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Miscellaneous warnings: ---------------------------------------------------------------------------- == The copyright year in the IETF Trust and authors Copyright Line does not match the current year == Line 704 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 (13 October 2020) is 1290 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 1236, 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) 13 October 2020 5 Intended status: Standards Track 6 Expires: 16 April 2021 8 RTP-mixer formatting of multi-party Real-time text 9 draft-ietf-avtcore-multi-party-rtt-mix-09 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-mixer". 36 The document updates RFC 4103[RFC4103] 38 A specifications of how a mixer can format text for the case when the 39 endpoint is not multi-party aware is also provided. 41 Status of This Memo 43 This Internet-Draft is submitted in full conformance with the 44 provisions of BCP 78 and BCP 79. 46 Internet-Drafts are working documents of the Internet Engineering 47 Task Force (IETF). Note that other groups may also distribute 48 working documents as Internet-Drafts. The list of current Internet- 49 Drafts is at https://datatracker.ietf.org/drafts/current/. 51 Internet-Drafts are draft documents valid for a maximum of six months 52 and may be updated, replaced, or obsoleted by other documents at any 53 time. It is inappropriate to use Internet-Drafts as reference 54 material or to cite them other than as "work in progress." 56 This Internet-Draft will expire on 16 April 2021. 58 Copyright Notice 60 Copyright (c) 2020 IETF Trust and the persons identified as the 61 document authors. All rights reserved. 63 This document is subject to BCP 78 and the IETF Trust's Legal 64 Provisions Relating to IETF Documents (https://trustee.ietf.org/ 65 license-info) in effect on the date of publication of this document. 66 Please review these documents carefully, as they describe your rights 67 and restrictions with respect to this document. Code Components 68 extracted from this document must include Simplified BSD License text 69 as described in Section 4.e of the Trust Legal Provisions and are 70 provided without warranty as described in the Simplified BSD License. 72 Table of Contents 74 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 75 1.1. Selected solution and considered alternative . . . . . . 5 76 1.2. Nomenclature . . . . . . . . . . . . . . . . . . . . . . 7 77 1.3. Intended application . . . . . . . . . . . . . . . . . . 7 78 2. Specified solutions . . . . . . . . . . . . . . . . . . . . . 8 79 2.1. Negotiated use of the RFC 4103 format for multi-party in a 80 single RTP stream . . . . . . . . . . . . . . . . . . . . 8 81 2.2. Mixing for multi-party unaware endpoints . . . . . . . . 8 82 3. Details for the multi-party aware mixing case . . . . . . . . 8 83 3.1. Offer/answer considerations . . . . . . . . . . . . . . . 9 84 3.2. Use of fields in the RTP packets . . . . . . . . . . . . 9 85 3.3. Initial transmission of a BOM character . . . . . . . . . 10 86 3.4. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 10 87 3.5. Transmission interval . . . . . . . . . . . . . . . . . . 10 88 3.6. Only one source per packet . . . . . . . . . . . . . . . 10 89 3.7. Do not send received text to the originating source . . . 10 90 3.8. Clean incoming text . . . . . . . . . . . . . . . . . . . 10 91 3.9. Redundancy . . . . . . . . . . . . . . . . . . . . . . . 11 92 3.10. Text placement in packets . . . . . . . . . . . . . . . . 11 93 3.11. Source switching . . . . . . . . . . . . . . . . . . . . 11 94 3.12. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 12 95 3.13. Creation of the redundancy . . . . . . . . . . . . . . . 12 96 3.14. Timer offset fields . . . . . . . . . . . . . . . . . . . 12 97 3.15. Other RTP header fields . . . . . . . . . . . . . . . . . 12 98 3.16. Pause in transmission . . . . . . . . . . . . . . . . . . 13 99 3.17. RTCP considerations . . . . . . . . . . . . . . . . . . . 13 100 3.18. Reception of multi-party contents . . . . . . . . . . . . 13 101 3.19. Performance considerations . . . . . . . . . . . . . . . 15 102 3.20. Security for session control and media . . . . . . . . . 15 103 3.21. SDP offer/answer examples . . . . . . . . . . . . . . . . 15 104 3.22. Packet sequence example from a source switch . . . . . . 17 105 3.23. Maximum character rate "CPS" . . . . . . . . . . . . . . 19 106 4. Presentation level considerations . . . . . . . . . . . . . . 19 107 4.1. Presentation by multi-party aware endpoints . . . . . . . 20 108 4.2. Multi-party mixing for multi-party unaware endpoints . . 22 109 5. Relation to Conference Control . . . . . . . . . . . . . . . 27 110 5.1. Use with SIP centralized conferencing framework . . . . . 28 111 5.2. Conference control . . . . . . . . . . . . . . . . . . . 28 112 6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 28 113 6.1. Gateway considerations with Textphones (e.g. TTYs). . . 28 114 6.2. Gateway considerations with WebRTC. . . . . . . . . . . . 29 115 7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 29 116 8. Congestion considerations . . . . . . . . . . . . . . . . . . 29 117 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 30 118 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 30 119 10.1. Registration of the "rtt-mixer" sdp media attribute . . 30 120 11. Security Considerations . . . . . . . . . . . . . . . . . . . 31 121 12. Change history . . . . . . . . . . . . . . . . . . . . . . . 31 122 12.1. Changes included in 123 draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 31 124 12.2. Changes included in 125 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 31 126 12.3. Changes included in 127 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 31 128 12.4. Changes included in 129 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 32 130 12.5. Changes included in 131 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 32 132 12.6. Changes included in 133 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 32 134 12.7. Changes included in 135 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 32 136 12.8. Changes included in 137 draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 33 138 12.9. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 33 139 12.10. Changes from 140 draft-hellstrom-avtcore-multi-party-rtt-source-03 to 141 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 34 143 12.11. Changes from 144 draft-hellstrom-avtcore-multi-party-rtt-source-02 to 145 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 34 146 12.12. Changes from 147 draft-hellstrom-avtcore-multi-party-rtt-source-01 to 148 -02 . . . . . . . . . . . . . . . . . . . . . . . . . . 34 149 12.13. Changes from 150 draft-hellstrom-avtcore-multi-party-rtt-source-00 to 151 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 35 152 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 35 153 13.1. Normative References . . . . . . . . . . . . . . . . . . 35 154 13.2. Informative References . . . . . . . . . . . . . . . . . 37 155 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 37 157 1. Introduction 159 RFC 4103[RFC4103] specifies use of RFC 3550 RTP [RFC3550] for 160 transmission of real-time text (RTT) and the "text/t140" format. It 161 also specifies a redundancy format "text/red" for increased 162 robustness. RFC 4102 [RFC4102] registers the "text/red" format. 163 Regional regulatory requirements specify provision of real-time text 164 in multi-party calls. 166 Real-time text is usually provided together with audio and sometimes 167 with video in conversational sessions. 169 The redundancy scheme of RFC 4103 [RFC4103] enables efficient 170 transmission of redundant text in packets together with new text. 171 However the redundancy header format has no source indicators for the 172 redundant transmissions. An assumption has to be made that the 173 redundant parts in a packet are from the same source as the new text. 174 The recommended transmission is one new and two redundant generations 175 of text (T140blocks) in each packet and the recommended transmission 176 interval is 300 ms. 178 A mixer, selecting between text input from different sources and 179 transmitting it in a common stream needs to make sure that the 180 receiver can assign the received text to the proper sources for 181 presentation. Therefore, using RFC 4103 without any extra rule for 182 source identification, the mixer needs to stop sending new text from 183 one source and then make sure that all text sent so far has been sent 184 with all intended redundancy levels (usually two) before switching to 185 another source. That causes the long time of one second to switch 186 between transmission of text from one source to text from another 187 source when using the default transmission interval 300 ms. Both the 188 total throughput and the switching performance in the mixer would be 189 too low for most applications. However by shortening the 190 transmission interval to 100 ms, good performance is achieved for up 191 to 3 simultaneously sending sources and usable performance for up to 192 5 simultaneously sending sources. Capability to use this method is 193 indicated by an sdp media attribute "rtt-mixer". 195 A negotiation mechanism can therefore be based on selection of the 196 "text/red" with media attribute "rtt-mixer" for verification that the 197 parties are able to handle a multi-party coded stream and agreeing on 198 using that method. 200 A fall-back mixing procedure is specified for cases when the 201 negotiation results in "text/red" without the "rtt-mixer" attribute 202 being the only common format for real-time text. 204 The document updates RFC 4103[RFC4103] by introducing an attribute 205 for indicating capability for the multi-party mixing case and rules 206 for source indications and source switching. 208 1.1. Selected solution and considered alternative 210 A number of alternatives were considered when searching an efficient 211 and easily implemented multi-party method for real-time text. This 212 section explains a few of them briefly. 214 One RTP stream per source, sent in the same RTP session with 215 "text/red" format. 216 From some points of view, use of multiple RTP streams, one for 217 each source, sent in the same RTP session, called the RTP 218 translator model in [RFC3550], would be efficient, and use exactly 219 the same packet format as [RFC4103], the same payload type and a 220 simple SDP declaration. However, there is currently lack of 221 support for multi-stream RTP in certain implementation 222 technologies. This fact made it not included in this document. 224 The "text/red" format in RFC 4103 with shorter transmission 225 interval, and indicating source in CSRC. 226 The "text/red" format with "text/t140" payload in a single RTP 227 stream can be sent with 100 ms packet intervals instead of the 228 regular 300 ms. The source is indicated in the CSRC field. 229 Source switching can then be done every 300 ms while simultaneous 230 transmission occurs. With two participants sending text 231 simultaneously, the switching and transmission performance is 232 good. With three simultaneously sending participants, there will 233 be a noticable jerkiness in text presentation. The jerkiness will 234 be more expressed the more participants who send text 235 simultaneously. With five sending participants, the jerkiness 236 will be about 1400 ms. Text sent from a source at the end of the 237 period its text is sent by the mixer will have close to zero extra 238 delay. Recent text will be presented with no or low delay. The 239 1400 ms jerkiness will be noticable and slightly unpleasant, but 240 corresponds in time to what typing humans often cause by 241 hesitation or changing position while typing. A benefit of this 242 method is that no new packet format needs to be introduced and 243 implemented. Since simultaneous typing by more than two parties 244 is rare, and in most applications also more than three parties in 245 a call is rare, this method can be used successfully without its 246 limitations becoming annoying. Negotiation is based on a new sdp 247 media attribute "rtt-mixer". This method is selected to be the 248 main one specified in this document. 250 A new "text" media subtype with up to 15 sources in each packet. 251 The mechanism makes use of the RTP mixer model specified in 252 RFC3550[RFC3550]. Text from up to 15 sources can be included in 253 each packet. Packets are normally sent every 300 ms. The mean 254 delay will be 150 ms. The sources are indicated in strict order 255 in the CSRC list of the RTP packets. A new redundancy packet 256 format is specified. This method would result in good 257 performance, but would require standardisation and implementation 258 of new releases in the target technologies that would take more 259 time than desirable to complete. It was therefore not selected to 260 be included in this document. 262 The presentation planned by the mixer for multi-party unaware 263 endpoints. 264 It is desirable to have a method that does not require any 265 modifications in existing user devices implementing RFC 4103 for 266 RTT without explicit support of multi-party sessions. This is 267 possible by having the mixer insert a new line and a text 268 formatted source label before each switch of text source in the 269 stream. Switch of source can only be done in places in the text 270 where it does not disturb the perception of the contents. Text 271 from only one source can be presented in real time at a time. The 272 delay will therefore be varying. The method has also other 273 limitations, but is included in this document as a fallback 274 method. In calls where parties take turns properly by ending 275 their entries with a new line, the limitations will have limited 276 influence on the user experience. while only two parties send 277 text, these two will see the text in real time with no delay. 278 This method is specified as a fallback method in this document. 280 RTT transport in WebRTC 281 Transport of real-time text in the WebRTC technology is specified 282 to use the WebRTC data channel in 283 [I-D.ietf-mmusic-t140-usage-data-channel]. That spcification 284 contains a section briefly describing its use in multi-party 285 sessions. The focus of this document is RTP transport. 286 Therefore, even if the WebRTC transport provides good multi-party 287 performance, it is just mentioned in this document in relation to 288 providing gateways with multi-party capabilities between RTP and 289 WebRTC technologies. 291 1.2. Nomenclature 293 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 294 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 295 document are to be interpreted as described in [RFC2119]. 297 The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP- 298 mixer, RTP-translator are explained in [RFC3550] 300 The term "T140block" is defined in RFC 4103 [RFC4103] to contain one 301 or more T.140 code elements. 303 "TTY" stands for a text telephone type used in North America. 305 "WebRTC" stands for web based communication specified by W3C and 306 IETF. 308 "DTLS-SRTP" stnds for security specified in RFC 5764 [RFC5764]. 310 1.3. Intended application 312 The method for multi-party real-time text specified in this document 313 is primarily intended for use in transmission between mixers and 314 endpoints in centralised mixing configurations. It is also 315 applicable between mixers. An often mentioned application is for 316 emergency service calls with real-time text and voice, where a 317 calltaker want to make an attended handover of a call to another 318 agent, and stay observing the session. Multimedia conference 319 sessions with support for participants to contribute in text is 320 another application. Conferences with central support for speech-to- 321 text conversion is yet another mentioned application. 323 In all these applications, normally only one participant at a time 324 will send long text utterances. In some cases, one other participant 325 will occasionally contribute with a longer comment simultaneously. 326 That may also happen in some rare cases when text is interpreted to 327 text in another language in a conference. Apart from these cases, 328 other participants are only expected to contribute with very brief 329 utterings while others are sending text. 331 Text is supposed to be human generated, by some text input means, 332 such as typing on a keyboard or using speech-to-text technology. 333 Occasional small cut-and-paste operations may appear even if that is 334 not the initial purpose of real-time text. 336 The real-time characteristics of real-time text is essential for the 337 participants to be able to contribute to a conversation. If the text 338 is too much delayed from typing a letter to its presentation, then, 339 in some conference situations, the opportunity to comment will be 340 gone and someone else will grab the turn. A delay of more than one 341 second in such situations is an obstacle for good conversation. 343 2. Specified solutions 345 2.1. Negotiated use of the RFC 4103 format for multi-party in a single 346 RTP stream 348 The main purpose of this document is to specify the use of the 349 current format for real-time text in [RFC4103] for true multi-party 350 real-time text. It is an update of RFC 4103 by a clarification on 351 one way to use it in the multi-party situation. It is done by 352 completing a negotiation for this kind of multi-party capability and 353 by indicating source in the CSRC element in the RTP packets. 355 The detailed procedures for the multi-party capable case are 356 specified in Section 3 358 Please use [RFC4103] as reference when reading the specification. 360 2.2. Mixing for multi-party unaware endpoints 362 A method is also specified in this document for cases when the 363 endpoint participating in a multi-party call does not implement any 364 solution for multi-party presentation of real-time text. The 365 solution requires the mixer to insert text dividers and readable 366 labels and only send text from one source at a time until a suitable 367 point appears for source change. This solution is a fallback method 368 with functional limitations that acts on the presentation level. 370 A party performing as a mixer, which has not negotiated the "rtt- 371 mixer" sdp media attribute, but negotiated a "text/red" or "text/ 372 t140" format in a session with a participant SHOULD, if nothing else 373 is specified for the application, format transmitted text to that 374 participant to be suitable to present on a multi-party unaware 375 endpoint as further specified in Section 4.2. 377 3. Details for the multi-party aware mixing case 378 3.1. Offer/answer considerations 380 RFC 4103[RFC4103] specifies use of RFC 3550 RTP[RFC3550], and a 381 redundancy format "text/red" for increased robustness of real-time 382 text transmission. This document updates RFC 4103[RFC4103] by 383 introducing a capability negotiation for handling multi-party real- 384 time text and a way to indicate the source of thansmitted text. The 385 capability negotiation is based on use of the sdp media attribute 386 "rtt-mixer". 388 The syntax is as follows: 389 "a=rtt-mixer" 391 A transmitting party SHALL send text according to the multi-party 392 format only when the negotiation for this method was successful and 393 when the CC field in the RTP packet is set to 1. In all other cases, 394 the packets SHALL be populated and interpreted as for a two-party 395 session. 397 A party which has negotiated the "rtt-mixer" sdp media attribute MUST 398 populate the CSRC-list and format the packets according to Section 3 399 if it acts as an rtp-mixer and sends multi-party text. 401 A party which has negotiated the "rtt-mixer" sdp media attribute MUST 402 interpret the contents of the "CC" field the CSRC-list and the 403 packets according to Section 3 in received rtp packets in the 404 corresponding RTP stream. 406 A party not performing as a mixer MUST not include the CSRC list. 408 3.2. Use of fields in the RTP packets 410 The CC field SHALL show the number of members in the CSRC list, which 411 SHALL be one (1) in transmissions from a mixer involved in a multi- 412 party session, and otherwise 0. 414 When transmitted from a mixer during a multi-party session, a CSRC 415 list SHALL be included in the packet. The single member in the CSRC- 416 list SHALL contain the SSRC of the source of the T140blocks in the 417 packet. When redundancy is used, the recommended level of redundancy 418 is to use one primary and two redundant generations of T140blocks. 419 In some cases, a primary or redundant T140block is empty, but is 420 still represented by a member in the redundancy header. 422 From other aspects, the contents of the RTP packets are equal to what 423 is specified in [RFC4103]. 425 3.3. Initial transmission of a BOM character 427 As soon as a participant is known to participate in a session and 428 being available for text reception, a Unicode BOM character SHALL be 429 sent to it according to the procedures in this section. If the 430 transmitter is a mixer, then the source of this character SHALL be 431 indicated to be the mixer itself. 433 Note that the BOM character SHALL be transmitted with the same 434 redundancy procedures as any other text. 436 3.4. Keep-alive 438 After that, the transmitter SHALL send keep-alive traffic to the 439 receivers at regular intervals when no other traffic has occurred 440 during that interval if that is decided for the actual connection. 441 Recommendations for keep-alive can be found in [RFC6263]. 443 3.5. Transmission interval 445 A "text/red" transmitter in a mixer SHOULD send packets distributed 446 in time as long as there is something (new or redundant T140blocks) 447 to transmit. The maximum transmission interval SHOULD then be 300 448 ms. It is RECOMMENDED to send next packet to a receiver as soon as 449 new text to that receiver is available, as long as the time after the 450 latest sent packet to the same receiver is more than or equal to 100 451 ms, and also the maximum character rate to the receiver is not 452 exceeded. The intention is to keep the latency low while keeping a 453 good protection against text loss in bursty packet loss conditions. 455 3.6. Only one source per packet 457 New and redundant text from one source MAY be transmitted in the same 458 packet. Text from different sources MUST NOT be transmitted in the 459 same packet. 461 3.7. Do not send received text to the originating source 463 Text received to a mixer from a participant SHOULD NOT be included in 464 transmission from the mixer to that participant. 466 3.8. Clean incoming text 468 A mixer SHALL handle reception, recovery of packet loss, marking of 469 possible text loss and deletion of 'BOM' characters from each 470 participant before queueing received text for transmission to 471 receiving participants. 473 3.9. Redundancy 475 A transmitting party using redundancy SHALL send redundant 476 repetitions of T140blocks aleady transmitted in earlier packets. 478 The number of redundant generations of T140blocks to include in 479 transmitted packets SHALL be deduced from the SDP negotiation. It 480 SHOULD be set to the minimum of the number declared by the two 481 parties negotiating a connection. 483 3.10. Text placement in packets 485 At time of transmission, the mixer SHALL populate the RTP packet with 486 all T140blocks queued for transmission originating from the source in 487 turn for transmission as long as this is not in conflict with the 488 allowed number of characters per second ("CPS") or the maximum packet 489 size. The SSRC of the source shall be placed as the only member in 490 the CSRC-list. 492 Note: The CSRC-list in an RTP packet only includes the participant 493 who's text is included in text blocks. It is not the same as the 494 total list of participants in a conference. With audio and video 495 media, the CSRC-list would often contain all participants who are not 496 muted whereas text participants that don't type are completely silent 497 and thus are not represented in RTP packet CSRC-lists. 499 3.11. Source switching 501 When text from more than one source is available for transmission, 502 the mixer SHALL let the sources take turns in having their text 503 transmitted. When switching from transmission of one source to allow 504 another source to get turn to have its text sent, all intended 505 redundant generations of the last text from the current source MUST 506 be transmitted before text from another source is transmitted. 508 Actively transmitting sources SHOULD be allowed to take turns as 509 frequently as possible to have their text transmitted. That implies 510 that with the recommended redundancy, the mixer SHALL send primary 511 text and two packets with redundant text from the current source 512 before text from another source is transmitted. The source with the 513 oldest text received in the mixer SHOULD be next in turn to get all 514 its available text transmitted. 516 3.12. Empty T140blocks 518 If no unsent T140blocks were available for a source at the time of 519 populating a packet, but T140blocks are available which have not yet 520 been sent the full intended number of redundant transmissions, then 521 the primary T140block for that source is composed of an empty 522 T140block, and populated (without taking up any length) in a packet 523 for transmission. The corresponding SSRC SHALL be placed as usual in 524 its place in the CSRC-list. 526 The first packet in the session, the first after a source switch and 527 the first after a pause SHALL be poulated with the available 528 T140blocks for the source in turn to be sent as primary, and empty 529 T140blocks for the agreed number of redundancy generations. 531 3.13. Creation of the redundancy 533 The primary T140block from a source in the latest transmitted packet 534 is used to populate the first redundant T140block for that source. 535 The first redundant T140block for that source from the latest 536 transmission is placed as the second redundant T140block. 538 Usually this is the level of redundancy used. If a higher number of 539 redundancy is negotiated, then the procedure SHALL be maintained 540 until all available redundant levels of T140blocks are placed in the 541 packet. If a receiver has negotiated a lower number of "text/red" 542 generations, then that level shall be the maximum used by the 543 transmitter. 545 3.14. Timer offset fields 547 The timestamp offset values are inserted in the redundancy header, 548 with the time offset from the RTP timestamp in the packet when the 549 corresponding T140block was sent from its original source as primary. 551 The timestamp offsets are expressed in the same clock tick units as 552 the RTP timestamp. 554 The timestamp offset values for empty T140blocks have no relevance 555 but SHOULD be assigned realistic values. 557 3.15. Other RTP header fields 559 The number of members in the CSRC list ( 0 or 1) shall be placed in 560 the "CC" header field. Only mixers place value 1 in the "CC" field. 562 The current time SHALL be inserted in the timestamp. 564 The SSRC of the mixer for the RTT session SHALL be inserted in the 565 SSRC field of the RTP header. 567 The M-bit shall be handled as specified in [RFC4103]. 569 3.16. Pause in transmission 571 When there is no new T140block to transmit, and no redundant 572 T140block that has not been retransmitted the intended number of 573 times from any source, the transmission process can stop until either 574 new T140blocks arrive, or a keep-alive method calls for transmission 575 of keep-alive packets. 577 3.17. RTCP considerations 579 A mixer SHALL send RTCP reports with SDES, CNAME and NAME information 580 about the sources in the multi-party call. This makes it possible 581 for participants to compose a suitable label for text from each 582 source. 584 Integrity considerations SHALL be considered when composing these 585 fields. 587 3.18. Reception of multi-party contents 589 The "text/red" receiver included in an endpoint with presentation 590 functions will receive RTP packets in the single stream from the 591 mixer, and SHALL distribute the T140blocks for presentation in 592 presentation areas for each source. Other receiver roles, such as 593 gateways or chained mixers are also feasible, and requires 594 consideration if the stream shall just be forwarded, or distributed 595 based on the different sources. 597 3.18.1. Multi-party vs two-party use 599 If the "CC" field value of a received packet is 1, it indicates that 600 multi-party transmission is active, and the receiver MUST be prepared 601 to act on the source according to its role. If the CC value is 0, 602 the connection is point-to-point. 604 3.18.2. Level of redundancy 606 The used level of redundancy generations SHALL be evaluated from the 607 received packet contents. The number of generations (including the 608 primary) is equal to the number of members in the redundancy header. 610 3.18.3. Extracting text and handling recovery and loss 612 The RTP sequence numbers of the received packets SHALL be monitored 613 for gaps and packets out of order. 615 As long as the sequence is correct, each packet SHALL be unpacked in 616 order. The T140blocks SHALL be extracted from the primary area, and 617 the corresponding SSRC SHALL be extracted from the CSRC list and used 618 for assigning the new T140block to the correct presentation area (or 619 correspondingly for other receiver roles). 621 If a sequence number gap appears and is still there after some 622 defined time for jitter resolution, T140data SHALL be recovered from 623 redundant data. If the gap is wider than the number of generations 624 of redundant T140blocks in the packet and it is concluded that 625 t140blocks MAY have been lost, then a t140block SHALL be created with 626 a marker for possible text loss [T140ad1] and assigned to the SSRC of 627 the transmitter as a general input from the mixer because in general 628 it is not possible to deduce from which source(s) text was lost. It 629 is in some cases possible to deduce that no text was lost even for a 630 gap wider than the redundancy generations, and in some cases it can 631 be concluded which source that likely had loss. Therefore, the 632 receiver MAY insert the marker for possible text loss [T140ad1] in 633 the presentation area corresponding to the source which may have had 634 loss. See Section 3.22 636 Then, the T140block in the received packet SHALL be retrieved 637 beginning with the highest redundant generation, and assigning it to 638 the presentation area of that source. Finally the primary T140block 639 SHALL be retrieved from the packet and similarly assigned to the 640 corresponding presentation area for the source. 642 If the sequence number gap was equal to or less than the number of 643 redundancy generations in the received packet, a missing text marker 644 SHALL NOT be inserted, and instead the T140block and the SSRC fully 645 recovered from the redundancy and primary information and the CSRC- 646 list in the way indicated above. 648 3.18.4. Delete 'BOM' 650 Unicode character 'BOM' is used as a start indication and sometimes 651 used as a filler or keep alive by transmission implementations. 652 These SHALL be deleted on reception. 654 3.18.5. Empty T140blocks 656 Empty T140blocks are included as fillers for unused redundancy levels 657 in the packets. They just do not provide any contents and do not 658 contribute to the received streams. 660 3.19. Performance considerations 662 This solution has good performance for up to three participants 663 simultaneously sending text. At higher numbers of participants 664 simultaneously sending text, a jerkiness is visible in the 665 presentation of text. With five participants simultaneously 666 transmitting text, the jerkiness is about 1400 ms. Evenso, the 667 transmission of text catches up, so there is no resulting total delay 668 introduced. The solution is therefore suitable for emergency service 669 use, relay service use, and small or well-managed larger multimedia 670 conferences. Only in large unmanaged conferences with a high number 671 of participants there may on very rare occasions appear situations 672 when many participants happen to send text simultaneously, resulting 673 in unpleasantly long switching times. It should be noted that it is 674 only the number of users sending text within the same moment that 675 causes jerkiness, not the total number of users with RTT capability. 677 3.20. Security for session control and media 679 Security SHOULD be applied on both session control and media. In 680 applications where legacy endpoints without security may exist, a 681 negotiation between security and no security SHOULD be applied. If 682 no other security solution is mandated by the application, then RFC 683 8643 OSRTP[RFC8643] SHOULD be applied to negotiate SRTP media 684 security with DTLS. Most SDP examples below are for simplicity 685 expressed without the security additions. The principles (but not 686 all details) for applying DTLS-SRTP security is shown in a couple of 687 the following examples. 689 3.21. SDP offer/answer examples 691 This sections shows some examples of SDP for session negotiation of 692 the real-time text media in SIP sessions. Audio is usually provided 693 in the same session, and sometimes also video. The examples only 694 show the part of importance for the real-time text media. 696 Offer example for "text/red" format and multi-party support: 698 m=text 11000 RTP/AVP 100 98 699 a=rtpmap:98 t140/1000 700 a=rtpmap:100 red/1000 701 a=fmtp:100 98/98/98 702 a=rtt-mixer 704 Answer example from a multi-party capable device 705 m=text 14000 RTP/AVP 100 98 706 a=rtpmap:98 t140/1000 707 a=rtpmap:100 red/1000 708 a=fmtp:100 98/98/98 709 a=rtt-mixer 711 Offer example for "text/red" format including multi-party 712 and security: 713 a=fingerprint: SHA-1 \ 714 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB 715 m=text 11000 RTP/AVP 100 98 716 a=rtpmap:98 t140/1000 717 a=rtpmap:100 red/1000 718 a=fmtp:100 98/98/98 719 a=rtt-mixer 721 The "Fingerprint" is sufficient to offer DTLS-SRTP, with the media 722 line still indicating RTP/AVP. 724 Answer example from a multi-party capable device with security 725 a=fingerprint: SHA-1 \ 726 FF:FF:FF:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB 727 m=text 16000 RTP/AVP 100 98 728 a=rtpmap:98 t140/1000 729 a=rtpmap:100 red/1000 730 a=fmtp:100 98/98/98 731 a=rtt-mixer 733 With the "fingerprint" the device acknowledges use of SRTP/DTLS. 735 Answer example from a multi-party unaware device that also 736 does not support security: 738 m=text 12000 RTP/AVP 100 98 739 a=rtpmap:98 t140/1000 740 a=rtpmap:100 red/1000 741 a=fmtp:100 98/98/98 743 3.22. Packet sequence example from a source switch 745 This example shows a symbolic flow of packets from a mixer including 746 loss and recovery. The sequence includes a source switch. A and B 747 are sources of RTT. P indicates primary data. R1 is first redundant 748 generation data and R2 is second redundant generation data. A1, B1, 749 A2 etc are text chunks (T140blocks) received from the respective 750 sources. X indicates dropped packet between the mixer and a 751 receiver. The session is assumed to use original and two redundant 752 generations of RTT. 754 |----------------| 755 |Seq no 101 | 756 |CC=1 | 757 |CSRC list A | 758 |R2: A1 | 759 |R1: A2 | 760 |P: A3 | 761 |----------------| 763 Assuming that earlier packets ( with text A1 and A2) were received in 764 sequence, text A3 is received from packet 101 and assigned to 765 reception area A. The mixer is now assumed to have received text 766 from source B and need to prepare for sending that text. First it 767 must send the redundant generations of text A2 and A3. 769 |----------------| 770 |Seq no 102 | 771 |CC=1 | 772 |CSRC list A | 773 |R2 A2 | 774 |R1: A3 | 775 |P: Empty | 776 |----------------| 777 Nothing needs to be retrieved from this packet. But the receiver 778 needs to remember temporarily that it was received. 780 X----------------| 781 X Seq no 103 | 782 X CC=1 | 783 X CSRC list A | 784 X R2: A3 | 785 X R1: Empty | 786 X P: Empty | 787 X----------------| 788 Packet 103 is assumed to be dropped in network problems. It was the 789 last packet with contents from A before the source switch. 791 X----------------| 792 X Seq no 104 | 793 X CC=1 | 794 X CSRC list B | 795 X R2: Empty | 796 X R1: Empty | 797 X P2: B1 | 798 X----------------| 799 Packet 104 contains text from B, assumed dropped in network 800 problems. The mixer is assumed to have received text from A 801 on turn to send. Sending of text from B must therefore be 802 temporarily ended by sending redundancy twice. 804 X----------------| 805 X Seq no 105 | 806 X CC=1 | 807 X CSRC list B | 808 X R2: Empty | 809 X R1: B1 | 810 X P: Empty | 811 X----------------| 812 Packet 105 is assumed to be dropped in network problems 814 |----------------| 815 |Seq no 106 | 816 |CC=1 | 817 |CSRC list B | 818 | R2: B1 | 819 | R1: Empty | 820 | P: Empty | 821 |----------------| 823 Packet 106 is received. The latest received sequence number was 102. 824 Recovery is therefore tried for 103,104,105. There is no coverage 825 for seq no 103. But knowing that A3 was received as R1 in packet 826 102, it can be concluded that packet 103 just contained A3 as R2 and 827 A3 was already received and nothing was lost in packet 103. 829 For seqno 104, text B1 is recovered from the second generation 830 redundancy in packet 106 and appended to the reception area of B. 831 For seqno 105, nothing needs to be recovered. No primary text is 832 available in packet 106. 834 After this sequence, A3 and B1 have been received. In this case no 835 text was lost. Even if also packet 102 was lost, it can be concluded 836 that no text was lost. 838 If also packets 101 and 102 were lost, there would be a need to 839 create a marker for possibly lost text (U'FFFD) [T140ad1], inserted 840 generally and possibly also in text sequences A and B. 842 The general conclusion is that if CSRC is equal before and after a 843 gap, only a gap of two can be recovered. If CSRC is different before 844 and after the gap, a gap of 4 can be recovered if the last packet 845 before the gap contained only R2 data, a gap of 3 can be recovered if 846 the last packet before the gap contained R1 data and a gap of 2 can 847 be recovered if the last packet received before the gap contained 848 primary data. For larger gaps than the above, a mark for possible 849 loss needs to be inserted in the received stream from the mixer. 851 3.23. Maximum character rate "CPS" 853 The default maximum rate of reception of "text/t140" real-time text 854 is in RFC 4103 [RFC4103] specified to be 30 characters per second. 855 The value MAY be modified in the CPS parameter of the FMTP attribute 856 in the media section for the "text/t140" media. A mixer combining 857 real-time text from a number of sources may occasionally have a 858 higher combined flow of text coming from the sources. Endpoints 859 SHOULD therefore specify a suitable higher value for the CPS 860 parameter, corresponding to its real reception capability. A value 861 for "CPS" of 90 is the default for the "text/t140" stream in the 862 "text/red" format when multi-party real-time text is negotiated. See 863 RFC 4103 [RFC4103] for the format and use of the CPS parameter. The 864 same rules apply for the multi-party case except for the default 865 value. 867 4. Presentation level considerations 869 ITU-T T.140 [T140] provides the presentation level requirements for 870 the RFC 4103 [RFC4103] transport. T.140 [T140] has functions for 871 erasure and other formatting functions and has the following general 872 statement for the presentation: 874 "The display of text from the members of the conversation should be 875 arranged so that the text from each participant is clearly readable, 876 and its source and the relative timing of entered text is visualized 877 in the display. Mechanisms for looking back in the contents from the 878 current session should be provided. The text should be displayed as 879 soon as it is received." 881 Strict application of T.140 [T140] is of essence for the 882 interoperability of real-time text implementations and to fulfill the 883 intention that the session participants have the same information of 884 the text contents of the conversation without necessarily having the 885 exact same layout of the conversation. 887 T.140 [T140] specifies a set of presentation control codes to include 888 in the stream. Some of them are optional. Implementations MUST be 889 able to ignore optional control codes that they do not support. 891 There is no strict "message" concept in real-time text. Line 892 Separator SHALL be used as a separator allowing a part of received 893 text to be grouped in presentation. The characters "CRLF" may be 894 used by other implementations as replacement for Line Separator. The 895 "CRLF" combination SHALL be erased by just one erasing action, just 896 as the Line Separator. Presentation functions are allowed to group 897 text for presentation in smaller groups than the line separators 898 imply and present such groups with source indication together with 899 text groups from other sources (see the following presentation 900 examples). Erasure has no specific limit by any delimiter in the 901 text stream. 903 4.1. Presentation by multi-party aware endpoints 905 A multi-party aware receiving party, presenting real-time text MUST 906 separate text from different sources and present them in separate 907 presentation fields. The receiving party MAY separate presentation 908 of parts of text from a source in readable groups based on other 909 criteria than line separator and merge these groups in the 910 presentation area when it benefits the user to most easily find and 911 read text from the different participants. The criteria MAY e.g. be 912 a received comma, full stop, or other phrase delimiters, or a long 913 pause. 915 When text is received from multiple original sources, the 916 presentation SHOULD provide a view where text is added in multiple 917 presentation fields. 919 If the presentation presents text from different sources in one 920 common area, the presenting endpoint SHOULD insert text from the 921 local user ended at suitable points merged with received text to 922 indicate the relative timing for when the text groups were completed. 923 In this presentation mode, the receiving endpoint SHALL present the 924 source of the different groups of text. 926 A view of a three-party RTT call in chat style is shown in this 927 example . 929 _________________________________________________ 930 | |^| 931 |[Alice] Hi, Alice here. |-| 932 | | | 933 |[Bob] Bob as well. | | 934 | | | 935 |[Eve] Hi, this is Eve, calling from Paris. | | 936 | I thought you should be here. | | 937 | | | 938 |[Alice] I am coming on Thursday, my | | 939 | performance is not until Friday morning.| | 940 | | | 941 |[Bob] And I on Wednesday evening. | | 942 | | | 943 |[Alice] Can we meet on Thursday evening? | | 944 | | | 945 |[Eve] Yes, definitely. How about 7pm. | | 946 | at the entrance of the restaurant | | 947 | Le Lion Blanc? | | 948 |[Eve] we can have dinner and then take a walk |-| 949 |______________________________________________|v| 950 | But I need to be back to |^| 951 | the hotel by 11 because I need |-| 952 | | | 953 | I wou |-| 954 |______________________________________________|v| 955 | of course, I underst | 956 |________________________________________________| 958 Figure 3: Example of a three-party RTT call presented in chat style 959 seen at participant 'Alice's endpoint. 961 Other presentation styles than the chat style may be arranged. 963 This figure shows how a coordinated column view MAY be presented. 965 _____________________________________________________________________ 966 | Bob | Eve | Alice | 967 |____________________|______________________|_______________________| 968 | | |I will arrive by TGV. | 969 |My flight is to Orly| |Convenient to the main | 970 | |Hi all, can we plan |station. | 971 | |for the seminar? | | 972 |Eve, will you do | | | 973 |your presentation on| | | 974 |Friday? |Yes, Friday at 10. | | 975 |Fine, wo | |We need to meet befo | 976 |___________________________________________________________________| 977 Figure 4: An example of a coordinated column-view of a three-party 978 session with entries ordered vertically in approximate time-order. 980 4.2. Multi-party mixing for multi-party unaware endpoints 982 When the mixer has indicated multi-party capability by the "rtt- 983 mixer" sdp attribute in an SDP negotiation, but the multi-party 984 capability negotiation fails with an endpoint, then the agreed "text/ 985 red" or "text/t140" format SHALL be used and the mixer SHOULD compose 986 a best-effort presentation of multi-party real-time text in one 987 stream intended to be presented by an endpoint with no multi-party 988 awareness. 990 This presentation format has functional limitations and SHOULD be 991 used only to enable participation in multi-party calls by legacy 992 deployed endpoints implementing only RFC 4103 without any multi-party 993 extensions specified in this document. 995 The principles and procedures below do not specify any new protocol 996 elements. They are instead composed from the information in ITU-T 997 T.140 [T140] and an ambition to provide a best effort presentation on 998 an endpoint which has functions only for two-party calls. 1000 The mixer mixing for multi-party unaware endpoints SHALL compose a 1001 simulated limited multi-party RTT view suitable for presentation in 1002 one presentation area. The mixer SHALL group text in suitable groups 1003 and prepare for presentation of them by inserting a new line between 1004 them if the transmitted text did not already end with a new line. A 1005 presentable label SHOULD be composed and sent for the source 1006 initially in the session and after each source switch. With this 1007 procedure the time for source switching is depending on the actions 1008 of the users. In order to expedite source switch, a user can for 1009 example end its turn with a new line. 1011 4.2.1. Actions by the mixer at reception from the call participants 1013 When text is received by the mixer from the different participants, 1014 the mixer SHALL recover text from redundancy if any packets are lost. 1015 The mark for lost text [T140ad1] SHOULD be inserted in the stream if 1016 unrecoverable loss appears. Any Unicode "BOM" characters, possibly 1017 used for keep-alive shall be deleted. The time of creation of text 1018 (retrieved from the RTP timestamp) SHALL be stored together with the 1019 received text from each source in queues for transmission to the 1020 recipients. 1022 4.2.2. Actions by the mixer for transmission to the recipients 1024 The following procedure SHOULD be applied for each recipient of 1025 multi-part text from the mixer. 1027 The text for transmission SHOULD be formatted by the mixer for each 1028 receiving user for presentation in one single presentation area. 1029 Text received from a participant SHOULD NOT be included in 1030 transmission to that participant. When there is text available for 1031 transmission from the mixer to a receiving party from more than one 1032 participant, the mixer SHOULD switch between transmission of text 1033 from the different sources at suitable points in the transmitted 1034 stream. 1036 When switching source, the mixer SHOULD insert a line separator if 1037 the already transmitted text did not end with a new line (line 1038 separator or CRLF). A label SHOULD be composed from information in 1039 the CNAME and NAME fields in RTCP reports from the participant to 1040 have its text transmitted, or from other session information for that 1041 user. The label SHOULD be delimited by suitable characters (e.g. '[ 1042 ]') and transmitted. The CSRC SHOULD indicate the selected source. 1043 Then text from that selected participant SHOULD be transmitted until 1044 a new suitable point for switching source is reached. 1046 Integrity considerations SHALL be taken when composing the label. 1048 Seeking a suitable point for switching source SHOULD be done when 1049 there is older text waiting for transmission from any party than the 1050 age of the last transmitted text. Suitable points for switching are: 1052 * A completed phrase ended by comma 1054 * A completed sentence 1056 * A new line (line separator or CRLF) 1058 * A long pause (e.g. > 10 seconds) in received text from the 1059 currently transmitted source 1061 * If text from one participant has been transmitted with text from 1062 other sources waiting for transmission for a long time (e.g. > 1 1063 minute) and none of the other suitable points for switching has 1064 occurred, a source switch MAY be forced by the mixer at next word 1065 delimiter, and also if even a word delimiter does not occur within 1066 a time (e.g. 15 seconds) after the scan for word delimiter 1067 started. 1069 When switching source, the source which has the oldest text in queue 1070 SHOULD be selected to be transmitted. A character display count 1071 SHOULD be maintained for the currently transmitted source, starting 1072 at zero after the label is transmitted for the currently transmitted 1073 source. 1075 The status SHOULD be maintained for the latest control code for 1076 Select Graphic Rendition (SGR) from each source. If there is an SGR 1077 code stored as the status for the current source before the source 1078 switch is done, a reset of SGR shall be sent by the sequence SGR 0 1079 [009B 0000 006D] after the new line and before the new label during a 1080 source switch. See SGR below for an explanation. This transmission 1081 does not influence the display count. 1083 If there is an SGR code stored for the new source after the source 1084 switch, that SGR code SHOULD be transmitted to the recipient before 1085 the label. This transmission does not influence the display count. 1087 4.2.3. Actions on transmission of text 1089 Text from a source sent to the recipient SHOULD increase the display 1090 count by one per transmitted character. 1092 4.2.4. Actions on transmission of control codes 1094 The following control codes specified by T.140 require specific 1095 actions. They SHOULD cause specific considerations in the mixer. 1096 Note that the codes presented here are expressed in UCS-16, while 1097 transmission is made in UTF-8 transform of these codes. 1099 BEL 0007 Bell Alert in session, provides for alerting during an 1100 active session. The display count SHOULD not be altered. 1102 NEW LINE 2028 Line separator. Check and perform a source switch if 1103 appropriate. Increase display count by 1. 1105 CR LF 000D 000A A supported, but not preferred way of requesting a 1106 new line. Check and perform a source switch if appropriate. 1107 Increase display count by 1. 1109 INT ESC 0061 Interrupt (used to initiate mode negotiation 1110 procedure). The display count SHOULD not be altered. 1112 SGR 009B Ps 006D Select graphic rendition. Ps is rendition 1113 parameters specified in ISO 6429. The display count SHOULD not be 1114 altered. The SGR code SHOULD be stored for the current source. 1116 SOS 0098 Start of string, used as a general protocol element 1117 introducer, followed by a maximum 256 bytes string and the ST. 1118 The display count SHOULD not be altered. 1120 ST 009C String terminator, end of SOS string. The display count 1121 SHOULD not be altered. 1123 ESC 001B Escape - used in control strings. The display count SHOULD 1124 not be altered for the complete escape code. 1126 Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used 1127 for synchronization and keep-alive. SHOULD be deleted from 1128 incoming streams. Shall be sent first after session establishment 1129 to the recipient. The display count shall not be altered. 1131 Missing text mark (U+FFFD) "Replacement character", represented as a 1132 question mark in a rhombus, or if that is not feasible, replaced 1133 by an apostrophe ', marks place in stream of possible text loss. 1134 SHOULD be inserted by the reception procedure in case of 1135 unrecoverable loss of packets. The display count SHOULD be 1136 increased by one when sent as for any other character. 1138 SGR If a control code for selecting graphic rendition (SGR), other 1139 than reset of the graphic rendition (SGR 0) is sent to a 1140 recipient, that control code shall also be stored as status for 1141 the source in the storage for SGR status. If a reset graphic 1142 rendition (SGR 0) originated from a source is sent, then the SGR 1143 status storage for that source shall be cleared. The display 1144 count shall not be increased. 1146 BS (U+0008) Back Space, intended to erase the last entered character 1147 by a source. Erasure by backspace cannot always be performed as 1148 the erasing party intended. If an erasing action erases all text 1149 up to the end of the leading label after a source switch, then the 1150 mixer must not transmit more backspaces. Instead it is 1151 RECOMMENDED that a letter "X" is inserted in the text stream for 1152 each backspace as an indication of the intent to erase more. A 1153 new line is usually coded by a Line Separator, but the character 1154 combination "CRLF" MAY be used instead. Erasure of a new line is 1155 in both cases done by just one erasing action (Backspace). If the 1156 display count has a positive value it is decreased by one when the 1157 BS is sent. If the display count is at zero, it is not altered. 1159 4.2.5. Packet transmission 1161 A mixer transmitting to a multi-party unaware terminal SHOULD send 1162 primary data only from one source per packet. The SSRC SHOULD be the 1163 SSRC of the mixer. The CSRC list SHOULD contain one member and be 1164 the SSRC of the source of the primary data. 1166 4.2.6. Functional limitations 1168 When a multi-party unaware endpoint presents a conversation in one 1169 display area in a chat style, it inserts source indications for 1170 remote text and local user text as they are merged in completed text 1171 groups. When an endpoint using this layout receives and presents 1172 text mixed for multi-party unaware endpoints, there will be two 1173 levels of source indicators for the received text; one generated by 1174 the mixer and inserted in a label after each source switch, and 1175 another generated by the receiving endpoint and inserted after each 1176 switch between local and remote source in the presentation area. 1177 This will waste display space and look inconsistent to the reader. 1179 New text can be presented only from one source at a time. Switch of 1180 source to be presented takes place at suitable places in the text, 1181 such as end of phrase, end of sentence, line separator and 1182 inactivity. Therefore the time to switch to present waiting text 1183 from other sources may become long and will vary and depend on the 1184 actions of the currently presented source. 1186 Erasure can only be done up to the latest source switch. If a user 1187 tries to erase more text, the erasing actions will be presented as 1188 letter X after the label. 1190 Text loss because of network errors may hit the label between entries 1191 from different parties, causing risk for misunderstanding from which 1192 source a piece of text is. 1194 These facts makes it strongly RECOMMENDED to implement multi-party 1195 awareness in RTT endpoints. The use of the mixing method for multi- 1196 party-unaware endpoints should be left for use with endpoints which 1197 are impossible to upgrade to become multi-party aware. 1199 4.2.7. Example views of presentation on multi-party unaware endpoints 1201 The following pictures are examples of the view on a participant's 1202 display for the multi-party-unaware case. 1204 _________________________________________________ 1205 | Conference | Alice | 1206 |________________________|_________________________| 1207 | |I will arrive by TGV. | 1208 |[Bob]:My flight is to |Convenient to the main | 1209 |Orly. |station. | 1210 |[Eve]:Hi all, can we | | 1211 |plan for the seminar. | | 1212 | | | 1213 |[Bob]:Eve, will you do | | 1214 |your presentation on | | 1215 |Friday? | | 1216 |[Eve]:Yes, Friday at 10.| | 1217 |[Bob]: Fine, wo |We need to meet befo | 1218 |________________________|_________________________| 1220 Figure 5: Alice who has a conference-unaware client is receiving the 1221 multi-party real-time text in a single-stream. This figure shows how 1222 a coordinated column view MAY be presented on Alice's device. 1224 _________________________________________________ 1225 | |^| 1226 |[Alice] Hi, Alice here. |-| 1227 | | | 1228 |[mix][Bob] Bob as well. | | 1229 | | | 1230 |[Eve] Hi, this is Eve, calling from Paris | | 1231 | I thought you should be here. | | 1232 | | | 1233 |[Alice] I am coming on Thursday, my | | 1234 | performance is not until Friday morning.| | 1235 | | | 1236 |[mix][Bob] And I on Wednesday evening. | | 1237 | | | 1238 |[Eve] we can have dinner and then walk | | 1239 | | | 1240 |[Eve] But I need to be back to | | 1241 | the hotel by 11 because I need | | 1242 | |-| 1243 |______________________________________________|v| 1244 | of course, I underst | 1245 |________________________________________________| 1247 Figure 6: An example of a view of the multi-party unaware 1248 presentation in chat style. Alice is the local user. 1250 5. Relation to Conference Control 1251 5.1. Use with SIP centralized conferencing framework 1253 The SIP conferencing framework, mainly specified in RFC 1254 4353[RFC4353], RFC 4579[RFC4579] and RFC 4575[RFC4575] is suitable 1255 for coordinating sessions including multi-party RTT. The RTT stream 1256 between the mixer and a participant is one and the same during the 1257 conference. Participants get announced by notifications when 1258 participants are joining or leaving, and further user information may 1259 be provided. The SSRC of the text to expect from joined users MAY be 1260 included in a notification. The notifications MAY be used both for 1261 security purposes and for translation to a label for presentation to 1262 other users. 1264 5.2. Conference control 1266 In managed conferences, control of the real-time text media SHOULD be 1267 provided in the same way as other for media, e.g. for muting and 1268 unmuting by the direction attributes in SDP [RFC4566]. 1270 Note that floor control functions may be of value for RTT users as 1271 well as for users of other media in a conference. 1273 6. Gateway Considerations 1275 6.1. Gateway considerations with Textphones (e.g. TTYs). 1277 Multi-party RTT sessions may involve gateways of different kinds. 1278 Gateways involved in setting up sessions SHALL correctly reflect the 1279 multi-party capability or unawareness of the combination of the 1280 gateway and the remote endpoint beyond the gateway. 1282 One case that may occur is a gateway to PSTN for communication with 1283 textphones (e.g. TTYs). Textphones are limited devices with no 1284 multi-party awareness, and it SHOULD therefore be suitable for the 1285 gateway to not indicate multi-party awareness for that case. Another 1286 solution is that the gateway indicates multi-party capability towards 1287 the mixer, and includes the multi-party mixer function for multi- 1288 party unaware endpoints itself. This solution makes it possible to 1289 make adaptations for the functional limitations of the textphone 1290 (TTY). 1292 More information on gateways to textphones (TTYs) is found in RFC 1293 5194[RFC5194] 1295 6.2. Gateway considerations with WebRTC. 1297 Gateway operation to real-time text in WebRTC may also be required. 1298 In WebRTC, RTT is specified in 1299 [I-D.ietf-mmusic-t140-usage-data-channel]. 1301 A multi-party bridge may have functionality for communicating by RTT 1302 both in RTP streams with RTT and WebRTC t140 data channels. Other 1303 configurations may consist of a multi-party bridge with either 1304 technology for RTT transport and a separate gateway for conversion of 1305 the text communication streams between RTP and t140 data channel. 1307 In WebRTC, it is assumed that for a multi-party session, one t140 1308 data channel is established for each source from a gateway or bridge 1309 to each participant. Each participant also has a data channel with 1310 two-way connection with the gateway or bridge. 1312 The t140 channel used both ways is for text from the WebRTC user and 1313 from the bridge or gateway itself to the WebRTC user. The label 1314 parameter of this t140 channel is used as NAME field in RTCP to 1315 participants on the RTP side. The other t140 channels are only for 1316 text from other participants to the WebRTC user. 1318 When a new participant has entered the session with RTP transport of 1319 rtt, a new t140 channel SHOULD be established to WebRTC users with 1320 the label parameter composed from the NAME field in RTCP on the RTP 1321 side. 1323 When a new participant has entered the multi-party session with RTT 1324 transport in a WebRTC t140 data channel, the new participant SHOULD 1325 be announced by a notification to RTP users. The label parameter 1326 from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP 1327 side, or other available session information. 1329 7. Updates to RFC 4103 1331 This document updates RFC 4103[RFC4103] by introducing an sdp media 1332 attribute "rtt-mixer" for negotiation of multi-party mixing 1333 capability with the [RFC4103] format, and by specifying the rules for 1334 packets when multi-party capability is negotiated and in use. 1336 8. Congestion considerations 1338 The congestion considerations and recommended actions from RFC 4103 1339 [RFC4103] are valid also in multi-party situations. 1341 The first action in case of congestion SHOULD be to temporarily 1342 increase the transmission interval up to two seconds. 1344 If the unlikely situation appears that more than 20 participants in a 1345 conference send text simultaneously, it will take more than 7 seconds 1346 between presentation of text from each of these participants. More 1347 time than that can cause confusion in the session. It is therefore 1348 RECOMMENDED that the mixer discards such text in excess inserts a 1349 general indication of possible text loss [T140ad1] in the session. 1350 If the main text contributor is indicated in any way, the mixer MAY 1351 avoid deleting text from that participant. 1353 9. Acknowledgements 1355 James Hamlin for format and performance aspects. 1357 10. IANA Considerations 1359 10.1. Registration of the "rtt-mixer" sdp media attribute 1361 [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the 1362 RFC number of this document.] 1364 IANA is asked to register the new sdp attribute "rtt-mixer". 1366 Contact name: IESG 1368 Contact email: iesg@ietf.org 1370 Attribute name: rtt-mixer 1372 Attribute semantics: See RFCXXXX Section 3.1 1374 Attribute value: none 1376 Usage level: media 1378 Purpose: Indicate support by mixer and endpoint of multi-party 1379 mixing for real-time text transmission, using a common RTP-stream 1380 for transmission of text from a number of sources mixed with one 1381 source at a time and the source indicated in a single CSRC-list 1382 member. 1384 Charset Dependent: no 1386 O/A procedure: See RFCXXXX Section 3.1 1388 Mux Category: normal 1390 Reference: RFCXXXX 1392 11. Security Considerations 1394 The RTP-mixer model requires the mixer to be allowed to decrypt, pack 1395 and encrypt secured text from the conference participants. Therefore 1396 the mixer needs to be trusted. This is similar to the situation for 1397 central mixers of audio and video. 1399 The requirement to transfer information about the user in RTCP 1400 reports in SDES, CNAME and NAME fields, and in conference 1401 notifications, for creation of labels may have privacy concerns as 1402 already stated in RFC 3550 [RFC3550], and may be restricted of 1403 privacy reasons. The receiving user will then get a more symbolic 1404 label for the source. 1406 12. Change history 1408 12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09 1410 Changed name on the SDP media attribute to "rtt-mixer" 1412 Restructure of section 2 for balance between aware and unaware cases. 1414 Moved conference control to own section. 1416 Improved clarification of recovery and loss in the packet sequence 1417 example. 1419 A number of editorial corrections and improvements. 1421 12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 1423 Deleted the method requiring a new packet format "text/rex" because 1424 of the longer standardization and implementation period it needs. 1426 Focus on use of RFC 4103 text/red format with shorter transmission 1427 interval, and source indicated in CSRC. 1429 12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 1431 Added a method based on the "text/red" format and single source per 1432 packet, negotiated by the "rtt-mixer" sdp attribute. 1434 Added reasoning and recommendation about indication of loss. 1436 The highest number of sources in one packet is 15, not 16. Changed. 1438 Added in information on update to RFC 4103 that RFC 4103 explicitly 1439 allows addition of FEC method. The redundancy is a kind of forward 1440 error correction.. 1442 12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 1444 Improved definitions list format. 1446 The format of the media subtype parameters is made to match the 1447 requirements. 1449 The mapping of media subtype parameters to sdp is included. 1451 The CPS parameter belongs to the t140 subtype and does not need to be 1452 registered here. 1454 12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 1456 nomenclature and editorial improvements 1458 "this document" used consistently to refer to this document. 1460 12.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 1462 'Redundancy header' renamed to 'data header'. 1464 More clarifications added. 1466 Language and figure number corrections. 1468 12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 1470 Mention possible need to mute and raise hands as for other media. 1471 ---done ---- 1473 Make sure that use in two-party calls is also possible and explained. 1474 - may need more wording - 1476 Clarify the RTT is often used together with other media. --done-- 1478 Tell that text mixing is N-1. A users own text is not received in 1479 the mix. -done- 1481 In 3. correct the interval to: A "text/rex" transmitter SHOULD send 1482 packets distributed in time as long as there is something (new or 1483 redundant T140blocks) to transmit. The maximum transmission interval 1484 SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 1485 receiver as soon as new text to that receiver is available, as long 1486 as the time after the latest sent packet to the same receiver is more 1487 than 150 ms, and also the maximum character rate to the receiver is 1488 not exceeded. The intention is to keep the latency low while keeping 1489 a good protection against text loss in bursty packet loss conditions. 1490 -done- 1492 In 1.3 say that the format is used both ways. -done- 1494 In 13.1 change presentation area to presentation field so that reader 1495 does not think it shall be totally separated. -done- 1497 In Performance and intro, tell the performance in number of 1498 simultaneous sending users and introduced delay 16, 150 vs 1499 requirements 5 vs 500. -done -- 1501 Clarify redundancy level per connection. -done- 1503 Timestamp also for the last data header. To make it possible for all 1504 text to have time offset as for transmission from the source. Make 1505 that header equal to the others. -done- 1507 Mixer always use the CSRC list, even for its own BOM. -done- 1509 Combine all talk about transmission interval (300 ms vs when text has 1510 arrived) in section 3 in one paragraph or close to each other. -done- 1512 Documents the goal of good performance with low delay for 5 1513 simultaneous typers in the introduction. -done- 1515 Describe better that only primary text shall be sent on to receivers. 1516 Redundancy and loss must be resolved by the mixer. -done- 1518 12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 1520 SDP and better description and visibility of security by OSRTP RFC 1521 8634 needed. 1523 The description of gatewaying to WebRTC extended. 1525 The description of the data header in the packet is improved. 1527 12.9. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 1529 2,5,6 More efficient format "text/rex" introduced and attribute 1530 a=rtt-mix deleted. 1532 3. Brief about use of OSRTP for security included- More needed. 1534 4. Brief motivation for the solution and why not rtp-translator is 1535 used added to intro. 1537 7. More limitations for the multi-party unaware mixing method 1538 inserted. 1540 8. Updates to RFC 4102 and 4103 more clearly expressed. 1542 9. Gateway to WebRTC started. More needed. 1544 12.10. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 1545 to draft-ietf-avtcore-multi-party-rtt-mix-00 1547 Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00 1549 Replaced CDATA in IANA registration table with better coding. 1551 Converted to xml2rfc version 3. 1553 12.11. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 1554 to -03 1556 Changed company and e-mail of the author. 1558 Changed title to "RTP-mixer formatting of multi-party Real-time text" 1559 to better match contents. 1561 Check and modification where needed of use of RFC 2119 words SHALL 1562 etc. 1564 More about the CC value in sections on transmitters and receivers so 1565 that 1-to-1 sessions do not use the mixer format. 1567 Enhanced section on presentation for multi-party-unaware endpoints 1569 A paragraph recommending CPS=150 inserted in the performance section. 1571 12.12. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 1572 to -02 1574 In Abstract and 1. Introduction: Introduced wording about regulatory 1575 requirements. 1577 In section 5: The transmission interval is decreased to 100 ms when 1578 there is text from more than one source to transmit. 1580 In section 11 about SDP negotiation, a SHOULD-requirement is 1581 introduced that the mixer should make a mix for multi-party unaware 1582 endpoints if the negotiation is not successful. And a reference to a 1583 later chapter about it. 1585 The presentation considerations chapter 14 is extended with more 1586 information about presentation on multi-party aware endpoints, and a 1587 new section on the multi-party unaware mixing with low functionality 1588 but SHOULD a be implemented in mixers. Presentation examples are 1589 added. 1591 A short chapter 15 on gateway considerations is introduced. 1593 Clarification about the text/t140 format included in chapter 10. 1595 This sentence added to the chapter 10 about use without redundancy. 1596 "The text/red format SHOULD be used unless some other protection 1597 against packet loss is utilized, for example a reliable network or 1598 transport." 1600 Note about deviation from RFC 2198 added in chapter 4. 1602 In chapter 9. "Use with SIP centralized conferencing framework" the 1603 following note is inserted: Note: The CSRC-list in an RTP packet only 1604 includes participants who's text is included in one or more text 1605 blocks. It is not the same as the list of participants in a 1606 conference. With audio and video media, the CSRC-list would often 1607 contain all participants who are not muted whereas text participants 1608 that don't type are completely silent and so don't show up in RTP 1609 packet CSRC-lists. 1611 12.13. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 1612 to -01 1614 Editorial cleanup. 1616 Changed capability indication from fmtp-parameter to SDP attribute 1617 "rtt-mix". 1619 Swapped order of redundancy elements in the example to match reality. 1621 Increased the SDP negotiation section 1623 13. References 1625 13.1. Normative References 1627 [I-D.ietf-mmusic-t140-usage-data-channel] 1628 Holmberg, C. and G. Hellstrom, "T.140 Real-time Text 1629 Conversation over WebRTC Data Channels", Work in Progress, 1630 Internet-Draft, draft-ietf-mmusic-t140-usage-data-channel- 1631 14, 10 April 2020, . 1634 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1635 Requirement Levels", BCP 14, RFC 2119, 1636 DOI 10.17487/RFC2119, March 1997, 1637 . 1639 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1640 Jacobson, "RTP: A Transport Protocol for Real-Time 1641 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1642 July 2003, . 1644 [RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type", 1645 RFC 4102, DOI 10.17487/RFC4102, June 2005, 1646 . 1648 [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text 1649 Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, 1650 . 1652 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1653 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 1654 July 2006, . 1656 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1657 Security (DTLS) Extension to Establish Keys for the Secure 1658 Real-time Transport Protocol (SRTP)", RFC 5764, 1659 DOI 10.17487/RFC5764, May 2010, 1660 . 1662 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 1663 Keeping Alive the NAT Mappings Associated with RTP / RTP 1664 Control Protocol (RTCP) Flows", RFC 6263, 1665 DOI 10.17487/RFC6263, June 2011, 1666 . 1668 [RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. 1669 Stach, "An Opportunistic Approach for Secure Real-time 1670 Transport Protocol (OSRTP)", RFC 8643, 1671 DOI 10.17487/RFC8643, August 2019, 1672 . 1674 [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for 1675 multimedia application text conversation", February 1998, 1676 . 1678 [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), 1679 Protocol for multimedia application text conversation", 1680 February 2000, 1681 . 1683 13.2. Informative References 1685 [RFC4353] Rosenberg, J., "A Framework for Conferencing with the 1686 Session Initiation Protocol (SIP)", RFC 4353, 1687 DOI 10.17487/RFC4353, February 2006, 1688 . 1690 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 1691 Session Initiation Protocol (SIP) Event Package for 1692 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 1693 2006, . 1695 [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol 1696 (SIP) Call Control - Conferencing for User Agents", 1697 BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, 1698 . 1700 [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- 1701 Time Text over IP Using the Session Initiation Protocol 1702 (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, 1703 . 1705 Author's Address 1707 Gunnar Hellstrom 1708 Gunnar Hellstrom Accessible Communication 1709 Esplanaden 30 1710 SE-13670 Vendelso 1711 Sweden 1713 Email: gunnar.hellstrom@ghaccess.se