idnits 2.17.1 draft-ietf-avtcore-multi-party-rtt-mix-11.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: ---------------------------------------------------------------------------- No issues found here. Checking nits according to https://www.ietf.org/id-info/checklist : ---------------------------------------------------------------------------- ** The abstract seems to contain references ([RFC4103]), which it shouldn't. Please replace those with straight textual mentions of the documents in question. == The 'Updates: ' line in the draft header should list only the _numbers_ of the RFCs which will be updated by this document (if approved); it should not include the word 'RFC' in the list. Miscellaneous warnings: ---------------------------------------------------------------------------- == The copyright year in the IETF Trust and authors Copyright Line does not match the current year == Line 813 has weird spacing: '...example from ...' == 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 'MUST not' in this paragraph: A party not performing as a mixer MUST not include the CSRC list. == 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: BEL 0007 Bell Alert in session, provides for alerting during an active session. The display count SHOULD not be altered. == 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: INT ESC 0061 Interrupt (used to initiate mode negotiation procedure). The display count SHOULD not be altered. == 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: SGR 009B Ps 006D Select graphic rendition. Ps is rendition parameters specified in ISO 6429. The display count SHOULD not be altered. The SGR code SHOULD be stored for the current source. == 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: SOS 0098 Start of string, used as a general protocol element introducer, followed by a maximum 256 bytes string and the ST. The display count SHOULD not be altered. == 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: ST 009C String terminator, end of SOS string. The display count SHOULD not be altered. == 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: ESC 001B Escape - used in control strings. The display count SHOULD not be altered for the complete escape code. (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 (22 November 2020) is 1243 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 1352, but not defined ** Obsolete normative reference: RFC 4566 (Obsoleted by RFC 8866) ** Downref: Normative reference to an Informational RFC: RFC 8643 -- Possible downref: Non-RFC (?) normative reference: ref. 'T140' -- Possible downref: Non-RFC (?) normative reference: ref. 'T140ad1' Summary: 3 errors (**), 0 flaws (~~), 11 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: RFC 4103 (if approved) 22 November 2020 5 Intended status: Standards Track 6 Expires: 26 May 2021 8 RTP-mixer formatting of multi-party Real-time text 9 draft-ietf-avtcore-multi-party-rtt-mix-11 11 Abstract 13 Real-time text mixers for multi-party sessions need to identify the 14 source of each transmitted group of text so that the text can be 15 presented by endpoints in suitable grouping with other text from the 16 same source, while new text from other sources is also presented in 17 readable grouping as received interleaved in real-time. 19 Regional regulatory requirements specify provision of real-time text 20 in multi-party calls. RFC 4103 mixer implementations can use 21 traditional RTP functions for source identification, but the mixer 22 source switching performance is limited when using the default 23 transmission characteristics with redundancy. 25 Enhancements for RFC 4103 real-time text mixing is provided in this 26 document, suitable for a centralized conference model that enables 27 source identification and source switching. The intended use is for 28 real-time text mixers and multi-party-aware participant endpoints. 29 The specified mechanism build on the standard use of the CSRC list in 30 the RTP packet for source identification. The method makes use of 31 the same "text/t140" and "text/red" formats as for two-party 32 sessions. 34 A capability exchange is specified so that it can be verified that a 35 participant can handle the multi-party coded real-time text stream. 36 The capability is indicated by use of a media attribute "rtt-mixer". 38 The document updates RFC 4103[RFC4103] 40 A specifications of how a mixer can format text for the case when the 41 endpoint is not multi-party aware is also provided. 43 Status of This Memo 45 This Internet-Draft is submitted in full conformance with the 46 provisions of BCP 78 and BCP 79. 48 Internet-Drafts are working documents of the Internet Engineering 49 Task Force (IETF). Note that other groups may also distribute 50 working documents as Internet-Drafts. The list of current Internet- 51 Drafts is at https://datatracker.ietf.org/drafts/current/. 53 Internet-Drafts are draft documents valid for a maximum of six months 54 and may be updated, replaced, or obsoleted by other documents at any 55 time. It is inappropriate to use Internet-Drafts as reference 56 material or to cite them other than as "work in progress." 58 This Internet-Draft will expire on 26 May 2021. 60 Copyright Notice 62 Copyright (c) 2020 IETF Trust and the persons identified as the 63 document authors. All rights reserved. 65 This document is subject to BCP 78 and the IETF Trust's Legal 66 Provisions Relating to IETF Documents (https://trustee.ietf.org/ 67 license-info) in effect on the date of publication of this document. 68 Please review these documents carefully, as they describe your rights 69 and restrictions with respect to this document. Code Components 70 extracted from this document must include Simplified BSD License text 71 as described in Section 4.e of the Trust Legal Provisions and are 72 provided without warranty as described in the Simplified BSD License. 74 Table of Contents 76 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 77 1.1. Selected solution and considered alternative . . . . . . 5 78 1.2. Nomenclature . . . . . . . . . . . . . . . . . . . . . . 7 79 1.3. Intended application . . . . . . . . . . . . . . . . . . 8 80 2. Overview over the two specified solutions . . . . . . . . . . 9 81 2.1. Negotiated use of the RFC 4103 format for multi-party 82 transmission in a single RTP stream . . . . . . . . . . . 9 83 2.2. Mixing for multi-party unaware endpoints . . . . . . . . 9 84 3. Details for the multi-party aware mixing case . . . . . . . . 9 85 3.1. Offer/answer considerations . . . . . . . . . . . . . . . 10 86 3.2. Actions depending on capability negotiation result . . . 10 87 3.3. Use of fields in the RTP packets . . . . . . . . . . . . 10 88 3.4. Initial transmission of a BOM character . . . . . . . . . 11 89 3.5. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 11 90 3.6. Transmission interval . . . . . . . . . . . . . . . . . . 11 91 3.7. Only one source per packet . . . . . . . . . . . . . . . 11 92 3.8. Do not send received text to the originating source . . . 12 93 3.9. Clean incoming text . . . . . . . . . . . . . . . . . . . 12 94 3.10. Redundant transmission principles . . . . . . . . . . . . 12 95 3.11. Interleaving text from different sources . . . . . . . . 12 96 3.12. Text placement in packets . . . . . . . . . . . . . . . . 12 97 3.13. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 13 98 3.14. Creation of the redundancy . . . . . . . . . . . . . . . 13 99 3.15. Timer offset fields . . . . . . . . . . . . . . . . . . . 14 100 3.16. Other RTP header fields . . . . . . . . . . . . . . . . . 14 101 3.17. Pause in transmission . . . . . . . . . . . . . . . . . . 14 102 3.18. RTCP considerations . . . . . . . . . . . . . . . . . . . 15 103 3.19. Reception of multi-party contents . . . . . . . . . . . . 15 104 3.20. Performance considerations . . . . . . . . . . . . . . . 17 105 3.21. Security for session control and media . . . . . . . . . 18 106 3.22. SDP offer/answer examples . . . . . . . . . . . . . . . . 18 107 3.23. Packet sequence example from interleaved transmission . . 19 108 3.24. Maximum character rate "CPS" . . . . . . . . . . . . . . 22 109 4. Presentation level considerations . . . . . . . . . . . . . . 22 110 4.1. Presentation by multi-party aware endpoints . . . . . . . 23 111 4.2. Multi-party mixing for multi-party unaware endpoints . . 25 112 5. Relation to Conference Control . . . . . . . . . . . . . . . 30 113 5.1. Use with SIP centralized conferencing framework . . . . . 31 114 5.2. Conference control . . . . . . . . . . . . . . . . . . . 31 115 6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 31 116 6.1. Gateway considerations with Textphones (e.g. TTYs). . . 31 117 6.2. Gateway considerations with WebRTC. . . . . . . . . . . . 32 118 7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 32 119 8. Congestion considerations . . . . . . . . . . . . . . . . . . 32 120 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33 121 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33 122 10.1. Registration of the "rtt-mixer" sdp media attribute . . 33 123 11. Security Considerations . . . . . . . . . . . . . . . . . . . 34 124 12. Change history . . . . . . . . . . . . . . . . . . . . . . . 34 125 12.1. Changes included in 126 draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 34 127 12.2. Changes included in 128 draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 34 129 12.3. Changes included in 130 draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 34 131 12.4. Changes included in 132 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 35 133 12.5. Changes included in 134 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 35 135 12.6. Changes included in 136 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 35 137 12.7. Changes included in 138 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 35 139 12.8. Changes included in 140 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 36 141 12.9. Changes included in 142 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 36 144 12.10. Changes included in 145 draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 37 146 12.11. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 37 147 12.12. Changes from 148 draft-hellstrom-avtcore-multi-party-rtt-source-03 to 149 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 37 150 12.13. Changes from 151 draft-hellstrom-avtcore-multi-party-rtt-source-02 to 152 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 38 153 12.14. Changes from 154 draft-hellstrom-avtcore-multi-party-rtt-source-01 to 155 -02 . . . . . . . . . . . . . . . . . . . . . . . . . . 38 156 12.15. Changes from 157 draft-hellstrom-avtcore-multi-party-rtt-source-00 to 158 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 39 159 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 39 160 13.1. Normative References . . . . . . . . . . . . . . . . . . 39 161 13.2. Informative References . . . . . . . . . . . . . . . . . 40 162 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 41 164 1. Introduction 166 RFC 4103[RFC4103] specifies use of RFC 3550 RTP [RFC3550] for 167 transmission of real-time text (RTT) and the "text/t140" format. It 168 also specifies a redundancy format "text/red" for increased 169 robustness. RFC 4102 [RFC4102] registers the "text/red" format. 170 Regional regulatory requirements specify provision of real-time text 171 in multi-party calls. 173 Real-time text is usually provided together with audio and sometimes 174 with video in conversational sessions. 176 A requirement related to multi-party sessions from the presentation 177 level standard T.140 for real-time text is: "The display of text from 178 the members of the conversation should be arranged so that the text 179 from each participant is clearly readable, and its source and the 180 relative timing of entered text is visualized in the display." 182 Another requirement is that the mixing procedure must not introduce 183 delays in the text streams that are experienced disturbing the real- 184 time experience of the receiving users. 186 The redundancy scheme of RFC 4103 [RFC4103] enables efficient 187 transmission of redundant text in packets together with new text. 188 However the redundancy header format has no source indicators for the 189 redundant transmissions. The redundant parts in a packet must 190 therefore be from the same source as the new text. The recommended 191 transmission is one new and two redundant generations of text 192 (T140blocks) in each packet and the recommended transmission interval 193 for two-party use is 300 ms. 195 Real-time text mixers for multi-party sessions therefore need to 196 insert the source of each transmitted group of text from a conference 197 participant so that the text can be transmitted interleaved with text 198 groups from different sources in the rate they are created. This 199 enables the text groups to be presented by endpoints in suitable 200 grouping with other text from the same source. The presentation can 201 then be arranged so that text from different sources can be presented 202 in real-time and easily read while it is possible for a reading user 203 to also perceive approximately when the text was created in real time 204 by the different parties. The transmission and mixing is intended to 205 be done in a general way so that presentation can be arranged in a 206 layout decided by the endpoint. 208 There are existing implementations of RFC 4103 without the updates 209 from this document. These will not be able to receive and present 210 real-time text mixed for multi-party aware endpoints. 212 A negotiation mechanism is therefore needed for verification if the 213 parties are able to handle a multi-party coded stream and agreeing on 214 using that method. 216 A fall-back mixing procedure is also needed for cases when the 217 negotiation result indicates that a receiving endpoint is not capable 218 of handling the mixed format. This method is called the mixing 219 procedure for multi-party unaware endpoints. The fall-back method is 220 naturally not expected to meet all performance requirements placed on 221 the mixing procedure for multi-party aware endpoints. 223 The document updates RFC 4103[RFC4103] by introducing an attribute 224 for indicating capability for the multi-party mixing case and rules 225 for source indications and source switching. 227 1.1. Selected solution and considered alternative 229 A number of alternatives were considered when searching an efficient 230 and easily implemented multi-party method for real-time text. This 231 section explains a few of them briefly. 233 One RTP stream per source, sent in the same RTP session with 234 "text/red" format. 235 From some points of view, use of multiple RTP streams, one for 236 each source, sent in the same RTP session, called the RTP 237 translator model in [RFC3550], would be efficient, and use exactly 238 the same packet format as [RFC4103], the same payload type and a 239 simple SDP declaration. However, the RTP implementation in both 240 mixers and endpoints need to support multiple streams in the same 241 RTP session in order to use this mechanism. For best deployment 242 opportunity, it should be possible to upgrade existing endpoint 243 solutions to be multi-party aware with a reasonable effort. There 244 is currently a lack of support for multi-stream RTP in certain 245 implementation technologies. This fact made this solution not 246 selected for inclusion in this document. 248 The "text/red" format in RFC 4103 with shorter transmission 249 interval, and indicating source in CSRC. 250 The "text/red" format with "text/t140" payload in a single RTP 251 stream can be sent with 100 ms packet intervals instead of the 252 regular 300 ms. The source is indicated in the CSRC field. 253 Source switching can then be done every 100 ms while simultaneous 254 transmission occurs. With five participants sending text 255 simultaneously, the switching and transmission performance is 256 good. With more simultaneously sending participants, there will 257 be a noticable jerkiness in text presentation. The jerkiness will 258 be more expressed the more participants who send text 259 simultaneously. With ten sending participants, the jerkiness will 260 be about one second. Text sent from a source at the end of the 261 period its text is sent by the mixer will have close to zero extra 262 delay. Recent text will be presented with no or low delay. The 263 one second jerkiness will be noticable and slightly unpleasant, 264 but corresponds in time to what typing humans often cause by 265 hesitation or changing position while typing. A benefit of this 266 method is that no new packet format needs to be introduced and 267 implemented. Since simultaneous typing by more than two parties 268 is very rare, and in most applications also more than three 269 parties in a call is rare, this method can be used successfully 270 with good performance. Recovery of text in case of packet loss is 271 based on analysis of timestamps of received redundancy versus 272 earlier received text. Negotiation is based on a new sdp media 273 attribute "rtt-mixer". This method is selected to be the main one 274 specified in this document. 276 A new "text" media subtype with up to 15 sources in each packet. 277 The mechanism makes use of the RTP mixer model specified in 278 RFC3550[RFC3550]. Text from up to 15 sources can be included in 279 each packet. Packets are normally sent every 300 ms. The mean 280 delay will be 150 ms. The sources are indicated in strict order 281 in the CSRC list of the RTP packets. A new redundancy packet 282 format is specified. This method would result in good 283 performance, but would require standardisation and implementation 284 of new releases in the target technologies that would take more 285 time than desirable to complete. It was therefore not selected to 286 be included in this document. 288 The presentation planned by the mixer for multi-party unaware 289 endpoints. 290 It is desirable to have a method that does not require any 291 modifications in existing user devices implementing RFC 4103 for 292 RTT without explicit support of multi-party sessions. This is 293 possible by having the mixer insert a new line and a text 294 formatted source label before each switch of text source in the 295 stream. Switch of source can only be done in places in the text 296 where it does not disturb the perception of the contents. Text 297 from only one source can be presented in real time at a time. The 298 delay will therefore be varying. The method has also other 299 limitations, but is included in this document as a fallback 300 method. In calls where parties take turns properly by ending 301 their entries with a new line, the limitations will have limited 302 influence on the user experience. while only two parties send 303 text, these two will see the text in real time with no delay. 304 This method is specified as a fallback method in this document. 306 RTT transport in WebRTC 307 Transport of real-time text in the WebRTC technology is specified 308 to use the WebRTC data channel in 309 [I-D.ietf-mmusic-t140-usage-data-channel]. That spcification 310 contains a section briefly describing its use in multi-party 311 sessions. The focus of this document is RTP transport. 312 Therefore, even if the WebRTC transport provides good multi-party 313 performance, it is just mentioned in this document in relation to 314 providing gateways with multi-party capabilities between RTP and 315 WebRTC technologies. 317 1.2. Nomenclature 319 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 320 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 321 document are to be interpreted as described in [RFC2119]. 323 The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP- 324 mixer, RTP-translator are explained in [RFC3550] 326 The term "T140block" is defined in RFC 4103 [RFC4103] to contain one 327 or more T.140 code elements. 329 "TTY" stands for a text telephone type used in North America. 331 "WebRTC" stands for web based communication specified by W3C and 332 IETF. 334 "DTLS-SRTP" stands for security specified in RFC 5764 [RFC5764]. 336 "multi-party aware" stands for an endpoint receiving real-time text 337 from multiple sources through a common conference mixer being able to 338 present the text in real-time separated by source and presented so 339 that a user can get an impression of the approximate relative timing 340 of text from different parties. 342 "multi-party unaware" stands for an endpoint not itself being able to 343 separate text from different sources when received through a common 344 conference mixer. 346 1.3. Intended application 348 The method for multi-party real-time text specified in this document 349 is primarily intended for use in transmission between mixers and 350 endpoints in centralised mixing configurations. It is also 351 applicable between mixers. An often mentioned application is for 352 emergency service calls with real-time text and voice, where a 353 calltaker want to make an attended handover of a call to another 354 agent, and stay observing the session. Multimedia conference 355 sessions with support for participants to contribute in text is 356 another application. Conferences with central support for speech-to- 357 text conversion is yet another mentioned application. 359 In all these applications, normally only one participant at a time 360 will send long text utterances. In some cases, one other participant 361 will occasionally contribute with a longer comment simultaneously. 362 That may also happen in some rare cases when text is interpreted to 363 text in another language in a conference. Apart from these cases, 364 other participants are only expected to contribute with very brief 365 utterings while others are sending text. 367 Text is supposed to be human generated, by some text input means, 368 such as typing on a keyboard or using speech-to-text technology. 369 Occasional small cut-and-paste operations may appear even if that is 370 not the initial purpose of real-time text. 372 The real-time characteristics of real-time text is essential for the 373 participants to be able to contribute to a conversation. If the text 374 is too much delayed from typing a letter to its presentation, then, 375 in some conference situations, the opportunity to comment will be 376 gone and someone else will grab the turn. A delay of more than one 377 second in such situations is an obstacle for good conversation. 379 2. Overview over the two specified solutions 381 This section contains a brief introduction of the two methods 382 specified in this document. 384 2.1. Negotiated use of the RFC 4103 format for multi-party transmission 385 in a single RTP stream 387 The main purpose of this document is to specify a method for true 388 multi-party real-time text mixing for multi-party aware endpoints. 389 The method makes use of the current format for real-time text in 390 [RFC4103]. It is an update of RFC 4103 by a clarification on one way 391 to use it in the multi-party situation. It is done by completing a 392 negotiation for this kind of multi-party capability and by indicating 393 source in the CSRC element in the RTP packets. Specific 394 considerations are made to be able to recover text after packet loss. 396 The detailed procedures for the multi-party aware case are specified 397 in Section 3 399 Please use [RFC4103] as reference when reading the specification. 401 2.2. Mixing for multi-party unaware endpoints 403 A method is also specified in this document for cases when the 404 endpoint participating in a multi-party call does not itself 405 implement any solution for multi-party handling of real-time text. 406 The solution requires the mixer to insert text dividers and readable 407 labels and only send text from one source at a time until a suitable 408 point appears for source change. This solution is a fallback method 409 with functional limitations. It acts on the presentation level. 411 A party performing as a mixer, which has not negotiated the "rtt- 412 mixer" sdp media attribute, but negotiated a "text/red" or "text/ 413 t140" format in a session with a participant SHOULD, if nothing else 414 is specified for the application, format transmitted text to that 415 participant to be suitable to present on a multi-party unaware 416 endpoint as further specified in Section 4.2. 418 3. Details for the multi-party aware mixing case 419 3.1. Offer/answer considerations 421 RFC 4103[RFC4103] specifies use of RFC 3550 RTP[RFC3550], and a 422 redundancy format "text/red" for increased robustness of real-time 423 text transmission. This document updates RFC 4103[RFC4103] by 424 introducing a capability negotiation for handling multi-party real- 425 time text, a way to indicate the source of transmitted text, and 426 rules for efficient timing of the transmissions interleaved from 427 different sources. 429 The capability negotiation is based on use of the sdp media attribute 430 "rtt-mixer". 432 Both parties shall indicate their capability in a session setup or 433 modification, and evaluate the capability of the counterpart. 435 The syntax is as follows: 436 "a=rtt-mixer" 438 3.2. Actions depending on capability negotiation result 440 A transmitting party SHALL send text according to the multi-party 441 format only when the negotiation for this method was successful and 442 when the CC field in the RTP packet is set to 1. In all other cases, 443 the packets SHALL be populated and interpreted as for a two-party 444 session. 446 A party which has negotiated the "rtt-mixer" sdp media attribute MUST 447 populate the CSRC-list and format the packets according to Section 3 448 if it acts as an rtp-mixer and sends multi-party text. 450 A party which has negotiated the "rtt-mixer" sdp media attribute MUST 451 interpret the contents of the "CC" field the CSRC-list and the 452 packets according to Section 3 in received rtp packets in the 453 corresponding RTP stream. 455 A party not performing as a mixer MUST not include the CSRC list. 457 3.3. Use of fields in the RTP packets 459 The CC field SHALL show the number of members in the CSRC list, which 460 SHALL be one (1) in transmissions from a mixer involved in a multi- 461 party session, and otherwise 0. 463 When transmitted from a mixer during a multi-party session, a CSRC 464 list SHALL be included in the packet. The single member in the CSRC- 465 list SHALL contain the SSRC of the source of the T140blocks in the 466 packet. When redundancy is used, the recommended level of redundancy 467 is to use one primary and two redundant generations of T140blocks. 468 In some cases, a primary or redundant T140block is empty, but is 469 still represented by a member in the redundancy header. 471 From other aspects, the contents of the RTP packets are equal to what 472 is specified in [RFC4103]. 474 3.4. Initial transmission of a BOM character 476 As soon as a participant is known to participate in a session with 477 another entity and being available for text reception, a Unicode BOM 478 character SHALL be sent to it by the other entity according to the 479 procedures in this section. If the transmitter is a mixer, then the 480 source of this character SHALL be indicated to be the mixer itself. 482 Note that the BOM character SHALL be transmitted with the same 483 redundancy procedures as any other text. 485 3.5. Keep-alive 487 After that, the transmitter SHALL send keep-alive traffic to the 488 receiver(s) at regular intervals when no other traffic has occurred 489 during that interval, if that is decided for the actual connection. 490 Recommendations for keep-alive can be found in [RFC6263]. 492 3.6. Transmission interval 494 A "text/red" or "text/t140" transmitter in a mixer SHOULD send 495 packets distributed in time as long as there is something (new or 496 redundant T140blocks) to transmit. The maximum transmission interval 497 SHOULD then be 330 ms. It is RECOMMENDED to send next packet to a 498 receiver as soon as new text to that receiver is available, as long 499 as the time after the latest sent packet to the same receiver is more 500 than or equal to 100 ms, and also the maximum character rate ("CPS") 501 to the receiver is not exceeded. The intention is to keep the 502 latency low and network load limited while keeping a good protection 503 against text loss in bursty packet loss conditions. 505 For a transmitter not acting in a mixer, the same transmission 506 interval principles apply, but the maximum transmission interval 507 SHOULD be 300 ms. 509 3.7. Only one source per packet 511 New and redundant text from one source SHALL be transmitted in the 512 same packet if available for transmission at the same time. Text 513 from different sources MUST NOT be transmitted in the same packet. 515 3.8. Do not send received text to the originating source 517 Text received to a mixer from a participant SHOULD NOT be included in 518 transmission from the mixer to that participant. 520 3.9. Clean incoming text 522 A mixer SHALL handle reception, recovery from packet loss, deletion 523 of superfluous redundancy, marking of possible text loss and deletion 524 of 'BOM' characters from each participant before queueing received 525 text for transmission to receiving participants. 527 3.10. Redundant transmission principles 529 A transmitting party using redundancy SHALL send redundant 530 repetitions of T140blocks already transmitted in earlier packets. 532 The number of redundant generations of T140blocks to include in 533 transmitted packets SHALL be deduced from the SDP negotiation. It 534 SHOULD be set to the minimum of the number declared by the two 535 parties negotiating a connection. It is RECOMMENDED to declare and 536 transmit one original and two redundant generations of the 537 T140blocks. 539 3.11. Interleaving text from different sources 541 When text from more than one source is available for transmission 542 from a mixer, the mixer SHALL let the sources take turns in having 543 their text transmitted. 545 The source with the oldest text received in the mixer or oldest 546 redundant text SHOULD be next in turn to get all its available unsent 547 text transmitted. The age of redundant text SHOULD then be 548 considered to be 330 ms after its previous transmission. 550 3.12. Text placement in packets 552 The mixer SHOULD compose and transmit an RTP packet to a receiver 553 when one of the following conditions has occurred: 555 * 100 ms has passed since the latest transmission to that receiver, 556 and there is unsent text available for transmission. 558 * New text has arrived and more than 100 ms has passed since latest 559 transmission to that receiver. 561 * 330 ms has passed since already transmitted text was queued for 562 transmission as redundant text, and more than 100 ms has passed 563 since the latest transmission to that receiver, and the redundant 564 text is still not sent. 566 At time of transmission, the mixer SHALL populate the RTP packet with 567 all T140blocks queued for transmission originating from the source in 568 turn for transmission as long as this is not in conflict with the 569 allowed number of characters per second ("CPS") or the maximum packet 570 size. In this way, the latency of the latest received text is kept 571 low even in moments of simultaneous transmission from many sources. 573 Redundant text SHALL also be included. See Section 3.14 575 The SSRC of the source shall be placed as the only member in the 576 CSRC-list. 578 Note: The CSRC-list in an RTP packet only includes the participant 579 who's text is included in text blocks. It is not the same as the 580 total list of participants in a conference. With audio and video 581 media, the CSRC-list would often contain all participants who are not 582 muted whereas text participants that don't type are completely silent 583 and thus are not represented in RTP packet CSRC-lists. 585 3.13. Empty T140blocks 587 If no unsent T140blocks were available for a source at the time of 588 populating a packet, but T140blocks are available which have not yet 589 been sent the full intended number of redundant transmissions, then 590 the primary T140block for that source is composed of an empty 591 T140block, and populated (without taking up any length) in a packet 592 for transmission. The corresponding SSRC SHALL be placed as usual in 593 its place in the CSRC-list. 595 The first packet in the session, the first after a source switch and 596 the first after a pause SHALL be poulated with the available 597 T140blocks for the source in turn to be sent as primary, and empty 598 T140blocks for the agreed number of redundancy generations. 600 3.14. Creation of the redundancy 602 The primary T140block from a source in the latest transmitted packet 603 is saved for populating the first redundant T140block for that source 604 in next transmission of text from that source. The first redundant 605 T140block for that source from the latest transmission is saved for 606 populating the second redundant T140block in next transmission of 607 text from that source. 609 Usually this is the level of redundancy used. If a higher number of 610 redundancy is negotiated, then the procedure SHALL be maintained 611 until all available redundant levels of T140blocks are placed in the 612 packet. If a receiver has negotiated a lower number of "text/red" 613 generations, then that level shall be the maximum used by the 614 transmitter. 616 The T140blocks saved for transmission as redundant data are assigned 617 a planned transmission time 330 ms after the current time, but SHOULD 618 be transmitted earlier if new text for the same source gets in turn 619 for transmission before that time. 621 3.15. Timer offset fields 623 The timestamp offset values are inserted in the redundancy header, 624 with the time offset from the RTP timestamp in the packet when the 625 corresponding T140block was sent as primary. 627 The timestamp offsets are expressed in the same clock tick units as 628 the RTP timestamp. 630 The timestamp offset values for empty T140blocks have no relevance 631 but SHOULD be assigned realistic values. 633 3.16. Other RTP header fields 635 The number of members in the CSRC list ( 0 or 1) shall be placed in 636 the "CC" header field. Only mixers place value 1 in the "CC" field. 637 A value of "0" indicates that the source is the transmitting device 638 itself and that the source is indicated by the SSRC field. This 639 value is used by endpoints, and by mixers sending data that it is 640 source of itself. 642 The current time SHALL be inserted in the timestamp. 644 The SSRC of the mixer for the RTT session SHALL be inserted in the 645 SSRC field of the RTP header. 647 The M-bit shall be handled as specified in [RFC4103]. 649 3.17. Pause in transmission 651 When there is no new T140block to transmit, and no redundant 652 T140block that has not been retransmitted the intended number of 653 times from any source, the transmission process can stop until either 654 new T140blocks arrive, or a keep-alive method calls for transmission 655 of keep-alive packets. 657 3.18. RTCP considerations 659 A mixer SHALL send RTCP reports with SDES, CNAME and NAME information 660 about the sources in the multi-party call. This makes it possible 661 for participants to compose a suitable label for text from each 662 source. 664 Integrity considerations SHALL be considered when composing these 665 fields. 667 3.19. Reception of multi-party contents 669 The "text/red" receiver included in an endpoint with presentation 670 functions will receive RTP packets in the single stream from the 671 mixer, and SHALL distribute the T140blocks for presentation in 672 presentation areas for each source. Other receiver roles, such as 673 gateways or chained mixers are also feasible, and requires 674 consideration if the stream shall just be forwarded, or distributed 675 based on the different sources. 677 3.19.1. Multi-party vs two-party use 679 If the "CC" field value of a received packet is 1, it indicates that 680 multi-party transmission is active, and the receiver MUST be prepared 681 to act on the source according to its role. If the CC value is 0, 682 the connection is point-to-point. 684 3.19.2. Level of redundancy 686 The used level of redundancy generations SHALL be evaluated from the 687 received packet contents. The number of generations (including the 688 primary) is equal to the number of members in the redundancy header. 690 3.19.3. Empty T140blocks 692 Empty T140blocks are included as fillers for unused primary or 693 redundancy levels in the packets. They just do not provide any 694 contents. 696 3.19.4. Detection and indication of possible text loss 698 The RTP sequence numbers of the received packets SHALL be monitored 699 for gaps and packets out of order. If a sequence number gap appears 700 and still exists after some defined short time for jitter resolution, 701 the packets in the gap SHALL be regarded lost. 703 If it is known that only one source is active in the RTP session, 704 then it is likely that a gap equal to or larger than the agreed 705 number of redundancy generations (including the primary) causes text 706 loss. In that case a t140block SHALL be created with a marker for 707 possible text loss [T140ad1] and assigned to the source and inserted 708 in the reception buffer for that source. 710 If it is known that more than one source is active in the RTP 711 session, then it is not possible in general to evaluate if text was 712 lost when packets were lost. With two active sources and the 713 recommended number of redundancy generations (3), it can take a gap 714 of five consecutive lost packets until any text may be lost, but text 715 loss can also appear if three non-consecutive packets are lost when 716 they contained consecutive data from the same source. A simple 717 method to decide when there is risk for resulting text loss is to 718 evaluate if three or more packets were lost within one second. Then 719 a t140block SHALL be created with a marker for possible text loss 720 [T140ad1] and assigned to the SSRC of the transmitter as a general 721 input from the mixer. 723 Implementations MAY apply more refined methods for more reliable 724 detection of if text was lost or not. Any refined method SHOULD 725 rather falsely mark possible loss when there was no loss instead of 726 not marking possible loss when there was loss. 728 3.19.5. Extracting text and handling recovery 730 When applying the following procedures, the effects MUST be 731 considered of possible timestamp wrap around and the RTP session 732 possibly changing SSRC. 734 When a packet is received in an RTP session using the packetization 735 for multi-party aware endpoints, its T140blocks SHALL be extracted in 736 the following way. The description is adapted to the default 737 redundancy case using the original and two redundant generations. 739 The source SHALL be extracted from the CSRC-list if available, 740 otherwise from the SSRC. 742 If the received packet is the first packet received from the source, 743 then all T140blocks in the packet SHALL be retrieved and assigned to 744 a receive buffer for the source beginning with the second generation 745 redundancy, continuing with the first generation redundancy and 746 finally the primary. 748 Note: The normal case is that in the first packet, only the primary 749 data has contents. The redundant data has contents in the first 750 received packet from a source only after initial packet loss. 752 If the packet is not the first packet from a source, then if the 753 second generation redundant data is available, its timestamp SHALL be 754 created by subtracting its timestamp offset from the RTP timestamp. 755 If the resulting timestamp is later than the latest retrieved data 756 from the same source, then the redundant data SHALL be retrieved and 757 appended to the receive buffer. The process SHALL be continued in 758 the same way for the first generation redundant data. After that, 759 the primary data SHALL be retrieved from the packet and appended to 760 the receive buffer for the source. 762 3.19.6. Delete 'BOM' 764 Unicode character 'BOM' is used as a start indication and sometimes 765 used as a filler or keep alive by transmission implementations. 766 These SHALL be deleted after extraction from received packets. 768 3.20. Performance considerations 770 This solution has good performance for up to five participants 771 simultaneously sending text. At higher numbers of participants 772 simultaneously sending text, a jerkiness is visible in the 773 presentation of text. With ten participants simultaneously 774 transmitting text, the jerkiness is about one second. Even so, the 775 transmission of text catches up, so there is limited delay of new 776 text. The solution is therefore suitable for emergency service use, 777 relay service use, and small or well-managed larger multimedia 778 conferences. Only in large unmanaged conferences with a high number 779 of participants there may on very rare occasions appear situations 780 when many participants happen to send text simultaneously, resulting 781 in unpleasantly jerky presentation of text from each sending 782 participant. It should be noted that it is only the number of users 783 sending text within the same moment that causes jerkiness, not the 784 total number of users with RTT capability. 786 3.21. Security for session control and media 788 Security SHOULD be applied on both session control and media. In 789 applications where legacy endpoints without security may exist, a 790 negotiation SHOULD be performed to decide if security by encryption 791 will be applied. If no other security solution is mandated for the 792 application, then RFC 8643 OSRTP [RFC8643] SHOULD be applied to 793 negotiate SRTP media security with DTLS. Most SDP examples below are 794 for simplicity expressed without the security additions. The 795 principles (but not all details) for applying DTLS-SRTP [RFC5764] 796 security is shown in a couple of the following examples. 798 3.22. SDP offer/answer examples 800 This sections shows some examples of SDP for session negotiation of 801 the real-time text media in SIP sessions. Audio is usually provided 802 in the same session, and sometimes also video. The examples only 803 show the part of importance for the real-time text media. 805 Offer example for "text/red" format and multi-party support: 807 m=text 11000 RTP/AVP 100 98 808 a=rtpmap:98 t140/1000 809 a=rtpmap:100 red/1000 810 a=fmtp:100 98/98/98 811 a=rtt-mixer 813 Answer example from a multi-party capable device 814 m=text 14000 RTP/AVP 100 98 815 a=rtpmap:98 t140/1000 816 a=rtpmap:100 red/1000 817 a=fmtp:100 98/98/98 818 a=rtt-mixer 820 Offer example for "text/red" format including multi-party 821 and security: 822 a=fingerprint: (fingerprint1) 823 m=text 11000 RTP/AVP 100 98 824 a=rtpmap:98 t140/1000 825 a=rtpmap:100 red/1000 826 a=fmtp:100 98/98/98 827 a=rtt-mixer 829 The "fingerprint" is sufficient to offer DTLS-SRTP, with the media 830 line still indicating RTP/AVP. 832 Note: For brevity, the entire value of the SDP fingerprint attribute 833 is not shown in this and the following example. 835 Answer example from a multi-party capable device with security 836 a=fingerprint: (fingerprint2) 837 m=text 16000 RTP/AVP 100 98 838 a=rtpmap:98 t140/1000 839 a=rtpmap:100 red/1000 840 a=fmtp:100 98/98/98 841 a=rtt-mixer 843 With the "fingerprint" the device acknowledges use of SRTP/DTLS. 845 Answer example from a multi-party unaware device that also 846 does not support security: 848 m=text 12000 RTP/AVP 100 98 849 a=rtpmap:98 t140/1000 850 a=rtpmap:100 red/1000 851 a=fmtp:100 98/98/98 853 3.23. Packet sequence example from interleaved transmission 855 This example shows a symbolic flow of packets from a mixer including 856 loss and recovery. The sequence includes interleaved transmission of 857 text from two RTT sources A and B. P indicates primary data. R1 is 858 first redundant generation data and R2 is the second redundant 859 generation data. A1, B1, A2 etc are text chunks (T140blocks) 860 received from the respective sources and sent on to the receiver by 861 the mixer. X indicates dropped packet between the mixer and a 862 receiver. The session is assumed to use original and two redundant 863 generations of RTT. 865 |-----------------------| 866 |Seq no 101, Time=20400 | 867 |CC=1 | 868 |CSRC list A | 869 |R2: A1, Offset=600 | 870 |R1: A2, Offset=300 | 871 |P: A3 | 872 |-----------------------| 874 Assuming that earlier packets ( with text A1 and A2) were received in 875 sequence, text A3 is received from packet 101 and assigned to 876 reception area A. The mixer is now assumed to have received text 877 from source B and will send that text 100 ms after packet 101. 878 Transmission of A2 and A3 as redundancy is planned for 330 ms after 879 packet 101 if no new text from A is ready to be sent before that. 881 |-----------------------| 882 |Seq no 102, Time=20500 | 883 |CC=1 | 884 |CSRC list B | 885 |R2 Empty, Offset=600 | 886 |R1: Empty, Offset=300 | 887 |P: B1 | 888 |-----------------------| 889 Packet 102 is received. 890 B1 is retrieved from this packet. Redundant transmission of 891 B1 is planned 330 ms after packet 102. 893 X------------------------| 894 X Seq no 103, Timer=20730| 895 X CC=1 | 896 X CSRC list A | 897 X R2: A2, Offset=630 | 898 X R1: A3, Offset=330 | 899 X P: Empty | 900 X------------------------| 901 Packet 103 is assumed to be dropped in network problems. It 902 contains redundancy for A. Sending A3 as second level 903 redundancy is planned for 330 ms after packet 104. 905 X------------------------| 906 X Seq no 104, Timer=20820| 907 X CC=1 | 908 X CSRC list B | 909 X R2: Empty, Offset=600 | 910 X R1: B1, Offset=300 | 911 X P: B2 | 912 X------------------------| 913 Packet 104 contains text from B, including new B2 and 914 redundant B1. It is assumed dropped in network 915 problems. 916 The mixer has A3 redundancy to send but no new text 917 appears from A and therefore the redundancy is sent 918 330 ms after the previous packet with text from A. 920 |------------------------| 921 | Seq no 105, Timer=21060| 922 | CC=1 | 923 | CSRC list A | 924 | R2: A3, Offset=660 | 925 | R1: Empty, Offset=330 | 926 | P: Empty | 927 |------------------------| 928 Packet 105 is received. 929 A gap for lost 103, and 104 is detected. 930 Assume that no other loss was detected the last second. 931 Then it can be concluded that nothing was totally lost. 933 R2 is checked. Its original time was 21040-660=20400. 934 A packet with text from A was received with that 935 timestamp, so nothing needs to be recovered. 937 B1 and B2 still needs to be transmitted as redundancy. 938 This is planned 330 ms after packet 105. But that 939 would be at 21150 which is only 90 ms after the 940 latest packet. It is instead transmitted at 941 time 21160. 943 |-----------------------| 944 |Seq no 106, Timer=21160| 945 |CC=1 | 946 |CSRC list B | 947 | R2: B1, Offset=660 | 948 | R1: B2, Offset=340 | 949 | P: Empty | 950 |-----------------------| 952 Packet 106 is received. 954 The second level redundancy in packet 106 is B1 and has timestamp 955 offset 660 ms. The timestamp of packet 106 minus 660 is 20500 which 956 is the timestamp of packet 101 THAT was received. So B1 does not 957 need to be retrieved. The first level redundancy in packet 106 has 958 offset 340. The timestamp of packet 106 minus 340 is 20820. That is 959 later than the latest received packet with source B. Therefore B2 is 960 retrieved and assigned to the input buffer for source B. No primary 961 is available in packet 106 963 After this sequence, A3 and B1 and B2 have been received. In this 964 case no text was lost. 966 3.24. Maximum character rate "CPS" 968 The default maximum rate of reception of "text/t140" real-time text 969 is in RFC 4103 [RFC4103] specified to be 30 characters per second. 970 The value MAY be modified in the CPS parameter of the FMTP attribute 971 in the media section for the "text/t140" media. A mixer combining 972 real-time text from a number of sources may occasionally have a 973 higher combined flow of text coming from the sources. Endpoints 974 SHOULD therefore specify a suitable higher value for the CPS 975 parameter, corresponding to its real reception capability. A value 976 for "CPS" of 90 is the default for the "text/t140" stream in the 977 "text/red" format when multi-party real-time text is negotiated. See 978 RFC 4103 [RFC4103] for the format and use of the CPS parameter. The 979 same rules apply for the multi-party case except for the default 980 value. 982 4. Presentation level considerations 984 ITU-T T.140 [T140] provides the presentation level requirements for 985 the RFC 4103 [RFC4103] transport. T.140 [T140] has functions for 986 erasure and other formatting functions and has the following general 987 statement for the presentation: 989 "The display of text from the members of the conversation should be 990 arranged so that the text from each participant is clearly readable, 991 and its source and the relative timing of entered text is visualized 992 in the display. Mechanisms for looking back in the contents from the 993 current session should be provided. The text should be displayed as 994 soon as it is received." 996 Strict application of T.140 [T140] is of essence for the 997 interoperability of real-time text implementations and to fulfill the 998 intention that the session participants have the same information of 999 the text contents of the conversation without necessarily having the 1000 exact same layout of the conversation. 1002 T.140 [T140] specifies a set of presentation control codes to include 1003 in the stream. Some of them are optional. Implementations MUST be 1004 able to ignore optional control codes that they do not support. 1006 There is no strict "message" concept in real-time text. The Unicode 1007 Line Separator character SHALL be used as a separator allowing a part 1008 of received text to be grouped in presentation. The characters 1009 "CRLF" may be used by other implementations as replacement for Line 1010 Separator. The "CRLF" combination SHALL be erased by just one 1011 erasing action, just as the Line Separator. Presentation functions 1012 are allowed to group text for presentation in smaller groups than the 1013 line separators imply and present such groups with source indication 1014 together with text groups from other sources (see the following 1015 presentation examples). Erasure has no specific limit by any 1016 delimiter in the text stream. 1018 4.1. Presentation by multi-party aware endpoints 1020 A multi-party aware receiving party, presenting real-time text MUST 1021 separate text from different sources and present them in separate 1022 presentation fields. The receiving party MAY separate presentation 1023 of parts of text from a source in readable groups based on other 1024 criteria than line separator and merge these groups in the 1025 presentation area when it benefits the user to most easily find and 1026 read text from the different participants. The criteria MAY e.g. be 1027 a received comma, full stop, or other phrase delimiters, or a long 1028 pause. 1030 When text is received from multiple original sources, the 1031 presentation SHOULD provide a view where text is added in multiple 1032 presentation fields. 1034 If the presentation presents text from different sources in one 1035 common area, the presenting endpoint SHOULD insert text from the 1036 local user ended at suitable points merged with received text to 1037 indicate the relative timing for when the text groups were completed. 1038 In this presentation mode, the receiving endpoint SHALL present the 1039 source of the different groups of text. This presentation style is 1040 called the "chat" style here. 1042 A view of a three-party RTT call in chat style is shown in this 1043 example . 1045 _________________________________________________ 1046 | |^| 1047 |[Alice] Hi, Alice here. |-| 1048 | | | 1049 |[Bob] Bob as well. | | 1050 | | | 1051 |[Eve] Hi, this is Eve, calling from Paris. | | 1052 | I thought you should be here. | | 1053 | | | 1054 |[Alice] I am coming on Thursday, my | | 1055 | performance is not until Friday morning.| | 1056 | | | 1057 |[Bob] And I on Wednesday evening. | | 1058 | | | 1059 |[Alice] Can we meet on Thursday evening? | | 1060 | | | 1061 |[Eve] Yes, definitely. How about 7pm. | | 1062 | at the entrance of the restaurant | | 1063 | Le Lion Blanc? | | 1064 |[Eve] we can have dinner and then take a walk |-| 1065 |______________________________________________|v| 1066 | But I need to be back to |^| 1067 | the hotel by 11 because I need |-| 1068 | | | 1069 | I wou |-| 1070 |______________________________________________|v| 1071 | of course, I underst | 1072 |________________________________________________| 1074 Figure 3: Example of a three-party RTT call presented in chat style 1075 seen at participant 'Alice's endpoint. 1077 Other presentation styles than the chat style may be arranged. 1079 This figure shows how a coordinated column view MAY be presented. 1081 _____________________________________________________________________ 1082 | Bob | Eve | Alice | 1083 |____________________|______________________|_______________________| 1084 | | |I will arrive by TGV. | 1085 |My flight is to Orly| |Convenient to the main | 1086 | |Hi all, can we plan |station. | 1087 | |for the seminar? | | 1088 |Eve, will you do | | | 1089 |your presentation on| | | 1090 |Friday? |Yes, Friday at 10. | | 1091 |Fine, wo | |We need to meet befo | 1092 |___________________________________________________________________| 1093 Figure 4: An example of a coordinated column-view of a three-party 1094 session with entries ordered vertically in approximate time-order. 1096 4.2. Multi-party mixing for multi-party unaware endpoints 1098 When the mixer has indicated multi-party capability by the "rtt- 1099 mixer" sdp attribute in an SDP negotiation, but the multi-party 1100 capability negotiation fails with an endpoint, then the agreed "text/ 1101 red" or "text/t140" format SHALL be used and the mixer SHOULD compose 1102 a best-effort presentation of multi-party real-time text in one 1103 stream intended to be presented by an endpoint with no multi-party 1104 awareness. 1106 This presentation format has functional limitations and SHOULD be 1107 used only to enable participation in multi-party calls by legacy 1108 deployed endpoints implementing only RFC 4103 without any multi-party 1109 extensions specified in this document. 1111 The principles and procedures below do not specify any new protocol 1112 elements. They are instead composed from the information in ITU-T 1113 T.140 [T140] and an ambition to provide a best effort presentation on 1114 an endpoint which has functions only for two-party calls. 1116 The mixer mixing for multi-party unaware endpoints SHALL compose a 1117 simulated limited multi-party RTT view suitable for presentation in 1118 one presentation area. The mixer SHALL group text in suitable groups 1119 and prepare for presentation of them by inserting a new line between 1120 them if the transmitted text did not already end with a new line. A 1121 presentable label SHOULD be composed and sent for the source 1122 initially in the session and after each source switch. With this 1123 procedure the time for source switching is depending on the actions 1124 of the users. In order to expedite source switch, a user can for 1125 example end its turn with a new line. 1127 4.2.1. Actions by the mixer at reception from the call participants 1129 When text is received by the mixer from the different participants, 1130 the mixer SHALL recover text from redundancy if any packets are lost. 1131 The mark for lost text [T140ad1] SHOULD be inserted in the stream if 1132 unrecoverable loss appears. Any Unicode "BOM" characters, possibly 1133 used for keep-alive shall be deleted. The time of creation of text 1134 (retrieved from the RTP timestamp) SHALL be stored together with the 1135 received text from each source in queues for transmission to the 1136 recipients. 1138 4.2.2. Actions by the mixer for transmission to the recipients 1140 The following procedure SHOULD be applied for each recipient of 1141 multi-part text from the mixer. 1143 The text for transmission SHOULD be formatted by the mixer for each 1144 receiving user for presentation in one single presentation area. 1145 Text received from a participant SHOULD NOT be included in 1146 transmission to that participant. When there is text available for 1147 transmission from the mixer to a receiving party from more than one 1148 participant, the mixer SHOULD switch between transmission of text 1149 from the different sources at suitable points in the transmitted 1150 stream. 1152 When switching source, the mixer SHOULD insert a line separator if 1153 the already transmitted text did not end with a new line (line 1154 separator or CRLF). A label SHOULD be composed from information in 1155 the CNAME and NAME fields in RTCP reports from the participant to 1156 have its text transmitted, or from other session information for that 1157 user. The label SHOULD be delimited by suitable characters (e.g. '[ 1158 ]') and transmitted. The CSRC SHOULD indicate the selected source. 1159 Then text from that selected participant SHOULD be transmitted until 1160 a new suitable point for switching source is reached. 1162 Integrity considerations SHALL be taken when composing the label. 1164 Seeking a suitable point for switching source SHOULD be done when 1165 there is older text waiting for transmission from any party than the 1166 age of the last transmitted text. Suitable points for switching are: 1168 * A completed phrase ended by comma 1170 * A completed sentence 1172 * A new line (line separator or CRLF) 1174 * A long pause (e.g. > 10 seconds) in received text from the 1175 currently transmitted source 1177 * If text from one participant has been transmitted with text from 1178 other sources waiting for transmission for a long time (e.g. > 1 1179 minute) and none of the other suitable points for switching has 1180 occurred, a source switch MAY be forced by the mixer at next word 1181 delimiter, and also if even a word delimiter does not occur within 1182 a time (e.g. 15 seconds) after the scan for word delimiter 1183 started. 1185 When switching source, the source which has the oldest text in queue 1186 SHOULD be selected to be transmitted. A character display count 1187 SHOULD be maintained for the currently transmitted source, starting 1188 at zero after the label is transmitted for the currently transmitted 1189 source. 1191 The status SHOULD be maintained for the latest control code for 1192 Select Graphic Rendition (SGR) from each source. If there is an SGR 1193 code stored as the status for the current source before the source 1194 switch is done, a reset of SGR shall be sent by the sequence SGR 0 1195 [009B 0000 006D] after the new line and before the new label during a 1196 source switch. See SGR below for an explanation. This transmission 1197 does not influence the display count. 1199 If there is an SGR code stored for the new source after the source 1200 switch, that SGR code SHOULD be transmitted to the recipient before 1201 the label. This transmission does not influence the display count. 1203 4.2.3. Actions on transmission of text 1205 Text from a source sent to the recipient SHOULD increase the display 1206 count by one per transmitted character. 1208 4.2.4. Actions on transmission of control codes 1210 The following control codes specified by T.140 require specific 1211 actions. They SHOULD cause specific considerations in the mixer. 1212 Note that the codes presented here are expressed in UCS-16, while 1213 transmission is made in UTF-8 transform of these codes. 1215 BEL 0007 Bell Alert in session, provides for alerting during an 1216 active session. The display count SHOULD not be altered. 1218 NEW LINE 2028 Line separator. Check and perform a source switch if 1219 appropriate. Increase display count by 1. 1221 CR LF 000D 000A A supported, but not preferred way of requesting a 1222 new line. Check and perform a source switch if appropriate. 1223 Increase display count by 1. 1225 INT ESC 0061 Interrupt (used to initiate mode negotiation 1226 procedure). The display count SHOULD not be altered. 1228 SGR 009B Ps 006D Select graphic rendition. Ps is rendition 1229 parameters specified in ISO 6429. The display count SHOULD not be 1230 altered. The SGR code SHOULD be stored for the current source. 1232 SOS 0098 Start of string, used as a general protocol element 1233 introducer, followed by a maximum 256 bytes string and the ST. 1234 The display count SHOULD not be altered. 1236 ST 009C String terminator, end of SOS string. The display count 1237 SHOULD not be altered. 1239 ESC 001B Escape - used in control strings. The display count SHOULD 1240 not be altered for the complete escape code. 1242 Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used 1243 for synchronization and keep-alive. SHOULD be deleted from 1244 incoming streams. Shall be sent first after session establishment 1245 to the recipient. The display count shall not be altered. 1247 Missing text mark (U+FFFD) "Replacement character", represented as a 1248 question mark in a rhombus, or if that is not feasible, replaced 1249 by an apostrophe ', marks place in stream of possible text loss. 1250 SHOULD be inserted by the reception procedure in case of 1251 unrecoverable loss of packets. The display count SHOULD be 1252 increased by one when sent as for any other character. 1254 SGR If a control code for selecting graphic rendition (SGR), other 1255 than reset of the graphic rendition (SGR 0) is sent to a 1256 recipient, that control code shall also be stored as status for 1257 the source in the storage for SGR status. If a reset graphic 1258 rendition (SGR 0) originated from a source is sent, then the SGR 1259 status storage for that source shall be cleared. The display 1260 count shall not be increased. 1262 BS (U+0008) Back Space, intended to erase the last entered character 1263 by a source. Erasure by backspace cannot always be performed as 1264 the erasing party intended. If an erasing action erases all text 1265 up to the end of the leading label after a source switch, then the 1266 mixer must not transmit more backspaces. Instead it is 1267 RECOMMENDED that a letter "X" is inserted in the text stream for 1268 each backspace as an indication of the intent to erase more. A 1269 new line is usually coded by a Line Separator, but the character 1270 combination "CRLF" MAY be used instead. Erasure of a new line is 1271 in both cases done by just one erasing action (Backspace). If the 1272 display count has a positive value it is decreased by one when the 1273 BS is sent. If the display count is at zero, it is not altered. 1275 4.2.5. Packet transmission 1277 A mixer transmitting to a multi-party unaware terminal SHOULD send 1278 primary data only from one source per packet. The SSRC SHOULD be the 1279 SSRC of the mixer. The CSRC list SHOULD contain one member and be 1280 the SSRC of the source of the primary data. 1282 4.2.6. Functional limitations 1284 When a multi-party unaware endpoint presents a conversation in one 1285 display area in a chat style, it inserts source indications for 1286 remote text and local user text as they are merged in completed text 1287 groups. When an endpoint using this layout receives and presents 1288 text mixed for multi-party unaware endpoints, there will be two 1289 levels of source indicators for the received text; one generated by 1290 the mixer and inserted in a label after each source switch, and 1291 another generated by the receiving endpoint and inserted after each 1292 switch between local and remote source in the presentation area. 1293 This will waste display space and look inconsistent to the reader. 1295 New text can be presented only from one source at a time. Switch of 1296 source to be presented takes place at suitable places in the text, 1297 such as end of phrase, end of sentence, line separator and 1298 inactivity. Therefore the time to switch to present waiting text 1299 from other sources may become long and will vary and depend on the 1300 actions of the currently presented source. 1302 Erasure can only be done up to the latest source switch. If a user 1303 tries to erase more text, the erasing actions will be presented as 1304 letter X after the label. 1306 Text loss because of network errors may hit the label between entries 1307 from different parties, causing risk for misunderstanding from which 1308 source a piece of text is. 1310 These facts makes it strongly RECOMMENDED to implement multi-party 1311 awareness in RTT endpoints. The use of the mixing method for multi- 1312 party-unaware endpoints should be left for use with endpoints which 1313 are impossible to upgrade to become multi-party aware. 1315 4.2.7. Example views of presentation on multi-party unaware endpoints 1317 The following pictures are examples of the view on a participant's 1318 display for the multi-party-unaware case. 1320 _________________________________________________ 1321 | Conference | Alice | 1322 |________________________|_________________________| 1323 | |I will arrive by TGV. | 1324 |[Bob]:My flight is to |Convenient to the main | 1325 |Orly. |station. | 1326 |[Eve]:Hi all, can we | | 1327 |plan for the seminar. | | 1328 | | | 1329 |[Bob]:Eve, will you do | | 1330 |your presentation on | | 1331 |Friday? | | 1332 |[Eve]:Yes, Friday at 10.| | 1333 |[Bob]: Fine, wo |We need to meet befo | 1334 |________________________|_________________________| 1336 Figure 5: Alice who has a conference-unaware client is receiving the 1337 multi-party real-time text in a single-stream. This figure shows how 1338 a coordinated column view MAY be presented on Alice's device. 1340 _________________________________________________ 1341 | |^| 1342 |[Alice] Hi, Alice here. |-| 1343 | | | 1344 |[mix][Bob] Bob as well. | | 1345 | | | 1346 |[Eve] Hi, this is Eve, calling from Paris | | 1347 | I thought you should be here. | | 1348 | | | 1349 |[Alice] I am coming on Thursday, my | | 1350 | performance is not until Friday morning.| | 1351 | | | 1352 |[mix][Bob] And I on Wednesday evening. | | 1353 | | | 1354 |[Eve] we can have dinner and then walk | | 1355 | | | 1356 |[Eve] But I need to be back to | | 1357 | the hotel by 11 because I need | | 1358 | |-| 1359 |______________________________________________|v| 1360 | of course, I underst | 1361 |________________________________________________| 1363 Figure 6: An example of a view of the multi-party unaware 1364 presentation in chat style. Alice is the local user. 1366 5. Relation to Conference Control 1367 5.1. Use with SIP centralized conferencing framework 1369 The SIP conferencing framework, mainly specified in RFC 1370 4353[RFC4353], RFC 4579[RFC4579] and RFC 4575[RFC4575] is suitable 1371 for coordinating sessions including multi-party RTT. The RTT stream 1372 between the mixer and a participant is one and the same during the 1373 conference. Participants get announced by notifications when 1374 participants are joining or leaving, and further user information may 1375 be provided. The SSRC of the text to expect from joined users MAY be 1376 included in a notification. The notifications MAY be used both for 1377 security purposes and for translation to a label for presentation to 1378 other users. 1380 5.2. Conference control 1382 In managed conferences, control of the real-time text media SHOULD be 1383 provided in the same way as other for media, e.g. for muting and 1384 unmuting by the direction attributes in SDP [RFC4566]. 1386 Note that floor control functions may be of value for RTT users as 1387 well as for users of other media in a conference. 1389 6. Gateway Considerations 1391 6.1. Gateway considerations with Textphones (e.g. TTYs). 1393 Multi-party RTT sessions may involve gateways of different kinds. 1394 Gateways involved in setting up sessions SHALL correctly reflect the 1395 multi-party capability or unawareness of the combination of the 1396 gateway and the remote endpoint beyond the gateway. 1398 One case that may occur is a gateway to PSTN for communication with 1399 textphones (e.g. TTYs). Textphones are limited devices with no 1400 multi-party awareness, and it SHOULD therefore be suitable for the 1401 gateway to not indicate multi-party awareness for that case. Another 1402 solution is that the gateway indicates multi-party capability towards 1403 the mixer, and includes the multi-party mixer function for multi- 1404 party unaware endpoints itself. This solution makes it possible to 1405 make adaptations for the functional limitations of the textphone 1406 (TTY). 1408 More information on gateways to textphones (TTYs) is found in RFC 1409 5194[RFC5194] 1411 6.2. Gateway considerations with WebRTC. 1413 Gateway operation to real-time text in WebRTC may also be required. 1414 In WebRTC, RTT is specified in 1415 [I-D.ietf-mmusic-t140-usage-data-channel]. 1417 A multi-party bridge may have functionality for communicating by RTT 1418 both in RTP streams with RTT and WebRTC t140 data channels. Other 1419 configurations may consist of a multi-party bridge with either 1420 technology for RTT transport and a separate gateway for conversion of 1421 the text communication streams between RTP and t140 data channel. 1423 In WebRTC, it is assumed that for a multi-party session, one t140 1424 data channel is established for each source from a gateway or bridge 1425 to each participant. Each participant also has a data channel with 1426 two-way connection with the gateway or bridge. 1428 The t140 channel used both ways is for text from the WebRTC user and 1429 from the bridge or gateway itself to the WebRTC user. The label 1430 parameter of this t140 channel is used as NAME field in RTCP to 1431 participants on the RTP side. The other t140 channels are only for 1432 text from other participants to the WebRTC user. 1434 When a new participant has entered the session with RTP transport of 1435 rtt, a new t140 channel SHOULD be established to WebRTC users with 1436 the label parameter composed from the NAME field in RTCP on the RTP 1437 side. 1439 When a new participant has entered the multi-party session with RTT 1440 transport in a WebRTC t140 data channel, the new participant SHOULD 1441 be announced by a notification to RTP users. The label parameter 1442 from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP 1443 side, or other available session information. 1445 7. Updates to RFC 4103 1447 This document updates RFC 4103[RFC4103] by introducing an sdp media 1448 attribute "rtt-mixer" for negotiation of multi-party mixing 1449 capability with the [RFC4103] format, and by specifying the rules for 1450 packets when multi-party capability is negotiated and in use. 1452 8. Congestion considerations 1454 The congestion considerations and recommended actions from RFC 4103 1455 [RFC4103] are valid also in multi-party situations. 1457 The first action in case of congestion SHOULD be to temporarily 1458 increase the transmission interval up to two seconds. 1460 If the unlikely situation appears that more than 20 participants in a 1461 conference send text simultaneously, it will take more than 7 seconds 1462 between presentation of text from each of these participants. More 1463 time than that can cause confusion in the session. It is therefore 1464 RECOMMENDED that the mixer discards such text in excess inserts a 1465 general indication of possible text loss [T140ad1] in the session. 1466 If the main text contributor is indicated in any way, the mixer MAY 1467 avoid deleting text from that participant. 1469 9. Acknowledgements 1471 James Hamlin for format and performance aspects. 1473 10. IANA Considerations 1475 10.1. Registration of the "rtt-mixer" sdp media attribute 1477 [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the 1478 RFC number of this document.] 1480 IANA is asked to register the new sdp attribute "rtt-mixer". 1482 Contact name: IESG 1484 Contact email: iesg@ietf.org 1486 Attribute name: rtt-mixer 1488 Attribute semantics: See RFCXXXX Section 3.1 1490 Attribute value: none 1492 Usage level: media 1494 Purpose: Indicate support by mixer and endpoint of multi-party 1495 mixing for real-time text transmission, using a common RTP-stream 1496 for transmission of text from a number of sources mixed with one 1497 source at a time and the source indicated in a single CSRC-list 1498 member. 1500 Charset Dependent: no 1502 O/A procedure: See RFCXXXX Section 3.1 1504 Mux Category: normal 1506 Reference: RFCXXXX 1508 11. Security Considerations 1510 The RTP-mixer model requires the mixer to be allowed to decrypt, pack 1511 and encrypt secured text from the conference participants. Therefore 1512 the mixer needs to be trusted. This is similar to the situation for 1513 central mixers of audio and video. 1515 The requirement to transfer information about the user in RTCP 1516 reports in SDES, CNAME and NAME fields, and in conference 1517 notifications, for creation of labels may have privacy concerns as 1518 already stated in RFC 3550 [RFC3550], and may be restricted of 1519 privacy reasons. The receiving user will then get a more symbolic 1520 label for the source. 1522 Participants with malicious intentions may appear and e.g. disturb 1523 the multi-party session by a continuous flow of text, or masquerading 1524 as text from other participants. Counteractions should be to require 1525 secure signaling, media and authentication, and to provide higher 1526 level conference functions e.g. for blocking and expelling 1527 participants. 1529 12. Change history 1531 12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11 1533 Timestamps and timestamp offsets added to the packet examples in 1534 section 3.23, and the description corrected. 1536 A number of minor corrections added in sections 3.10 - 3.23. 1538 12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10 1540 The packet composition was modified for interleaving packets from 1541 different sources. 1543 The packet reception was modified for the new interleaving method. 1545 The packet sequence examples was adjusted for the new interleaving 1546 method. 1548 Modifications according to responses to Brian Rosen of 2020-11-03 1550 12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09 1552 Changed name on the SDP media attribute to "rtt-mixer" 1554 Restructure of section 2 for balance between aware and unaware cases. 1556 Moved conference control to own section. 1558 Improved clarification of recovery and loss in the packet sequence 1559 example. 1561 A number of editorial corrections and improvements. 1563 12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 1565 Deleted the method requiring a new packet format "text/rex" because 1566 of the longer standardization and implementation period it needs. 1568 Focus on use of RFC 4103 text/red format with shorter transmission 1569 interval, and source indicated in CSRC. 1571 12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 1573 Added a method based on the "text/red" format and single source per 1574 packet, negotiated by the "rtt-mixer" sdp attribute. 1576 Added reasoning and recommendation about indication of loss. 1578 The highest number of sources in one packet is 15, not 16. Changed. 1580 Added in information on update to RFC 4103 that RFC 4103 explicitly 1581 allows addition of FEC method. The redundancy is a kind of forward 1582 error correction.. 1584 12.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 1586 Improved definitions list format. 1588 The format of the media subtype parameters is made to match the 1589 requirements. 1591 The mapping of media subtype parameters to sdp is included. 1593 The CPS parameter belongs to the t140 subtype and does not need to be 1594 registered here. 1596 12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 1598 nomenclature and editorial improvements 1600 "this document" used consistently to refer to this document. 1602 12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 1604 'Redundancy header' renamed to 'data header'. 1606 More clarifications added. 1608 Language and figure number corrections. 1610 12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 1612 Mention possible need to mute and raise hands as for other media. 1613 ---done ---- 1615 Make sure that use in two-party calls is also possible and explained. 1616 - may need more wording - 1618 Clarify the RTT is often used together with other media. --done-- 1620 Tell that text mixing is N-1. A users own text is not received in 1621 the mix. -done- 1623 In 3. correct the interval to: A "text/rex" transmitter SHOULD send 1624 packets distributed in time as long as there is something (new or 1625 redundant T140blocks) to transmit. The maximum transmission interval 1626 SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 1627 receiver as soon as new text to that receiver is available, as long 1628 as the time after the latest sent packet to the same receiver is more 1629 than 150 ms, and also the maximum character rate to the receiver is 1630 not exceeded. The intention is to keep the latency low while keeping 1631 a good protection against text loss in bursty packet loss conditions. 1632 -done- 1634 In 1.3 say that the format is used both ways. -done- 1636 In 13.1 change presentation area to presentation field so that reader 1637 does not think it shall be totally separated. -done- 1639 In Performance and intro, tell the performance in number of 1640 simultaneous sending users and introduced delay 16, 150 vs 1641 requirements 5 vs 500. -done -- 1643 Clarify redundancy level per connection. -done- 1645 Timestamp also for the last data header. To make it possible for all 1646 text to have time offset as for transmission from the source. Make 1647 that header equal to the others. -done- 1649 Mixer always use the CSRC list, even for its own BOM. -done- 1650 Combine all talk about transmission interval (300 ms vs when text has 1651 arrived) in section 3 in one paragraph or close to each other. -done- 1653 Documents the goal of good performance with low delay for 5 1654 simultaneous typers in the introduction. -done- 1656 Describe better that only primary text shall be sent on to receivers. 1657 Redundancy and loss must be resolved by the mixer. -done- 1659 12.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 1661 SDP and better description and visibility of security by OSRTP RFC 1662 8634 needed. 1664 The description of gatewaying to WebRTC extended. 1666 The description of the data header in the packet is improved. 1668 12.11. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 1670 2,5,6 More efficient format "text/rex" introduced and attribute 1671 a=rtt-mix deleted. 1673 3. Brief about use of OSRTP for security included- More needed. 1675 4. Brief motivation for the solution and why not rtp-translator is 1676 used added to intro. 1678 7. More limitations for the multi-party unaware mixing method 1679 inserted. 1681 8. Updates to RFC 4102 and 4103 more clearly expressed. 1683 9. Gateway to WebRTC started. More needed. 1685 12.12. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 1686 to draft-ietf-avtcore-multi-party-rtt-mix-00 1688 Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00 1690 Replaced CDATA in IANA registration table with better coding. 1692 Converted to xml2rfc version 3. 1694 12.13. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 1695 to -03 1697 Changed company and e-mail of the author. 1699 Changed title to "RTP-mixer formatting of multi-party Real-time text" 1700 to better match contents. 1702 Check and modification where needed of use of RFC 2119 words SHALL 1703 etc. 1705 More about the CC value in sections on transmitters and receivers so 1706 that 1-to-1 sessions do not use the mixer format. 1708 Enhanced section on presentation for multi-party-unaware endpoints 1710 A paragraph recommending CPS=150 inserted in the performance section. 1712 12.14. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 1713 to -02 1715 In Abstract and 1. Introduction: Introduced wording about regulatory 1716 requirements. 1718 In section 5: The transmission interval is decreased to 100 ms when 1719 there is text from more than one source to transmit. 1721 In section 11 about SDP negotiation, a SHOULD-requirement is 1722 introduced that the mixer should make a mix for multi-party unaware 1723 endpoints if the negotiation is not successful. And a reference to a 1724 later chapter about it. 1726 The presentation considerations chapter 14 is extended with more 1727 information about presentation on multi-party aware endpoints, and a 1728 new section on the multi-party unaware mixing with low functionality 1729 but SHOULD a be implemented in mixers. Presentation examples are 1730 added. 1732 A short chapter 15 on gateway considerations is introduced. 1734 Clarification about the text/t140 format included in chapter 10. 1736 This sentence added to the chapter 10 about use without redundancy. 1737 "The text/red format SHOULD be used unless some other protection 1738 against packet loss is utilized, for example a reliable network or 1739 transport." 1741 Note about deviation from RFC 2198 added in chapter 4. 1743 In chapter 9. "Use with SIP centralized conferencing framework" the 1744 following note is inserted: Note: The CSRC-list in an RTP packet only 1745 includes participants who's text is included in one or more text 1746 blocks. It is not the same as the list of participants in a 1747 conference. With audio and video media, the CSRC-list would often 1748 contain all participants who are not muted whereas text participants 1749 that don't type are completely silent and so don't show up in RTP 1750 packet CSRC-lists. 1752 12.15. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 1753 to -01 1755 Editorial cleanup. 1757 Changed capability indication from fmtp-parameter to SDP attribute 1758 "rtt-mix". 1760 Swapped order of redundancy elements in the example to match reality. 1762 Increased the SDP negotiation section 1764 13. References 1766 13.1. Normative References 1768 [I-D.ietf-mmusic-t140-usage-data-channel] 1769 Holmberg, C. and G. Hellstrom, "T.140 Real-time Text 1770 Conversation over WebRTC Data Channels", Work in Progress, 1771 Internet-Draft, draft-ietf-mmusic-t140-usage-data-channel- 1772 14, 10 April 2020, . 1775 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1776 Requirement Levels", BCP 14, RFC 2119, 1777 DOI 10.17487/RFC2119, March 1997, 1778 . 1780 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1781 Jacobson, "RTP: A Transport Protocol for Real-Time 1782 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1783 July 2003, . 1785 [RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type", 1786 RFC 4102, DOI 10.17487/RFC4102, June 2005, 1787 . 1789 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 1790 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 1791 . 1793 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1794 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 1795 July 2006, . 1797 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1798 Security (DTLS) Extension to Establish Keys for the Secure 1799 Real-time Transport Protocol (SRTP)", RFC 5764, 1800 DOI 10.17487/RFC5764, May 2010, 1801 . 1803 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 1804 Keeping Alive the NAT Mappings Associated with RTP / RTP 1805 Control Protocol (RTCP) Flows", RFC 6263, 1806 DOI 10.17487/RFC6263, June 2011, 1807 . 1809 [RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. 1810 Stach, "An Opportunistic Approach for Secure Real-time 1811 Transport Protocol (OSRTP)", RFC 8643, 1812 DOI 10.17487/RFC8643, August 2019, 1813 . 1815 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 1816 multimedia application text conversation", February 1998, 1817 . 1819 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 1820 Protocol for multimedia application text conversation", 1821 February 2000, 1822 . 1824 13.2. Informative References 1826 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 1827 Session Initiation Protocol (SIP)", RFC 4353, 1828 DOI 10.17487/RFC4353, February 2006, 1829 . 1831 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 1832 Session Initiation Protocol (SIP) Event Package for 1833 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 1834 2006, . 1836 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 1837 (SIP) Call Control - Conferencing for User Agents", 1838 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 1839 . 1841 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 1842 Time Text over IP Using the Session Initiation Protocol 1843 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 1844 . 1846 Author's Address 1848 Gunnar Hellstrom 1849 Gunnar Hellstrom Accessible Communication 1850 SE-13670 Vendelso 1851 Sweden 1853 Email: gunnar.hellstrom@ghaccess.se