idnits 2.17.1 draft-ietf-avtcore-multi-party-rtt-mix-08.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 661 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 (12 August 2020) is 1354 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 1214, 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) 12 August 2020 5 Intended status: Standards Track 6 Expires: 13 February 2021 8 RTP-mixer formatting of multi-party Real-time text 9 draft-ietf-avtcore-multi-party-rtt-mix-08 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. 18 Regional regulatory requirements specify provision of real-time text 19 in multi-party calls. RFC 4103 mixer implementations can use 20 traditional RTP functions for source identification, but the mixer 21 source switching performance is limited when using the default 22 transmission characteristics with redundancy. 24 Enhancements for RFC 4103 real-time text mixing is provided in this 25 document, suitable for a centralized conference model that enables 26 source identification and source switching. The intended use is for 27 real-time text mixers and multi-party-aware participant endpoints. 28 The specified mechanism build on the standard use of the CSRC list in 29 the RTP packet for source identification. The method makes use of 30 the same "text/red" format as for two-party sessions. 32 A capability exchange is specified so that it can be verified that a 33 participant can handle the multi-party coded real-time text stream. 34 The capability is indicated by use of a media attribute "rtt-mix-rtp- 35 mixer". 37 The document updates RFC 4103[RFC4103] 39 A specifications of how a mixer can format text for the case when the 40 endpoint is not multi-party aware is also provided. 42 Status of This Memo 44 This Internet-Draft is submitted in full conformance with the 45 provisions of BCP 78 and BCP 79. 47 Internet-Drafts are working documents of the Internet Engineering 48 Task Force (IETF). Note that other groups may also distribute 49 working documents as Internet-Drafts. The list of current Internet- 50 Drafts is at https://datatracker.ietf.org/drafts/current/. 52 Internet-Drafts are draft documents valid for a maximum of six months 53 and may be updated, replaced, or obsoleted by other documents at any 54 time. It is inappropriate to use Internet-Drafts as reference 55 material or to cite them other than as "work in progress." 57 This Internet-Draft will expire on 13 February 2021. 59 Copyright Notice 61 Copyright (c) 2020 IETF Trust and the persons identified as the 62 document authors. All rights reserved. 64 This document is subject to BCP 78 and the IETF Trust's Legal 65 Provisions Relating to IETF Documents (https://trustee.ietf.org/ 66 license-info) in effect on the date of publication of this document. 67 Please review these documents carefully, as they describe your rights 68 and restrictions with respect to this document. Code Components 69 extracted from this document must include Simplified BSD License text 70 as described in Section 4.e of the Trust Legal Provisions and are 71 provided without warranty as described in the Simplified BSD License. 73 Table of Contents 75 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 76 1.1. Selected solution and considered alternative . . . . . . 4 77 1.2. Nomenclature . . . . . . . . . . . . . . . . . . . . . . 6 78 1.3. Intended application . . . . . . . . . . . . . . . . . . 7 79 2. Specified solutions . . . . . . . . . . . . . . . . . . . . . 7 80 2.1. Negotiated use of the RFC 4103 format for multi-party in a 81 single RTP stream . . . . . . . . . . . . . . . . . . . . 7 82 2.2. Mixing for multi-party unaware endpoints . . . . . . . . 19 83 3. Presentation level considerations . . . . . . . . . . . . . . 19 84 3.1. Presentation by multi-party aware endpoints . . . . . . . 20 85 3.2. Multi-party mixing for multi-party unaware endpoints . . 22 86 4. Gateway Considerations . . . . . . . . . . . . . . . . . . . 27 87 4.1. Gateway considerations with Textphones (e.g. TTYs). . . 28 88 4.2. Gateway considerations with WebRTC. . . . . . . . . . . . 28 89 5. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 29 90 6. Congestion considerations . . . . . . . . . . . . . . . . . . 29 91 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 29 92 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 29 93 8.1. Registration of the "rtt-mix-rtp-mixer" sdp media 94 attribute . . . . . . . . . . . . . . . . . . . . . . . . 29 96 9. Security Considerations . . . . . . . . . . . . . . . . . . . 30 97 10. Change history . . . . . . . . . . . . . . . . . . . . . . . 30 98 10.1. Changes included in 99 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 30 100 10.2. Changes included in 101 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 31 102 10.3. Changes included in 103 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 31 104 10.4. Changes included in 105 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 31 106 10.5. Changes included in 107 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 31 108 10.6. Changes included in 109 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 31 110 10.7. Changes included in 111 draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 32 112 10.8. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 33 113 10.9. Changes from 114 draft-hellstrom-avtcore-multi-party-rtt-source-03 to 115 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 33 116 10.10. Changes from 117 draft-hellstrom-avtcore-multi-party-rtt-source-02 to 118 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 33 119 10.11. Changes from 120 draft-hellstrom-avtcore-multi-party-rtt-source-01 to 121 -02 . . . . . . . . . . . . . . . . . . . . . . . . . . 34 122 10.12. Changes from 123 draft-hellstrom-avtcore-multi-party-rtt-source-00 to 124 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 34 125 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 35 126 11.1. Normative References . . . . . . . . . . . . . . . . . . 35 127 11.2. Informative References . . . . . . . . . . . . . . . . . 36 128 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 36 130 1. Introduction 132 RFC 4103[RFC4103] specifies use of RFC 3550 RTP [RFC3550] for 133 transmission of real-time text (RTT) and the "text/t140" format. It 134 also specifies a redundancy format "text/red" for increased 135 robustness. RFC 4102 [RFC4102] registers the "text/red" format. 136 Regional regulatory requirements specify provision of real-time text 137 in multi-party calls. 139 Real-time text is usually provided together with audio and sometimes 140 with video in conversational sessions. 142 The redundancy scheme of RFC 4103 [RFC4103] enables efficient 143 transmission of redundant text in packets together with new text. 144 However the redundancy header format has no source indicators for the 145 redundant transmissions. An assumption has to be made that the 146 redundant parts in a packet are from the same source as the new text. 147 The recommended transmission is one new and two redundant generations 148 of text (T140blocks) in each packet and the recommended transmission 149 interval is 300 ms. 151 A mixer, selecting between text input from different sources and 152 transmitting it in a common stream needs to make sure that the 153 receiver can assign the received text to the proper sources for 154 presentation. Therefore, using RFC 4103 without any extra rule for 155 source identification, the mixer needs to stop sending new text from 156 one source and then make sure that all text sent so far has been sent 157 with all intended redundancy levels (usually two) before switching to 158 another source. That causes the long time of one second to switch 159 between transmission of text from one source to text from another 160 source when using the default transmission interval 300 ms. Both the 161 total throughput and the switching performance in the mixer would be 162 too low for most applications. However by shorting the transmission 163 interval to 100 ms, good performance is achieved for up to 3 164 simultaneously sending sources and usable performance for up to 5 165 simultaneously sending sources. Capability to use this method is 166 indicated by an sdp media attribute "rtt-mix-rtp-mixer". 168 A negotiation mechanism can therefore be based on selection of the 169 "text/red" with media attribute "rtt-mix-rtp-mixer" for verification 170 that the parties are able to handle a multi-party coded stream and 171 agreeing on using that method. 173 A fall-back mixing procedure is specified for cases when the 174 negotiation results in "text/red" without the "rtt-mix-rtp-mixer" 175 attribute being the only common format for real-time text. 177 The document updates RFC 4103[RFC4103] by introducing an attribute 178 for indicating capability for the multi-party mixing case and rules 179 for source indications and source switching. 181 1.1. Selected solution and considered alternative 183 A number of alternatives were considered when searching an efficient 184 and easily implemented multi-party method for real-time text. This 185 section explains a few of them briefly. 187 One RTP stream per source, sent in the same RTP session with 188 "text/red" format. 189 From some points of view, use of multiple RTP streams, one for 190 each source, sent in the same RTP session, called the RTP 191 translator model in [RFC3550], would be efficient, and use exactly 192 the same packet format as [RFC4103], the same payload type and a 193 simple SDP declaration. However, there is currently lack of 194 support for multi-stream RTP in certain implementation 195 technologies. This fact made it not included in this 196 specification. 198 The "text/red" format in RFC 4103 with shorter transmission 199 interval, and indicating source in CSRC. 200 The "text/red" format with "text/t140" payload in a single RTP 201 stream can be sent with 100 ms packet intervals instead of the 202 regular 300 ms. The source is indicated in the CSRC field. 203 Source switching can then be done every 300 ms while simultaneous 204 transmission occurs. With two participants sending text 205 simultaneously, the switching and transmission performance is 206 good. With three simultaneously sending participants, there will 207 be a noticable jerkiness in text presentation. The jerkiness will 208 be more expressed the more participants who send text 209 simultaneously. With five sending participants, the jerkiness 210 will be about 1400 ms. Text sent from a source at the end of the 211 period its text is sent by the mixer will have close to zero extra 212 delay. Recent text will be presented with no or low delay. The 213 1400 ms jerkiness will be noticable and slightly unpleasant, but 214 corresponds in time to what typing humans often cause by 215 hesitation or changing position while typing. A benefit of this 216 method is that no new packet format needs to be introduced and 217 implemented. Since simultaneous typing by more than two parties 218 is rare, and in most applications also more than three parties in 219 a call is rare, this method can be used successfully without its 220 limitations becoming annoying. Negotiation is based on a new sdp 221 media attribute "rtt-mix-rtp-mixer". This method is selected to 222 be the main one specified in this document. 224 A new "text" media subtype with up to 15 sources in each packet. 225 The mechanism makes use of the RTP mixer model specified in 226 RFC3550[RFC3550]. Text from up to 15 sources can be included in 227 each packet. Packets are normally sent every 300 ms. The mean 228 delay will be 150 ms. The sources are indicated in strict order 229 in the CSRC list of the RTP packets. A new redundancy packet 230 format is specified. This method would result in good 231 performance, but would require standardisation and implementation 232 of new releases in the target technologies that would take more 233 time than desirable to complete. It was therefore not selected to 234 be included in this specification. 236 The presentation planned by the mixer for multi-party unaware 237 endpoints. 238 It is desirable to have a method that does not require any 239 modifications in existing user devices implementing RFC 4103 for 240 RTT without explicit support of multi-party sessions. This is 241 possible by having the mixer insert a new line and a text 242 formatted source label before each switch of text source in the 243 stream. Switch of source can only be done in places in the text 244 where it does not disturb the perception of the contents. Text 245 from only one source can be presented in real time at a time. The 246 delay will therefore be varying. The method has also other 247 limitations, but is included in this document as a fallback 248 method. In calls where parties take turns properly by ending 249 their entries with a new line, the limitations will have limited 250 influence on the user experience. while only two parties send 251 text, these two will see the text in real time with no delay. 252 This method is specified as a fallback method in this 253 specification. 255 RTT transport in WebRTC 256 Transport of real-time text in the WebRTC technology is specified 257 to use the WebRTC data channel in 258 [I-D.ietf-mmusic-t140-usage-data-channel]. That spcification 259 contains a section briefly describing its use in multi-party 260 sessions. The focus of this specification is RTP transport. 261 Therefore, even if the WebRTC transport provides good multi-party 262 performance, it is just mentioned in this specification in 263 relation to providing gateways with multi-party capabilities 264 between RTP and WebRTC technologies. 266 1.2. Nomenclature 268 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 269 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 270 document are to be interpreted as described in [RFC2119]. 272 The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP- 273 mixer, RTP-translator are explained in [RFC3550] 275 The term "T140block" is defined in RFC 4103 [RFC4103] to contain one 276 or more T.140 code elements. 278 "TTY" stands for a text telephone type used in North America. 280 "WebRTC" stands for web based communication specified by W3C and 281 IETF. 283 "DTLS-SRTP" stnds for security specified in RFC 5764 [RFC5764]. 285 1.3. Intended application 287 The method for multi-party real-time text documented in this 288 specification is primarily intended for use in transmission between 289 mixers and endpoints in centralised mixing configurations. It is 290 also applicable between mixers. An often mentioned application is 291 for emergency service calls with real-time text and voice, where a 292 calltaker want to make an attended handover of a call to another 293 agent, and stay observing the session. Multimedia conference 294 sessions with support for participants to contribute in text is 295 another application. Conferences with central support for speech-to- 296 text conversion is yet another mentioned application. 298 In all these applications, normally only one participant at a time 299 will send long text utterances. In some cases, one other participant 300 will occasionally contribute with a longer comment simultaneously. 301 That may also happen in some rare cases when text is interpreted to 302 text in another language in a conference. Apart from these cases, 303 other participants are only expected to contribute with very brief 304 utterings while others are sending text. 306 Text is supposed to be human generated, by some text input means, 307 such as typing on a keyboard or using speech-to-text technology. 308 Occasional small cut-and-paste operations may appear even if that is 309 not the initial purpose of real-time text. 311 The real-time characteristics of real-time text is essential for the 312 participants to be able to contribute to a conversation. If the text 313 is too much delayed from typing a letter to its presentation, then, 314 in some conference situations, the opportunity to comment will be 315 gone and someone else will grab the turn. A delay of more than one 316 second in such situations is an obstacle for good conversation. 318 2. Specified solutions 320 2.1. Negotiated use of the RFC 4103 format for multi-party in a single 321 RTP stream 323 This section specifies use of the current format specified in 324 [RFC4103] for true multi-party real-time text. It is an update of 325 RFC 4103 by a clarification on one way to use it in the multi-party 326 situation. It is done by completing a negotiation for this kind of 327 multi-party capability and by indicating source in the CSRC element 328 in the RTP packets. 330 Please use [RFC4103] as reference when reading the following 331 description. 333 2.1.1. Negotiation for use of this method 335 RFC 4103[RFC4103] specifies use of RFC 3550 RTP[RFC3550], and a 336 redundancy format "text/red" for increased robustness of real-time 337 text transmission. This document updates RFC 4103[RFC4103] by 338 introducing a capability negotiation for handling multi-party real- 339 time text. The capability negotiation is based on use of the sdp 340 media attribute "rtt-mix-rtp-mixer". 342 The syntax is as follows: 343 "a=rtt-mix-rtp-mixer" 345 A transmitting party SHALL send text according to the multi-party 346 format only when the negotiation for this method was successful and 347 when the CC field in the RTP packet is set to 1. In all other cases, 348 the packets SHALL be populated and interpreted as for a two-party 349 session. 351 2.1.2. Use of fields in the RTP packets 353 The CC field SHALL show the number of members in the CSRC list, which 354 SHALL be one (1) in transmissions from a mixer involved in a multi- 355 party session, and otherwise 0. 357 When transmitted from a mixer during a multi-party session, a CSRC 358 list is included in the packet. The single member in the CSRC-list 359 SHALL contain the SSRC of the source of the T140blocks in the packet. 360 When redundancy is used, the recommended level of redundancy is to 361 use one primary and two redundant generations of T140blocks. In some 362 cases, a primary or redundant T140block is empty, but is still 363 represented by a member in the redundancy header. 365 From other aspects, the contents of the RTP packts are equal to what 366 is specified in [RFC4103]. 368 2.1.3. Transmission of multi-party contents 370 As soon as a participant is known to participate in a session and 371 being available for text reception, a Unicode BOM character SHALL be 372 sent to it according to the procedures in this section. If the 373 transmitter is a mixer, then the source of this character SHALL be 374 indicated to be the mixer itself. 376 2.1.4. Keep-alive 378 After that, the transmitter SHALL send keep-alive traffic to the 379 receivers at regular intervals when no other traffic has occurred 380 during that interval if that is decided for the actual connection. 381 Recommendations for keep-alive can be found in [RFC6263]. 383 2.1.5. Transmission interval 385 A "text/red" transmitter in a mixer SHOULD send packets distributed 386 in time as long as there is something (new or redundant T140blocks) 387 to transmit. The maximum transmission interval SHOULD then be 300 388 ms. It is RECOMMENDED to send next packet to a receiver as soon as 389 new text to that receiver is available, as long as the time after the 390 latest sent packet to the same receiver is more than or equal to 100 391 ms, and also the maximum character rate to the receiver is not 392 exceeded. The intention is to keep the latency low while keeping a 393 good protection against text loss in bursty packet loss conditions. 395 2.1.6. Only one source per packet 397 New and redundant text from one source MAY be transmitted in the same 398 packet. Text from different sources MUST NOT be transmitted in the 399 same packet. 401 2.1.7. Do not send received text to the originating source 403 Text received from a participant SHOULD NOT be included in 404 transmission to that participant. 406 2.1.8. Clean incoming text 408 A mixer SHALL handle reception and recovery of packet loss, marking 409 of possible text loss and deletion of 'BOM' characters from each 410 participant before queueing received text for transmission to 411 receiving participants. 413 2.1.9. Redundancy 415 The transmitting party using redundancy SHALL send redundant 416 repetitions of T140blocks aleady transmitted in earlier packets. 418 The number of redundant generations of T140blocks to include in 419 transmitted packets SHALL be deducted from the SDP negotiation. It 420 SHOULD be set to the minimum of the number declared by the two 421 parties negotiating a connection. 423 2.1.10. Text placement in packets 425 At time of transmission, the mixer SHALL populate the RTP packet with 426 all T140blocks queued for transmission originating from the source in 427 turn for transmission as long as this is not in conflict with the 428 allowed number of characters per second ("CPS") or the maximum packet 429 size. The SSRC of the source shall be placed as the only member in 430 the CSRC-list. 432 Note: The CSRC-list in an RTP packet only includes the participant 433 who's text is included in text blocks. It is not the same as the 434 total list of participants in a conference. With audio and video 435 media, the CSRC-list would often contain all participants who are not 436 muted whereas text participants that don't type are completely silent 437 and thus are not represented in RTP packet CSRC-lists once their text 438 have been transmitted as primary and the intended number of redundant 439 generations. 441 2.1.11. Source switching 443 When text from more than one source is available for transmission, 444 the mixer SHALL let the sources take turns in having their text 445 transmitted. When switching from transmission of one source to allow 446 another source to have its text sent, all intended redundant 447 generations of the last text from the current source MUST be 448 transmitted before text from another source can be transmitted. 450 Actively transmitting sources SHOULD be allowed to take turns as 451 frequently as possible to have their text transmitted. That implies 452 that with the recommended redundancy, the mixer SHALL send primary 453 text and two packets with redundant text from the current source 454 before text from another source is transmitted. The source with the 455 oldest text received in the mixer SHOULD be next in turn to get all 456 its available text transmitted. 458 2.1.12. Empty T140blocks 460 If no unsent T140blocks were available for a source at the time of 461 populating a packet, but T140blocks are available which have not yet 462 been sent the full intended number of redundant transmissions, then 463 the primary T140block for that source is composed of an empty 464 T140block, and populated (without taking up any length) in a packet 465 for transmission. The corresponding SSRC SHALL be placed as usual in 466 its place in the CSRC-list. 468 2.1.13. Creation of the redundancy 470 The primary T140block from a source in the latest transmitted packet 471 is used to populate the first redundant T140block for that source. 472 The first redundant T140block for that source from the latest 473 transmission is placed as the second redundant T140block. 475 Usually this is the level of redundancy used. If a higher number of 476 redundancy is negotiated, then the procedure SHALL be maintained 477 until all available redundant levels of T140blocks are placed in the 478 packet. If a receiver has negotiated a lower number of "text/red" 479 generations, then that level shall be the maximum used by the 480 transmitter. 482 2.1.14. Timer offset fields 484 The timestamp offset values are inserted in the redundancy header, 485 with the time offset from the RTP timestamp in the packet when the 486 corresponding T140block was sent from its original source as primary. 488 The timestamp offsets are expressed in the same clock tick units as 489 the RTP timestamp. 491 The timestamp offset values for empty T140blocks have no relevance 492 but SHOULD be assigned realistic values. 494 2.1.15. Other RTP header fields 496 The number of members in the CSRC list ( 0 or 1) shall be placed in 497 the "CC" header field. Only mixers place value 1 in the "CC" field. 499 The current time SHALL be inserted in the timestamp. 501 The SSRC of the mixer for the RTT session SHALL be inserted in the 502 SSRC field of the RTP header. 504 The M-bit shall be handled as specified in [RFC4103]. 506 2.1.16. Pause in transmission 508 When there is no new T140block to transmit, and no redundant 509 T140block that has not been retransmitted the intended number of 510 times from any source, the transmission process can stop until either 511 new T140blocks arrive, or a keep-alive method calls for transmission 512 of keep-alive packets. 514 2.1.17. RTCP considerations 516 A mixer SHALL send RTCP reports with SDES, CNAME and NAME information 517 about the sources in the multi-party call. This makes it possible 518 for participants to compose a suitable label for text from each 519 source. 521 Integrity considerations SHALL be considered when composing these 522 fields. 524 2.1.18. Reception of multi-party contents 526 The "text/red" receiver included in an endpoint with presentation 527 functions will receive RTP packets in the single stream from the 528 mixer, and SHALL distribute the T140blocks for presentation in 529 presentation areas for each source. Other receiver roles, such as 530 gateways or chained mixers are also feasible, and requires 531 consideration if the stream shall just be forwarded, or distributed 532 based on the different sources. 534 2.1.18.1. Multi-party vs two-party use 536 If the "CC" field value of a received packet is 1, it indicates that 537 multi-party transmission is active, and the receiver MUST be prepared 538 to act on the source according to its role. If the CC value is 0, 539 the connection is point-to-point. 541 2.1.18.2. Level of redundancy 543 The used level of redundancy generations SHALL be evaluated from the 544 received packet contents. The number of generations (including the 545 primary) is equal to the number of members in the redundancy header. 547 2.1.18.3. Extracting text and handling recovery and loss 549 The RTP sequence numbers of the received packets SHALL be monitored 550 for gaps and packets out of order. 552 As long as the sequence is correct, each packet SHALL be unpacked in 553 order. The T140blocks SHALL be extracted from the primary area, and 554 the corresponding SSRC SHALL be extracted from the CSRC list and used 555 for assigning the new T140block to the correct presentation areas (or 556 correspondingly for other receiver roles). 558 If a sequence number gap appears and is still there after some 559 defined time for jitter resolution, T140data SHALL be recovered from 560 redundant data. If the gap is wider than the number of generations 561 of redundant T140blocks in the packet, then a t140block SHALL be 562 created with a marker for possible text loss [T140ad1] and assigned 563 to the SSRC of the transmitter as a general input from the mixer 564 because in general it is not possible to deduct from which source(s) 565 text was lost. It is in some cases possible to deduct that no text 566 was lost even for a gap wider than the redundancy generations, and in 567 some cases it can be concluded which source that likely had loss. 568 Therefore, the receiver MAY insert the marker for possible text loss 569 [T140ad1] in the presentation area corresponding to the source which 570 may have had loss. 572 Then, the T140block in the received packet SHALL be retrieved 573 beginning with the highest redundant generation, and assigning it to 574 the presentation area of that source. Finally the primary T140block 575 SHALL be retrieved from the packet and similarly assigned to the 576 corresponding presentation area for the source. 578 If the sequence number gap was equal to or less than the number of 579 redundancy generations in the received packet, a missing text marker 580 SHALL NOT be inserted, and instead the T140block and the SSRC fully 581 recovered from the redundancy information and the CSRC-list in the 582 way indicated above. 584 2.1.18.4. Delete BOM 586 Unicode character "BOM" is used as a start indication and sometimes 587 used as a filler or keep alive by transmission implementations. 588 These SHALL be deleted on reception. 590 2.1.18.5. Empty T140blocks 592 Empty T140blocks are included as fillers for unused redundancy levels 593 in the packets. They just do not provide any contents and do not 594 contribute to the received streams. 596 2.1.19. Performance considerations 598 This solution has good performance for up to three participants 599 simultaneously sending text. At higher numbers of participants 600 simultaneously sending text, a jerkiness is visible in the 601 presentation of text. With five participants simultaneously 602 transmitting text, the jerkiness is about 1400 ms. Evenso, the 603 transmission of text catches up, so there is no resulting total delay 604 introduced. The solution is therefore suitable for emergency service 605 use, relay service use, and small or well-managed larger multimedia 606 conferences. Only in large unmanaged conferences with a high number 607 of participants there may on very rare occasions appear situations 608 when many participants happen to send text simultaneously, resulting 609 in unpleasantly long switching times. It should be noted that it is 610 only the number of users sending text within the same moment that 611 causes jerkiness, not the total number of users with RTT capability. 613 2.1.20. Offer/answer considerations 615 A party which has negotiated the "rtt-mix-rtp-mixer" sdp media 616 attribute MUST populate the CSRC-list and format the packets 617 according to this section if it acts as an rtp-mixer and sends multi- 618 party text. 620 A party which has negotiated the the "rtt-mix-rtp-mixer" sdp media 621 attribute MUST interpret the contents of the "CC" field the CSRC-list 622 and the packets according to this section in received rtp packets in 623 the corresponding RTP stream. 625 A party performing as a mixer, which has not negotiated the "rtt-mix- 626 rtp-mixer" sdp media attribute, but negotiated a "text/red" or "text/ 627 t140" format in a session with a participant SHOULD, if nothing else 628 is specified for the application, format transmitted text to that 629 participant to be suitable to present on a multi-party unaware 630 endpoint as further specified in section Section 3.2. 632 A party not performing as a mixer MUST not include the CSRC list. 634 2.1.21. Security for session control and media 636 Security SHOULD be applied on both session control and media. In 637 applications where legacy endpoints without security may exist, a 638 negotiation between security and no security SHOULD be applied. If 639 no other security solution is mandated by the application, then RFC 640 8643 OSRTP[RFC8643] SHOULD be applied to negotiate SRTP media 641 security with DTLS. Most SDP examples below are for simplicity 642 expressed without the security additions. The principles (but not 643 all details) for applying DTLS-SRTP security is shown in a couple of 644 the following examples. 646 2.1.22. SDP offer/answer examples 648 This sections shows some examples of SDP for session negotiation of 649 the real-time text media in SIP sessions. Audio is usually provided 650 in the same session, and sometimes also video. The examples only 651 show the part of importance for the real-time text media. 653 Offer example for "text/red" format and multi-party support: 655 m=text 11000 RTP/AVP 100 98 656 a=rtpmap:98 t140/1000 657 a=rtpmap:100 red/1000 658 a=fmtp:100 98/98/98 659 a=rtt-mix-rtp-mixer 661 Answer example from a multi-party capable device 662 m=text 14000 RTP/AVP 100 98 663 a=rtpmap:98 t140/1000 664 a=rtpmap:100 red/1000 665 a=fmtp:100 98/98/98 666 a=rtt-mix-rtp-mixer 668 Offer example for "text/red" format including multi-party 669 and security: 670 a=fingerprint: SHA-1 \ 671 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB 672 m=text 11000 RTP/AVP 100 98 673 a=rtpmap:98 t140/1000 674 a=rtpmap:100 red/1000 675 a=fmtp:100 98/98/98 676 a=rtt-mix-rtp-mixer 678 The "Fingerprint" is sufficient to offer DTLS-SRTP, with the media 679 line still indicating RTP/AVP. 681 Answer example from a multi-party capable device with security 682 a=fingerprint: SHA-1 \ 683 FF:FF:FF:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB 684 m=text 16000 RTP/AVP 100 98 685 a=rtpmap:98 t140/1000 686 a=rtpmap:100 red/1000 687 a=fmtp:100 98/98/98 688 a=rtt-mix-rtp-mixer 690 With the "fingerprint" the device acknowledges use of SRTP/DTLS. 692 Answer example from a multi-party unaware device that also 693 does not support security: 695 m=text 12000 RTP/AVP 100 98 696 a=rtpmap:98 t140/1000 697 a=rtpmap:100 red/1000 698 a=fmtp:100 98/98/98 700 2.1.23. Packet sequence example from a source switch 702 This example shows a symbolic flow of packets from a mixer including 703 loss and recovery. The sequence includes a source switch. A and B 704 are sources of RTT. P indicates primary data. R1 is first redundant 705 generation data and R2 is second redundant generation data. A1, B1, 706 A2 etc are text chunks (T140blocks) received from the respective 707 sources. X indicates dropped packet between the mixer and a 708 receiver. 710 |----------------| 711 |Seq no 1 | 712 |CC=1 | 713 |CSRC list A | 714 |R2: A1 | 715 |R1: A2 | 716 |P: A3 | 717 |----------------| 719 Assuming that earlier packets ( with text A1 and A2) were received in 720 sequence, text A3 is received from packet 1 and assigned to reception 721 area A. The mixer is now assumed to have received text from source B 722 and need to prepare for sending that text. First it must send the 723 redundant generations of text A2 and A3. 725 |----------------| 726 |Seq no 2 | 727 |CC=1 | 728 |CSRC list A | 729 |R2 A2 | 730 |R1: A3 | 731 |P: Empty | 732 |----------------| 733 Nothing needs to be retrieved from this packet. 735 X----------------| 736 X Seq no 3 | 737 X CC=1 | 738 X CSRC list A | 739 X R2: A3 | 740 X R1: Empty | 741 X P: Empty | 742 X----------------| 743 Packet 3 is assumed to be dropped in network problems. It was the 744 last packet with contents from A before the source switch. 746 X----------------| 747 X Seq no 4 | 748 X CC=1 | 749 X CSRC list B | 750 X R2: Empty | 751 X R1: Empty | 752 X P2: B1 | 753 X----------------| 754 Packet 4 contains text from B, assumed dropped in network problems. 755 The mixer is assumed to have received text from A on turn to send. 756 Sending of text from B must therefore be temporarily ended by 757 sending redundancy twice. 759 X----------------| 760 X Seq no 5 | 761 X CC=1 | 762 X CSRC list B | 763 X R2: Empty | 764 X R1: B1 | 765 X P: Empty | 766 X----------------| 767 Packet 5 is assumed to be dropped in network problems 769 |----------------| 770 |Seq no 6 | 771 |CC=1 | 772 |CSRC list B | 773 | R2: B1 | 774 | R1: Empty | 775 | P: Empty | 776 |----------------| 778 Packet 6 is received. The latest received sequence number was 2. 779 Recovery is therefore tried for 3,4,5. There is no coverage for seq 780 no 3. But knowing that A1 must have been sent as R2 in packet 3, it 781 can be concluded that nothing was lost. 783 For seqno 4, text B1 is recovered from the second generation 784 redundancy and appended to the reception area of B. For seqno 5, 785 nothing needs to be recovered. No primary text is available in 786 packet 6. 788 After this sequence, A3 and B1 have been received. In this case no 789 text was lost. Even if also packet 2 was lost, it can be concluded 790 that no text was lost. 792 If also packets 1 and 2 were lost, there would be a need to create a 793 marker for possibly lost text (U'FFFD) [T140ad1], inserted generally 794 and possibly also in text sequences A and B. 796 2.1.24. Use with SIP centralized conferencing framework 798 The SIP conferencing framework, mainly specified in RFC 799 4353[RFC4353], RFC 4579[RFC4579] and RFC 4575[RFC4575] is suitable 800 for coordinating sessions including multi-party RTT. The RTT stream 801 between the mixer and a participant is one and the same during the 802 conference. Participants get announced by notifications when 803 participants are joining or leaving, and further user information may 804 be provided. The SSRC of the text to expect from joined users MAY be 805 included in a notification. The notifications MAY be used both for 806 security purposes and for translation to a label for presentation to 807 other users. 809 2.1.25. Conference control 811 In managed conferences, control of the real-time text media SHOULD be 812 provided in the same way as other for media, e.g. for muting and 813 unmuting by the direction attributes in SDP [RFC4566]. 815 Note that floor control functions may be of value for RTT users as 816 well as for users of other media in a conference. 818 2.1.26. Maximum character rate "CPS" 820 The default maximum rate of reception of "text/t140" real-time text 821 is in RFC 4103 [RFC4103] specified to be 30 characters per second. 822 The value MAY be modified in the CPS parameter of the FMTP attribute 823 in the media section for the "text/t140" media. A mixer combining 824 real-time text from a number of sources may occasionally have a 825 higher combined flow of text coming from the sources. Endpoints 826 SHOULD therefore specify a suitable higher value for the CPS 827 parameter, corresponding to its real reception capability. A value 828 for "CPS" of 90 is the default for the "text/t140" stream in the 829 "text/red" format when multi-party real-time text is negotiated. See 830 RFC 4103 [RFC4103] for the format and use of the CPS parameter. The 831 same rules apply for the multi-party case except for the default 832 value. 834 2.2. Mixing for multi-party unaware endpoints 836 A method is specified in this section for cases when the 837 participating endpoint does not implement any solution for multi- 838 party presentation of real-time text. The solution requires the 839 mixer to insert text dividers and readable labels and only send text 840 from one source at a time until a suitable point appears for source 841 change. This solution is a fallback method with functional 842 limitations that acts on the presentation level and is further 843 specified in Section 3.2. 845 3. Presentation level considerations 847 ITU-T T.140 [T140] provides the presentation level requirements for 848 the RFC 4103 [RFC4103] transport. T.140 [T140] has functions for 849 erasure and other formatting functions and has the following general 850 statement for the presentation: 852 "The display of text from the members of the conversation should be 853 arranged so that the text from each participant is clearly readable, 854 and its source and the relative timing of entered text is visualized 855 in the display. Mechanisms for looking back in the contents from the 856 current session should be provided. The text should be displayed as 857 soon as it is received." 859 Strict application of T.140 [T140] is of essence for the 860 interoperability of real-time text implementations and to fulfill the 861 intention that the session participants have the same information of 862 the text contents of the conversation without necessarily having the 863 exact same layout of the conversation. 865 T.140 [T140] specifies a set of presentation control codes to include 866 in the stream. Some of them are optional. Implementations MUST be 867 able to ignore optional control codes that they do not support. 869 There is no strict "message" concept in real-time text. Line 870 Separator SHALL be used as a separator allowing a part of received 871 text to be grouped in presentation. The characters "CRLF" may be 872 used by other implementations as replacement for Line Separator. The 873 "CRLF" combination SHALL be erased by just one erasing action, just 874 as the Line Separator. Presentation functions are allowed to group 875 text for presentation in smaller groups than the line separators 876 imply and present such groups with source indication together with 877 text groups from other sources (see the following presentation 878 examples). Erasure has no specific limit by any delimiter in the 879 text stream. 881 3.1. Presentation by multi-party aware endpoints 883 A multi-party aware receiving party, presenting real-time text MUST 884 separate text from different sources and present them in separate 885 presentation fields. The receiving party MAY separate presentation 886 of parts of text from a source in readable groups based on other 887 criteria than line separator and merge these groups in the 888 presentation area when it benefits the user to most easily find and 889 read text from the different participants. The criteria MAY e.g. be 890 a received comma, full stop, or other phrase delimiters, or a long 891 pause. 893 When text is received from multiple original sources simultaneously, 894 the presentation SHOULD provide a view where text is added in 895 multiple places simultaneously. 897 If the presentation presents text from different sources in one 898 common area, the presenting endpoint SHOULD insert text from the 899 local user ended at suitable points merged with received text to 900 indicate the relative timing for when the text groups were completed. 901 In this presentation mode, the receiving endpoint SHALL present the 902 source of the different groups of text. 904 A view of a three-party RTT call in chat style is shown in this 905 example . 907 _________________________________________________ 908 | |^| 909 |[Alice] Hi, Alice here. |-| 910 | | | 911 |[Bob] Bob as well. | | 912 | | | 913 |[Eve] Hi, this is Eve, calling from Paris. | | 914 | I thought you should be here. | | 915 | | | 916 |[Alice] I am coming on Thursday, my | | 917 | performance is not until Friday morning.| | 918 | | | 919 |[Bob] And I on Wednesday evening. | | 920 | | | 921 |[Alice] Can we meet on Thursday evening? | | 922 | | | 923 |[Eve] Yes, definitely. How about 7pm. | | 924 | at the entrance of the restaurant | | 925 | Le Lion Blanc? | | 926 |[Eve] we can have dinner and then take a walk |-| 927 |______________________________________________|v| 928 | But I need to be back to |^| 929 | the hotel by 11 because I need |-| 930 | | | 931 | I wou |-| 932 |______________________________________________|v| 933 | of course, I underst | 934 |________________________________________________| 936 Figure 3: Example of a three-party RTT call presented in chat style 937 seen at participant 'Alice's endpoint. 939 Other presentation styles than the chat style may be arranged. 941 This figure shows how a coordinated column view MAY be presented. 943 _____________________________________________________________________ 944 | Bob | Eve | Alice | 945 |____________________|______________________|_______________________| 946 | | |I will arrive by TGV. | 947 |My flight is to Orly| |Convenient to the main | 948 | |Hi all, can we plan |station. | 949 | |for the seminar? | | 950 |Eve, will you do | | | 951 |your presentation on| | | 952 |Friday? |Yes, Friday at 10. | | 953 |Fine, wo | |We need to meet befo | 954 |___________________________________________________________________| 955 Figure 4: An example of a coordinated column-view of a three-party 956 session with entries ordered vertically in approximate time-order. 958 3.2. Multi-party mixing for multi-party unaware endpoints 960 When the mixer has indicated multi-party capability by the "rtt-mix- 961 rtp-mixer" sdp attribute in an SDP negotiation, but the multi-party 962 capability negotiation fails with an endpoint, then the agreed "text/ 963 red" or "text/t140" format SHALL be used and the mixer SHOULD compose 964 a best-effort presentation of multi-party real-time text in one 965 stream intended to be presented by an endpoint with no multi-party 966 awareness. 968 This presentation format has functional limitations and SHOULD be 969 used only to enable participation in multi-party calls by legacy 970 deployed endpoints implementing only RFC 4103 without any multi-party 971 extensions specified in this document. 973 The principles and procedures below do not specify any new protocol 974 elements. They are instead composed from the information in ITU-T 975 T.140 [T140] and an ambition to provide a best effort presentation on 976 an endpoint which has functions only for two-party calls. 978 The mixer mixing for multi-party unaware endpoints SHALL compose a 979 simulated limited multi-party RTT view suitable for presentation in 980 one presentation area. The mixer SHALL group text in suitable groups 981 and prepare for presentation of them by inserting a new line between 982 them if the transmitted text did not already end with a new line. A 983 presentable label SHOULD be composed and sent for the source 984 initially in the session and after each source switch. With this 985 procedure the time for source switching is depending on the actions 986 of the users. In order to expedite source switch, a user can for 987 example end its turn with a new line. 989 3.2.1. Actions by the mixer at reception from the call participants 991 When text is received by the mixer from the different participants, 992 the mixer SHALL recover text from redundancy if any packets are lost. 993 The mark for lost text [T140ad1] SHOULD be inserted in the stream if 994 unrecoverable loss appears. Any Unicode "BOM" characters, possibly 995 used for keep-alive shall be deleted. The time of creation of text 996 (retrieved from the RTP timestamp) SHALL be stored together with the 997 received text from each source in queues for transmission to the 998 recipients. 1000 3.2.2. Actions by the mixer for transmission to the recipients 1002 The following procedure SHOULD be applied for each recipient of 1003 multi-part text from the mixer. 1005 The text for transmission SHOULD be formatted by the mixer for each 1006 receiving user for presentation in one single presentation area. 1007 Text received from a participant SHOULD NOT be included in 1008 transmission to that participant. When there is text available for 1009 transmission from the mixer to a receiving party from more than one 1010 participant, the mixer SHOULD switch between transmission of text 1011 from the different sources at suitable points in the transmitted 1012 stream. 1014 When switching source, the mixer SHOULD insert a line separator if 1015 the already transmitted text did not end with a new line (line 1016 separator or CRLF). A label SHOULD be composed from information in 1017 the CNAME and NAME fields in RTCP reports from the participant to 1018 have its text transmitted, or from other session information for that 1019 user. The label SHOULD be delimited by suitable characters (e.g. '[ 1020 ]') and transmitted. The CSRC SHOULD indicate the selected source. 1021 Then text from that selected participant SHOULD be transmitted until 1022 a new suitable point for switching source is reached. 1024 Integrity considerations SHALL be taken when composing the label. 1026 Seeking a suitable point for switching source SHOULD be done when 1027 there is older text waiting for transmission from any party than the 1028 age of the last transmitted text. Suitable points for switching are: 1030 * A completed phrase ended by comma 1032 * A completed sentence 1034 * A new line (line separator or CRLF) 1036 * A long pause (e.g. > 10 seconds) in received text from the 1037 currently transmitted source 1039 * If text from one participant has been transmitted with text from 1040 other sources waiting for transmission for a long time (e.g. > 1 1041 minute) and none of the other suitable points for switching has 1042 occurred, a source switch MAY be forced by the mixer at next word 1043 delimiter, and also if even a word delimiter does not occur within 1044 a time (e.g. 15 seconds) after the scan for word delimiter 1045 started. 1047 When switching source, the source which has the oldest text in queue 1048 SHOULD be selected to be transmitted. A character display count 1049 SHOULD be maintained for the currently transmitted source, starting 1050 at zero after the label is transmitted for the currently transmitted 1051 source. 1053 The status SHOULD be maintained for the latest control code for 1054 Select Graphic Rendition (SGR) from each source. If there is an SGR 1055 code stored as the status for the current source before the source 1056 switch is done, a reset of SGR shall be sent by the sequence SGR 0 1057 [009B 0000 006D] after the new line and before the new label during a 1058 source switch. See SGR below for an explanation. This transmission 1059 does not influence the display count. 1061 If there is an SGR code stored for the new source after the source 1062 switch, that SGR code SHOULD be transmitted to the recipient before 1063 the label. This transmission does not influence the display count. 1065 3.2.3. Actions on transmission of text 1067 Text from a source sent to the recipient SHOULD increase the display 1068 count by one per transmitted character. 1070 3.2.4. Actions on transmission of control codes 1072 The following control codes specified by T.140 require specific 1073 actions. They SHOULD cause specific considerations in the mixer. 1074 Note that the codes presented here are expressed in UCS-16, while 1075 transmission is made in UTF-8 transform of these codes. 1077 BEL 0007 Bell Alert in session, provides for alerting during an 1078 active session. The display count SHOULD not be altered. 1080 NEW LINE 2028 Line separator. Check and perform a source switch if 1081 appropriate. Increase display count by 1. 1083 CR LF 000D 000A A supported, but not preferred way of requesting a 1084 new line. Check and perform a source switch if appropriate. 1085 Increase display count by 1. 1087 INT ESC 0061 Interrupt (used to initiate mode negotiation 1088 procedure). The display count SHOULD not be altered. 1090 SGR 009B Ps 006D Select graphic rendition. Ps is rendition 1091 parameters specified in ISO 6429. The display count SHOULD not be 1092 altered. The SGR code SHOULD be stored for the current source. 1094 SOS 0098 Start of string, used as a general protocol element 1095 introducer, followed by a maximum 256 bytes string and the ST. 1096 The display count SHOULD not be altered. 1098 ST 009C String terminator, end of SOS string. The display count 1099 SHOULD not be altered. 1101 ESC 001B Escape - used in control strings. The display count SHOULD 1102 not be altered for the complete escape code. 1104 Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used 1105 for synchronization and keep-alive. SHOULD be deleted from 1106 incoming streams. Shall be sent first after session establishment 1107 to the recipient. The display count shall not be altered. 1109 Missing text mark (U+FFFD) "Replacement character", represented as a 1110 question mark in a rhombus, or if that is not feasible, replaced 1111 by an apostrophe ', marks place in stream of possible text loss. 1112 SHOULD be inserted by the reception procedure in case of 1113 unrecoverable loss of packets. The display count SHOULD be 1114 increased by one when sent as for any other character. 1116 SGR If a control code for selecting graphic rendition (SGR), other 1117 than reset of the graphic rendition (SGR 0) is sent to a 1118 recipient, that control code shall also be stored as status for 1119 the source in the storage for SGR status. If a reset graphic 1120 rendition (SGR 0) originated from a source is sent, then the SGR 1121 status storage for that source shall be cleared. The display 1122 count shall not be increased. 1124 BS (U+0008) Back Space, intended to erase the last entered character 1125 by a source. Erasure by backspace cannot always be performed as 1126 the erasing party intended. If an erasing action erases all text 1127 up to the end of the leading label after a source switch, then the 1128 mixer must not transmit more backspaces. Instead it is 1129 RECOMMENDED that a letter "X" is inserted in the text stream for 1130 each backspace as an indication of the intent to erase more. A 1131 new line is usually coded by a Line Separator, but the character 1132 combination "CRLF" MAY be used instead. Erasure of a new line is 1133 in both cases done by just one erasing action (Backspace). If the 1134 display count has a positive value it is decreased by one when the 1135 BS is sent. If the display count is at zero, it is not altered. 1137 3.2.5. Packet transmission 1139 A mixer transmitting to a multi-party unaware terminal SHOULD send 1140 primary data only from one source per packet. The SSRC SHOULD be the 1141 SSRC of the mixer. The CSRC list SHOULD contain one member and be 1142 the SSRC of the source of the primary data. 1144 3.2.6. Functional limitations 1146 When a multi-party unaware endpoint presents a conversation in one 1147 display area in a chat style, it inserts source indications for 1148 remote text and local user text as they are merged in completed text 1149 groups. When an endpoint using this layout receives and presents 1150 text mixed for multi-party unaware endpoints, there will be two 1151 levels of source indicators for the received text; one generated by 1152 the mixer and inserted in a label after each source switch, and 1153 another generated by the receiving endpoint and inserted after each 1154 switch between local and remote source in the presentation area. 1155 This will waste display space and look inconsistent to the reader. 1157 New text can be presented only from one source at a time. Switch of 1158 source to be presented takes place at suitable places in the text, 1159 such as end of phrase, end of sentence, line separator and 1160 inactivity. Therefore the time to switch to present waiting text 1161 from other sources may become long and will vary and depend on the 1162 actions of the currently presented source. 1164 Erasure can only be done up to the latest source switch. If a user 1165 tries to erase more text, the erasing actions will be presented as 1166 letter X after the label. 1168 Text loss because of network errors may hit the label between entries 1169 from different parties, causing risk for misunderstanding from which 1170 source a piece of text is. 1172 These facts makes it strongly RECOMMENDED to implement multi-party 1173 awareness in RTT endpoints. The use of the mixing method for multi- 1174 party-unaware endpoints should be left for use with endpoints which 1175 are impossible to upgrade to become multi-party aware. 1177 3.2.7. Example views of presentation on multi-party unaware endpoints 1179 The following pictures are examples of the view on a participant's 1180 display for the multi-party-unaware case. 1182 _________________________________________________ 1183 | Conference | Alice | 1184 |________________________|_________________________| 1185 | |I will arrive by TGV. | 1186 |[Bob]:My flight is to |Convenient to the main | 1187 |Orly. |station. | 1188 |[Eve]:Hi all, can we | | 1189 |plan for the seminar. | | 1190 | | | 1191 |[Bob]:Eve, will you do | | 1192 |your presentation on | | 1193 |Friday? | | 1194 |[Eve]:Yes, Friday at 10.| | 1195 |[Bob]: Fine, wo |We need to meet befo | 1196 |________________________|_________________________| 1198 Figure 5: Alice who has a conference-unaware client is receiving the 1199 multi-party real-time text in a single-stream. This figure shows how 1200 a coordinated column view MAY be presented on Alice's device. 1202 _________________________________________________ 1203 | |^| 1204 |[Alice] Hi, Alice here. |-| 1205 | | | 1206 |[mix][Bob] Bob as well. | | 1207 | | | 1208 |[Eve] Hi, this is Eve, calling from Paris | | 1209 | I thought you should be here. | | 1210 | | | 1211 |[Alice] I am coming on Thursday, my | | 1212 | performance is not until Friday morning.| | 1213 | | | 1214 |[mix][Bob] And I on Wednesday evening. | | 1215 | | | 1216 |[Eve] we can have dinner and then walk | | 1217 | | | 1218 |[Eve] But I need to be back to | | 1219 | the hotel by 11 because I need | | 1220 | |-| 1221 |______________________________________________|v| 1222 | of course, I underst | 1223 |________________________________________________| 1225 Figure 6: An example of a view of the multi-party unaware 1226 presentation in chat style. Alice is the local user. 1228 4. Gateway Considerations 1229 4.1. Gateway considerations with Textphones (e.g. TTYs). 1231 Multi-party RTT sessions may involve gateways of different kinds. 1232 Gateways involved in setting up sessions SHALL correctly reflect the 1233 multi-party capability or unawareness of the combination of the 1234 gateway and the remote endpoint beyond the gateway. 1236 One case that may occur is a gateway to PSTN for communication with 1237 textphones (e.g. TTYs). Textphones are limited devices with no 1238 multi-party awareness, and it SHOULD therefore be suitable for the 1239 gateway to not indicate multi-party awareness for that case. Another 1240 solution is that the gateway indicates multi-party capability towards 1241 the mixer, and includes the multi-party mixer function for multi- 1242 party unaware endpoints itself. This solution makes it possible to 1243 make adaptations for the functional limitations of the textphone 1244 (TTY). 1246 More information on gateways to textphones (TTYs) is found in RFC 1247 5194[RFC5194] 1249 4.2. Gateway considerations with WebRTC. 1251 Gateway operation to real-time text in WebRTC may also be required. 1252 In WebRTC, RTT is specified in 1253 [I-D.ietf-mmusic-t140-usage-data-channel]. 1255 A multi-party bridge may have functionality for communicating by RTT 1256 both in RTP streams with RTT and WebRTC t140 data channels. Other 1257 configurations may consist of a multi-party bridge with either 1258 technology for RTT transport and a separate gateway for conversion of 1259 the text communication streams between RTP and t140 data channel. 1261 In WebRTC, it is assumed that for a multi-party session, one t140 1262 data channel is established for each source from a gateway or bridge 1263 to each participant. Each participant also has a data channel with 1264 two-way connection with the gateway or bridge. 1266 The t140 channel used both ways is for text from the WebRTC user and 1267 from the bridge or gateway itself to the WebRTC user. The label 1268 parameter of this t140 channel is used as NAME field in RTCP to 1269 participants on the RTP side. The other t140 channels are only for 1270 text from other participants to the WebRTC user. 1272 When a new participant has entered the session with RTP transport of 1273 rtt, a new t140 channel SHOULD be established to WebRTC users with 1274 the label parameter composed from the NAME field in RTCP on the RTP 1275 side. 1277 When a new participant has entered the multi-party session with RTT 1278 transport in a WebRTC t140 data channel, the new participant SHOULD 1279 be announced by a notification to RTP users. The label parameter 1280 from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP 1281 side, or other available session information. 1283 5. Updates to RFC 4103 1285 This document updates RFC 4103[RFC4103] by introducing an sdp media 1286 attribute "rtt-mix-rtp-mixer" for negotiation of multi-party mixing 1287 capability with the [RFC4103] format, and by specifying the rules for 1288 packets when multi-party capability is negotiated and in use. 1290 6. Congestion considerations 1292 The congestion considerations and recommended actions from RFC 4103 1293 [RFC4103] are valid also in multi-party situations. 1295 The first action in case of congestion SHOULD be to temporarily 1296 increase the transmission interval up to two seconds. 1298 If the unlikely situation appears that more than 20 participants in a 1299 conference send text simultaneously, it will take more than 7 seconds 1300 between presentation of text from each of these participants. More 1301 time than that can cause confusion in the session. It is therefore 1302 RECOMMENDED that the mixer discards such text in excess inserts a 1303 general indication of possible text loss [T140ad1] in the session. 1304 If the main text contributor is indicated in any way, the mixer MAY 1305 avoid deleting text from that participant. 1307 7. Acknowledgements 1309 James Hamlin for format and performance aspects. 1311 8. IANA Considerations 1313 8.1. Registration of the "rtt-mix-rtp-mixer" sdp media attribute 1315 [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the 1316 RFC number of this document.] 1318 IANA is asked to register the new sdp attribute "rtt-mix-rtp-mixer". 1320 Contact name: IESG 1322 Contact email: iesg@ietf.org 1324 Attribute name: rtt-mix-rtp-mixer 1325 Attribute syntax: a=rtt-mix-rtp-mixer 1327 Attribute semantics: See RFCXXXX Section 2.1.1 1329 Attribute value: none 1331 Usage level: media 1333 Purpose: Indicate support by mixer and endpoint of multi-party 1334 mixing for real-time text transmission, using a common RTP-stream 1335 for transmission of text from a number of sources mixed with one 1336 source at a time and the source indicated in a single CSRC-list 1337 member. 1339 Charset Dependent: no 1341 O/A procedure: See RFCXXXX Section 2.1.20 1343 Mux Category: normal 1345 Reference: RFCXXXX 1347 9. Security Considerations 1349 The RTP-mixer model requires the mixer to be allowed to decrypt, pack 1350 and encrypt secured text from the conference participants. Therefore 1351 the mixer needs to be trusted. This is similar to the situation for 1352 central mixers of audio and video. 1354 The requirement to transfer information about the user in RTCP 1355 reports in SDES, CNAME and NAME fields, and in conference 1356 notifications, for creation of labels may have privacy concerns as 1357 already stated in RFC 3550 [RFC3550], and may be restricted of 1358 privacy reasons. The receiving user will then get a more symbolic 1359 label for the source. 1361 10. Change history 1363 10.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 1365 Deleted the method requiring a new packet format "text/rex" because 1366 of the longer standardization and implementation period it needs. 1368 Focus on use of RFC 4103 text/red format with shorter transmission 1369 interval, and source indicated in CSRC. 1371 10.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 1373 Added a method based on the "text/red" format and single source per 1374 packet, negotiated by the "rtt-mix-rtp-mixer" sdp attribute. 1376 Added reasoning and recommendation about indication of loss. 1378 The highest number of sources in one packet is 15, not 16. Changed. 1380 Added in information on update to RFC 4103 that RFC 4103 explicitly 1381 allows addition of FEC method. The redundancy is a kind of forward 1382 error correction.. 1384 10.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 1386 Improved definitions list format. 1388 The format of the media subtype parameters is made to match the 1389 requirements. 1391 The mapping of media subtype parameters to sdp is included. 1393 The CPS parameter belongs to the t140 subtype and does not need to be 1394 registered here. 1396 10.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 1398 nomenclature and editorial improvements 1400 "this document" used consistently to refer to this document. 1402 10.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 1404 'Redundancy header' renamed to 'data header'. 1406 More clarifications added. 1408 Language and figure number corrections. 1410 10.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 1412 Mention possible need to mute and raise hands as for other media. 1413 ---done ---- 1415 Make sure that use in two-party calls is also possible and explained. 1416 - may need more wording - 1418 Clarify the RTT is often used together with other media. --done-- 1419 Tell that text mixing is N-1. A users own text is not received in 1420 the mix. -done- 1422 In 3. correct the interval to: A "text/rex" transmitter SHOULD send 1423 packets distributed in time as long as there is something (new or 1424 redundant T140blocks) to transmit. The maximum transmission interval 1425 SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 1426 receiver as soon as new text to that receiver is available, as long 1427 as the time after the latest sent packet to the same receiver is more 1428 than 150 ms, and also the maximum character rate to the receiver is 1429 not exceeded. The intention is to keep the latency low while keeping 1430 a good protection against text loss in bursty packet loss conditions. 1431 -done- 1433 In 1.3 say that the format is used both ways. -done- 1435 In 13.1 change presentation area to presentation field so that reader 1436 does not think it shall be totally separated. -done- 1438 In Performance and intro, tell the performance in number of 1439 simultaneous sending users and introduced delay 16, 150 vs 1440 requirements 5 vs 500. -done -- 1442 Clarify redundancy level per connection. -done- 1444 Timestamp also for the last data header. To make it possible for all 1445 text to have time offset as for transmission from the source. Make 1446 that header equal to the others. -done- 1448 Mixer always use the CSRC list, even for its own BOM. -done- 1450 Combine all talk about transmission interval (300 ms vs when text has 1451 arrived) in section 3 in one paragraph or close to each other. -done- 1453 Documents the goal of good performance with low delay for 5 1454 simultaneous typers in the introduction. -done- 1456 Describe better that only primary text shall be sent on to receivers. 1457 Redundancy and loss must be resolved by the mixer. -done- 1459 10.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 1461 SDP and better description and visibility of security by OSRTP RFC 1462 8634 needed. 1464 The description of gatewaying to WebRTC extended. 1466 The description of the data header in the packet is improved. 1468 10.8. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 1470 2,5,6 More efficient format "text/rex" introduced and attribute 1471 a=rtt-mix deleted. 1473 3. Brief about use of OSRTP for security included- More needed. 1475 4. Brief motivation for the solution and why not rtp-translator is 1476 used added to intro. 1478 7. More limitations for the multi-party unaware mixing method 1479 inserted. 1481 8. Updates to RFC 4102 and 4103 more clearly expressed. 1483 9. Gateway to WebRTC started. More needed. 1485 10.9. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 to 1486 draft-ietf-avtcore-multi-party-rtt-mix-00 1488 Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00 1490 Replaced CDATA in IANA registration table with better coding. 1492 Converted to xml2rfc version 3. 1494 10.10. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 1495 to -03 1497 Changed company and e-mail of the author. 1499 Changed title to "RTP-mixer formatting of multi-party Real-time text" 1500 to better match contents. 1502 Check and modification where needed of use of RFC 2119 words SHALL 1503 etc. 1505 More about the CC value in sections on transmitters and receivers so 1506 that 1-to-1 sessions do not use the mixer format. 1508 Enhanced section on presentation for multi-party-unaware endpoints 1510 A paragraph recommending CPS=150 inserted in the performance section. 1512 10.11. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 1513 to -02 1515 In Abstract and 1. Introduction: Introduced wording about regulatory 1516 requirements. 1518 In section 5: The transmission interval is decreased to 100 ms when 1519 there is text from more than one source to transmit. 1521 In section 11 about SDP negotiation, a SHOULD-requirement is 1522 introduced that the mixer should make a mix for multi-party unaware 1523 endpoints if the negotiation is not successful. And a reference to a 1524 later chapter about it. 1526 The presentation considerations chapter 14 is extended with more 1527 information about presentation on multi-party aware endpoints, and a 1528 new section on the multi-party unaware mixing with low functionality 1529 but SHOULD a be implemented in mixers. Presentation examples are 1530 added. 1532 A short chapter 15 on gateway considerations is introduced. 1534 Clarification about the text/t140 format included in chapter 10. 1536 This sentence added to the chapter 10 about use without redundancy. 1537 "The text/red format SHOULD be used unless some other protection 1538 against packet loss is utilized, for example a reliable network or 1539 transport." 1541 Note about deviation from RFC 2198 added in chapter 4. 1543 In chapter 9. "Use with SIP centralized conferencing framework" the 1544 following note is inserted: Note: The CSRC-list in an RTP packet only 1545 includes participants who's text is included in one or more text 1546 blocks. It is not the same as the list of participants in a 1547 conference. With audio and video media, the CSRC-list would often 1548 contain all participants who are not muted whereas text participants 1549 that don't type are completely silent and so don't show up in RTP 1550 packet CSRC-lists. 1552 10.12. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 1553 to -01 1555 Editorial cleanup. 1557 Changed capability indication from fmtp-parameter to SDP attribute 1558 "rtt-mix". 1560 Swapped order of redundancy elements in the example to match reality. 1562 Increased the SDP negotiation section 1564 11. References 1566 11.1. Normative References 1568 [I-D.ietf-mmusic-t140-usage-data-channel] 1569 Holmberg, C. and G. Hellstrom, "T.140 Real-time Text 1570 Conversation over WebRTC Data Channels", Work in Progress, 1571 Internet-Draft, draft-ietf-mmusic-t140-usage-data-channel- 1572 14, 10 April 2020, . 1575 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1576 Requirement Levels", BCP 14, RFC 2119, 1577 DOI 10.17487/RFC2119, March 1997, 1578 . 1580 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1581 Jacobson, "RTP: A Transport Protocol for Real-Time 1582 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1583 July 2003, . 1585 [RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type", 1586 RFC 4102, DOI 10.17487/RFC4102, June 2005, 1587 . 1589 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 1590 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 1591 . 1593 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1594 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 1595 July 2006, . 1597 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1598 Security (DTLS) Extension to Establish Keys for the Secure 1599 Real-time Transport Protocol (SRTP)", RFC 5764, 1600 DOI 10.17487/RFC5764, May 2010, 1601 . 1603 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 1604 Keeping Alive the NAT Mappings Associated with RTP / RTP 1605 Control Protocol (RTCP) Flows", RFC 6263, 1606 DOI 10.17487/RFC6263, June 2011, 1607 . 1609 [RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. 1610 Stach, "An Opportunistic Approach for Secure Real-time 1611 Transport Protocol (OSRTP)", RFC 8643, 1612 DOI 10.17487/RFC8643, August 2019, 1613 . 1615 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 1616 multimedia application text conversation", February 1998, 1617 . 1619 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 1620 Protocol for multimedia application text conversation", 1621 February 2000, 1622 . 1624 11.2. Informative References 1626 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 1627 Session Initiation Protocol (SIP)", RFC 4353, 1628 DOI 10.17487/RFC4353, February 2006, 1629 . 1631 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 1632 Session Initiation Protocol (SIP) Event Package for 1633 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 1634 2006, . 1636 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 1637 (SIP) Call Control - Conferencing for User Agents", 1638 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 1639 . 1641 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 1642 Time Text over IP Using the Session Initiation Protocol 1643 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 1644 . 1646 Author's Address 1648 Gunnar Hellstrom 1649 Gunnar Hellstrom Accessible Communication 1650 Esplanaden 30 1651 SE-13670 Vendelso 1652 Sweden 1654 Email: gunnar.hellstrom@ghaccess.se