idnits 2.17.1 draft-ietf-avtcore-multi-party-rtt-mix-18.txt: Checking boilerplate required by RFC 5378 and the IETF Trust (see https://trustee.ietf.org/license-info): ---------------------------------------------------------------------------- No issues found here. Checking nits according to https://www.ietf.org/id-info/1id-guidelines.txt: ---------------------------------------------------------------------------- == There is 1 instance of lines with non-ascii characters in the document. Checking nits according to https://www.ietf.org/id-info/checklist : ---------------------------------------------------------------------------- No issues found here. Miscellaneous warnings: ---------------------------------------------------------------------------- == The copyright year in the IETF Trust and authors Copyright Line does not match the current year == The document seems to lack the recommended RFC 2119 boilerplate, even if it appears to use RFC 2119 keywords -- however, there's a paragraph with a matching beginning. Boilerplate error? (The document does seem to have the reference to RFC 2119 which the ID-Checklist requires). == Using lowercase 'not' together with uppercase 'MUST', 'SHALL', 'SHOULD', or 'RECOMMENDED' is not an accepted usage according to RFC 2119. Please use uppercase 'NOT' together with RFC 2119 keywords (if that is what you mean). Found 'SHOULD not' in this paragraph: Information available to the mixer for composing the label may contain sensitive personal information that SHOULD not be revealed in sessions not securely authenticated and protected. Integrity considerations regarding how much personal information is included in the label SHOULD therefore be taken when composing the label. (Using the creation date from RFC4103, updated by this document, for RFC5378 checks: 2003-11-21) -- The document seems to lack a disclaimer for pre-RFC5378 work, but may have content which was first submitted before 10 November 2008. If you have contacted all the original authors and they are all willing to grant the BCP78 rights to the IETF Trust, then this is fine, and you can ignore this comment. If not, you may need to add the pre-RFC5378 disclaimer. (See the Legal Provisions document at https://trustee.ietf.org/license-info for more information.) -- The document date (17 May 2021) is 1074 days in the past. Is this intentional? Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Missing Reference: 'Bob' is mentioned on line 1529, 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) 17 May 2021 5 Intended status: Standards Track 6 Expires: 18 November 2021 8 RTP-mixer formatting of multiparty Real-time text 9 draft-ietf-avtcore-multi-party-rtt-mix-18 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 CSRC list in the RTP packet for source identification. The method 21 makes use of the same "text/t140" and "text/red" formats as for two- 22 party sessions. 24 Solutions using multiple RTP streams in the same RTP session are 25 briefly mentioned, as they could have some benefits over the RTP- 26 mixer model. The possibility to implement the solution in a wide 27 range of existing RTP implementations made the RTP-mixer model be 28 selected to be fully specified in this document. 30 A capability exchange is specified so that it can be verified that a 31 mixer and a participant can handle the multiparty-coded real-time 32 text stream using the RTP-mixer method. The capability is indicated 33 by use of an SDP media attribute "rtt-mixer". 35 The document updates RFC 4103 "RTP Payload for Text Conversation". 37 A specification of how a mixer can format text for the case when the 38 endpoint is not multiparty-aware is also provided. 40 Status of This Memo 42 This Internet-Draft is submitted in full conformance with the 43 provisions of BCP 78 and BCP 79. 45 Internet-Drafts are working documents of the Internet Engineering 46 Task Force (IETF). Note that other groups may also distribute 47 working documents as Internet-Drafts. The list of current Internet- 48 Drafts is at https://datatracker.ietf.org/drafts/current/. 50 Internet-Drafts are draft documents valid for a maximum of six months 51 and may be updated, replaced, or obsoleted by other documents at any 52 time. It is inappropriate to use Internet-Drafts as reference 53 material or to cite them other than as "work in progress." 55 This Internet-Draft will expire on 18 November 2021. 57 Copyright Notice 59 Copyright (c) 2021 IETF Trust and the persons identified as the 60 document authors. All rights reserved. 62 This document is subject to BCP 78 and the IETF Trust's Legal 63 Provisions Relating to IETF Documents (https://trustee.ietf.org/ 64 license-info) in effect on the date of publication of this document. 65 Please review these documents carefully, as they describe your rights 66 and restrictions with respect to this document. Code Components 67 extracted from this document must include Simplified BSD License text 68 as described in Section 4.e of the Trust Legal Provisions and are 69 provided without warranty as described in the Simplified BSD License. 71 Table of Contents 73 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 74 1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 75 1.2. Selected solution and considered alternatives . . . . . . 7 76 1.3. Intended application . . . . . . . . . . . . . . . . . . 9 77 2. Overview of the two specified solutions and selection of 78 method . . . . . . . . . . . . . . . . . . . . . . . . . 10 79 2.1. The RTP-mixer-based solution for multiparty-aware 80 endpoints . . . . . . . . . . . . . . . . . . . . . . . . 10 81 2.2. Mixing for multiparty-unaware endpoints . . . . . . . . . 11 82 2.3. Offer/answer considerations . . . . . . . . . . . . . . . 11 83 2.4. Actions depending on capability negotiation result . . . 13 84 3. Details for the RTP-mixer-based mixing method for 85 multiparty-aware endpoints . . . . . . . . . . . . . . . 13 86 3.1. Use of fields in the RTP packets . . . . . . . . . . . . 13 87 3.2. Initial transmission of a BOM character . . . . . . . . . 14 88 3.3. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 14 89 3.4. Transmission interval . . . . . . . . . . . . . . . . . . 14 90 3.5. Only one source per packet . . . . . . . . . . . . . . . 15 91 3.6. Do not send received text to the originating source . . . 15 92 3.7. Clean incoming text . . . . . . . . . . . . . . . . . . . 15 93 3.8. Redundant transmission principles . . . . . . . . . . . . 15 94 3.9. Interleaving text from different sources . . . . . . . . 15 95 3.10. Text placement in packets . . . . . . . . . . . . . . . . 16 96 3.11. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 16 97 3.12. Creation of the redundancy . . . . . . . . . . . . . . . 17 98 3.13. Timer offset fields . . . . . . . . . . . . . . . . . . . 17 99 3.14. Other RTP header fields . . . . . . . . . . . . . . . . . 17 100 3.15. Pause in transmission . . . . . . . . . . . . . . . . . . 18 101 3.16. RTCP considerations . . . . . . . . . . . . . . . . . . . 18 102 3.17. Reception of multiparty contents . . . . . . . . . . . . 18 103 3.18. Performance considerations . . . . . . . . . . . . . . . 20 104 3.19. Security for session control and media . . . . . . . . . 20 105 3.20. SDP offer/answer examples . . . . . . . . . . . . . . . . 21 106 3.21. Packet sequence example from interleaved transmission . . 22 107 3.22. Maximum character rate "CPS" . . . . . . . . . . . . . . 25 108 4. Presentation level considerations . . . . . . . . . . . . . . 25 109 4.1. Presentation by multiparty-aware endpoints . . . . . . . 26 110 4.2. multiparty mixing for multiparty-unaware endpoints . . . 28 111 5. Relation to Conference Control . . . . . . . . . . . . . . . 34 112 5.1. Use with SIP centralized conferencing framework . . . . . 34 113 5.2. Conference control . . . . . . . . . . . . . . . . . . . 35 114 6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 35 115 6.1. Gateway considerations with Textphones . . . . . . . . . 35 116 6.2. Gateway considerations with WebRTC . . . . . . . . . . . 35 117 7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 36 118 8. Congestion considerations . . . . . . . . . . . . . . . . . . 36 119 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 37 120 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 37 121 10.1. Registration of the "rtt-mixer" SDP media attribute . . 37 122 11. Security Considerations . . . . . . . . . . . . . . . . . . . 38 123 12. Change history . . . . . . . . . . . . . . . . . . . . . . . 38 124 12.1. Changes included in 125 draft-ietf-avtcore-multi-party-rtt-mix-18 . . . . . . . 38 126 12.2. Changes included in 127 draft-ietf-avtcore-multi-party-rtt-mix-17 . . . . . . . 38 128 12.3. Changes included in 129 draft-ietf-avtcore-multi-party-rtt-mix-16 . . . . . . . 38 130 12.4. Changes included in 131 draft-ietf-avtcore-multi-party-rtt-mix-15 . . . . . . . 39 132 12.5. Changes included in 133 draft-ietf-avtcore-multi-party-rtt-mix-14 . . . . . . . 39 134 12.6. Changes included in 135 draft-ietf-avtcore-multi-party-rtt-mix-13 . . . . . . . 39 136 12.7. Changes included in 137 draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . . 40 138 12.8. Changes included in 139 draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 40 140 12.9. Changes included in 141 draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 40 142 12.10. Changes included in 143 draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 40 144 12.11. Changes included in 145 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 41 147 12.12. Changes included in 148 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 41 149 12.13. Changes included in 150 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 41 151 12.14. Changes included in 152 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 41 153 12.15. Changes included in 154 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 41 155 12.16. Changes included in 156 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 42 157 12.17. Changes included in 158 draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 43 159 12.18. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 43 160 12.19. Changes from 161 draft-hellstrom-avtcore-multi-party-rtt-source-03 to 162 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 43 163 12.20. Changes from 164 draft-hellstrom-avtcore-multi-party-rtt-source-02 to 165 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 43 166 12.21. Changes from 167 draft-hellstrom-avtcore-multi-party-rtt-source-01 to 168 -02 . . . . . . . . . . . . . . . . . . . . . . . . . . 44 169 12.22. Changes from 170 draft-hellstrom-avtcore-multi-party-rtt-source-00 to 171 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 45 172 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 45 173 13.1. Normative References . . . . . . . . . . . . . . . . . . 45 174 13.2. Informative References . . . . . . . . . . . . . . . . . 46 175 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 47 177 1. Introduction 179 "RTP Payload for Text Conversation" [RFC4103] specifies use of RTP 180 [RFC3550] for transmission of real-time text (RTT) and the "text/ 181 t140" format. It also specifies a redundancy format "text/red" for 182 increased robustness. The "text/red" format is registered in 183 [RFC4102]. 185 Real-time text is usually provided together with audio and sometimes 186 with video in conversational sessions. 188 A requirement related to multiparty sessions from the presentation 189 level standard T.140 [T140] for real-time text is: "The display of 190 text from the members of the conversation should be arranged so that 191 the text from each participant is clearly readable, and its source 192 and the relative timing of entered text is visualized in the 193 display." 194 Another requirement is that the mixing procedure must not introduce 195 delays in the text streams that are experienced to be disturbing the 196 real-time experience of the receiving users. 198 Use of RTT is increasing, and specifically, use in emergency calls is 199 increasing. Emergency call use requires multiparty mixing. RFC 4103 200 "RTP Payload for Text Conversation" mixer implementations can use 201 traditional RTP functions for source identification, but the 202 performance of the mixer when giving turns for the different sources 203 to transmit is limited when using the default transmission 204 characteristics with redundancy. 206 The redundancy scheme of [RFC4103] enables efficient transmission of 207 earlier transmitted redundant text in packets together with new text. 208 However, the redundancy header format has no source indicators for 209 the redundant transmissions. The redundant parts in a packet must 210 therefore be from the same source as the new text. The recommended 211 transmission is one new and two redundant generations of text 212 (T140blocks) in each packet and the recommended transmission interval 213 for two-party use is 300 ms. 215 Real-time text mixers for multiparty sessions need to include the 216 source with each transmitted group of text from a conference 217 participant so that the text can be transmitted interleaved with text 218 groups from different sources at the rate they are created. This 219 enables the text groups to be presented by endpoints in suitable 220 grouping with other text from the same source. 222 The presentation can then be arranged so that text from different 223 sources can be presented in real-time and easily read. At the same 224 time it is possible for a reading user to perceive approximately when 225 the text was created in real time by the different parties. The 226 transmission and mixing is intended to be done in a general way, so 227 that presentation can be arranged in a layout decided by the 228 endpoint. 230 There are existing implementations of RFC 4103 in endpoints without 231 the updates from this document. These will not be able to receive 232 and present real-time text mixed for multiparty-aware endpoints. 234 A negotiation mechanism is therefore needed for verification if the 235 parties are able to handle a common method for multiparty 236 transmission and agreeing on using that method. 238 A fallback mixing procedure is also needed for cases when the 239 negotiation result indicates that a receiving endpoint is not capable 240 of handling the mixed format. Multiparty-unaware endpoints would 241 possibly otherwise present all received multiparty mixed text as if 242 it came from the same source regardless of any accompanying source 243 indication coded in fields in the packet. Or they may have other 244 undesirable ways of acting on the multiparty content. The fallback 245 method is called the mixing procedure for multiparty-unaware 246 endpoints. The fallback method is naturally not expected to meet all 247 performance requirements placed on the mixing procedure for 248 multiparty-aware endpoints. 250 The document updates [RFC4103] by introducing an attribute for 251 indicating capability for the RTP-mixer-based multiparty mixing case 252 and rules for source indications and interleaving of text from 253 different sources. 255 1.1. Terminology 257 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 258 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 259 "OPTIONAL" in this document are to be interpreted as described in BCP 260 14 [RFC2119] [RFC8174] when, and only when, they appear in all 261 capitals, as shown above. 263 The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP- 264 mixer, RTP-translator are defined in [RFC3550]. 266 The term "T140block" is defined in [RFC4103] to contain one or more 267 T.140 code elements. 269 "TTY" stands for a textphone type used in North America. 271 "WebRTC" stands for web-based real-time communication specified by 272 W3C and IETF. See [RFC8825]. 274 "DTLS-SRTP" is a DTLS extension for use with SRTP/SRTCP specified in 275 [RFC5764]. 277 "multiparty-aware" describes an endpoint receiving real-time text 278 from multiple sources through a common conference mixer being able to 279 present the text in real-time, separated by source, and presented so 280 that a user can get an impression of the approximate relative timing 281 of text from different parties. 283 "multiparty-unaware" describes an endpoint not itself being able to 284 separate text from different sources when received through a common 285 conference mixer. 287 1.2. Selected solution and considered alternatives 289 A number of alternatives were considered when searching an efficient 290 and easily implemented multiparty method for real-time text. This 291 section explains a few of them briefly. 293 Multiple RTP streams, one per participant 294 One RTP stream per source would be sent in the same RTP session 295 with the "text/red" format. From some points of view, use of 296 multiple RTP streams, one for each source, sent in the same RTP 297 session would be efficient, and would use exactly the same packet 298 format as [RFC4103] and the same payload type. A couple of 299 relevant scenarios using multiple RTP-streams are specified in 300 "RTP Topologies" [RFC7667]. One possibility of special interest 301 is the Selective Forwarding Middlebox (SFM) topology specified in 302 RFC 7667 section 3.7 that could enable end-to-end encryption. In 303 contrast to audio and video, real-time text is only transmitted 304 when the users actually transmit information. Thus, an SFM 305 solution would not need to exclude any party from transmission 306 under normal conditions. In order to allow the mixer to convey 307 the packets with the payload preserved and encrypted, an SFM 308 solution would need to act on some specific characteristics of the 309 "text/red" format. The redundancy headers are part of the 310 payload, so the receiver would need to just assume that the 311 payload type number in the redundancy header is for "text/t140". 312 The characters per second parameter (CPS) would need to act per 313 stream. The relation between the SSRC and the source would need 314 to be conveyed in some specified way, e.g., in the CSRC. Recovery 315 and loss detection would preferably be based on sequence number 316 gap detection. Thus, sequence number gaps in the incoming stream 317 to the mixer would need to be reflected in the stream to the 318 participant, with no new gaps created by the mixer. However, the 319 RTP implementation in both mixers and endpoints need to support 320 multiple streams in the same RTP session in order to use this 321 mechanism. For best deployment opportunity, it should be possible 322 to upgrade existing endpoint solutions to be multiparty-aware with 323 a reasonable effort. There is currently a lack of support for 324 multi-stream RTP in certain implementations. This fact led to 325 this solution being only briefly mentioned in this document as an 326 option for further study. 328 RTP-mixer-based method for multiparty-aware endpoints 329 The "text/red" format in RFC 4103 is sent with a shorter 330 transmission interval with the RTP-mixer method and indicating the 331 source in the CSRC field. The "text/red" format with a "text/ 332 t140" payload in a single RTP stream can be sent when text is 333 available from the call participants instead of at the regular 300 334 ms. Transmission of packets with text from different sources can 335 then be done smoothly while simultaneous transmission occurs as 336 long as it is not limited by the maximum character rate "CPS". 337 With ten participants sending text simultaneously, the switching 338 and transmission performance is good. With more simultaneously 339 sending participants, and with receivers having the default 340 capacity there will be a noticeable jerkiness and delay in text 341 presentation. The jerkiness will be more expressed the more 342 participants who send text simultaneously. Two seconds jerkiness 343 will be noticeable and slightly unpleasant, but it corresponds in 344 time to what typing humans often cause by hesitation or changing 345 position while typing. A benefit of this method is that no new 346 packet format needs to be introduced and implemented. Since 347 simultaneous typing by more than two parties is very rare, this 348 method can be used successfully with good performance. Recovery 349 of text in case of packet loss is based on analysis of timestamps 350 of received redundancy versus earlier received text. Negotiation 351 is based on a new SDP media attribute "rtt-mixer". This method is 352 selected to be the main one specified in this document. 354 Multiple sources per packet 355 A new "text" media subtype would be specified with up to 15 356 sources in each packet. The mechanism would make use of the RTP 357 mixer model specified in RTP [RFC3550]. Text from up to 15 358 sources can be included in each packet. Packets are normally sent 359 with 300 ms intervals. The mean delay will be 150 ms. The 360 sources are indicated in strict order in the CSRC list of the RTP 361 packets. A new redundancy packet format is specified. This 362 method would result in good performance, but would require 363 standardization and implementation of new releases in the target 364 technologies that would take more time than desirable to complete. 365 It was therefore not selected to be included in this document. 367 Mixing for multiparty-unaware endpoints 368 Presentation of text from multiple parties is prepared by the 369 mixer in one single stream. It is desirable to have a method that 370 does not require any modifications in existing user devices 371 implementing RFC 4103 for RTT without explicit support of 372 multiparty sessions. This is possible by having the mixer insert 373 a new line and a text formatted source label before each switch of 374 text source in the stream. Switch of source can only be done in 375 places in the text where it does not disturb the perception of the 376 contents. Text from only one source can be presented in real time 377 at a time. The delay will therefore vary. The method also has 378 other limitations, but is included in this document as a fallback 379 method. In calls where parties take turns properly by ending 380 their entries with a new line, the limitations will have limited 381 influence on the user experience. when only two parties send text, 382 these two will see the text in real time with no delay. This 383 method is specified as a fallback method in this document. 385 RTT transport in WebRTC 386 Transport of real-time text in the WebRTC technology is specified 387 to use the WebRTC data channel in [RFC8865]. That specification 388 contains a section briefly describing its use in multiparty 389 sessions. The focus of this document is RTP transport. 390 Therefore, even if the WebRTC transport provides good multiparty 391 performance, it is just mentioned in this document in relation to 392 providing gateways with multiparty capabilities between RTP and 393 WebRTC technologies. 395 1.3. Intended application 397 The method for multiparty real-time text specified in this document 398 is primarily intended for use in transmission between mixers and 399 endpoints in centralized mixing configurations. It is also 400 applicable between mixers. An often mentioned application is for 401 emergency service calls with real-time text and voice, where a call 402 taker wants to make an attended handover of a call to another agent, 403 and stay to observe the session. Multimedia conference sessions with 404 support for participants to contribute in text is another 405 application. Conferences with central support for speech-to-text 406 conversion is yet another mentioned application. 408 In all these applications, normally only one participant at a time 409 will send long text utterances. In some cases, one other participant 410 will occasionally contribute with a longer comment simultaneously. 411 That may also happen in some rare cases when text is interpreted to 412 text in another language in a conference. Apart from these cases, 413 other participants are only expected to contribute with very brief 414 utterings while others are sending text. 416 Users expect that the text they send is presented in real-time in a 417 readable way to the other participants even if they send 418 simultaneously with other users and even when they make brief edit 419 operations of their text by backspacing and correcting their text. 421 Text is supposed to be human generated, by some text input means, 422 such as typing on a keyboard or using speech-to-text technology. 423 Occasional small cut-and-paste operations may appear even if that is 424 not the initial purpose of real-time text. 426 The real-time characteristics of real-time text is essential for the 427 participants to be able to contribute to a conversation. If the text 428 is too much delayed from typing a letter to its presentation, then, 429 in some conference situations, the opportunity to comment will be 430 gone and someone else will grab the turn. A delay of more than one 431 second in such situations is an obstacle for good conversation. 433 2. Overview of the two specified solutions and selection of method 435 This section contains a brief introduction of the two methods 436 specified in this document. 438 2.1. The RTP-mixer-based solution for multiparty-aware endpoints 440 This method specifies negotiated use of the RFC 4103 format for 441 multiparty transmission in a single RTP stream. The main purpose of 442 this document is to specify a method for true multiparty real-time 443 text mixing for multiparty-aware endpoints that can be widely 444 deployed. The RTP-mixer-based method makes use of the current format 445 for real-time text in [RFC4103]. It is an update of RFC 4103 by a 446 clarification on one way to use it in the multiparty situation. That 447 is done by completing a negotiation for this kind of multiparty 448 capability and by interleaving packets from different sources. The 449 source is indicated in the CSRC element in the RTP packets. Specific 450 considerations are made to be able to recover text after packet loss. 452 The detailed procedures for the RTP-mixer-based multiparty-aware case 453 are specified in Section 3. 455 Please use [RFC4103] as reference when reading the specification. 457 2.2. Mixing for multiparty-unaware endpoints 459 A method is also specified in this document for cases when the 460 endpoint participating in a multiparty call does not itself implement 461 any solution, or not the same, as the mixer. The method requires the 462 mixer to insert text dividers and readable labels and only send text 463 from one source at a time until a suitable point appears for source 464 change. This solution is a fallback method with functional 465 limitations. It acts on the presentation level. 467 A party acting as a mixer, which has not negotiated any method for 468 true multiparty RTT handling, but negotiated a "text/red" or "text/ 469 t140" format in a session with a participant SHOULD in order to 470 maintain interoperability, if nothing else is specified for the 471 application, format transmitted text to that participant to be 472 suitable to present on a multiparty-unaware endpoint as further 473 specified in Section 4.2. 475 2.3. Offer/answer considerations 477 RTP Payload for Text Conversation [RFC4103] specifies use of RTP 478 [RFC3550], and a redundancy format "text/red" for increased 479 robustness of real-time text transmission. This document updates 480 [RFC4103] by introducing a capability negotiation for handling 481 multiparty real-time text, a way to indicate the source of 482 transmitted text, and rules for efficient timing of the transmissions 483 interleaved from different sources. 485 The capability negotiation for the "RTP-mixer-based multiparty 486 method" is based on use of the SDP media attribute "rtt-mixer". 488 The syntax is as follows: 489 "a=rtt-mixer" 491 If any other method for RTP-based multiparty real-time text gets 492 specified by additional work, it is assumed that it will be 493 recognized by some specific SDP feature exchange. 495 2.3.1. Initial offer 497 A party intending to set up a session and being willing to use the 498 RTP-mixer-based method of this specification for sending or receiving 499 or both sending and receiving real-time text SHALL include the "rtt- 500 mixer" SDP attribute in the corresponding "text" media section in the 501 initial offer. 503 The party MAY indicate capability for both the RTP-mixer-based method 504 of this specification and other methods. 506 When the offeror has sent the offer including the "rtt-mixer" 507 attribute, it MUST be prepared to receive and handle real-time text 508 formatted according to both the method for multiparty-aware parties 509 specified in Section 3 in this specification and two-party formatted 510 real-time text. 512 2.3.2. Answering the offer 514 A party receiving an offer containing the "rtt-mixer" SDP attribute 515 and being willing to use the RTP-mixer-based method of this 516 specification for sending or receiving or both sending and receiving 517 SHALL include the "rtt-mixer" SDP attribute in the corresponding 518 "text" media section in the answer. 520 If the offer did not contain the "rtt-mixer" attribute, the answer 521 MUST NOT contain the "rtt-mixer" attribute. 523 An answer MUST NOT include acceptance of more than one method for 524 multiparty real-time text in the same RTP session. 526 When the answer including acceptance is transmitted, the answerer 527 MUST be prepared to act on received text in the negotiated session 528 according to the method for multiparty-aware parties specified in 529 Section 3 of this specification. Reception of text for a two-party 530 session SHALL also be supported. 532 2.3.3. Offeror processing the answer 534 When the answer is processed by the offeror, it MUST act as specified 535 in Section 2.4 537 2.3.4. Modifying a session 539 A session MAY be modified at any time by any party offering a 540 modified SDP with or without the "rtt-mixer" SDP attribute expressing 541 a desired change in the support of multiparty real-time text. 543 If the modified offer adds indication of support for multiparty real- 544 time text by including the "rtt-mixer" SDP attribute, the procedures 545 specified in the previous subsections SHALL be applied. 547 If the modified offer deletes indication of support for multiparty 548 real-time text by excluding the "rtt-mixer" SDP attribute, the answer 549 MUST NOT contain the "rtt-mixer" attribute, and both parties SHALL 550 after processing the SDP exchange NOT send real-time text formatted 551 for multiparty-aware parties according to this specification. 553 2.4. Actions depending on capability negotiation result 555 A transmitting party SHALL send text according to the RTP-mixer-based 556 multiparty method only when the negotiation for that method was 557 successful and when it conveys text for another source. In all other 558 cases, the packets SHALL be populated and interpreted as for a two- 559 party session. 561 A party which has negotiated the "rtt-mixer" SDP media attribute MUST 562 populate the CSRC-list, and format the packets according to Section 3 563 if it acts as an rtp-mixer and sends multiparty text. 565 A party which has negotiated the "rtt-mixer" SDP media attribute MUST 566 interpret the contents of the "CC" field, the CSRC-list and the 567 packets according to Section 3 in received RTP packets in the 568 corresponding RTP stream. 570 A party which has not successfully completed the negotiation of the 571 "rtt-mixer" SDP media attribute MUST NOT transmit packets interleaved 572 from different sources in the same RTP stream as specified in 573 Section 3. If the party is a mixer and did declare the "rtt-mixer" 574 SDP media attribute, it SHOULD perform the procedure for multiparty- 575 unaware endpoints. If the party is not a mixer, it SHOULD transmit 576 as in a two-party session according to [RFC4103]. 578 3. Details for the RTP-mixer-based mixing method for multiparty-aware 579 endpoints 581 3.1. Use of fields in the RTP packets 583 The CC field SHALL show the number of members in the CSRC list, which 584 SHALL be one (1) in transmissions from a mixer when conveying text 585 from other sources in a multiparty session, and otherwise 0. 587 When text is conveyed by a mixer during a multiparty session, a CSRC 588 list SHALL be included in the packet. The single member in the CSRC- 589 list SHALL contain the SSRC of the source of the T140blocks in the 590 packet. 592 When redundancy is used, the RECOMMENDED level of redundancy is to 593 use one primary and two redundant generations of T140blocks. In some 594 cases, a primary or redundant T140block is empty, but is still 595 represented by a member in the redundancy header. 597 In other regards, the contents of the RTP packets are equal to what 598 is specified in [RFC4103]. 600 3.2. Initial transmission of a BOM character 602 As soon as a participant is known to participate in a session with 603 another entity and is available for text reception, a Unicode BOM 604 character SHALL be sent to it by the other entity according to the 605 procedures in this section. If the transmitter is a mixer, then the 606 source of this character SHALL be indicated to be the mixer itself. 608 Note that the BOM character SHALL be transmitted with the same 609 redundancy procedures as any other text. 611 3.3. Keep-alive 613 After that, the transmitter SHALL send keep-alive traffic to the 614 receiver(s) at regular intervals when no other traffic has occurred 615 during that interval, if that is decided for the actual connection. 616 It is RECOMMENDED to use the keep-alive solution from [RFC6263]. The 617 consent check of [RFC7675] is a possible alternative if it is used 618 anyway for other reasons. 620 3.4. Transmission interval 622 A "text/red" or "text/t140" transmitter in a mixer SHALL send packets 623 distributed in time as long as there is something (new or redundant 624 T140blocks) to transmit. The maximum transmission interval SHALL 625 then be 330 ms, when no other limitations cause a longer interval to 626 be temporarily used. It is RECOMMENDED to send the next packet to a 627 receiver as soon as new text to that receiver is available, as long 628 as the maximum character rate ("CPS") to the receiver is not exceeded 629 during any 10-second interval. The intention of these time intervals 630 is to keep the latency low and network load limited while keeping 631 good protection against text loss in bursty packet loss conditions. 632 The main purpose of the 330 ms interval is for timing of redundant 633 transmission, when no new text from the same source is available. 635 If the "CPS" value is reached, longer transmission intervals SHALL be 636 applied and only as much of the text queued for transmission SHALL be 637 sent at the end of each transmission interval that can be allowed 638 without exceeding the "CPS" value, until the transmission rate falls 639 under the "CPS" value again. Division of text for partial 640 transmission MUST then be made at T140block borders. See also 641 Section 8 643 For a transmitter not acting as a mixer, the transmission interval 644 principles from [RFC4103] apply, and the transmission interval SHALL 645 be 300 ms. 647 3.5. Only one source per packet 649 New text and redundant copies of earlier text from one source SHALL 650 be transmitted in the same packet if available for transmission at 651 the same time. Text from different sources MUST NOT be transmitted 652 in the same packet. 654 3.6. Do not send received text to the originating source 656 Text received by a mixer from a participant SHOULD NOT be included in 657 transmission from the mixer to that participant, because the normal 658 behavior of the endpoint is to present locally-produced text locally. 660 3.7. Clean incoming text 662 A mixer SHALL handle reception, recovery from packet loss, deletion 663 of superfluous redundancy, marking of possible text loss and deletion 664 of 'BOM' characters from each participant before queueing received 665 text for transmission to receiving participants. 667 3.8. Redundant transmission principles 669 A transmitting party using redundancy SHALL send redundant 670 repetitions of T140blocks already transmitted in earlier packets. 672 The number of redundant generations of T140blocks to include in 673 transmitted packets SHALL be deduced from the SDP negotiation. It 674 SHALL be set to the minimum of the number declared by the two parties 675 negotiating a connection. It is RECOMMENDED to declare and transmit 676 one original and two redundant generations of the T140blocks because 677 that provides good protection against text loss in case of packet 678 loss, and low overhead. 680 3.9. Interleaving text from different sources 682 When text from more than one source is available for transmission 683 from a mixer, the mixer SHALL let the sources take turns in having 684 their text transmitted. 686 The source with the oldest text received in the mixer or oldest 687 redundant text SHALL be next in turn to get all its available unsent 688 text transmitted. Any redundant repetitions of earlier transmitted 689 text not yet sent the intended number of times SHALL be included as 690 redundant retransmission in the transmission. 692 3.10. Text placement in packets 694 The mixer SHALL compose and transmit an RTP packet to a receiver when 695 one of the following conditions has occurred: 697 * There is unsent text available for transmission to that receiver. 699 * 330 ms has passed since already transmitted text was queued for 700 transmission as redundant text. 702 At the time of transmission, the mixer SHALL populate the RTP packet 703 with all T140blocks queued for transmission originating from the 704 source in turn for transmission as long as this is not in conflict 705 with the allowed number of characters per second ("CPS") or the 706 maximum packet size. In this way, the latency of the latest received 707 text is kept low even in moments of simultaneous transmission from 708 many sources. 710 Redundant text SHALL also be included, and the assessment of how much 711 new text can be included within the maximum packet size MUST take 712 into account that the redundancy has priority to be transmitted in 713 its entirety. See Section 3.12 715 The SSRC of the source SHALL be placed as the only member in the 716 CSRC-list. 718 Note: The CSRC-list in an RTP packet only includes the participant 719 whose text is included in text blocks. It is not the same as the 720 total list of participants in a conference. With audio and video 721 media, the CSRC-list would often contain all participants who are not 722 muted whereas text participants that don't type are completely silent 723 and thus are not represented in RTP packet CSRC-lists. 725 3.11. Empty T140blocks 727 If no unsent T140blocks were available for a source at the time of 728 populating a packet, but T140blocks are available which have not yet 729 been sent the full intended number of redundant transmissions, then 730 the primary T140block for that source is composed of an empty 731 T140block, and populated (without taking up any length) in a packet 732 for transmission. The corresponding SSRC SHALL be placed as usual in 733 its place in the CSRC-list. 735 The first packet in the session, the first after a source switch, and 736 the first after a pause SHALL be populated with the available 737 T140blocks for the source in turn to be sent as primary, and empty 738 T140blocks for the agreed number of redundancy generations. 740 3.12. Creation of the redundancy 742 The primary T140block from a source in the latest transmitted packet 743 is saved for populating the first redundant T140block for that source 744 in the next transmission of text from that source. The first 745 redundant T140block for that source from the latest transmission is 746 saved for populating the second redundant T140block in the next 747 transmission of text from that source. 749 Usually this is the level of redundancy used. If a higher level of 750 redundancy is negotiated, then the procedure SHALL be maintained 751 until all available redundant levels of T140blocks are placed in the 752 packet. If a receiver has negotiated a lower number of "text/red" 753 generations, then that level SHALL be the maximum used by the 754 transmitter. 756 The T140blocks saved for transmission as redundant data are assigned 757 a planned transmission time 330 ms after the current time, but SHOULD 758 be transmitted earlier if new text for the same source gets in turn 759 for transmission before that time. 761 3.13. Timer offset fields 763 The timestamp offset values SHALL be inserted in the redundancy 764 header, with the time offset from the RTP timestamp in the packet 765 when the corresponding T140block was sent as primary. 767 The timestamp offsets are expressed in the same clock tick units as 768 the RTP timestamp. 770 The timestamp offset values for empty T140blocks have no relevance 771 but SHOULD be assigned realistic values. 773 3.14. Other RTP header fields 775 The number of members in the CSRC list (0 or 1) SHALL be placed in 776 the "CC" header field. Only mixers place value 1 in the "CC" field. 777 A value of "0" indicates that the source is the transmitting device 778 itself and that the source is indicated by the SSRC field. This 779 value is used by endpoints, and by mixers sending self-sourced data. 781 The current time SHALL be inserted in the timestamp. 783 The SSRC of the mixer for the RTT session SHALL be inserted in the 784 SSRC field of the RTP header. 786 The M-bit SHALL be handled as specified in [RFC4103]. 788 3.15. Pause in transmission 790 When there is no new T140block to transmit, and no redundant 791 T140block that has not been retransmitted the intended number of 792 times from any source, the transmission process SHALL be stopped 793 until either new T140blocks arrive, or a keep-alive method calls for 794 transmission of keep-alive packets. 796 3.16. RTCP considerations 798 A mixer SHALL send RTCP reports with SDES, CNAME, and NAME 799 information about the sources in the multiparty call. This makes it 800 possible for participants to compose a suitable label for text from 801 each source. 803 Confidentiality SHALL be considered when composing these fields. 804 They contain name and address information that may be sensitive to 805 transmit in its entirety e.g., to unauthenticated participants. 806 Similar considerations SHALL be taken as for other media. 808 3.17. Reception of multiparty contents 810 The "text/red" receiver included in an endpoint with presentation 811 functions will receive RTP packets in the single stream from the 812 mixer, and SHALL distribute the T140blocks for presentation in 813 presentation areas for each source. Other receiver roles, such as 814 gateways or chained mixers are also feasible, and requires 815 consideration if the stream shall just be forwarded, or distributed 816 based on the different sources. 818 3.17.1. Acting on the source of the packet contents 820 If the "CC" field value of a received packet is 1, it indicates that 821 the text is conveyed from a source indicated in the single member in 822 the CSRC-list, and the receiver MUST act on the source according to 823 its role. If the CC value is 0, the source is indicated in the SSRC 824 field. 826 3.17.2. Detection and indication of possible text loss 828 The receiver SHALL monitor the RTP sequence numbers of the received 829 packets for gaps and packets out of order. If a sequence number gap 830 appears and still exists after some defined short time for jitter 831 resolution, the packets in the gap SHALL be regarded as lost. 833 If it is known that only one source is active in the RTP session, 834 then it is likely that a gap equal to or larger than the agreed 835 number of redundancy generations (including the primary) causes text 836 loss. In that case, the receiver SHALL create a t140block with a 837 marker for possible text loss [T140ad1] and associate it with the 838 source and insert it in the reception buffer for that source. 840 If it is known that more than one source is active in the RTP 841 session, then it is not possible in general to evaluate if text was 842 lost when packets were lost. With two active sources and the 843 recommended number of redundancy generations (3), it can take a gap 844 of five consecutive lost packets until any text may be lost, but text 845 loss can also appear if three non-consecutive packets are lost when 846 they contained consecutive data from the same source. A simple 847 method to decide when there is risk for resulting text loss is to 848 evaluate if three or more packets were lost within one second. If 849 this simple method is used, then a t140block SHOULD be created with a 850 marker for possible text loss [T140ad1] and associated with the SSRC 851 of the transmitter as a general input from the mixer. 853 Implementations MAY apply more refined methods for more reliable 854 detection of whether text was lost or not. Any refined method SHOULD 855 prefer marking possible loss rather than not marking when it is 856 uncertain if there was loss. 858 3.17.3. Extracting text and handling recovery 860 When applying the following procedures, the effects MUST be 861 considered of possible timestamp wrap around and the RTP session 862 possibly changing SSRC. 864 When a packet is received in an RTP session using the packetization 865 for multiparty-aware endpoints, its T140blocks SHALL be extracted in 866 the following way. The description is adapted to the default 867 redundancy case using the original and two redundant generations. 869 The source SHALL be extracted from the CSRC-list if available, 870 otherwise from the SSRC. 872 If the received packet is the first packet received from the source, 873 then all T140blocks in the packet SHALL be retrieved and assigned to 874 a receive buffer for the source beginning with the second generation 875 redundancy, continuing with the first generation redundancy and 876 finally the primary. 878 Note: The normal case is that in the first packet, only the primary 879 data has contents. The redundant data has contents in the first 880 received packet from a source only after initial packet loss. 882 If the packet is not the first packet from a source, then if the 883 second generation redundant data is available, its timestamp SHALL be 884 created by subtracting its timestamp offset from the RTP timestamp. 885 If the resulting timestamp is later than the latest retrieved data 886 from the same source, then the redundant data SHALL be retrieved and 887 appended to the receive buffer. The process SHALL be continued in 888 the same way for the first generation redundant data. After that, 889 the primary data SHALL be retrieved from the packet and appended to 890 the receive buffer for the source. 892 3.17.4. Delete 'BOM' 894 Unicode character 'BOM' is used as a start indication and sometimes 895 used as a filler or keep alive by transmission implementations. 896 These SHALL be deleted after extraction from received packets. 898 3.18. Performance considerations 900 This solution has good performance with low text delays as long as 901 the sum of characters per second during any 10-second interval sent 902 from a number of simultaneously sending participants to a receiving 903 participant does not reach the 'CPS' value. At higher numbers of 904 characters per second sent, a jerkiness is visible in the 905 presentation of text. The solution is therefore suitable for 906 emergency service use, relay service use, and small or well-managed 907 larger multimedia conferences. Only in large unmanaged conferences 908 with a high number of participants there may on very rare occasions 909 appear situations when many participants happen to send text 910 simultaneously, resulting in unpleasantly jerky presentation of text 911 from each sending participant. It should be noted that it is only 912 the number of users sending text within the same moment that causes 913 jerkiness, not the total number of users with RTT capability. 915 3.19. Security for session control and media 917 Security SHOULD be applied when possible regarding the capabilities 918 of the participating devices by use of SIP over TLS by default 919 according to [RFC5630] section 3.1.3 on the session control level and 920 by default using DTLS-SRTP [RFC5764] on the media level. In 921 applications where legacy endpoints without security may exist, a 922 negotiation SHOULD be performed to decide if encryption on the media 923 level will be applied. If no other security solution is mandated for 924 the application, then OSRTP [RFC8643] is a suitable method to be 925 applied to negotiate SRTP media security with DTLS. Most SDP 926 examples below are for simplicity expressed without the security 927 additions. The principles (but not all details) for applying DTLS- 928 SRTP [RFC5764] security are shown in a couple of the following 929 examples. 931 This document contains two mixing procedures which imply different 932 security levels. The mixing for conference-unaware endpoints has 933 lower security level than the mixing method for conference-aware 934 endpoints, because there may be an opportunity for a malicious mixer 935 or a middleman to masquerade the source labels accompanying the text 936 streams in text format. This is especially true if support of un- 937 encrypted SIP and media is supported because of lack of such support 938 in the target endpoints. However, the mixing for conference-aware 939 endpoints as specified here also requires that the mixer can be 940 trusted. Further general security considerations are covered in 941 Section 11. 943 End-to-end encryption would require further work and could be based 944 on WebRTC as specified in Section 1.2. 946 3.20. SDP offer/answer examples 948 This section shows some examples of SDP for session negotiation of 949 the real-time text media in SIP sessions. Audio is usually provided 950 in the same session, and sometimes also video. The examples only 951 show the part of importance for the real-time text media. The 952 examples relate to the single RTP stream mixing for multiparty-aware 953 endpoints and for multiparty-unaware endpoints. 955 Note: Multiparty RTT MAY also be provided through other methods, 956 e.g., by a Selective Forwarding Middlebox (SFM). In that case, the 957 SDP of the offer will include something specific for that method, 958 e.g., an SDP attribute or another media format. An answer selecting 959 the use of that method would accept it by a corresponding 960 acknowledgement included in the SDP. The offer may contain also the 961 "rtt-mixer" SDP media attribute for the main RTT media when the 962 offeror has capability for both multiparty methods, while an answer, 963 selecting to use SFM will not include the "rtt-mixer" SDP media 964 attribute. 966 Offer example for "text/red" format and multiparty support: 968 m=text 11000 RTP/AVP 100 98 969 a=rtpmap:98 t140/1000 970 a=rtpmap:100 red/1000 971 a=fmtp:100 98/98/98 972 a=rtt-mixer 974 Answer example from a multiparty-aware device 975 m=text 14000 RTP/AVP 100 98 976 a=rtpmap:98 t140/1000 977 a=rtpmap:100 red/1000 978 a=fmtp:100 98/98/98 979 a=rtt-mixer 981 Offer example for "text/red" format including multiparty 982 and security: 983 a=fingerprint: (fingerprint1) 984 m=text 11000 RTP/AVP 100 98 985 a=rtpmap:98 t140/1000 986 a=rtpmap:100 red/1000 987 a=fmtp:100 98/98/98 988 a=rtt-mixer 990 The "fingerprint" is sufficient to offer DTLS-SRTP, with the media 991 line still indicating RTP/AVP. 993 Note: For brevity, the entire value of the SDP fingerprint attribute 994 is not shown in this and the following example. 996 Answer example from a multiparty-aware device with security 997 a=fingerprint: (fingerprint2) 998 m=text 16000 RTP/AVP 100 98 999 a=rtpmap:98 t140/1000 1000 a=rtpmap:100 red/1000 1001 a=fmtp:100 98/98/98 1002 a=rtt-mixer 1004 With the "fingerprint" the device acknowledges use of SRTP/DTLS. 1006 Answer example from a multiparty-unaware device that also 1007 does not support security: 1009 m=text 12000 RTP/AVP 100 98 1010 a=rtpmap:98 t140/1000 1011 a=rtpmap:100 red/1000 1012 a=fmtp:100 98/98/98 1014 3.21. Packet sequence example from interleaved transmission 1016 This example shows a symbolic flow of packets from a mixer including 1017 loss and recovery. The sequence includes interleaved transmission of 1018 text from two RTT sources A and B. P indicates primary data. R1 is 1019 first redundant generation data and R2 is the second redundant 1020 generation data. A1, B1, A2 etc. are text chunks (T140blocks) 1021 received from the respective sources and sent on to the receiver by 1022 the mixer. X indicates a dropped packet between the mixer and a 1023 receiver. The session is assumed to use original and two redundant 1024 generations of RTT. 1026 |-----------------------| 1027 |Seq no 101, Time=20400 | 1028 |CC=1 | 1029 |CSRC list A | 1030 |R2: A1, Offset=600 | 1031 |R1: A2, Offset=300 | 1032 |P: A3 | 1033 |-----------------------| 1035 Assuming that earlier packets (with text A1 and A2) were received in 1036 sequence, text A3 is received from packet 101 and assigned to 1037 reception buffer A. The mixer is now assumed to have received 1038 initial text from source B 100 ms after packet 101 and will send that 1039 text. Transmission of A2 and A3 as redundancy is planned for 330 ms 1040 after packet 101 if no new text from A is ready to be sent before 1041 that. 1043 |-----------------------| 1044 |Seq no 102, Time=20500 | 1045 |CC=1 | 1046 |CSRC list B | 1047 |R2 Empty, Offset=600 | 1048 |R1: Empty, Offset=300 | 1049 |P: B1 | 1050 |-----------------------| 1051 Packet 102 is received. 1052 B1 is retrieved from this packet. Redundant transmission of 1053 B1 is planned 330 ms after packet 102. 1055 X------------------------| 1056 X Seq no 103, Timer=20730| 1057 X CC=1 | 1058 X CSRC list A | 1059 X R2: A2, Offset=630 | 1060 X R1: A3, Offset=330 | 1061 X P: Empty | 1062 X------------------------| 1063 Packet 103 is assumed to be lost due to network problems. 1064 It contains redundancy for A. Sending A3 as second level 1065 redundancy is planned for 330 ms after packet 103. 1067 X------------------------| 1068 X Seq no 104, Timer=20830| 1069 X CC=1 | 1070 X CSRC list B | 1071 X R2: Empty, Offset=600 | 1072 X R1: B1, Offset=300 | 1073 X P: B2 | 1074 X------------------------| 1075 Packet 104 contains text from B, including new B2 and 1076 redundant B1. It is assumed dropped due to network 1077 problems. 1078 The mixer has A3 redundancy to send, but no new text 1079 appears from A and therefore the redundancy is sent 1080 330 ms after the previous packet with text from A. 1082 |------------------------| 1083 | Seq no 105, Timer=21060| 1084 | CC=1 | 1085 | CSRC list A | 1086 | R2: A3, Offset=660 | 1087 | R1: Empty, Offset=330 | 1088 | P: Empty | 1089 |------------------------| 1090 Packet 105 is received. 1091 A gap for lost packets 103 and 104 is detected. 1092 Assume that no other loss was detected during the last second. 1093 Then it can be concluded that nothing was totally lost. 1095 R2 is checked. Its original time was 21060-660=20400. 1096 A packet with text from A was received with that 1097 timestamp, so nothing needs to be recovered. 1099 B1 and B2 still need to be transmitted as redundancy. 1100 This is planned 330 ms after packet 105. That 1101 would be at 21160. 1103 |-----------------------| 1104 |Seq no 106, Timer=21160| 1105 |CC=1 | 1106 |CSRC list B | 1107 | R2: B1, Offset=660 | 1108 | R1: B2, Offset=330 | 1109 | P: Empty | 1110 |-----------------------| 1112 Packet 106 is received. 1114 The second level redundancy in packet 106 is B1 and has timestamp 1115 offset 660 ms. The timestamp of packet 106 minus 660 is 20500 which 1116 is the timestamp of packet 102 THAT was received. So B1 does not 1117 need to be retrieved. The first level redundancy in packet 106 has 1118 offset 330. The timestamp of packet 106 minus 330 is 20830. That is 1119 later than the latest received packet with source B. Therefore B2 is 1120 retrieved and assigned to the input buffer for source B. No primary 1121 is available in packet 106. 1123 After this sequence, A3 and B1 and B2 have been received. In this 1124 case no text was lost. 1126 3.22. Maximum character rate "CPS" 1128 The default maximum rate of reception of "text/t140" real-time text 1129 is in [RFC4103] specified to be 30 characters per second. The actual 1130 rate is calculated without regard to any redundant text transmission 1131 and is in the multiparty case evaluated for all sources contributing 1132 to transmission to a receiver. The value MAY be modified in the 1133 "CPS" parameter of the FMTP attribute in the media section for the 1134 "text/t140" media. A mixer combining real-time text from a number of 1135 sources may occasionally have a higher combined flow of text coming 1136 from the sources. Endpoints SHOULD therefore specify a suitable 1137 higher value for the "CPS" parameter, corresponding to its real 1138 reception capability. A value for "CPS" of 90 SHALL be the default 1139 for the "text/t140" stream in the "text/red" format when multiparty 1140 real-time text is negotiated. See [RFC4103] for the format and use 1141 of the "CPS" parameter. The same rules apply for the multiparty case 1142 except for the default value. 1144 4. Presentation level considerations 1146 "Protocol for multimedia application text conversation" [T140] 1147 provides the presentation level requirements for the [RFC4103] 1148 transport. Functions for erasure and other formatting functions are 1149 specified in [T140] which has the following general statement for the 1150 presentation: 1152 "The display of text from the members of the conversation should be 1153 arranged so that the text from each participant is clearly readable, 1154 and its source and the relative timing of entered text is visualized 1155 in the display. Mechanisms for looking back in the contents from the 1156 current session should be provided. The text should be displayed as 1157 soon as it is received." 1158 Strict application of [T140] is of essence for the interoperability 1159 of real-time text implementations and to fulfill the intention that 1160 the session participants have the same information conveyed in the 1161 text contents of the conversation without necessarily having the 1162 exact same layout of the conversation. 1164 [T140] specifies a set of presentation control codes to include in 1165 the stream. Some of them are optional. Implementations MUST be able 1166 to ignore optional control codes that they do not support. 1168 There is no strict "message" concept in real-time text. The Unicode 1169 Line Separator character SHALL be used as a separator allowing a part 1170 of received text to be grouped in presentation. The characters 1171 "CRLF" may be used by other implementations as a replacement for Line 1172 Separator. The "CRLF" combination SHALL be erased by just one 1173 erasing action, the same as the Line Separator. Presentation 1174 functions are allowed to group text for presentation in smaller 1175 groups than the line separators imply and present such groups with 1176 source indication together with text groups from other sources (see 1177 the following presentation examples). Erasure has no specific limit 1178 by any delimiter in the text stream. 1180 4.1. Presentation by multiparty-aware endpoints 1182 A multiparty-aware receiving party, presenting real-time text MUST 1183 separate text from different sources and present them in separate 1184 presentation fields. The receiving party MAY separate presentation 1185 of parts of text from a source in readable groups based on other 1186 criteria than line separator and merge these groups in the 1187 presentation area when it benefits the user to most easily find and 1188 read text from the different participants. The criteria MAY e.g., be 1189 a received comma, full stop, or other phrase delimiters, or a long 1190 pause. 1192 When text is received from multiple original sources, the 1193 presentation SHALL provide a view where text is added in multiple 1194 presentation fields. 1196 If the presentation presents text from different sources in one 1197 common area, the presenting endpoint SHOULD insert text from the 1198 local user ended at suitable points merged with received text to 1199 indicate the relative timing for when the text groups were completed. 1200 In this presentation mode, the receiving endpoint SHALL present the 1201 source of the different groups of text. This presentation style is 1202 called the "chat" style here and provides a possibility to follow 1203 text arriving from multiple parties and the approximate relative time 1204 that text is received related to text from the local user. 1206 A view of a three-party RTT call in chat style is shown in this 1207 example . 1209 _________________________________________________ 1210 | |^| 1211 |[Alice] Hi, Alice here. |-| 1212 | | | 1213 |[Bob] Bob as well. | | 1214 | | | 1215 |[Eve] Hi, this is Eve, calling from Paris. | | 1216 | I thought you should be here. | | 1217 | | | 1218 |[Alice] I am coming on Thursday, my | | 1219 | performance is not until Friday morning.| | 1220 | | | 1221 |[Bob] And I on Wednesday evening. | | 1222 | | | 1223 |[Alice] Can we meet on Thursday evening? | | 1224 | | | 1225 |[Eve] Yes, definitely. How about 7pm. | | 1226 | at the entrance of the restaurant | | 1227 | Le Lion Blanc? | | 1228 |[Eve] we can have dinner and then take a walk |-| 1229 |______________________________________________|v| 1230 | But I need to be back to |^| 1231 | the hotel by 11 because I need |-| 1232 | | | 1233 | I wou |-| 1234 |______________________________________________|v| 1235 | of course, I underst | 1236 |________________________________________________| 1238 Figure 3: Example of a three-party RTT call presented in chat style 1239 seen at participant 'Alice's endpoint. 1241 Other presentation styles than the chat style MAY be arranged. 1243 This figure shows how a coordinated column view MAY be presented. 1245 _____________________________________________________________________ 1246 | Bob | Eve | Alice | 1247 |____________________|______________________|_______________________| 1248 | | |I will arrive by TGV. | 1249 |My flight is to Orly| |Convenient to the main | 1250 | |Hi all, can we plan |station. | 1251 | |for the seminar? | | 1252 |Eve, will you do | | | 1253 |your presentation on| | | 1254 |Friday? |Yes, Friday at 10. | | 1255 |Fine, wo | |We need to meet befo | 1256 |___________________________________________________________________| 1258 Figure 4: An example of a coordinated column-view of a three-party 1259 session with entries ordered vertically in approximate time-order. 1261 4.2. multiparty mixing for multiparty-unaware endpoints 1263 When the mixer has indicated RTT multiparty capability in an SDP 1264 negotiation, but the multiparty capability negotiation fails with an 1265 endpoint, then the agreed "text/red" or "text/t140" format SHALL be 1266 used and the mixer SHOULD compose a best-effort presentation of 1267 multiparty real-time text in one stream intended to be presented by 1268 an endpoint with no multiparty awareness, when that is desired in the 1269 actual implementation. The following specifies a procedure which MAY 1270 be applied in that situation. 1272 This presentation format has functional limitations and SHOULD be 1273 used only to enable participation in multiparty calls by legacy 1274 deployed endpoints implementing only RFC 4103 without any multiparty 1275 extensions specified in this document. 1277 The principles and procedures below do not specify any new protocol 1278 elements. They are instead composed of information from [T140] and 1279 an ambition to provide a best-effort presentation on an endpoint 1280 which has functions originally intended only for two-party calls. 1282 The mixer mixing for multiparty-unaware endpoints SHALL compose a 1283 simulated, limited multiparty RTT view suitable for presentation in 1284 one presentation area. The mixer SHALL group text in suitable groups 1285 and prepare for presentation of them by inserting a new line between 1286 them if the transmitted text did not already end with a new line. A 1287 presentable label SHALL be composed and sent for the source initially 1288 in the session and after each source switch. With this procedure the 1289 time for switching from transmission of text from one source to 1290 transmission of text from another source depends on the actions of 1291 the users. In order to expedite source switching, a user can, for 1292 example, end its turn with a new line. 1294 4.2.1. Actions by the mixer at reception from the call participants 1296 When text is received by the mixer from the different participants, 1297 the mixer SHALL recover text from redundancy if any packets are lost. 1298 The mark for lost text [T140ad1] SHALL be inserted in the stream if 1299 unrecoverable loss appears. Any Unicode "BOM" characters, possibly 1300 used for keep-alive, SHALL be deleted. The time of creation of text 1301 (retrieved from the RTP timestamp) SHALL be stored together with the 1302 received text from each source in queues for transmission to the 1303 recipients in order to be able to evaluate text loss. 1305 4.2.2. Actions by the mixer for transmission to the recipients 1307 The following procedure SHALL be applied for each multiparty-unaware 1308 recipient of multiparty text from the mixer. 1310 The text for transmission SHALL be formatted by the mixer for each 1311 receiving user for presentation in one single presentation area. 1312 Text received from a participant SHOULD NOT be included in 1313 transmission to that participant because it is usually presented 1314 locally at transmission time. When there is text available for 1315 transmission from the mixer to a receiving party from more than one 1316 participant, the mixer SHALL switch between transmission of text from 1317 the different sources at suitable points in the transmitted stream. 1319 When switching source, the mixer SHALL insert a line separator if the 1320 already transmitted text did not end with a new line (line separator 1321 or CRLF). A label SHALL be composed of information in the CNAME and 1322 NAME fields in RTCP reports from the participant to have its text 1323 transmitted, or from other session information for that user. The 1324 label SHALL be delimited by suitable characters (e.g., '[ ]') and 1325 transmitted. The CSRC SHALL indicate the selected source. Then text 1326 from that selected participant SHALL be transmitted until a new 1327 suitable point for switching source is reached. 1329 Information available to the mixer for composing the label may 1330 contain sensitive personal information that SHOULD not be revealed in 1331 sessions not securely authenticated and protected. Integrity 1332 considerations regarding how much personal information is included in 1333 the label SHOULD therefore be taken when composing the label. 1335 Seeking a suitable point for switching source SHALL be done when 1336 there is older text waiting for transmission from any party than the 1337 age of the last transmitted text. Suitable points for switching are: 1339 * A completed phrase ended by comma 1341 * A completed sentence 1342 * A new line (line separator or CRLF) 1344 * A long pause (e.g., > 10 seconds) in received text from the 1345 currently transmitted source 1347 * If text from one participant has been transmitted with text from 1348 other sources waiting for transmission for a long time (e.g., > 1 1349 minute) and none of the other suitable points for switching has 1350 occurred, a source switch MAY be forced by the mixer at the next 1351 word delimiter, and also even if a word delimiter does not occur 1352 within a time (e.g., 15 seconds) after the scan for a word 1353 delimiter started. 1355 When switching source, the source which has the oldest text in queue 1356 SHALL be selected to be transmitted. A character display count SHALL 1357 be maintained for the currently transmitted source, starting at zero 1358 after the label is transmitted for the currently transmitted source. 1360 The status SHALL be maintained for the latest control code for Select 1361 Graphic Rendition (SGR) from each source. If there is an SGR code 1362 stored as the status for the current source before the source switch 1363 is done, a reset of SGR SHALL be sent by the sequence SGR 0 [009B 1364 0000 006D] after the new line and before the new label during a 1365 source switch. See SGR below for an explanation. This transmission 1366 does not influence the display count. 1368 If there is an SGR code stored for the new source after the source 1369 switch, that SGR code SHALL be transmitted to the recipient before 1370 the label. This transmission does not influence the display count. 1372 4.2.3. Actions on transmission of text 1374 Text from a source sent to the recipient SHALL increase the display 1375 count by one per transmitted character. 1377 4.2.4. Actions on transmission of control codes 1379 The following control codes specified by T.140 require specific 1380 actions. They SHALL cause specific considerations in the mixer. 1381 Note that the codes presented here are expressed in UCS-16, while 1382 transmission is made in the UTF-8 encoding of these codes. 1384 BEL 0007 Bell Alert in session. Provides for alerting during an 1385 active session. The display count SHALL NOT be altered. 1387 NEW LINE 2028 Line separator. Check and perform a source switch if 1388 appropriate. Increase the display count by 1. 1390 CR LF 000D 000A A supported but not preferred way of requesting a 1391 new line. Check and perform a source switch if appropriate. 1392 Increase the display count by 1. 1394 INT ESC 0061 Interrupt (used to initiate the mode negotiation 1395 procedure). The display count SHALL NOT be altered. 1397 SGR 009B Ps 006D Select graphic rendition. Ps is the rendition 1398 parameters specified in ISO 6429. The display count SHALL NOT be 1399 altered. The SGR code SHOULD be stored for the current source. 1401 SOS 0098 Start of string, used as a general protocol element 1402 introducer, followed by a maximum 256-byte string and the ST. The 1403 display count SHALL NOT be altered. 1405 ST 009C String terminator, end of SOS string. The display count 1406 SHALL NOT be altered. 1408 ESC 001B Escape - used in control strings. The display count SHALL 1409 NOT be altered for the complete escape code. 1411 Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used 1412 for synchronization and keep-alive. It SHALL be deleted from 1413 incoming streams. It SHALL also be sent first after session 1414 establishment to the recipient. The display count SHALL NOT be 1415 altered. 1417 Missing text mark (U+FFFD) "Replacement character", represented as a 1418 question mark in a rhombus, or if that is not feasible, replaced 1419 by an apostrophe '. It marks the place in the stream of possible 1420 text loss. This mark SHALL be inserted by the reception procedure 1421 in case of unrecoverable loss of packets. The display count SHALL 1422 be increased by one when sent as for any other character. 1424 SGR If a control code for selecting graphic rendition (SGR) other 1425 than reset of the graphic rendition (SGR 0) is sent to a 1426 recipient, that control code SHALL also be stored as the status 1427 for the source in the storage for SGR status. If a reset graphic 1428 rendition (SGR 0) originating from a source is sent, then the SGR 1429 status storage for that source SHALL be cleared. The display 1430 count SHALL NOT be increased. 1432 BS (U+0008) Back Space, intended to erase the last entered character 1433 by a source. Erasure by backspace cannot always be performed as 1434 the erasing party intended. If an erasing action erases all text 1435 up to the end of the leading label after a source switch, then the 1436 mixer MUST NOT transmit more backspaces. Instead, it is 1437 RECOMMENDED that a letter "X" is inserted in the text stream for 1438 each backspace as an indication of the intent to erase more. A 1439 new line is usually coded by a Line Separator, but the character 1440 combination "CRLF" MAY be used instead. Erasure of a new line is 1441 in both cases done by just one erasing action (Backspace). If the 1442 display count has a positive value it SHALL be decreased by one 1443 when the BS is sent. If the display count is at zero, it SHALL 1444 NOT be altered. 1446 4.2.5. Packet transmission 1448 A mixer transmitting to a multiparty-unaware terminal SHALL send 1449 primary data only from one source per packet. The SSRC SHALL be the 1450 SSRC of the mixer. The CSRC list SHALL contain one member and be the 1451 SSRC of the source of the primary data. 1453 4.2.6. Functional limitations 1455 When a multiparty-unaware endpoint presents a conversation in one 1456 display area in a chat style, it inserts source indications for 1457 remote text and local user text as they are merged in completed text 1458 groups. When an endpoint using this layout receives and presents 1459 text mixed for multiparty-unaware endpoints, there will be two levels 1460 of source indicators for the received text; one generated by the 1461 mixer and inserted in a label after each source switch, and another 1462 generated by the receiving endpoint and inserted after each switch 1463 between local and remote source in the presentation area. This will 1464 waste display space and look inconsistent to the reader. 1466 New text can be presented only from one source at a time. Switch of 1467 source to be presented takes place at suitable places in the text, 1468 such as end of phrase, end of sentence, line separator and 1469 inactivity. Therefore, the time to switch to present waiting text 1470 from other sources may become long and will vary and depend on the 1471 actions of the currently presented source. 1473 Erasure can only be done up to the latest source switch. If a user 1474 tries to erase more text, the erasing actions will be presented as 1475 letter X after the label. 1477 Text loss because of network errors may hit the label between entries 1478 from different parties, causing risk for misunderstanding from which 1479 source a piece of text is. 1481 These facts make it strongly RECOMMENDED implementing multiparty 1482 awareness in RTT endpoints. The use of the mixing method for 1483 multiparty-unaware endpoints should be left for use with endpoints 1484 which are impossible to upgrade to become multiparty-aware. 1486 4.2.7. Example views of presentation on multiparty-unaware endpoints 1488 The following pictures are examples of the view on a participant's 1489 display for the multiparty-unaware case. 1491 _________________________________________________ 1492 | Conference | Alice | 1493 |________________________|_________________________| 1494 | |I will arrive by TGV. | 1495 |[Bob]:My flight is to |Convenient to the main | 1496 |Orly. |station. | 1497 |[Eve]:Hi all, can we | | 1498 |plan for the seminar. | | 1499 | | | 1500 |[Bob]:Eve, will you do | | 1501 |your presentation on | | 1502 |Friday? | | 1503 |[Eve]:Yes, Friday at 10.| | 1504 |[Bob]: Fine, wo |We need to meet befo | 1505 |________________________|_________________________| 1507 Figure 5: Alice who has a conference-unaware client is receiving the 1508 multiparty real-time text in a single-stream. 1510 This figure shows how a coordinated column view MAY be presented on 1511 Alice's device in a view with two-columns. The mixer inserts labels 1512 to show how the sources alternate in the column with received text. 1513 The mixer alternates between the sources at suitable points in the 1514 text exchange so that text entries from each party can be 1515 conveniently read. 1517 _________________________________________________ 1518 | |^| 1519 |(Alice) Hi, Alice here. |-| 1520 | | | 1521 |(mix)[Bob)] Bob as well. | | 1522 | | | 1523 |[Eve] Hi, this is Eve, calling from Paris | | 1524 | I thought you should be here. | | 1525 | | | 1526 |(Alice) I am coming on Thursday, my | | 1527 | performance is not until Friday morning.| | 1528 | | | 1529 |(mix)[Bob] And I on Wednesday evening. | | 1530 | | | 1531 |[Eve] we can have dinner and then walk | | 1532 | | | 1533 |[Eve] But I need to be back to | | 1534 | the hotel by 11 because I need | | 1535 | |-| 1536 |______________________________________________|v| 1537 | of course, I underst | 1538 |________________________________________________| 1540 Figure 6: An example of a view of the multiparty-unaware presentation 1541 in chat style. Alice is the local user. 1543 In this view, there is a tradition in receiving applications to 1544 include a label showing the source of the text, here shown with 1545 parenthesis "()". The mixer also inserts source labels for the 1546 multiparty call participants, here shown with brackets "[]". 1548 5. Relation to Conference Control 1550 5.1. Use with SIP centralized conferencing framework 1552 The SIP conferencing framework, mainly specified in [RFC4353], 1553 [RFC4579] and [RFC4575] is suitable for coordinating sessions 1554 including multiparty RTT. The RTT stream between the mixer and a 1555 participant is one and the same during the conference. Participants 1556 get announced by notifications when participants are joining or 1557 leaving, and further user information may be provided. The SSRC of 1558 the text to expect from joined users MAY be included in a 1559 notification. The notifications MAY be used both for security 1560 purposes and for translation to a label for presentation to other 1561 users. 1563 5.2. Conference control 1565 In managed conferences, control of the real-time text media SHOULD be 1566 provided in the same way as other for media, e.g., for muting and 1567 unmuting by the direction attributes in SDP [RFC8866]. 1569 Note that floor control functions may be of value for RTT users as 1570 well as for users of other media in a conference. 1572 6. Gateway Considerations 1574 6.1. Gateway considerations with Textphones 1576 multiparty RTT sessions may involve gateways of different kinds. 1577 Gateways involved in setting up sessions SHALL correctly reflect the 1578 multiparty capability or unawareness of the combination of the 1579 gateway and the remote endpoint beyond the gateway. 1581 One case that may occur is a gateway to PSTN for communication with 1582 textphones (e.g., TTYs). Textphones are limited devices with no 1583 multiparty awareness, and it SHOULD therefore be suitable for the 1584 gateway to not indicate multiparty awareness for that case. Another 1585 solution is that the gateway indicates multiparty capability towards 1586 the mixer, and includes the multiparty mixer function for multiparty- 1587 unaware endpoints itself. This solution makes it possible to adapt 1588 to the functional limitations of the textphone. 1590 More information on gateways to textphones is found in [RFC5194] 1592 6.2. Gateway considerations with WebRTC 1594 Gateway operation to real-time text in WebRTC may also be required. 1595 In WebRTC, RTT is specified in [RFC8865]. 1597 A multiparty bridge may have functionality for communicating by RTT 1598 both in RTP streams with RTT and WebRTC T.140 data channels. Other 1599 configurations may consist of a multiparty bridge with either 1600 technology for RTT transport and a separate gateway for conversion of 1601 the text communication streams between RTP and T.140 data channel. 1603 In WebRTC, it is assumed that for a multiparty session, one T.140 1604 data channel is established for each source from a gateway or bridge 1605 to each participant. Each participant also has a data channel with a 1606 two-way connection with the gateway or bridge. 1608 The T.140 data channel used both ways is for text from the WebRTC 1609 user and from the bridge or gateway itself to the WebRTC user. The 1610 label parameter of this T.140 data channel is used as the NAME field 1611 in RTCP to participants on the RTP side. The other T.140 data 1612 channels are only for text from other participants to the WebRTC 1613 user. 1615 When a new participant has entered the session with RTP transport of 1616 RTT, a new T.140 channel SHOULD be established to WebRTC users with 1617 the label parameter composed of information from the NAME field in 1618 RTCP on the RTP side. 1620 When a new participant has entered the multiparty session with RTT 1621 transport in a WebRTC T.140 data channel, the new participant SHOULD 1622 be announced by a notification to RTP users. The label parameter 1623 from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP 1624 side, or other available session information. 1626 When a participant on the RTP side is disconnected from the 1627 multiparty session, the corresponding T.140 data channel(s) SHOULD be 1628 closed. 1630 When a WebRTC user of T.140 data channels disconnects from the mixer, 1631 the corresponding RTP streams or sources in an RTP-mixed stream 1632 SHOULD be closed. 1634 T.140 data channels MAY be opened and closed by negotiation or 1635 renegotiation of the session or by any other valid means as specified 1636 in section 1 of [RFC8865]. 1638 7. Updates to RFC 4103 1640 This document updates [RFC4103] by introducing an SDP media attribute 1641 "rtt-mixer" for negotiation of multiparty-mixing capability with the 1642 [RFC4103] format, and by specifying the rules for packets when 1643 multiparty capability is negotiated and in use. 1645 8. Congestion considerations 1647 The congestion considerations and recommended actions from [RFC4103] 1648 are also valid in multiparty situations. 1650 The time values SHALL then be applied per source of text sent to a 1651 receiver. 1653 If the very unlikely situation appears that many participants in a 1654 conference send text simultaneously for a long period, a delay may 1655 build up for presentation of text at the receivers if the limitation 1656 in characters per second ("CPS") to be transmitted to the 1657 participants is exceeded. More delay than 7 seconds can cause 1658 confusion in the session. It is therefore RECOMMENDED that an RTP- 1659 mixer-based mixer discards such text causing excessive delays and 1660 inserts a general indication of possible text loss [T140ad1] in the 1661 session. If the main text contributor is indicated in any way, the 1662 mixer MAY avoid deleting text from that participant. It should 1663 however be noted that human creation of text normally contains 1664 pauses, when the transmission can catch up, so that the transmission 1665 overload situations are expected to be very rare. 1667 9. Acknowledgements 1669 James Hamlin for format and performance aspects. 1671 10. IANA Considerations 1673 10.1. Registration of the "rtt-mixer" SDP media attribute 1675 [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the 1676 RFC number of this document.] 1678 IANA is asked to register the new SDP attribute "rtt-mixer". 1680 Contact name: IESG 1682 Contact email: iesg@ietf.org 1684 Attribute name: rtt-mixer 1686 Attribute semantics: See RFCXXXX Section 2.3 1688 Attribute value: none 1690 Usage level: media 1692 Purpose: Indicate support by mixer and endpoint of multiparty mixing 1693 for real-time text transmission, using a common RTP-stream for 1694 transmission of text from a number of sources mixed with one 1695 source at a time and the source indicated in a single CSRC-list 1696 member. 1698 Charset Dependent: no 1700 O/A procedure: See RFCXXXX Section 2.3 1702 Mux Category: normal 1704 Reference: RFCXXXX 1706 11. Security Considerations 1708 The RTP-mixer model requires the mixer to be allowed to decrypt, 1709 pack, and encrypt secured text from the conference participants. 1710 Therefore, the mixer needs to be trusted to achieve security in 1711 confidentiality and integrity. This situation is similar to the 1712 situation for handling audio and video media in centralized mixers. 1714 The requirement to transfer information about the user in RTCP 1715 reports in SDES, CNAME, and NAME fields, and in conference 1716 notifications, for creation of labels may have privacy concerns as 1717 already stated in RFC 3550 [RFC3550], and may be restricted for 1718 privacy reasons. The receiving user will then get a more symbolic 1719 label for the source. 1721 Participants with malicious intentions may appear and e.g., disturb 1722 the multiparty session by emitting a continuous flow of text. They 1723 may also send text that appears to originate from other participants. 1724 Counteractions should be to require secure signaling, media and 1725 authentication, and to provide higher-layer conference functions 1726 e.g., for blocking, muting, and expelling participants. 1728 Further security considerations specific for this application are 1729 specified in Section 3.19. 1731 12. Change history 1733 [RFC Editor: Please remove this section prior to publication.] 1735 12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-18 1737 Edits of nits as proposed in a review by Lars Eggert. 1739 Edits as response to review by Martin Duke. 1741 12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-17 1743 Actions on Gen-ART review comments. 1745 Actions on SecDir review comments. 1747 12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-16 1749 Improvements in the offer/answer considerations section by adding 1750 subsections for each phase in the negotiation as requested by IANA 1751 expert review. 1753 12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-15 1755 Actions on review comments from Jurgen Schonwalder: 1757 A bit more about congestion situations and that they are expected to 1758 be very rare. 1760 Explanation of differences in security between the conference-aware 1761 and the conference-unaware case added in security section. 1763 Presentation examples with suource labels made less confusing, and 1764 explained. 1766 Reference to T.140 inserted at first mentioning of T.140. 1768 Reference to RFC 8825 inserted to explain WebRTC 1770 Nit in wording in terminology section adjusted. 1772 12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-14 1774 Changes from comments by Murray Kucherawy during AD review. 1776 Many SHOULD in section 4.2 on multiparty-unaware mixing changed to 1777 SHALL, and the whole section instead specified to be optional 1778 depending on the application. 1780 Some SHOULD in section 3 either explained or changed to SHALL. 1782 In order to have explainable conditions behind SHOULDs, the 1783 transmission interval in 3.4 is changed to as soon as text is 1784 available as a main principle. The call participants send with 300 1785 ms interval so that will create realistic load conditions anyway. 1787 12.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-13 1789 Changed year to 2021. 1791 Changed reference to draft on RTT in WebRTC to recently published RFC 1792 8865. 1794 Changed label brackets in example from "[]" to "()" to avoid nits 1795 comment. 1797 Changed reference "RFC 4566" to recently published "RFC 8866" 1799 12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-12 1801 Changes according to responses on comments from Brian Rosen in 1802 Avtcore list on 2020-12-05 and -06. 1804 Changes according to responses to comments by Bernard Aboba in 1805 avtcore list 2020-12-06. 1807 Introduction of an optiona RTP multi-stream mixing method for further 1808 study as proposed by Bernard Aboba. 1810 Changes clarifying how to open and close T.140 data channels included 1811 in 6.2 after comments by Lorenzo Miniero. 1813 Changes to satisfy nits check. Some "not" changed to "NOT" in 1814 normative wording combinations. Some lower case normative words 1815 changed to upper case. A normative reference deleted from the 1816 abstract. Two informative documents moved from normative references 1817 to informative references. 1819 12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11 1821 Timestamps and timestamp offsets added to the packet examples in 1822 section 3.23, and the description corrected. 1824 A number of minor corrections added in sections 3.10 - 3.23. 1826 12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10 1828 The packet composition was modified for interleaving packets from 1829 different sources. 1831 The packet reception was modified for the new interleaving method. 1833 The packet sequence examples was adjusted for the new interleaving 1834 method. 1836 Modifications according to responses to Brian Rosen of 2020-11-03 1838 12.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09 1840 Changed name on the SDP media attribute to "rtt-mixer" 1842 Restructure of section 2 for balance between aware and unaware cases. 1844 Moved conference control to own section. 1846 Improved clarification of recovery and loss in the packet sequence 1847 example. 1849 A number of editorial corrections and improvements. 1851 12.11. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 1853 Deleted the method requiring a new packet format "text/rex" because 1854 of the longer standardization and implementation period it needs. 1856 Focus on use of RFC 4103 text/red format with shorter transmission 1857 interval, and source indicated in CSRC. 1859 12.12. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 1861 Added a method based on the "text/red" format and single source per 1862 packet, negotiated by the "rtt-mixer" SDP attribute. 1864 Added reasoning and recommendation about indication of loss. 1866 The highest number of sources in one packet is 15, not 16. Changed. 1868 Added in information on update to RFC 4103 that RFC 4103 explicitly 1869 allows addition of FEC method. The redundancy is a kind of forward 1870 error correction. 1872 12.13. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 1874 Improved definitions list format. 1876 The format of the media subtype parameters is made to match the 1877 requirements. 1879 The mapping of media subtype parameters to SDP is included. 1881 The "CPS" parameter belongs to the t140 subtype and does not need to 1882 be registered here. 1884 12.14. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 1886 nomenclature and editorial improvements 1888 "this document" used consistently to refer to this document. 1890 12.15. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 1892 'Redundancy header' renamed to 'data header'. 1894 More clarifications added. 1896 Language and figure number corrections. 1898 12.16. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 1900 Mention possible need to mute and raise hands as for other media. 1901 ---done ---- 1903 Make sure that use in two-party calls is also possible and explained. 1904 - may need more wording - 1906 Clarify the RTT is often used together with other media. --done-- 1908 Tell that text mixing is N-1. A users own text is not received in 1909 the mix. -done- 1911 In 3. correct the interval to: A "text/rex" transmitter SHOULD send 1912 packets distributed in time as long as there is something (new or 1913 redundant T140blocks) to transmit. The maximum transmission interval 1914 SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 1915 receiver as soon as new text to that receiver is available, as long 1916 as the time after the latest sent packet to the same receiver is more 1917 than 150 ms, and also the maximum character rate to the receiver is 1918 not exceeded. The intention is to keep the latency low while keeping 1919 a good protection against text loss in bursty packet loss conditions. 1920 -done- 1922 In 1.3 say that the format is used both ways. -done- 1924 In 13.1 change presentation area to presentation field so that reader 1925 does not think it shall be totally separated. -done- 1927 In Performance and intro, tell the performance in number of 1928 simultaneous sending users and introduced delay 16, 150 vs 1929 requirements 5 vs 500. -done -- 1931 Clarify redundancy level per connection. -done- 1933 Timestamp also for the last data header. To make it possible for all 1934 text to have time offset as for transmission from the source. Make 1935 that header equal to the others. -done- 1937 Mixer always use the CSRC list, even for its own BOM. -done- 1939 Combine all talk about transmission interval (300 ms vs when text has 1940 arrived) in section 3 in one paragraph or close to each other. -done- 1941 Documents the goal of good performance with low delay for 5 1942 simultaneous typers in the introduction. -done- 1944 Describe better that only primary text shall be sent on to receivers. 1945 Redundancy and loss must be resolved by the mixer. -done- 1947 12.17. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 1949 SDP and better description and visibility of security by OSRTP RFC 1950 8634 needed. 1952 The description of gatewaying to WebRTC extended. 1954 The description of the data header in the packet is improved. 1956 12.18. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 1958 2,5,6 More efficient format "text/rex" introduced and attribute 1959 a=rtt-mix deleted. 1961 3. Brief about use of OSRTP for security included- More needed. 1963 4. Brief motivation for the solution and why not rtp-translator is 1964 used added to intro. 1966 7. More limitations for the multiparty-unaware mixing method 1967 inserted. 1969 8. Updates to RFC 4102 and 4103 more clearly expressed. 1971 9. Gateway to WebRTC started. More needed. 1973 12.19. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 1974 to draft-ietf-avtcore-multi-party-rtt-mix-00 1976 Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00 1978 Replaced CDATA in IANA registration table with better coding. 1980 Converted to xml2rfc version 3. 1982 12.20. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 1983 to -03 1985 Changed company and e-mail of the author. 1987 Changed title to "RTP-mixer formatting of multi-party Real-time text" 1988 to better match contents. 1990 Check and modification where needed of use of RFC 2119 words SHALL 1991 etc. 1993 More about the CC value in sections on transmitters and receivers so 1994 that 1-to-1 sessions do not use the mixer format. 1996 Enhanced section on presentation for multiparty-unaware endpoints 1998 A paragraph recommending CPS=150 inserted in the performance section. 2000 12.21. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 2001 to -02 2003 In Abstract and 1. Introduction: Introduced wording about regulatory 2004 requirements. 2006 In section 5: The transmission interval is decreased to 100 ms when 2007 there is text from more than one source to transmit. 2009 In section 11 about SDP negotiation, a SHOULD-requirement is 2010 introduced that the mixer should make a mix for multiparty-unaware 2011 endpoints if the negotiation is not successful. And a reference to a 2012 later chapter about it. 2014 The presentation considerations chapter 14 is extended with more 2015 information about presentation on multiparty-aware endpoints, and a 2016 new section on the multiparty-unaware mixing with low functionality 2017 but SHOULD be implemented in mixers. Presentation examples are 2018 added. 2020 A short chapter 15 on gateway considerations is introduced. 2022 Clarification about the text/t140 format included in chapter 10. 2024 This sentence added to the chapter 10 about use without redundancy. 2025 "The text/red format SHOULD be used unless some other protection 2026 against packet loss is utilized, for example a reliable network or 2027 transport." 2029 Note about deviation from RFC 2198 added in chapter 4. 2031 In chapter 9. "Use with SIP centralized conferencing framework" the 2032 following note is inserted: Note: The CSRC-list in an RTP packet only 2033 includes participants whose text is included in one or more text 2034 blocks. It is not the same as the list of participants in a 2035 conference. With audio and video media, the CSRC-list would often 2036 contain all participants who are not muted whereas text participants 2037 that don't type are completely silent and so don't show up in RTP 2038 packet CSRC-lists. 2040 12.22. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 2041 to -01 2043 Editorial cleanup. 2045 Changed capability indication from fmtp-parameter to SDP attribute 2046 "rtt-mix". 2048 Swapped order of redundancy elements in the example to match reality. 2050 Increased the SDP negotiation section 2052 13. References 2054 13.1. Normative References 2056 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 2057 Requirement Levels", BCP 14, RFC 2119, 2058 DOI 10.17487/RFC2119, March 1997, 2059 . 2061 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 2062 Jacobson, "RTP: A Transport Protocol for Real-Time 2063 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 2064 July 2003, . 2066 [RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type", 2067 RFC 4102, DOI 10.17487/RFC4102, June 2005, 2068 . 2070 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 2071 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 2072 . 2074 [RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session 2075 Initiation Protocol (SIP)", RFC 5630, 2076 DOI 10.17487/RFC5630, October 2009, 2077 . 2079 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 2080 Security (DTLS) Extension to Establish Keys for the Secure 2081 Real-time Transport Protocol (SRTP)", RFC 5764, 2082 DOI 10.17487/RFC5764, May 2010, 2083 . 2085 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 2086 Keeping Alive the NAT Mappings Associated with RTP / RTP 2087 Control Protocol (RTCP) Flows", RFC 6263, 2088 DOI 10.17487/RFC6263, June 2011, 2089 . 2091 [RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M. 2092 Thomson, "Session Traversal Utilities for NAT (STUN) Usage 2093 for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675, 2094 October 2015, . 2096 [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2097 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, 2098 May 2017, . 2100 [RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for 2101 Browser-Based Applications", RFC 8825, 2102 DOI 10.17487/RFC8825, January 2021, 2103 . 2105 [RFC8865] Holmberg, C. and G. Hellström, "T.140 Real-Time Text 2106 Conversation over WebRTC Data Channels", RFC 8865, 2107 DOI 10.17487/RFC8865, January 2021, 2108 . 2110 [RFC8866] Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP: 2111 Session Description Protocol", RFC 8866, 2112 DOI 10.17487/RFC8866, January 2021, 2113 . 2115 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 2116 multimedia application text conversation", February 1998, 2117 . 2119 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 2120 Protocol for multimedia application text conversation", 2121 February 2000, 2122 . 2124 13.2. Informative References 2126 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 2127 Session Initiation Protocol (SIP)", RFC 4353, 2128 DOI 10.17487/RFC4353, February 2006, 2129 . 2131 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 2132 Session Initiation Protocol (SIP) Event Package for 2133 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 2134 2006, . 2136 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 2137 (SIP) Call Control - Conferencing for User Agents", 2138 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 2139 . 2141 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 2142 Time Text over IP Using the Session Initiation Protocol 2143 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 2144 . 2146 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 2147 DOI 10.17487/RFC7667, November 2015, 2148 . 2150 [RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. 2151 Stach, "An Opportunistic Approach for Secure Real-time 2152 Transport Protocol (OSRTP)", RFC 8643, 2153 DOI 10.17487/RFC8643, August 2019, 2154 . 2156 Author's Address 2158 Gunnar Hellstrom 2159 Gunnar Hellstrom Accessible Communication 2160 SE-13670 Vendelso 2161 Sweden 2163 Email: gunnar.hellstrom@ghaccess.se