that source 780 in the next transmission of text from that source. The first 781 redundant T140block for that source from the latest transmission is 782 saved for populating the second redundant T140block in the next 783 transmission of text from that source. 785 Usually this is the level of redundancy used. If a higher level of 786 redundancy is negotiated, then the procedure SHALL be maintained 787 until all available redundant levels of T140blocks are placed in the 788 packet. If a receiver has negotiated a lower number of "text/red" 789 generations, then that level SHALL be the maximum used by the 790 transmitter. 792 The T140blocks saved for transmission as redundant data are assigned 793 a planned transmission time 330 ms after the current time, but SHOULD 794 be transmitted earlier if new text for the same source gets in turn 795 for transmission before that time. 797 3.12. Timer offset fields 799 The timestamp offset values SHALL be inserted in the redundancy 800 header, with the time offset from the RTP timestamp in the packet 801 when the corresponding T140block was sent as primary. 803 The timestamp offsets are expressed in the same clock tick units as 804 the RTP timestamp. 806 The timestamp offset values for empty T140blocks have no relevance 807 but SHOULD be assigned realistic values. 809 3.13. Other RTP header fields 811 The number of members in the CSRC list (0 or 1) SHALL be placed in 812 the "CC" header field. Only mixers place value 1 in the "CC" field. 813 A value of "0" indicates that the source is the transmitting device 814 itself and that the source is indicated by the SSRC field. This 815 value is used by endpoints, and by mixers sending self-sourced data. 817 The current time SHALL be inserted in the timestamp. 819 The SSRC header field SHALL contain the SSRC of the RTP session where 820 the packet will be transmitted. 822 The M-bit SHALL be handled as specified in [RFC4103]. 824 3.14. Pause in transmission 826 When there is no new T140block to transmit, and no redundant 827 T140block that has not been retransmitted the intended number of 828 times from any source, the transmission process SHALL be stopped 829 until either new T140blocks arrive, or a keep-alive method calls for 830 transmission of keep-alive packets. 832 3.15. RTCP considerations 834 A mixer SHALL send RTCP reports with SDES, CNAME, and NAME 835 information about the sources in the multiparty call. This makes it 836 possible for participants to compose a suitable label for text from 837 each source. 839 Privacy considerations SHALL be taken when composing these fields. 840 They contain name and address information that may be sensitive to 841 transmit in its entirety, e.g., to unauthenticated participants. 843 3.16. Reception of multiparty contents 845 The "text/red" receiver included in an endpoint with presentation 846 functions will receive RTP packets in the single stream from the 847 mixer, and SHALL distribute the T140blocks for presentation in 848 presentation areas for each source. Other receiver roles, such as 849 gateways or chained mixers, are also feasible. They require 850 considerations if the stream shall just be forwarded, or distributed 851 based on the different sources. 853 3.16.1. Acting on the source of the packet contents 855 If the "CC" field value of a received packet is 1, it indicates that 856 the text is conveyed from a source indicated in the single member in 857 the CSRC-list, and the receiver MUST act on the source according to 858 its role. If the CC value is 0, the source is indicated in the SSRC 859 field. 861 3.16.2. Detection and indication of possible text loss 863 The receiver SHALL monitor the RTP sequence numbers of the received 864 packets for gaps and packets out of order. If a sequence number gap 865 appears and still exists after some defined short time for jitter and 866 reordering resolution, the packets in the gap SHALL be regarded as 867 lost. 869 If it is known that only one source is active in the RTP session, 870 then it is likely that a gap equal to or larger than the agreed 871 number of redundancy generations (including the primary) causes text 872 loss. In that case, the receiver SHALL create a t140block with a 873 marker for possible text loss [T140ad1] and associate it with the 874 source and insert it in the reception buffer for that source. 876 If it is known that more than one source is active in the RTP 877 session, then it is not possible in general to evaluate if text was 878 lost when packets were lost. With two active sources and the 879 recommended number of redundancy generations (3), it can take a gap 880 of five consecutive lost packets until any text may be lost, but text 881 loss can also appear if three non-consecutive packets are lost when 882 they contained consecutive data from the same source. A simple 883 method to decide when there is risk for resulting text loss is to 884 evaluate if three or more packets were lost within one second. If 885 this simple method is used, then a t140block SHOULD be created with a 886 marker for possible text loss [T140ad1] and associated with the SSRC 887 of the RTP session as a general input from the mixer. 889 Implementations MAY apply more refined methods for more reliable 890 detection of whether text was lost or not. Any refined method SHOULD 891 prefer marking possible loss rather than not marking when it is 892 uncertain if there was loss. 894 3.16.3. Extracting text and handling recovery 896 When applying the following procedures, the effects MUST be 897 considered of possible timestamp wrap around and the RTP session 898 possibly changing SSRC. 900 When a packet is received in an RTP session using the packetization 901 for multiparty-aware endpoints, its T140blocks SHALL be extracted in 902 the following way. The description is adapted to the default 903 redundancy case using the original and two redundant generations. 905 The source SHALL be extracted from the CSRC-list if available, 906 otherwise from the SSRC. 908 If the received packet is the first packet received from the source, 909 then all T140blocks in the packet SHALL be retrieved and assigned to 910 a receive buffer for the source beginning with the second generation 911 redundancy, continuing with the first generation redundancy and 912 finally the primary. 914 Note: The normal case is that in the first packet, only the primary 915 data has contents. The redundant data has contents in the first 916 received packet from a source only after initial packet loss. 918 If the packet is not the first packet from a source, then if the 919 second generation redundant data is available, its timestamp SHALL be 920 created by subtracting its timestamp offset from the RTP timestamp. 921 If the resulting timestamp is later than the latest retrieved data 922 from the same source, then the redundant data SHALL be retrieved and 923 appended to the receive buffer. The process SHALL be continued in 924 the same way for the first generation redundant data. After that, 925 the timestamp of the packet SHALL be compared with the timestamp of 926 the latest retrieved data from the same source and if it is later, 927 then the primary data SHALL be retrieved from the packet and appended 928 to the receive buffer for the source. 930 3.16.4. Delete 'BOM' 932 Unicode character 'BOM' is used as a start indication and sometimes 933 used as a filler or keep alive by transmission implementations. 934 These SHALL be deleted after extraction from received packets. 936 3.17. Performance considerations 938 This solution has good performance with low text delays, as long as 939 the mean number of characters per second sent during any 10-second 940 interval from a number of simultaneously sending participants to a 941 receiving participant, does not reach the "cps" value. At higher 942 numbers of sent characters per second, a jerkiness is visible in the 943 presentation of text. The solution is therefore suitable for 944 emergency service use, relay service use, and small or well-managed 945 larger multimedia conferences. Only in large unmanaged conferences 946 with a high number of participants there may on very rare occasions 947 appear situations when many participants happen to send text 948 simultaneously. In such circumstances, the result may be 949 unpleasantly jerky presentation of text from each sending 950 participant. It should be noted that it is only the number of users 951 sending text within the same moment that causes jerkiness, not the 952 total number of users with RTT capability. 954 3.18. Security for session control and media 956 Security mechanisms to provide confidentiality and integrity 957 protection and peer authentication SHOULD be applied when possible 958 regarding the capabilities of the participating devices by use of SIP 959 over TLS by default according to [RFC5630] section 3.1.3 on the 960 session control level and by default using DTLS-SRTP [RFC5764] on the 961 media level. In applications where legacy endpoints without security 962 are allowed, a negotiation SHOULD be performed to decide if 963 encryption on the media level will be applied. If no other security 964 solution is mandated for the application, then OSRTP [RFC8643] is a 965 suitable method to be applied to negotiate SRTP media security with 966 DTLS. Most SDP examples below are for simplicity expressed without 967 the security additions. The principles (but not all details) for 968 applying DTLS-SRTP [RFC5764] security are shown in a couple of the 969 following examples. 971 Further general security considerations are covered in Section 11. 973 End-to-end encryption would require further work and could be based 974 on WebRTC as specified in Section 1.2 or on double encryption as 975 specified in [RFC8723]. 977 3.19. SDP offer/answer examples 979 This section shows some examples of SDP for session negotiation of 980 the real-time text media in SIP sessions. Audio is usually provided 981 in the same session, and sometimes also video. The examples only 982 show the part of importance for the real-time text media. The 983 examples relate to the single RTP stream mixing for multiparty-aware 984 endpoints and for multiparty-unaware endpoints. 986 Note: Multiparty RTT MAY also be provided through other methods, 987 e.g., by a Selective Forwarding Middlebox (SFM). In that case, the 988 SDP of the offer will include something specific for that method, 989 e.g., an SDP attribute or another media format. An answer selecting 990 the use of that method would accept it by a corresponding 991 acknowledgement included in the SDP. The offer may contain also the 992 "rtt-mixer" SDP media attribute for the main RTT media when the 993 offeror has capability for both multiparty methods, while an answer, 994 selecting to use SFM will not include the "rtt-mixer" SDP media 995 attribute. 997 Offer example for "text/red" format and multiparty support: 999 m=text 11000 RTP/AVP 100 98 1000 a=rtpmap:98 t140/1000 1001 a=rtpmap:100 red/1000 1002 a=fmtp:100 98/98/98 1003 a=rtt-mixer 1005 Answer example from a multiparty-aware device 1006 m=text 14000 RTP/AVP 100 98 1007 a=rtpmap:98 t140/1000 1008 a=rtpmap:100 red/1000 1009 a=fmtp:100 98/98/98 1010 a=rtt-mixer 1012 Offer example for "text/red" format including multiparty 1013 and security: 1014 a=fingerprint: (fingerprint1) 1015 m=text 11000 RTP/AVP 100 98 1016 a=rtpmap:98 t140/1000 1017 a=rtpmap:100 red/1000 1018 a=fmtp:100 98/98/98 1019 a=rtt-mixer 1021 The "fingerprint" is sufficient to offer DTLS-SRTP, with the media 1022 line still indicating RTP/AVP. 1024 Note: For brevity, the entire value of the SDP fingerprint attribute 1025 is not shown in this and the following example. 1027 Answer example from a multiparty-aware device with security 1028 a=fingerprint: (fingerprint2) 1029 m=text 16000 RTP/AVP 100 98 1030 a=rtpmap:98 t140/1000 1031 a=rtpmap:100 red/1000 1032 a=fmtp:100 98/98/98 1033 a=rtt-mixer 1035 With the "fingerprint" the device acknowledges use of SRTP/DTLS. 1037 Answer example from a multiparty-unaware device that also 1038 does not support security: 1040 m=text 12000 RTP/AVP 100 98 1041 a=rtpmap:98 t140/1000 1042 a=rtpmap:100 red/1000 1043 a=fmtp:100 98/98/98 1045 3.20. Packet sequence example from interleaved transmission 1047 This example shows a symbolic flow of packets from a mixer including 1048 loss and recovery. The sequence includes interleaved transmission of 1049 text from two RTT sources A and B. P indicates primary data. R1 is 1050 first redundant generation data and R2 is the second redundant 1051 generation data. A1, B1, A2 etc. are text chunks (T140blocks) 1052 received from the respective sources and sent on to the receiver by 1053 the mixer. X indicates a dropped packet between the mixer and a 1054 receiver. The session is assumed to use original and two redundant 1055 generations of RTT. 1057 |-----------------------| 1058 |Seq no 101, Time=20400 | 1059 |CC=1 | 1060 |CSRC list A | 1061 |R2: A1, Offset=600 | 1062 |R1: A2, Offset=300 | 1063 |P: A3 | 1064 |-----------------------| 1066 Assuming that earlier packets (with text A1 and A2) were received in 1067 sequence, text A3 is received from packet 101 and assigned to 1068 reception buffer A. The mixer is now assumed to have received 1069 initial text from source B 100 ms after packet 101 and will send that 1070 text. Transmission of A2 and A3 as redundancy is planned for 330 ms 1071 after packet 101 if no new text from A is ready to be sent before 1072 that. 1074 |-----------------------| 1075 |Seq no 102, Time=20500 | 1076 |CC=1 | 1077 |CSRC list B | 1078 |R2 Empty, Offset=600 | 1079 |R1: Empty, Offset=300 | 1080 |P: B1 | 1081 |-----------------------| 1082 Packet 102 is received. 1083 B1 is retrieved from this packet. Redundant transmission of 1084 B1 is planned 330 ms after packet 102. 1086 X------------------------| 1087 X Seq no 103, Timer=20730| 1088 X CC=1 | 1089 X CSRC list A | 1090 X R2: A2, Offset=630 | 1091 X R1: A3, Offset=330 | 1092 X P: Empty | 1093 X------------------------| 1094 Packet 103 is assumed to be lost due to network problems. 1095 It contains redundancy for A. Sending A3 as second level 1096 redundancy is planned for 330 ms after packet 103. 1098 X------------------------| 1099 X Seq no 104, Timer=20800| 1100 X CC=1 | 1101 X CSRC list B | 1102 X R2: Empty, Offset=600 | 1103 X R1: B1, Offset=300 | 1104 X P: B2 | 1105 X------------------------| 1106 Packet 104 contains text from B, including new B2 and 1107 redundant B1. It is assumed dropped due to network 1108 problems. 1109 The mixer has A3 redundancy to send, but no new text 1110 appears from A and therefore the redundancy is sent 1111 330 ms after the previous packet with text from A. 1113 |------------------------| 1114 | Seq no 105, Timer=21060| 1115 | CC=1 | 1116 | CSRC list A | 1117 | R2: A3, Offset=660 | 1118 | R1: Empty, Offset=330 | 1119 | P: Empty | 1120 |------------------------| 1121 Packet 105 is received. 1122 A gap for lost packets 103 and 104 is detected. 1123 Assume that no other loss was detected during the last second. 1124 Then it can be concluded that nothing was totally lost. 1126 R2 is checked. Its original time was 21060-660=20400. 1127 A packet with text from A was received with that 1128 timestamp, so nothing needs to be recovered. 1130 B1 and B2 still need to be transmitted as redundancy. 1131 This is planned 330 ms after packet 104. That 1132 would be at 21130. 1134 |-----------------------| 1135 |Seq no 106, Timer=21130| 1136 |CC=1 | 1137 |CSRC list B | 1138 | R2: B1, Offset=630 | 1139 | R1: B2, Offset=330 | 1140 | P: Empty | 1141 |-----------------------| 1143 Packet 106 is received. 1145 The second level redundancy in packet 106 is B1 and has timestamp 1146 offset 630 ms. The timestamp of packet 106 minus 630 is 20500 which 1147 is the timestamp of packet 102 that was received. So B1 does not 1148 need to be retrieved. The first level redundancy in packet 106 has 1149 offset 330. The timestamp of packet 106 minus 330 is 20800. That is 1150 later than the latest received packet with source B. Therefore B2 is 1151 retrieved and assigned to the input buffer for source B. No primary 1152 is available in packet 106. 1154 After this sequence, A3 and B1 and B2 have been received. In this 1155 case no text was lost. 1157 3.21. Maximum character rate "cps" 1159 The default maximum rate of reception of "text/t140" real-time text 1160 is in [RFC4103] specified to be 30 characters per second. The actual 1161 rate is calculated without regard to any redundant text transmission 1162 and is in the multiparty case evaluated for all sources contributing 1163 to transmission to a receiver. The value MAY be modified in the 1164 "cps" parameter of the FMTP attribute in the media section for the 1165 "text/t140" media. A mixer combining real-time text from a number of 1166 sources may occasionally have a higher combined flow of text coming 1167 from the sources. Endpoints SHOULD therefore specify a suitable 1168 higher value for the "cps" parameter, corresponding to its real 1169 reception capability. A value for "cps" of 90 SHALL be the default 1170 for the "text/t140" stream in the "text/red" format when multiparty 1171 real-time text is negotiated. See [RFC4103] for the format and use 1172 of the "cps" parameter. The same rules apply for the multiparty case 1173 except for the default value. 1175 4. Presentation level considerations 1177 "Protocol for multimedia application text conversation" [T140] 1178 provides the presentation level requirements for the [RFC4103] 1179 transport. Functions for erasure and other formatting functions are 1180 specified in [T140] which has the following general statement for the 1181 presentation: 1183 "The display of text from the members of the conversation should be 1184 arranged so that the text from each participant is clearly readable, 1185 and its source and the relative timing of entered text is visualized 1186 in the display. Mechanisms for looking back in the contents from the 1187 current session should be provided. The text should be displayed as 1188 soon as it is received." 1189 Strict application of [T140] is of essence for the interoperability 1190 of real-time text implementations and to fulfill the intention that 1191 the session participants have the same information conveyed in the 1192 text contents of the conversation without necessarily having the 1193 exact same layout of the conversation. 1195 [T140] specifies a set of presentation control codes to include in 1196 the stream. Some of them are optional. Implementations MUST ignore 1197 optional control codes that they do not support. 1199 There is no strict "message" concept in real-time text. The Unicode 1200 Line Separator character SHALL be used as a separator allowing a part 1201 of received text to be grouped in presentation. The characters 1202 "CRLF" may be used by other implementations as a replacement for Line 1203 Separator. The "CRLF" combination SHALL be erased by just one 1204 erasing action, the same as the Line Separator. Presentation 1205 functions are allowed to group text for presentation in smaller 1206 groups than the line separators imply and present such groups with 1207 source indication together with text groups from other sources (see 1208 the following presentation examples). Erasure has no specific limit 1209 by any delimiter in the text stream. 1211 4.1. Presentation by multiparty-aware endpoints 1213 A multiparty-aware receiving party, presenting real-time text MUST 1214 separate text from different sources and present them in separate 1215 presentation fields. The receiving party MAY separate presentation 1216 of parts of text from a source in readable groups based on other 1217 criteria than line separator and merge these groups in the 1218 presentation area when it benefits the user to most easily find and 1219 read text from the different participants. The criteria MAY e.g., be 1220 a received comma, full stop, or other phrase delimiters, or a long 1221 pause. 1223 When text is received from multiple original sources, the 1224 presentation SHALL provide a view where text is added in multiple 1225 presentation fields. 1227 If the presentation presents text from different sources in one 1228 common area, the presenting endpoint SHOULD insert text from the 1229 local user ended at suitable points merged with received text to 1230 indicate the relative timing for when the text groups were completed. 1231 In this presentation mode, the receiving endpoint SHALL present the 1232 source of the different groups of text. This presentation style is 1233 called the "chat" style here and provides a possibility to follow 1234 text arriving from multiple parties and the approximate relative time 1235 that text is received related to text from the local user. 1237 A view of a three-party RTT call in chat style is shown in this 1238 example . 1240 _________________________________________________ 1241 | |^| 1242 |[Alice] Hi, Alice here. |-| 1243 | | | 1244 |[Bob] Bob as well. | | 1245 | | | 1246 |[Eve] Hi, this is Eve, calling from Paris. | | 1247 | I thought you should be here. | | 1248 | | | 1249 |[Alice] I am coming on Thursday, my | | 1250 | performance is not until Friday morning.| | 1251 | | | 1252 |[Bob] And I on Wednesday evening. | | 1253 | | | 1254 |[Alice] Can we meet on Thursday evening? | | 1255 | | | 1256 |[Eve] Yes, definitely. How about 7pm. | | 1257 | at the entrance of the restaurant | | 1258 | Le Lion Blanc? | | 1259 |[Eve] we can have dinner and then take a walk |-| 1260 |______________________________________________|v| 1261 | But I need to be back to |^| 1262 | the hotel by 11 because I need |-| 1263 | | | 1264 | I wou |-| 1265 |______________________________________________|v| 1266 | of course, I underst | 1267 |________________________________________________| 1269 Figure 3: Example of a three-party RTT call presented in chat style 1270 seen at participant 'Alice's endpoint. 1272 Other presentation styles than the chat style MAY be arranged. 1274 This figure shows how a coordinated column view MAY be presented. 1276 _____________________________________________________________________ 1277 | Bob | Eve | Alice | 1278 |____________________|______________________|_______________________| 1279 | | |I will arrive by TGV. | 1280 |My flight is to Orly| |Convenient to the main | 1281 | |Hi all, can we plan |station. | 1282 | |for the seminar? | | 1283 |Eve, will you do | | | 1284 |your presentation on| | | 1285 |Friday? |Yes, Friday at 10. | | 1286 |Fine, wo | |We need to meet befo | 1287 |___________________________________________________________________| 1289 Figure 4: An example of a coordinated column-view of a three-party 1290 session with entries ordered vertically in approximate time-order. 1292 4.2. Multiparty mixing for multiparty-unaware endpoints 1294 When the mixer has indicated RTT multiparty capability in an SDP 1295 negotiation, but the multiparty capability negotiation fails with an 1296 endpoint, then the agreed "text/red" or "text/t140" format SHALL be 1297 used and the mixer SHOULD compose a best-effort presentation of 1298 multiparty real-time text in one stream intended to be presented by 1299 an endpoint with no multiparty awareness, when that is desired in the 1300 actual implementation. The following specifies a procedure which MAY 1301 be applied in that situation. 1303 This presentation format has functional limitations and SHOULD be 1304 used only to enable participation in multiparty calls by legacy 1305 deployed endpoints implementing only RFC 4103 without any multiparty 1306 extensions specified in this document. 1308 The principles and procedures below do not specify any new protocol 1309 elements. They are instead composed of information from [T140] and 1310 an ambition to provide a best-effort presentation on an endpoint 1311 which has functions originally intended only for two-party calls. 1313 The mixer mixing for multiparty-unaware endpoints SHALL compose a 1314 simulated, limited multiparty RTT view suitable for presentation in 1315 one presentation area. The mixer SHALL group text in suitable groups 1316 and prepare for presentation of them by inserting a line separator 1317 between them if the transmitted text did not already end with a new 1318 line (line separator or CRLF). A presentable label SHALL be composed 1319 and sent for the source initially in the session and after each 1320 source switch. With this procedure the time for switching from 1321 transmission of text from one source to transmission of text from 1322 another source depends on the actions of the users. In order to 1323 expedite source switching, a user can, for example, end its turn with 1324 a new line. 1326 4.2.1. Actions by the mixer at reception from the call participants 1328 When text is received by the mixer from the different participants, 1329 the mixer SHALL recover text from redundancy if any packets are lost. 1330 The mark for lost text [T140ad1] SHALL be inserted in the stream if 1331 unrecoverable loss appears. Any Unicode "BOM" characters, possibly 1332 used for keep-alive, SHALL be deleted. The time of creation of text 1333 (retrieved from the RTP timestamp) SHALL be stored together with the 1334 received text from each source in queues for transmission to the 1335 recipients in order to be able to evaluate text loss. 1337 4.2.2. Actions by the mixer for transmission to the recipients 1339 The following procedure SHALL be applied for each multiparty-unaware 1340 recipient of multiparty text from the mixer. 1342 The text for transmission SHALL be formatted by the mixer for each 1343 receiving user for presentation in one single presentation area. 1344 Text received from a participant SHOULD NOT be included in 1345 transmission to that participant because it is usually presented 1346 locally at transmission time. When there is text available for 1347 transmission from the mixer to a receiving party from more than one 1348 participant, the mixer SHALL switch between transmission of text from 1349 the different sources at suitable points in the transmitted stream. 1351 When switching source, the mixer SHALL insert a line separator if the 1352 already transmitted text did not end with a new line (line separator 1353 or CRLF). A label SHALL be composed of information in the CNAME and 1354 NAME fields in RTCP reports from the participant to have its text 1355 transmitted, or from other session information for that user. The 1356 label SHALL be delimited by suitable characters (e.g., '[ ]') and 1357 transmitted. The CSRC SHALL indicate the selected source. Then text 1358 from that selected participant SHALL be transmitted until a new 1359 suitable point for switching source is reached. 1361 Information available to the mixer for composing the label may 1362 contain sensitive personal information that SHOULD NOT be revealed in 1363 sessions not securely authenticated and confidentiality protected. 1364 Privacy considerations regarding how much personal information is 1365 included in the label SHOULD therefore be taken when composing the 1366 label. 1368 Seeking a suitable point for switching source SHALL be done when 1369 there is older text waiting for transmission from any party than the 1370 age of the last transmitted text. Suitable points for switching are: 1372 * A completed phrase ended by comma 1374 * A completed sentence 1376 * A new line (line separator or CRLF) 1378 * A long pause (e.g., > 10 seconds) in received text from the 1379 currently transmitted source 1381 * If text from one participant has been transmitted with text from 1382 other sources waiting for transmission for a long time (e.g., > 1 1383 minute) and none of the other suitable points for switching has 1384 occurred, a source switch MAY be forced by the mixer at the next 1385 word delimiter, and also even if a word delimiter does not occur 1386 within a time (e.g., 15 seconds) after the scan for a word 1387 delimiter started. 1389 When switching source, the source which has the oldest text in queue 1390 SHALL be selected to be transmitted. A character display count SHALL 1391 be maintained for the currently transmitted source, starting at zero 1392 after the label is transmitted for the currently transmitted source. 1394 The status SHALL be maintained for the latest control code for Select 1395 Graphic Rendition (SGR) from each source. If there is an SGR code 1396 stored as the status for the current source before the source switch 1397 is done, a reset of SGR SHALL be sent by the sequence SGR 0 [009B 1398 0000 006D] after the new line and before the new label during a 1399 source switch. See SGR below for an explanation. This transmission 1400 does not influence the display count. 1402 If there is an SGR code stored for the new source after the source 1403 switch, that SGR code SHALL be transmitted to the recipient before 1404 the label. This transmission does not influence the display count. 1406 4.2.3. Actions on transmission of text 1408 Text from a source sent to the recipient SHALL increase the display 1409 count by one per transmitted character. 1411 4.2.4. Actions on transmission of control codes 1413 The following control codes specified by T.140 require specific 1414 actions. They SHALL cause specific considerations in the mixer. 1415 Note that the codes presented here are expressed in UCS-16, while 1416 transmission is made in the UTF-8 encoding of these codes. 1418 BEL 0007 Bell Alert in session. Provides for alerting during an 1419 active session. The display count SHALL NOT be altered. 1421 NEW LINE 2028 Line separator. Check and perform a source switch if 1422 appropriate. Increase the display count by 1. 1424 CR LF 000D 000A A supported but not preferred way of requesting a 1425 new line. Check and perform a source switch if appropriate. 1426 Increase the display count by 1. 1428 INT ESC 0061 Interrupt (used to initiate the mode negotiation 1429 procedure). The display count SHALL NOT be altered. 1431 SGR 009B Ps 006D Select graphic rendition. Ps is the rendition 1432 parameters specified in ISO 6429. The display count SHALL NOT be 1433 altered. The SGR code SHOULD be stored for the current source. 1435 SOS 0098 Start of string, used as a general protocol element 1436 introducer, followed by a maximum 256-byte string and the ST. The 1437 display count SHALL NOT be altered. 1439 ST 009C String terminator, end of SOS string. The display count 1440 SHALL NOT be altered. 1442 ESC 001B Escape - used in control strings. The display count SHALL 1443 NOT be altered for the complete escape code. 1445 Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used 1446 for synchronization and keep-alive. It SHALL be deleted from 1447 incoming streams. It SHALL also be sent first after session 1448 establishment to the recipient. The display count SHALL NOT be 1449 altered. 1451 Missing text mark (U+FFFD) "Replacement character", represented as a 1452 question mark in a rhombus, or if that is not feasible, replaced 1453 by an apostrophe '. It marks the place in the stream of possible 1454 text loss. This mark SHALL be inserted by the reception procedure 1455 in case of unrecoverable loss of packets. The display count SHALL 1456 be increased by one when sent as for any other character. 1458 SGR If a control code for selecting graphic rendition (SGR) other 1459 than reset of the graphic rendition (SGR 0) is sent to a 1460 recipient, that control code SHALL also be stored as the status 1461 for the source in the storage for SGR status. If a reset graphic 1462 rendition (SGR 0) originating from a source is sent, then the SGR 1463 status storage for that source SHALL be cleared. The display 1464 count SHALL NOT be increased. 1466 BS (U+0008) Back Space, intended to erase the last entered character 1467 by a source. Erasure by backspace cannot always be performed as 1468 the erasing party intended. If an erasing action erases all text 1469 up to the end of the leading label after a source switch, then the 1470 mixer MUST NOT transmit more backspaces. Instead, it is 1471 RECOMMENDED that a letter "X" is inserted in the text stream for 1472 each backspace as an indication of the intent to erase more. A 1473 new line is usually coded by a Line Separator, but the character 1474 combination "CRLF" MAY be used instead. Erasure of a new line is 1475 in both cases done by just one erasing action (Backspace). If the 1476 display count has a positive value it SHALL be decreased by one 1477 when the BS is sent. If the display count is at zero, it SHALL 1478 NOT be altered. 1480 4.2.5. Packet transmission 1482 A mixer transmitting to a multiparty-unaware terminal SHALL send 1483 primary data only from one source per packet. The SSRC SHALL be the 1484 SSRC of the mixer. The CSRC list SHALL contain one member and be the 1485 SSRC of the source of the primary data. 1487 4.2.6. Functional limitations 1489 When a multiparty-unaware endpoint presents a conversation in one 1490 display area in a chat style, it inserts source indications for 1491 remote text and local user text as they are merged in completed text 1492 groups. When an endpoint using this layout receives and presents 1493 text mixed for multiparty-unaware endpoints, there will be two levels 1494 of source indicators for the received text; one generated by the 1495 mixer and inserted in a label after each source switch, and another 1496 generated by the receiving endpoint and inserted after each switch 1497 between local and remote source in the presentation area. This will 1498 waste display space and look inconsistent to the reader. 1500 New text can be presented only from one source at a time. Switch of 1501 source to be presented takes place at suitable places in the text, 1502 such as end of phrase, end of sentence, line separator and 1503 inactivity. Therefore, the time to switch to present waiting text 1504 from other sources may become long and will vary and depend on the 1505 actions of the currently presented source. 1507 Erasure can only be done up to the latest source switch. If a user 1508 tries to erase more text, the erasing actions will be presented as 1509 letter X after the label. 1511 Text loss because of network errors may hit the label between entries 1512 from different parties, causing risk for misunderstanding from which 1513 source a piece of text is. 1515 These facts make it strongly RECOMMENDED implementing multiparty 1516 awareness in RTT endpoints. The use of the mixing method for 1517 multiparty-unaware endpoints should be left for use with endpoints 1518 which are impossible to upgrade to become multiparty-aware. 1520 4.2.7. Example views of presentation on multiparty-unaware endpoints 1522 The following pictures are examples of the view on a participant's 1523 display for the multiparty-unaware case. 1525 _________________________________________________ 1526 | Conference | Alice | 1527 |________________________|_________________________| 1528 | |I will arrive by TGV. | 1529 |[Bob]:My flight is to |Convenient to the main | 1530 |Orly. |station. | 1531 |[Eve]:Hi all, can we | | 1532 |plan for the seminar. | | 1533 | | | 1534 |[Bob]:Eve, will you do | | 1535 |your presentation on | | 1536 |Friday? | | 1537 |[Eve]:Yes, Friday at 10.| | 1538 |[Bob]: Fine, wo |We need to meet befo | 1539 |________________________|_________________________| 1541 Figure 5: Alice who has a conference-unaware client is receiving the 1542 multiparty real-time text in a single-stream. 1544 This figure shows how a coordinated column view MAY be presented on 1545 Alice's device in a view with two-columns. The mixer inserts labels 1546 to show how the sources alternate in the column with received text. 1548 The mixer alternates between the sources at suitable points in the 1549 text exchange so that text entries from each party can be 1550 conveniently read. 1552 _________________________________________________ 1553 | |^| 1554 |(Alice) Hi, Alice here. |-| 1555 | | | 1556 |(mix)[Bob)] Bob as well. | | 1557 | | | 1558 |[Eve] Hi, this is Eve, calling from Paris | | 1559 | I thought you should be here. | | 1560 | | | 1561 |(Alice) I am coming on Thursday, my | | 1562 | performance is not until Friday morning.| | 1563 | | | 1564 |(mix)[Bob] And I on Wednesday evening. | | 1565 | | | 1566 |[Eve] we can have dinner and then walk | | 1567 | | | 1568 |[Eve] But I need to be back to | | 1569 | the hotel by 11 because I need | | 1570 | |-| 1571 |______________________________________________|v| 1572 | of course, I underst | 1573 |________________________________________________| 1575 Figure 6: An example of a view of the multiparty-unaware presentation 1576 in chat style. Alice is the local user. 1578 In this view, there is a tradition in receiving applications to 1579 include a label showing the source of the text, here shown with 1580 parenthesis "()". The mixer also inserts source labels for the 1581 multiparty call participants, here shown with brackets "[]". 1583 5. Relation to Conference Control 1584 5.1. Use with SIP centralized conferencing framework 1586 The Session Initiation Protocol (SIP) conferencing framework, mainly 1587 specified in [RFC4353], [RFC4579] and [RFC4575] is suitable for 1588 coordinating sessions including multiparty RTT. The RTT stream 1589 between the mixer and a participant is one and the same during the 1590 conference. Participants get announced by notifications when 1591 participants are joining or leaving, and further user information may 1592 be provided. The SSRC of the text to expect from joined users MAY be 1593 included in a notification. The notifications MAY be used both for 1594 security purposes and for translation to a label for presentation to 1595 other users. 1597 5.2. Conference control 1599 In managed conferences, control of the real-time text media SHOULD be 1600 provided in the same way as other for media, e.g., for muting and 1601 unmuting by the direction attributes in SDP [RFC8866]. 1603 Note that floor control functions may be of value for RTT users as 1604 well as for users of other media in a conference. 1606 6. Gateway Considerations 1608 6.1. Gateway considerations with Textphones 1610 multiparty RTT sessions may involve gateways of different kinds. 1611 Gateways involved in setting up sessions SHALL correctly reflect the 1612 multiparty capability or unawareness of the combination of the 1613 gateway and the remote endpoint beyond the gateway. 1615 One case that may occur is a gateway to Public Switched Telephone 1616 Network (PSTN) for communication with textphones (e.g., TTYs). 1617 Textphones are limited devices with no multiparty awareness, and it 1618 SHOULD therefore be suitable for the gateway to not indicate 1619 multiparty awareness for that case. Another solution is that the 1620 gateway indicates multiparty capability towards the mixer, and 1621 includes the multiparty mixer function for multiparty-unaware 1622 endpoints itself. This solution makes it possible to adapt to the 1623 functional limitations of the textphone. 1625 More information on gateways to textphones is found in [RFC5194] 1627 6.2. Gateway considerations with WebRTC 1629 Gateway operation to real-time text in WebRTC may also be required. 1630 In WebRTC, RTT is specified in [RFC8865]. 1632 A multiparty bridge may have functionality for communicating by RTT 1633 both in RTP streams with RTT and WebRTC T.140 data channels. Other 1634 configurations may consist of a multiparty bridge with either 1635 technology for RTT transport and a separate gateway for conversion of 1636 the text communication streams between RTP and T.140 data channel. 1638 In WebRTC, it is assumed that for a multiparty session, one T.140 1639 data channel is established for each source from a gateway or bridge 1640 to each participant. Each participant also has a data channel with a 1641 two-way connection with the gateway or bridge. 1643 The T.140 data channel used both ways is for text from the WebRTC 1644 user and from the bridge or gateway itself to the WebRTC user. The 1645 label parameter of this T.140 data channel is used as the NAME field 1646 in RTCP to participants on the RTP side. The other T.140 data 1647 channels are only for text from other participants to the WebRTC 1648 user. 1650 When a new participant has entered the session with RTP transport of 1651 RTT, a new T.140 channel SHOULD be established to WebRTC users with 1652 the label parameter composed of information from the NAME field in 1653 RTCP on the RTP side. 1655 When a new participant has entered the multiparty session with RTT 1656 transport in a WebRTC T.140 data channel, the new participant SHOULD 1657 be announced by a notification to RTP users. The label parameter 1658 from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP 1659 side, or other available session information. 1661 When a participant on the RTP side is disconnected from the 1662 multiparty session, the corresponding T.140 data channel(s) SHOULD be 1663 closed. 1665 When a WebRTC user of T.140 data channels disconnects from the mixer, 1666 the corresponding RTP streams or sources in an RTP-mixed stream 1667 SHOULD be closed. 1669 T.140 data channels MAY be opened and closed by negotiation or 1670 renegotiation of the session or by any other valid means as specified 1671 in section 1 of [RFC8865]. 1673 7. Updates to RFC 4103 1675 This document updates [RFC4103] by introducing an SDP media attribute 1676 "rtt-mixer" for negotiation of multiparty-mixing capability with the 1677 [RFC4103] format, and by specifying the rules for packets when 1678 multiparty capability is negotiated and in use. 1680 8. Congestion considerations 1682 The congestion considerations and recommended actions from [RFC4103] 1683 are also valid in multiparty situations. 1685 The time values SHALL then be applied per source of text sent to a 1686 receiver. 1688 If the very unlikely situation appears that many participants in a 1689 conference send text simultaneously for a long period, a delay may 1690 build up for presentation of text at the receivers if the limitation 1691 in characters per second ("cps") to be transmitted to the 1692 participants is exceeded. More delay than 7 seconds can cause 1693 confusion in the session. It is therefore RECOMMENDED that an RTP- 1694 mixer-based mixer discards such text causing excessive delays and 1695 inserts a general indication of possible text loss [T140ad1] in the 1696 session. If the main text contributor is indicated in any way, the 1697 mixer MAY avoid deleting text from that participant. It should 1698 however be noted that human creation of text normally contains 1699 pauses, when the transmission can catch up, so that the transmission 1700 overload situations are expected to be very rare. 1702 9. Acknowledgements 1704 James Hamlin for format and performance aspects. 1706 10. IANA Considerations 1708 10.1. Registration of the "rtt-mixer" SDP media attribute 1710 [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the 1711 RFC number of this document.] 1713 IANA is asked to register the new SDP attribute "rtt-mixer". 1715 Contact name: IESG 1717 Contact email: iesg@ietf.org 1719 Attribute name: rtt-mixer 1721 Attribute semantics: See RFCXXXX Section 2.3 1723 Attribute value: none 1725 Usage level: media 1727 Purpose: Indicate support by mixer and endpoint of multiparty mixing 1728 for real-time text transmission, using a common RTP-stream for 1729 transmission of text from a number of sources mixed with one 1730 source at a time and the source indicated in a single CSRC-list 1731 member. 1733 Charset Dependent: no 1735 O/A procedure: See RFCXXXX Section 2.3 1737 Mux Category: normal 1739 Reference: RFCXXXX 1741 11. Security Considerations 1743 The RTP-mixer model requires the mixer to be allowed to decrypt, 1744 pack, and encrypt secured text from the conference participants. 1745 Therefore, the mixer needs to be trusted to maintain confidentiality 1746 and integrity of the RTT data. This situation is similar to the 1747 situation for handling audio and video media in centralized mixers. 1749 The requirement to transfer information about the user in RTCP 1750 reports in SDES, CNAME, and NAME fields, and in conference 1751 notifications, may have privacy concerns as already stated in RFC 1752 3550 [RFC3550], and may be restricted for privacy reasons. When used 1753 for creation of readable labels in the presentation, the receiving 1754 user will then get a more symbolic label for the source. 1756 The services available through the RTT mixer may have special 1757 interest for deaf and hard-of-hearing persons. Some users may want 1758 to refrain from revealing such characteristics broadly in 1759 conferences. The design of the conference systems where the mixer is 1760 included MAY need to be made with confidentiality of such 1761 characteristics in mind. 1763 Participants with malicious intentions may appear and e.g., disturb 1764 the multiparty session by emitting a continuous flow of text. They 1765 may also send text that appears to originate from other participants. 1766 Counteractions should be to require secure signaling, media and 1767 authentication, and to provide higher-layer conference functions 1768 e.g., for blocking, muting, and expelling participants. 1770 Participants with malicious intentions may also try to disturb the 1771 presentation by sending incomplete or malformed control codes. 1772 Handling of text from the different sources by the receivers MUST 1773 therefore be well separated so that the effects of such actions only 1774 affect text from the source causing the action. 1776 Care should be taken that if use of the mixer is allowed for users 1777 both with and without security procedures, opens for possible attacks 1778 by both unauthenticated call participants and even eavesdropping and 1779 manipulating of content non-participants. 1781 As already stated in Section 3.18, security in media SHOULD be 1782 applied by using DTLS-SRTP [RFC5764] on the media level. 1784 Further security considerations specific for this application are 1785 specified in Section 3.18. 1787 12. Change history 1789 [RFC Editor: Please remove this section prior to publication.] 1791 12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-19 1793 Edits because of comments in a review by Francesca Palombini. 1795 Edits because of comments from Benjamin Kaduk. 1797 Proposed to not change anything because of Robert Wilton's comments. 1799 Two added sentences in the security section to meet comments by Roman 1800 Danyliw. 1802 12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-18 1804 Edits of nits as proposed in a review by Lars Eggert. 1806 Edits as response to review by Martin Duke. 1808 12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-17 1810 Actions on Gen-ART review comments. 1812 Actions on SecDir review comments. 1814 12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-16 1816 Improvements in the offer/answer considerations section by adding 1817 subsections for each phase in the negotiation as requested by IANA 1818 expert review. 1820 12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-15 1822 Actions on review comments from Jurgen Schonwalder: 1824 A bit more about congestion situations and that they are expected to 1825 be very rare. 1827 Explanation of differences in security between the conference-aware 1828 and the conference-unaware case added in security section. 1830 Presentation examples with suource labels made less confusing, and 1831 explained. 1833 Reference to T.140 inserted at first mentioning of T.140. 1835 Reference to RFC 8825 inserted to explain WebRTC 1837 Nit in wording in terminology section adjusted. 1839 12.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-14 1841 Changes from comments by Murray Kucherawy during AD review. 1843 Many SHOULD in section 4.2 on multiparty-unaware mixing changed to 1844 SHALL, and the whole section instead specified to be optional 1845 depending on the application. 1847 Some SHOULD in section 3 either explained or changed to SHALL. 1849 In order to have explainable conditions behind SHOULDs, the 1850 transmission interval in 3.4 is changed to as soon as text is 1851 available as a main principle. The call participants send with 300 1852 ms interval so that will create realistic load conditions anyway. 1854 12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-13 1856 Changed year to 2021. 1858 Changed reference to draft on RTT in WebRTC to recently published RFC 1859 8865. 1861 Changed label brackets in example from "[]" to "()" to avoid nits 1862 comment. 1864 Changed reference "RFC 4566" to recently published "RFC 8866" 1866 12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-12 1868 Changes according to responses on comments from Brian Rosen in 1869 Avtcore list on 2020-12-05 and -06. 1871 Changes according to responses to comments by Bernard Aboba in 1872 avtcore list 2020-12-06. 1874 Introduction of an optiona RTP multi-stream mixing method for further 1875 study as proposed by Bernard Aboba. 1877 Changes clarifying how to open and close T.140 data channels included 1878 in 6.2 after comments by Lorenzo Miniero. 1880 Changes to satisfy nits check. Some "not" changed to "NOT" in 1881 normative wording combinations. Some lower case normative words 1882 changed to upper case. A normative reference deleted from the 1883 abstract. Two informative documents moved from normative references 1884 to informative references. 1886 12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11 1888 Timestamps and timestamp offsets added to the packet examples in 1889 section 3.23, and the description corrected. 1891 A number of minor corrections added in sections 3.10 - 3.23. 1893 12.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10 1895 The packet composition was modified for interleaving packets from 1896 different sources. 1898 The packet reception was modified for the new interleaving method. 1900 The packet sequence examples was adjusted for the new interleaving 1901 method. 1903 Modifications according to responses to Brian Rosen of 2020-11-03 1905 12.11. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09 1907 Changed name on the SDP media attribute to "rtt-mixer" 1909 Restructure of section 2 for balance between aware and unaware cases. 1911 Moved conference control to own section. 1913 Improved clarification of recovery and loss in the packet sequence 1914 example. 1916 A number of editorial corrections and improvements. 1918 12.12. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 1920 Deleted the method requiring a new packet format "text/rex" because 1921 of the longer standardization and implementation period it needs. 1923 Focus on use of RFC 4103 text/red format with shorter transmission 1924 interval, and source indicated in CSRC. 1926 12.13. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 1928 Added a method based on the "text/red" format and single source per 1929 packet, negotiated by the "rtt-mixer" SDP attribute. 1931 Added reasoning and recommendation about indication of loss. 1933 The highest number of sources in one packet is 15, not 16. Changed. 1935 Added in information on update to RFC 4103 that RFC 4103 explicitly 1936 allows addition of FEC method. The redundancy is a kind of forward 1937 error correction. 1939 12.14. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 1941 Improved definitions list format. 1943 The format of the media subtype parameters is made to match the 1944 requirements. 1946 The mapping of media subtype parameters to SDP is included. 1948 The "cps" parameter belongs to the t140 subtype and does not need to 1949 be registered here. 1951 12.15. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 1953 nomenclature and editorial improvements 1955 "this document" used consistently to refer to this document. 1957 12.16. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 1959 'Redundancy header' renamed to 'data header'. 1961 More clarifications added. 1963 Language and figure number corrections. 1965 12.17. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 1967 Mention possible need to mute and raise hands as for other media. 1968 ---done ---- 1970 Make sure that use in two-party calls is also possible and explained. 1971 - may need more wording - 1973 Clarify the RTT is often used together with other media. --done-- 1975 Tell that text mixing is N-1. A users own text is not received in 1976 the mix. -done- 1978 In 3. correct the interval to: A "text/rex" transmitter SHOULD send 1979 packets distributed in time as long as there is something (new or 1980 redundant T140blocks) to transmit. The maximum transmission interval 1981 SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 1982 receiver as soon as new text to that receiver is available, as long 1983 as the time after the latest sent packet to the same receiver is more 1984 than 150 ms, and also the maximum character rate to the receiver is 1985 not exceeded. The intention is to keep the latency low while keeping 1986 a good protection against text loss in bursty packet loss conditions. 1987 -done- 1989 In 1.3 say that the format is used both ways. -done- 1991 In 13.1 change presentation area to presentation field so that reader 1992 does not think it shall be totally separated. -done- 1994 In Performance and intro, tell the performance in number of 1995 simultaneous sending users and introduced delay 16, 150 vs 1996 requirements 5 vs 500. -done -- 1998 Clarify redundancy level per connection. -done- 2000 Timestamp also for the last data header. To make it possible for all 2001 text to have time offset as for transmission from the source. Make 2002 that header equal to the others. -done- 2004 Mixer always use the CSRC list, even for its own BOM. -done- 2006 Combine all talk about transmission interval (300 ms vs when text has 2007 arrived) in section 3 in one paragraph or close to each other. -done- 2009 Documents the goal of good performance with low delay for 5 2010 simultaneous typers in the introduction. -done- 2011 Describe better that only primary text shall be sent on to receivers. 2012 Redundancy and loss must be resolved by the mixer. -done- 2014 12.18. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 2016 SDP and better description and visibility of security by OSRTP RFC 2017 8634 needed. 2019 The description of gatewaying to WebRTC extended. 2021 The description of the data header in the packet is improved. 2023 12.19. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 2025 2,5,6 More efficient format "text/rex" introduced and attribute 2026 a=rtt-mix deleted. 2028 3. Brief about use of OSRTP for security included- More needed. 2030 4. Brief motivation for the solution and why not rtp-translator is 2031 used added to intro. 2033 7. More limitations for the multiparty-unaware mixing method 2034 inserted. 2036 8. Updates to RFC 4102 and 4103 more clearly expressed. 2038 9. Gateway to WebRTC started. More needed. 2040 12.20. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 2041 to draft-ietf-avtcore-multi-party-rtt-mix-00 2043 Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00 2045 Replaced CDATA in IANA registration table with better coding. 2047 Converted to xml2rfc version 3. 2049 12.21. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 2050 to -03 2052 Changed company and e-mail of the author. 2054 Changed title to "RTP-mixer formatting of multi-party Real-time text" 2055 to better match contents. 2057 Check and modification where needed of use of RFC 2119 words SHALL 2058 etc. 2060 More about the CC value in sections on transmitters and receivers so 2061 that 1-to-1 sessions do not use the mixer format. 2063 Enhanced section on presentation for multiparty-unaware endpoints 2065 A paragraph recommending cps=150 inserted in the performance section. 2067 12.22. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 2068 to -02 2070 In Abstract and 1. Introduction: Introduced wording about regulatory 2071 requirements. 2073 In section 5: The transmission interval is decreased to 100 ms when 2074 there is text from more than one source to transmit. 2076 In section 11 about SDP negotiation, a SHOULD-requirement is 2077 introduced that the mixer should make a mix for multiparty-unaware 2078 endpoints if the negotiation is not successful. And a reference to a 2079 later chapter about it. 2081 The presentation considerations chapter 14 is extended with more 2082 information about presentation on multiparty-aware endpoints, and a 2083 new section on the multiparty-unaware mixing with low functionality 2084 but SHOULD be implemented in mixers. Presentation examples are 2085 added. 2087 A short chapter 15 on gateway considerations is introduced. 2089 Clarification about the text/t140 format included in chapter 10. 2091 This sentence added to the chapter 10 about use without redundancy. 2092 "The text/red format SHOULD be used unless some other protection 2093 against packet loss is utilized, for example a reliable network or 2094 transport." 2096 Note about deviation from RFC 2198 added in chapter 4. 2098 In chapter 9. "Use with SIP centralized conferencing framework" the 2099 following note is inserted: Note: The CSRC-list in an RTP packet only 2100 includes participants whose text is included in one or more text 2101 blocks. It is not the same as the list of participants in a 2102 conference. With audio and video media, the CSRC-list would often 2103 contain all participants who are not muted whereas text participants 2104 that don't type are completely silent and so don't show up in RTP 2105 packet CSRC-lists. 2107 12.23. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 2108 to -01 2110 Editorial cleanup. 2112 Changed capability indication from fmtp-parameter to SDP attribute 2113 "rtt-mix". 2115 Swapped order of redundancy elements in the example to match reality. 2117 Increased the SDP negotiation section 2119 13. References 2121 13.1. Normative References 2123 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 2124 Requirement Levels", BCP 14, RFC 2119, 2125 DOI 10.17487/RFC2119, March 1997, 2126 . 2128 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 2129 Jacobson, "RTP: A Transport Protocol for Real-Time 2130 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 2131 July 2003, . 2133 [RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type", 2134 RFC 4102, DOI 10.17487/RFC4102, June 2005, 2135 . 2137 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 2138 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 2139 . 2141 [RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session 2142 Initiation Protocol (SIP)", RFC 5630, 2143 DOI 10.17487/RFC5630, October 2009, 2144 . 2146 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 2147 Security (DTLS) Extension to Establish Keys for the Secure 2148 Real-time Transport Protocol (SRTP)", RFC 5764, 2149 DOI 10.17487/RFC5764, May 2010, 2150 . 2152 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 2153 Keeping Alive the NAT Mappings Associated with RTP / RTP 2154 Control Protocol (RTCP) Flows", RFC 6263, 2155 DOI 10.17487/RFC6263, June 2011, 2156 . 2158 [RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M. 2159 Thomson, "Session Traversal Utilities for NAT (STUN) Usage 2160 for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675, 2161 October 2015, . 2163 [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2164 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, 2165 May 2017, . 2167 [RFC8865] Holmberg, C. and G. Hellström, "T.140 Real-Time Text 2168 Conversation over WebRTC Data Channels", RFC 8865, 2169 DOI 10.17487/RFC8865, January 2021, 2170 . 2172 [RFC8866] Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP: 2173 Session Description Protocol", RFC 8866, 2174 DOI 10.17487/RFC8866, January 2021, 2175 . 2177 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 2178 multimedia application text conversation", February 1998, 2179 . 2181 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 2182 Protocol for multimedia application text conversation", 2183 February 2000, 2184 . 2186 13.2. Informative References 2188 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 2189 Session Initiation Protocol (SIP)", RFC 4353, 2190 DOI 10.17487/RFC4353, February 2006, 2191 . 2193 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 2194 Session Initiation Protocol (SIP) Event Package for 2195 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 2196 2006, . 2198 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 2199 (SIP) Call Control - Conferencing for User Agents", 2200 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 2201 . 2203 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 2204 Time Text over IP Using the Session Initiation Protocol 2205 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 2206 . 2208 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 2209 DOI 10.17487/RFC7667, November 2015, 2210 . 2212 [RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. 2213 Stach, "An Opportunistic Approach for Secure Real-time 2214 Transport Protocol (OSRTP)", RFC 8643, 2215 DOI 10.17487/RFC8643, August 2019, 2216 . 2218 [RFC8723] Jennings, C., Jones, P., Barnes, R., and A.B. Roach, 2219 "Double Encryption Procedures for the Secure Real-Time 2220 Transport Protocol (SRTP)", RFC 8723, 2221 DOI 10.17487/RFC8723, April 2020, 2222 . 2224 [RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for 2225 Browser-Based Applications", RFC 8825, 2226 DOI 10.17487/RFC8825, January 2021, 2227 . 2229 Author's Address 2231 Gunnar Hellstrom 2232 Gunnar Hellstrom Accessible Communication 2233 SE-13670 Vendelso 2234 Sweden 2236 Email: gunnar.hellstrom@ghaccess.se