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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group B. Burman 3 Internet-Draft M. Westerlund 4 Intended status: Standards Track Ericsson 5 Expires: October 12, 2018 S. Nandakumar 6 M. Zanaty 7 Cisco 8 April 10, 2018 10 Using Simulcast in SDP and RTP Sessions 11 draft-ietf-mmusic-sdp-simulcast-12 13 Abstract 15 In some application scenarios it may be desirable to send multiple 16 differently encoded versions of the same media source in different 17 RTP streams. This is called simulcast. This document describes how 18 to accomplish simulcast in RTP and how to signal it in SDP. The 19 described solution uses an RTP/RTCP identification method to identify 20 RTP streams belonging to the same media source, and makes an 21 extension to SDP to relate those RTP streams as being different 22 simulcast formats of that media source. The SDP extension consists 23 of a new media level SDP attribute that expresses capability to send 24 and/or receive simulcast RTP streams. 26 Status of This Memo 28 This Internet-Draft is submitted in full conformance with the 29 provisions of BCP 78 and BCP 79. 31 Internet-Drafts are working documents of the Internet Engineering 32 Task Force (IETF). Note that other groups may also distribute 33 working documents as Internet-Drafts. The list of current Internet- 34 Drafts is at https://datatracker.ietf.org/drafts/current/. 36 Internet-Drafts are draft documents valid for a maximum of six months 37 and may be updated, replaced, or obsoleted by other documents at any 38 time. It is inappropriate to use Internet-Drafts as reference 39 material or to cite them other than as "work in progress." 41 This Internet-Draft will expire on October 12, 2018. 43 Copyright Notice 45 Copyright (c) 2018 IETF Trust and the persons identified as the 46 document authors. All rights reserved. 48 This document is subject to BCP 78 and the IETF Trust's Legal 49 Provisions Relating to IETF Documents 50 (https://trustee.ietf.org/license-info) in effect on the date of 51 publication of this document. Please review these documents 52 carefully, as they describe your rights and restrictions with respect 53 to this document. Code Components extracted from this document must 54 include Simplified BSD License text as described in Section 4.e of 55 the Trust Legal Provisions and are provided without warranty as 56 described in the Simplified BSD License. 58 Table of Contents 60 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 61 2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 4 62 2.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 4 63 2.2. Requirements Language . . . . . . . . . . . . . . . . . . 5 64 3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 5 65 3.1. Reaching a Diverse Set of Receivers . . . . . . . . . . . 6 66 3.2. Application Specific Media Source Handling . . . . . . . 7 67 3.3. Receiver Media Source Preferences . . . . . . . . . . . . 7 68 4. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . 7 69 5. Detailed Description . . . . . . . . . . . . . . . . . . . . 9 70 5.1. Simulcast Attribute . . . . . . . . . . . . . . . . . . . 10 71 5.2. Simulcast Capability . . . . . . . . . . . . . . . . . . 11 72 5.3. Offer/Answer Use . . . . . . . . . . . . . . . . . . . . 13 73 5.3.1. Generating the Initial SDP Offer . . . . . . . . . . 13 74 5.3.2. Creating the SDP Answer . . . . . . . . . . . . . . . 13 75 5.3.3. Offerer Processing the SDP Answer . . . . . . . . . . 14 76 5.3.4. Modifying the Session . . . . . . . . . . . . . . . . 15 77 5.4. Use with Declarative SDP . . . . . . . . . . . . . . . . 15 78 5.5. Relating Simulcast Streams . . . . . . . . . . . . . . . 15 79 5.6. Signaling Examples . . . . . . . . . . . . . . . . . . . 16 80 5.6.1. Single-Source Client . . . . . . . . . . . . . . . . 16 81 5.6.2. Multi-Source Client . . . . . . . . . . . . . . . . . 18 82 5.6.3. Simulcast and Redundancy . . . . . . . . . . . . . . 20 83 6. RTP Aspects . . . . . . . . . . . . . . . . . . . . . . . . . 23 84 6.1. Outgoing from Endpoint with Media Source . . . . . . . . 23 85 6.2. RTP Middlebox to Receiver . . . . . . . . . . . . . . . . 23 86 6.2.1. Media-Switching Mixer . . . . . . . . . . . . . . . . 24 87 6.2.2. Selective Forwarding Middlebox . . . . . . . . . . . 26 88 6.3. RTP Middlebox to RTP Middlebox . . . . . . . . . . . . . 27 89 7. Network Aspects . . . . . . . . . . . . . . . . . . . . . . . 28 90 7.1. Bitrate Adaptation . . . . . . . . . . . . . . . . . . . 28 91 8. Limitation . . . . . . . . . . . . . . . . . . . . . . . . . 29 92 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 29 93 10. Security Considerations . . . . . . . . . . . . . . . . . . . 30 94 11. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 30 95 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 30 96 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 30 97 13.1. Normative References . . . . . . . . . . . . . . . . . . 31 98 13.2. Informative References . . . . . . . . . . . . . . . . . 32 99 Appendix A. Requirements . . . . . . . . . . . . . . . . . . . . 34 100 Appendix B. Changes From Earlier Versions . . . . . . . . . . . 35 101 B.1. Modifications Between WG Version -11 and -12 . . . . . . 35 102 B.2. Modifications Between WG Version -10 and -11 . . . . . . 36 103 B.3. Modifications Between WG Version -09 and -10 . . . . . . 36 104 B.4. Modifications Between WG Version -08 and -09 . . . . . . 36 105 B.5. Modifications Between WG Version -07 and -08 . . . . . . 37 106 B.6. Modifications Between WG Version -06 and -07 . . . . . . 37 107 B.7. Modifications Between WG Version -05 and -06 . . . . . . 37 108 B.8. Modifications Between WG Version -04 and -05 . . . . . . 38 109 B.9. Modifications Between WG Version -03 and -04 . . . . . . 38 110 B.10. Modifications Between WG Version -02 and -03 . . . . . . 39 111 B.11. Modifications Between WG Version -01 and -02 . . . . . . 39 112 B.12. Modifications Between WG Version -00 and -01 . . . . . . 39 113 B.13. Modifications Between Individual Version -00 and WG 114 Version -00 . . . . . . . . . . . . . . . . . . . . . . . 40 115 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 40 117 1. Introduction 119 Most of today's multiparty video conference solutions make use of 120 centralized servers to reduce the bandwidth and CPU consumption in 121 the endpoints. Those servers receive RTP streams from each 122 participant and send some suitable set of possibly modified RTP 123 streams to the rest of the participants, which usually have 124 heterogeneous capabilities (screen size, CPU, bandwidth, codec, etc). 125 One of the biggest issues is how to perform RTP stream adaptation to 126 different participants' constraints with the minimum possible impact 127 on both video quality and server performance. 129 Simulcast is defined in this memo as the act of simultaneously 130 sending multiple different encoded streams of the same media source, 131 e.g. the same video source encoded with different video encoder types 132 or image resolutions. This can be done in several ways and for 133 different purposes. This document focuses on the case where it is 134 desirable to provide a media source as multiple encoded streams over 135 RTP [RFC3550] towards an intermediary so that the intermediary can 136 provide the wanted functionality by selecting which RTP stream(s) to 137 forward to other participants in the session, and more specifically 138 how the identification and grouping of the involved RTP streams are 139 done. 141 The intended scope of the defined mechanism is to support negotiation 142 and usage of simulcast when using SDP offer/answer and media 143 transport over RTP. The media transport topologies considered are 144 point to point RTP sessions as well as centralized multi-party RTP 145 sessions, where a media sender will provide the simulcasted streams 146 to an RTP middlebox or endpoint, and middleboxes may further 147 distribute the simulcast streams to other middleboxes or endpoints. 148 Usage of multicast or broadcast transport is out of scope and left 149 for future extension. 151 This document describes a few scenarios that motivates the use of 152 simulcast, and also defines the needed RTP/RTCP and SDP signaling for 153 it. 155 2. Definitions 157 2.1. Terminology 159 This document makes use of the terminology defined in RTP Taxonomy 160 [RFC7656], and RTP Topologies [RFC7667]. The following terms are 161 especially noted or here defined: 163 RTP Mixer: An RTP middle node, defined in [RFC7667] (Section 3.6 to 164 3.9). 166 RTP Session: An association among a group of participants 167 communicating with RTP, as defined in [RFC3550] and amended by 168 [RFC7656]. 170 RTP Stream: A stream of RTP packets containing media data, as 171 defined in [RFC7656]. 173 RTP Switch: A common short term for the terms "switching RTP mixer", 174 "source projecting middlebox", and "video switching MCU" as 175 discussed in [RFC7667]. 177 Simulcast Stream: One encoded stream or dependent stream from a set 178 of concurrently transmitted encoded streams and optional dependent 179 streams, all sharing a common media source, as defined in 180 [RFC7656]. For example, HD and thumbnail video simulcast versions 181 of a single media source sent concurrently as separate RTP 182 Streams. 184 Simulcast Format: Different formats of a simulcast stream serve the 185 same purpose as alternative RTP payload types in non-simulcast 186 SDP: to allow multiple alternative media formats for a given RTP 187 stream. As for multiple RTP payload types on the m-line in offer/ 188 answer [RFC3264], any one of the negotiated alternative formats 189 can be used in a single RTP stream at a given point in time, but 190 not more than one (based on RTP timestamp). What format is used 191 can change dynamically from one RTP packet to another. 193 2.2. Requirements Language 195 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 196 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 197 "OPTIONAL" in this document are to be interpreted as described in BCP 198 14 [RFC2119] [RFC8174] when, and only when, they appear in all 199 capitals, as shown here. 201 3. Use Cases 203 The use cases of simulcast described in this document relate to a 204 multi-party communication session where one or more central nodes are 205 used to adapt the view of the communication session towards 206 individual participants, and facilitate the media transport between 207 participants. Thus, these cases target the RTP Mixer type of 208 topology. 210 There are two principle approaches for an RTP Mixer to provide this 211 adapted view of the communication session to each receiving 212 participant: 214 o Transcoding (decoding and re-encoding) received RTP streams with 215 characteristics adapted to each receiving participant. This often 216 include mixing or composition of media sources from multiple 217 participants into a mixed media source originated by the RTP 218 Mixer. The main advantage of this approach is that it achieves 219 close to optimal adaptation to individual receiving participants. 220 The main disadvantages are that it can be very computationally 221 expensive to the RTP Mixer, typically degrades media Quality of 222 Experience (QoE) such as end-to-end delay for the receiving 223 participants, and requires RTP Mixer access to media content. 225 o Switching a subset of all received RTP streams or sub-streams to 226 each receiving participant, where the used subset is typically 227 specific to each receiving participant. The main advantages of 228 this approach are that it is computationally cheap to the RTP 229 Mixer, has very limited impact on media QoE, and does not require 230 RTP Mixer (full) access to media content. The main disadvantage 231 is that it can be difficult to combine a subset of received RTP 232 streams into a perfect fit to the resource situation of a 233 receiving participant. 235 The use of simulcast relates to the latter approach, where it is more 236 important to reduce the load on the RTP Mixer and/or minimize QoE 237 impact than to achieve an optimal adaptation of resource usage. 239 3.1. Reaching a Diverse Set of Receivers 241 The media sources provided by a sending participant potentially need 242 to reach several receiving participants that differ in terms of 243 available resources. The receiver resources that typically differ 244 include, but are not limited to: 246 Codec: This includes codec type (such as RTP payload format MIME 247 type) and can include codec configuration. A couple of codec 248 resources that differ only in codec configuration will be 249 "different" if they are somehow not "compatible", like if they 250 differ in video codec profile, or the transport packetization 251 configuration. 253 Sampling: This relates to how the media source is sampled, in 254 spatial as well as in temporal domain. For video streams, spatial 255 sampling affects image resolution and temporal sampling affects 256 video frame rate. For audio, spatial sampling relates to the 257 number of audio channels and temporal sampling affects audio 258 bandwidth. This may be used to suit different rendering 259 capabilities or needs at the receiving endpoints. 261 Bitrate: This relates to the number of bits sent per second to 262 transmit the media source as an RTP stream, which typically also 263 affects the Quality of Experience (QoE) for the receiving user. 265 Letting the sending participant create a simulcast of a few 266 differently configured RTP streams per media source can be a good 267 tradeoff when using an RTP switch as middlebox, instead of sending a 268 single RTP stream and using an RTP mixer to create individual 269 transcodings to each receiving participant. 271 This requires that the receiving participants can be categorized in 272 terms of available resources and that the sending participant can 273 choose a matching configuration for a single RTP stream per category 274 and media source. For example, a set of receiving participants 275 differ only in screen resolution; some are able to display video with 276 at most 360p resolution and some support 720p resolution. A sending 277 participant can then reach all receivers with best possible 278 resolution by creating a simulcast of RTP streams with 360p and 720p 279 resolution for each sent video media source. 281 The maximum number of simulcasted RTP streams that can be sent is 282 mainly limited by the amount of processing and uplink network 283 resources available to the sending participant. 285 3.2. Application Specific Media Source Handling 287 The application logic that controls the communication session may 288 include special handling of some media sources. It is, for example, 289 commonly the case that the media from a sending participant is not 290 sent back to itself. 292 It is also common that a currently active speaker participant is 293 shown in larger size or higher quality than other participants (the 294 sampling or bitrate aspects of Section 3.1). Not sending the active 295 speaker media back to itself means there is some other participant's 296 media that instead has to receive special handling towards the active 297 speaker; typically the previous active speaker. This way, the 298 previously active speaker is needed both in larger size (to current 299 active speaker) and in small size (to the rest of the participants), 300 which can be solved with a simulcast from the previously active 301 speaker to the RTP switch. 303 3.3. Receiver Media Source Preferences 305 The application logic that controls the communication session may 306 allow receiving participants to state preferences on the 307 characteristics of the RTP stream they like to receive, for example 308 in terms of the aspects listed in Section 3.1. Sending a simulcast 309 of RTP streams is one way of accommodating receivers with conflicting 310 or otherwise incompatible preferences. 312 4. Overview 314 This memo defines SDP [RFC4566] signaling that covers the above 315 described simulcast use cases and functionalities. A number of 316 requirements for such signaling are elaborated in Appendix A. 318 The RID mechanism, as defined in [I-D.ietf-mmusic-rid], enables an 319 SDP offerer or answerer to specify a number of different RTP stream 320 restrictions for a rid-id by using the "a=rid" line. Examples of 321 such restrictions are maximum bitrate, maximum spatial video 322 resolution (width and height), maximum video framerate, etc. Each 323 rid-id may also be restricted to use only a subset of the RTP payload 324 types in the associated SDP media description. Those RTP payload 325 types can have their own configurations and parameters affecting what 326 can be sent or received, using the "a=fmtp" line as well as other SDP 327 attributes. 329 A new SDP media level attribute "a=simulcast" is defined. The 330 attribute describes, independently for send and receive directions, 331 the number of simulcast RTP streams as well as potential alternative 332 formats for each simulcast RTP stream. Each simulcast RTP stream, 333 including alternatives, is identified using the RID identifier (rid- 334 id), defined in [I-D.ietf-mmusic-rid]. 336 a=simulcast:send 1;2,3 recv 4 338 If the above line is included in an SDP offer, the "send" part 339 indicates the offerer's capability and proposal to send two simulcast 340 RTP streams. Each simulcast stream is described by one or more RTP 341 stream identifiers (rid-id), each group of rid-ids for a simulcast 342 stream is separated by a semicolon (";"). When a simulcast stream 343 has multiple rid-ids that are separated by a comma (","), they 344 describe alternative representations for that particular simulcast 345 RTP stream. Thus, the above "send" part is interpreted as an 346 intention to send two simulcast RTP streams. The first simulcast RTP 347 stream is identified and restricted according to rid-id 1. The 348 second simulcast RTP stream can be sent as two alternatives, 349 identified and restricted according to rid-ids 2 and 3. The "recv" 350 part of the above line indicates that the offerer desires to receive 351 a single RTP stream (no simulcast) according to rid-id 4. 353 A more complete example SDP offer media description is provided 354 below: 356 m=video 49300 RTP/AVP 97 98 99 357 a=rtpmap:97 H264/90000 358 a=rtpmap:98 H264/90000 359 a=rtpmap:99 VP8/90000 360 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 361 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 362 a=fmtp:99 max-fs=240; max-fr=30 363 a=rid:1 send pt=97 max-width=1280;max-height=720 364 a=rid:2 send pt=98 max-width=320;max-height=180 365 a=rid:3 send pt=99 max-width=320;max-height=180 366 a=rid:4 recv pt=97 367 a=simulcast:send 1;2,3 recv 4 368 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 370 Figure 1: Example Simulcast Media Description in Offer 372 The above SDP media description can be interpreted at a high level to 373 say that the offerer is capable of sending two simulcast RTP streams, 374 one H.264 encoded stream in up to 720p resolution, and one additional 375 stream encoded as either H.264 or VP8 with a maximum resolution of 376 320x180 pixels. The offerer can receive one H.264 stream with 377 maximum 720p resolution. 379 The receiver of this SDP offer can generate an SDP answer that 380 indicates what it accepts. It uses the "a=simulcast" attribute to 381 indicate simulcast capability and specify what simulcast RTP streams 382 and alternatives to receive and/or send. An example of such 383 answering "a=simulcast" attribute, corresponding to the above offer, 384 is: 386 a=simulcast:recv 1;2 send 4 388 With this SDP answer, the answerer indicates in the "recv" part that 389 it wants to receive the two simulcast RTP streams. It has removed an 390 alternative that it doesn't support (rid-id 3). The send part 391 confirms to the offerer that it will receive one stream for this 392 media source according to rid-id 4. The corresponding, more complete 393 example SDP answer media description could look like: 395 m=video 49674 RTP/AVP 97 98 396 a=rtpmap:97 H264/90000 397 a=rtpmap:98 H264/90000 398 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 399 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 400 a=rid:1 recv pt=97 max-width=1280;max-height=720 401 a=rid:2 recv pt=98 max-width=320;max-height=180 402 a=rid:4 send pt=97 403 a=simulcast:recv 1;2 send 4 404 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 406 Figure 2: Example Simulcast Media Description in Answer 408 It is assumed that a single SDP media description is used to describe 409 a single media source. This is aligned with the concepts defined in 410 [RFC7656] and will work in a WebRTC context, both with and without 411 BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation] grouping of media 412 descriptions. 414 To summarize, the "a=simulcast" line describes send and receive 415 direction simulcast streams separately. Each direction can in turn 416 describe one or more simulcast streams, separated by semicolon. The 417 identifiers describing simulcast streams on the "a=simulcast" line 418 are rid-id, as defined by "a=rid" lines in [I-D.ietf-mmusic-rid]. 419 Each simulcast stream can be offered as a list of alternative rid-id, 420 with each alternative separated by comma (not in the examples above). 421 A detailed specification can be found in Section 5 and more detailed 422 examples are outlined in Section 5.6. 424 5. Detailed Description 426 This section further details the overview above (Section 4). First, 427 formal syntax is provided (Section 5.1), followed by the rest of the 428 SDP attribute definition in Section 5.2. Relating Simulcast Streams 429 (Section 5.5) provides the definition of the RTP/RTCP mechanisms 430 used. The section is concluded with a number of examples. 432 5.1. Simulcast Attribute 434 This document defines a new SDP media-level "a=simulcast" attribute, 435 with value according to the following ABNF [RFC5234] syntax: 437 sc-value = ( sc-send [SP sc-recv] ) / ( sc-recv [SP sc-send] ) 438 sc-send = "send" SP sc-str-list 439 sc-recv = "recv" SP sc-str-list 440 sc-str-list = sc-alt-list *( ";" sc-alt-list ) 441 sc-alt-list = sc-id *( "," sc-id ) 442 sc-id-paused = "~" 443 sc-id = [sc-id-paused] rid-id 444 ; SP defined in [RFC5234] 445 ; rid-id defined in [I-D.ietf-mmusic-rid] 447 Figure 3: ABNF for Simulcast Value 449 Note to RFC Editor: Replace "I-D.ietf-mmusic-rid" in the above 450 figure with RFC number of draft-ietf-mmusic-rid before publication 451 of this document. 453 The "a=simulcast" attribute has a parameter in the form of one or two 454 simulcast stream descriptions, each consisting of a direction ("send" 455 or "recv"), followed by a list of one or more simulcast streams. 456 Each simulcast stream consists of one or more alternative simulcast 457 formats. Each simulcast format is identified by a simulcast stream 458 identifier (rid-id). The rid-id MUST have the form of an RTP stream 459 identifier, as described by RTP Payload Format Restrictions 460 [I-D.ietf-mmusic-rid]. 462 In the list of simulcast streams, each simulcast stream is separated 463 by a semicolon (";"). Each simulcast stream can in turn be offered 464 in one or more alternative formats, represented by rid-ids, separated 465 by a comma (","). Each rid-id can also be specified as initially 466 paused [RFC7728], indicated by prepending a "~" to the rid-id. The 467 reason to allow separate initial pause states for each rid-id is that 468 pause capability can be specified individually for each RTP payload 469 type referenced by an rid-id. Since pause capability specified via 470 the "a=rtcp-fb" attribute and rid-id specified by "a=rid" can refer 471 to common payload types, it is unfeasible to pause streams with rid- 472 id where any of the related RTP payload type(s) do not have pause 473 capability. 475 5.2. Simulcast Capability 477 Simulcast capability is expressed through a new media level SDP 478 attribute, "a=simulcast" (Section 5.1). The use of this attribute at 479 the session level is undefined. Implementations of this 480 specification MUST NOT use it at the session level and MUST ignore it 481 if received at the session level. Extensions to this specification 482 may define such session level usage. Each SDP media description MUST 483 contain at most one "a=simulcast" line. 485 There are separate and independent sets of simulcast streams in send 486 and receive directions. When listing multiple directions, each 487 direction MUST NOT occur more than once on the same line. 489 Simulcast streams using undefined rid-id MUST NOT be used as valid 490 simulcast streams by an RTP stream receiver. The direction for an 491 rid-id MUST be aligned with the direction specified for the 492 corresponding RTP stream identifier on the "a=rid" line. 494 The listed number of simulcast streams for a direction sets a limit 495 to the number of supported simulcast streams in that direction. The 496 order of the listed simulcast streams in the "send" direction 497 suggests a proposed order of preference, in decreasing order: the 498 rid-id listed first is the most preferred and subsequent streams have 499 progressively lower preference. The order of the listed rid-id in 500 the "recv" direction expresses which simulcast streams that are 501 preferred, with the leftmost being most preferred. This can be of 502 importance if the number of actually sent simulcast streams have to 503 be reduced for some reason. 505 rid-id that have explicit dependencies [RFC5583] 506 [I-D.ietf-mmusic-rid] to other rid-id (even in the same media 507 description) MAY be used. 509 Use of more than a single, alternative simulcast format for a 510 simulcast stream MAY be specified as part of the attribute parameters 511 by expressing the simulcast stream as a comma-separated list of 512 alternative rid-id. The order of the rid-id alternatives within a 513 simulcast stream is significant; the rid-id alternatives are listed 514 from (left) most preferred to (right) least preferred. For the use 515 of simulcast, this overrides the normal codec preference as expressed 516 by format type ordering on the "m=" line, using regular SDP rules. 517 This is to enable a separation of general codec preferences and 518 simulcast stream configuration preferences. However, the choice of 519 which alternative to use per simulcast stream is independent, and 520 there is currently no mechanism for align the choice between 521 alternative rid-ids between different simulcast streams. 523 A simulcast stream can use a codec defined such that the same RTP 524 SSRC can change RTP payload type multiple times during a session, 525 possibly even on a per-packet basis. A typical example can be a 526 speech codec that makes use of Comfort Noise [RFC3389] and/or DTMF 527 [RFC4733] formats. 529 If RTP stream pause/resume [RFC7728] is supported, any rid-id MAY be 530 prefixed by a "~" character to indicate that the corresponding 531 simulcast stream is initially paused already from start of the RTP 532 session. In this case, support for RTP stream pause/resume MUST also 533 be included under the same "m=" line where "a=simulcast" is included. 534 All RTP payload types related to such initially paused simulcast 535 stream MUST be listed in the SDP as pause/resume capable as specified 536 by [RFC7728], e.g. by using the "*" wildcard format for "a=rtcp-fb". 538 An initially paused simulcast stream in "send" direction for the 539 endpoint sending the SDP MUST be considered equivalent to an 540 unsolicited locally paused stream, and be handled accordingly. 541 Initially paused simulcast streams are resumed as described by the 542 RTP pause/resume specification. An RTP stream receiver that wishes 543 to resume an unsolicited locally paused stream needs to know the SSRC 544 of that stream. The SSRC of an initially paused simulcast stream can 545 be obtained from an RTP stream sender RTCP Sender Report (SR) 546 including both the desired SSRC as "SSRC of sender", and the rid-id 547 value in an RtpStreamId RTCP SDES item [I-D.ietf-avtext-rid]. 549 If the endpoint sending the SDP includes an "recv" direction 550 simulcast stream that is initially paused, then the remote RTP sender 551 receiving the SDP SHOULD put its RTP stream in a unsolicited locally 552 paused state. However, this does not apply if there are other RTP 553 stream receivers that do not mark the simulcast stream as initially 554 paused. The reason to require an initially paused "recv" stream to 555 be considered locally paused by the remote RTP sender, instead of 556 making it equivalent to implicitly sending a pause request, is 557 because the pausing RTP sender cannot know which receiving SSRC owns 558 the restriction when Temporary Maximum Media Stream Bit Rate Request 559 (TMMBR) and Temporary Maximum Media Stream Bit Rate Notification 560 (TMMBN) are used for pause/resume signaling (Section 5.6 of 561 [RFC7728]) since the RTP receiver's SSRC in send direction is 562 sometimes not yet known. 564 Use of the redundant audio data [RFC2198] format could be seen as a 565 form of simulcast for loss protection purposes, but is not considered 566 conflicting with the mechanisms described in this memo and MAY 567 therefore be used as any other format. In this case the "red" 568 format, rather than the carried formats, SHOULD be the one to list as 569 a simulcast stream on the "a=simulcast" line. 571 The media formats and corresponding characteristics of simulcast 572 streams SHOULD be chosen such that they are different, e.g. as 573 different SDP formats with differing "a=rtpmap" and/or "a=fmtp" 574 lines, or as differently defined RTP payload format restrictions. If 575 this difference is not required, it is RECOMMENDED to use RTP 576 duplication [RFC7104] procedures instead of simulcast. To avoid 577 complications in implementations, a single rid-id MUST NOT occur more 578 than once per "a=simulcast" line. Note that this does not eliminate 579 use of simulcast as an RTP duplication mechanism, since it is 580 possible to define multiple different rid-id that are effectively 581 equivalent. 583 5.3. Offer/Answer Use 585 Note: The inclusion of "a=simulcast" or the use of simulcast does 586 not change any of the interpretation or Offer/Answer procedures 587 for other SDP attributes, like "a=fmtp" or "a=rid". 589 5.3.1. Generating the Initial SDP Offer 591 An offerer wanting to use simulcast for a media description SHALL 592 include one "a=simulcast" attribute in that media description in the 593 offer. An offerer listing a set of receive simulcast streams and/or 594 alternative formats as rid-id in the offer MUST be prepared to 595 receive RTP streams for any of those simulcast streams and/or 596 alternative formats from the answerer. 598 5.3.2. Creating the SDP Answer 600 An answerer that does not understand the concept of simulcast will 601 also not know the attribute and will remove it in the SDP answer, as 602 defined in existing SDP Offer/Answer [RFC3264] procedures. Since SDP 603 session level simulcast is undefined in this memo, an answerer that 604 receives an offer with the "a=simulcast" attribute on SDP session 605 level SHALL remove it in the answer. An answerer that understands 606 the attribute but receives multiple "a=simulcast" attributes in the 607 same media description SHALL disable use of simulcast by removing all 608 "a=simulcast" lines for that media description in the answer. 610 An answerer that does understand the attribute and that wants to 611 support simulcast in an indicated direction SHALL reverse 612 directionality of the unidirectional direction parameters; "send" 613 becomes "recv" and vice versa, and include it in the answer. 615 An answerer that receives an offer with simulcast containing an 616 "a=simulcast" attribute listing alternative rid-id MAY keep all the 617 alternative rid-id in the answer, but it MAY also choose to remove 618 any non-desirable alternative rid-id in the answer. The answerer 619 MUST NOT add any alternative rid-id in send direction in the answer 620 that were not present in the offer receive direction. The answerer 621 MUST be prepared to receive any of the receive direction rid-id 622 alternatives and MAY send any of the send direction alternatives that 623 are part of the answer. 625 An answerer that receives an offer with simulcast that lists a number 626 of simulcast streams, MAY reduce the number of simulcast streams in 627 the answer, but MUST NOT add simulcast streams. 629 An answerer that receives an offer without RTP stream pause/resume 630 capability MUST NOT mark any simulcast streams as initially paused in 631 the answer. 633 An RTP stream pause/resume capable answerer that receives an offer 634 with RTP stream pause/resume capability MAY mark any rid-id that 635 refer to pause/resume capable formats as initially paused in the 636 answer. 638 An answerer that receives indication in an offer of an rid-id being 639 initially paused SHOULD mark that rid-id as initially paused also in 640 the answer, regardless of direction, unless it has good reason for 641 the rid-id not being initially paused. One reason to remove an 642 initial pause in the answer compared to the offer could, for example, 643 be that all receive direction simulcast streams for a media source 644 the answerer accepts in the answer would otherwise be paused. 646 5.3.3. Offerer Processing the SDP Answer 648 An offerer that receives an answer without "a=simulcast" MUST NOT use 649 simulcast towards the answerer. An offerer that receives an answer 650 with "a=simulcast" without any rid-id in a specified direction MUST 651 NOT use simulcast in that direction. 653 An offerer that receives an answer where some rid-id alternatives are 654 kept MUST be prepared to receive any of the kept send direction rid- 655 id alternatives, and MAY send any of the kept receive direction rid- 656 id alternatives. 658 An offerer that receives an answer where some of the rid-id are 659 removed compared to the offer MAY release the corresponding resources 660 (codec, transport, etc) in its receive direction and MUST NOT send 661 any RTP packets corresponding to the removed rid-id. 663 An offerer that offered some of its rid-id as initially paused and 664 that receives an answer that does not indicate RTP stream pause/ 665 resume capability, MUST NOT initially pause any simulcast streams. 667 An offerer with RTP stream pause/resume capability that receives an 668 answer where some rid-id are marked as initially paused, SHOULD 669 initially pause those RTP streams regardless if they were marked as 670 initially paused also in the offer, unless it has good reason for 671 those RTP streams not being initially paused. One such reason could, 672 for example, be that the answerer would otherwise initially not 673 receive any media of that type at all. 675 5.3.4. Modifying the Session 677 Offers inside an existing session follow the same rules as for 678 initial SDP offer, with these additions: 680 1. rid-id marked as initially paused in the offerer's send direction 681 SHALL reflect the offerer's opinion of the current pause state at 682 the time of creating the offer. This is purely informational, 683 and RTP stream pause/resume [RFC7728] signaling in the ongoing 684 session SHALL take precedence in case of any conflict or 685 ambiguity. 687 2. rid-id marked as initially paused in the offerer's receive 688 direction SHALL (as in an initial offer) reflect the offerer's 689 desired rid-id pause state. Except for the case where the 690 offerer already paused the corresponding RTP stream through RTP 691 stream pause/resume [RFC7728] signaling , this is identical to 692 the conditions at an initial offer. 694 Creation of SDP answers and processing of SDP answers inside an 695 existing session follow the same rules as described above for initial 696 SDP offer/answer. 698 Session modification restrictions in section 6.5 of RTP payload 699 format restrictions [I-D.ietf-mmusic-rid] also apply. 701 5.4. Use with Declarative SDP 703 This document does not define the use of "a=simulcast" in declarative 704 SDP, partly motivated by use of the simulcast format identification 705 [I-D.ietf-mmusic-rid] not being defined for use in declarative SDP. 706 If concrete use cases for simulcast in declarative SDP are identified 707 in the future, the authors of this memo expect that additional 708 specifications will address such use. 710 5.5. Relating Simulcast Streams 712 Simulcast RTP streams MUST be related on RTP level through 713 RtpStreamId [I-D.ietf-avtext-rid], as specified in the SDP 714 "a=simulcast" attribute (Section 5.2) parameters. This is sufficient 715 as long as there is only a single media source per SDP media 716 description. When using BUNDLE 717 [I-D.ietf-mmusic-sdp-bundle-negotiation], where multiple SDP media 718 descriptions jointly specify a single RTP session, the SDES MID 719 identification mechanism in BUNDLE allows relating RTP streams back 720 to individual media descriptions, after which the above described 721 RtpStreamId relations can be used. Use of the RTP header extension 722 [RFC8285] for both MID and RtpStreamId identifications can be 723 important to ensure rapid initial reception, required to correctly 724 interpret and process the RTP streams. Implementers of this 725 specification MUST support the RTCP source description (SDES) item 726 method and SHOULD support RTP header extension method to signal 727 RtpStreamId on RTP level. 729 NOTE: For the case where it is clear from SDP that RTP PT uniquely 730 maps to corresponding RtpStreamId, an RTP receiver can use RTP PT 731 to relate simulcast streams. This can sometimes enable decoding 732 even in advance to receiving RtpStreamId information in RTCP SDES 733 and/or RTP header extensions. 735 RTP streams MUST only use a single alternative rid-id at a time 736 (based on RTP timestamps), but MAY change format (and rid-id) on a 737 per-RTP packet basis. This corresponds to the existing (non- 738 simulcast) SDP offer/answer case when multiple formats are included 739 on the "m=" line in the SDP answer, enabling per-RTP packet change of 740 RTP payload type. 742 5.6. Signaling Examples 744 These examples describe a client to video conference service, using a 745 centralized media topology with an RTP mixer. 747 +---+ +-----------+ +---+ 748 | A |<---->| |<---->| B | 749 +---+ | | +---+ 750 | Mixer | 751 +---+ | | +---+ 752 | F |<---->| |<---->| J | 753 +---+ +-----------+ +---+ 755 Figure 4: Four-party Mixer-based Conference 757 5.6.1. Single-Source Client 759 Alice is calling in to the mixer with a simulcast-enabled client 760 capable of a single media source per media type. The client can send 761 a simulcast of 2 video resolutions and frame rates: HD 1280x720p 762 30fps and thumbnail 320x180p 15fps. This is defined below using the 763 "imageattr" [RFC6236]. In this example, only the "pt" "a=rid" 764 parameter is used, effectively achieving a 1:1 mapping between 765 RtpStreamId and media formats (RTP payload types), to describe 766 simulcast stream formats. Alice's Offer: 768 v=0 769 o=alice 2362969037 2362969040 IN IP4 192.0.2.156 770 s=Simulcast Enabled Client 771 t=0 0 772 c=IN IP4 192.0.2.156 773 m=audio 49200 RTP/AVP 0 774 a=rtpmap:0 PCMU/8000 775 m=video 49300 RTP/AVP 97 98 776 a=rtpmap:97 H264/90000 777 a=rtpmap:98 H264/90000 778 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 779 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 780 a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720] 781 a=imageattr:98 send [x=320,y=180] recv [x=320,y=180] 782 a=rid:1 send pt=97 783 a=rid:2 send pt=98 784 a=rid:3 recv pt=97 785 a=simulcast:send 1;2 recv 3 786 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 788 Figure 5: Single-Source Simulcast Offer 790 The only thing in the SDP that indicates simulcast capability is the 791 line in the video media description containing the "simulcast" 792 attribute. The included "a=fmtp" and "a=imageattr" parameters 793 indicates that sent simulcast streams can differ in video resolution. 794 The RTP header extension for RtpStreamId is offered to avoid issues 795 with the initial binding between RTP streams (SSRCs) and the 796 RtpStreamId identifying the simulcast stream and its format. 798 The Answer from the server indicates that it too is simulcast 799 capable. Should it not have been simulcast capable, the 800 "a=simulcast" line would not have been present and communication 801 would have started with the media negotiated in the SDP. Also the 802 usage of the RtpStreamId RTP header extension is accepted. 804 v=0 805 o=server 823479283 1209384938 IN IP4 192.0.2.2 806 s=Answer to Simulcast Enabled Client 807 t=0 0 808 c=IN IP4 192.0.2.43 809 m=audio 49672 RTP/AVP 0 810 a=rtpmap:0 PCMU/8000 811 m=video 49674 RTP/AVP 97 98 812 a=rtpmap:97 H264/90000 813 a=rtpmap:98 H264/90000 814 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 815 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 816 a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720] 817 a=imageattr:98 send [x=320,y=180] recv [x=320,y=180] 818 a=rid:1 recv pt=97 819 a=rid:2 recv pt=98 820 a=rid:3 send pt=97 821 a=simulcast:recv 1;2 send 3 822 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 824 Figure 6: Single-Source Simulcast Answer 826 Since the server is the simulcast media receiver, it reverses the 827 direction of the "simulcast" and "rid" attribute parameters. 829 5.6.2. Multi-Source Client 831 Fred is calling in to the same conference as in the example above 832 with a two-camera, two-display system, thus capable of handling two 833 separate media sources in each direction, where each media source is 834 simulcast-enabled in the send direction. Fred's client is restricted 835 to a single media source per media description. 837 The first two simulcast streams for the first media source use 838 different codecs, H264-SVC [RFC6190] and H264 [RFC6184]. These two 839 simulcast streams also have a temporal dependency. Two different 840 video codecs, VP8 [RFC7741] and H264, are offered as alternatives for 841 the third simulcast stream for the first media source. Only the 842 highest fidelity simulcast stream is sent from start, the lower 843 fidelity streams being initially paused. 845 The second media source is offered with three different simulcast 846 streams. All video streams of this second media source are loss 847 protected by RTP retransmission [RFC4588]. Also here, all but the 848 highest fidelity simulcast stream are initially paused. 850 Fred's client is also using BUNDLE to send all RTP streams from all 851 media descriptions in the same RTP session on a single media 852 transport. Although using many different simulcast streams in this 853 example, the use of RtpStreamId as simulcast stream identification 854 enables use of a low number of RTP payload types. Note that the use 855 of both BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation] and "a=rid" 856 [I-D.ietf-mmusic-rid] recommends using the RTP header extension 857 [RFC8285] for carrying these RTP stream identification fields, which 858 is consequently also included in the SDP. Note also that for 859 "a=rid", the corresponding SDES attribute is named RtpStreamId 860 [I-D.ietf-avtext-rid]. 862 v=0 863 o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d 864 s=Offer from Simulcast Enabled Multi-Source Client 865 t=0 0 866 c=IN IP6 2001:db8::c000:27d 867 a=group:BUNDLE foo bar zen 868 m=audio 49200 RTP/AVP 99 869 a=mid:foo 870 a=rtpmap:99 G722/8000 871 m=video 49600 RTP/AVPF 100 101 103 872 a=mid:bar 873 a=rtpmap:100 H264-SVC/90000 874 a=rtpmap:101 H264/90000 875 a=rtpmap:103 VP8/90000 876 a=fmtp:100 profile-level-id=42400d;max-fs=3600;max-mbps=216000; \ 877 mst-mode=NI-TC 878 a=fmtp:101 profile-level-id=42c00d;max-fs=3600;max-mbps=108000 879 a=fmtp:103 max-fs=900; max-fr=30 880 a=rid:1 send pt=100;max-width=1280;max-height=720;max-fps=60;depend=2 881 a=rid:2 send pt=101;max-width=1280;max-height=720;max-fps=30 882 a=rid:3 send pt=101;max-width=640;max-height=360 883 a=rid:4 send pt=103;max-width=640;max-height=360 884 a=depend:100 lay bar:101 885 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid 886 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 887 a=rtcp-fb:* ccm pause nowait 888 a=simulcast:send 1;2;~4,3 889 m=video 49602 RTP/AVPF 96 104 890 a=mid:zen 891 a=rtpmap:96 VP8/90000 892 a=fmtp:96 max-fs=3600; max-fr=30 893 a=rtpmap:104 rtx/90000 894 a=fmtp:104 apt=96;rtx-time=200 895 a=rid:1 send pt=96;max-fs=921600;max-fps=30 896 a=rid:2 send pt=96;max-fs=614400;max-fps=15 897 a=rid:3 send pt=96;max-fs=230400;max-fps=30 898 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid 899 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 900 a=rtcp-fb:* ccm pause nowait 901 a=simulcast:send 1;~2;~3 903 Figure 7: Fred's Multi-Source Simulcast Offer 905 5.6.3. Simulcast and Redundancy 907 The example in this section looks at applying simulcast with audio 908 and video redundancy formats. The audio media description uses codec 909 and bitrate restrictions, combining it with RTP Payload for Redundant 910 Audio Data [RFC2198] for enhanced packet loss resilience. The video 911 media description applies both resolution and bitrate restrictions, 912 combining it with FEC in the form of Flexible FEC 913 [I-D.ietf-payload-flexible-fec-scheme] and RTP Retransmission 914 [RFC4588]. 916 The audio source is offered to be sent as two simulcast streams. The 917 first simulcast stream is encoded with Opus, restricted to 50 kbps 918 (rid-id=5), and the second simulcast stream is encoded either with 919 G.711 (rid-id=7) or with G.711 combined with LPC for redundancy (rid- 920 id=6). In this example, stand-alone LPC is not offered as an 921 possible payload type for the second simulcast stream's RID, which 922 could e.g. be motivated by not providing sufficient quality. 924 The video source is offered to be sent as two simulcast streams, both 925 with two alternative simulcast formats. Redundancy and repair are 926 offered in the form of both Flexible FEC and RTP Retransmission. The 927 Flexible FEC is not bound to any particular RTP streams and is 928 therefore possible to use across all RTP streams that are being sent 929 as part of this media description. 931 v=0 932 o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d 933 s=Offer from Simulcast Enabled Client using Redundancy 934 t=0 0 935 c=IN IP6 2001:db8::c000:27d 936 a=group:BUNDLE foo bar 937 m=audio 49200 RTP/AVP 97 98 99 100 101 102 938 a=mid:foo 939 a=rtpmap:97 G711/8000 940 a=rtpmap:98 LPC/8000 941 a=rtpmap:99 OPUS/48000/1 942 a=rtpmap:100 RED/8000/1 943 a=rtpmap:101 CN/8000 944 a=rtpmap:102 telephone-event/8000 945 a=fmtp:99 useinbandfec=1; usedtx=0 946 a=fmtp:100 97/98 947 a=fmtp:102 0-15 948 a=ptime:20 949 a=maxptime:40 950 a=rid:5 send pt=99,102;max-br=64000 951 a=rid:6 send pt=100,97,101,102 952 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid 953 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 954 a=simulcast:send 5;6 955 m=video 49600 RTP/AVPF 103 104 105 106 107 956 a=mid:bar 957 a=rtpmap:103 H264/90000 958 a=rtpmap:104 VP8/90000 959 a=rtpmap:105 rtx/90000 960 a=rtpmap:106 rtx/90000 961 a=rtpmap:107 flexfec/90000 962 a=fmtp:103 profile-level-id=42c00d;max-fs=3600;max-mbps=108000 963 a=fmtp:104 max-fs=3600; max-fr=30 964 a=fmtp:105 apt=103;rtx-time=200 965 a=fmtp:106 apt=104;rtx-time=200 966 a=fmtp:107 repair-window=2000 967 a=rid:1 send pt=103;max-width=1280;max-height=720;max-fps=30 968 a=rid:2 send pt=104;max-width=1280;max-height=720;max-fps=30 969 a=rid:3 send pt=103;max-width=640;max-height=360;max-br=300000 970 a=rid:4 send pt=104;max-width=640;max-height=360;max-br=300000 971 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid 972 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 973 a=rtcp-fb:* ccm pause nowait 974 a=simulcast:send 1,2;3,4 976 6. RTP Aspects 978 This section discusses what the different entities in a simulcast 979 media path can expect to happen on RTP level. This is explored from 980 source to sink by starting in an endpoint with a media source that is 981 simulcasted to an RTP middlebox. That RTP middlebox sends media 982 sources both to other RTP middleboxes (cascaded middleboxes), as well 983 as selecting some simulcast format of the media source and sending it 984 to receiving endpoints. Different types of RTP middleboxes and their 985 usage of the different simulcast formats results in several different 986 behaviors. 988 6.1. Outgoing from Endpoint with Media Source 990 The most straightforward simulcast case is the RTP streams being 991 emitted from the endpoint that originates a media source. When 992 simulcast has been negotiated in the sending direction, the endpoint 993 can transmit up to the number of RTP streams needed for the 994 negotiated simulcast streams for that media source. Each RTP stream 995 (SSRC) is identified by associating (Section 5.5) it with an 996 RtpStreamId SDES item, transmitted in RTCP and possibly also as an 997 RTP header extension. In cases where multiple media sources have 998 been negotiated for the same RTP session and thus BUNDLE 999 [I-D.ietf-mmusic-sdp-bundle-negotiation] is used, also the MID SDES 1000 item will be sent similarly to the RtpStreamId. 1002 Each RTP stream might not be continuously transmitted due to any of 1003 the following reasons; temporarily paused using Pause/Resume 1004 [RFC7728], sender side application logic temporarily pausing it, or 1005 lack of network resources to transmit this simulcast stream. 1006 However, all simulcast streams that have been negotiated have active 1007 and maintained SSRC (at least in regular RTCP reports), even if no 1008 RTP packets are currently transmitted. The relation between an RTP 1009 Stream (SSRC) and a particular simulcast stream is not expected to 1010 change, except in exceptional situations such as SSRC collisions. At 1011 SSRC changes, the usage of MID and RtpStreamId should enable the 1012 receiver to correctly identify the RTP streams even after an SSRC 1013 change. 1015 6.2. RTP Middlebox to Receiver 1017 RTP streams in a multi-party RTP session can be used in multiple 1018 different ways, when the session utilizes simulcast at least on the 1019 media source to middlebox legs. This is to a large degree due to the 1020 different RTP middlebox behaviors, but also the needs of the 1021 application. This text assumes that the RTP middlebox will select a 1022 media source and choose which simulcast stream for that media source 1023 to deliver to a specific receiver. In many cases, at most one 1024 simulcast stream per media source will be forwarded to a particular 1025 receiver at any instant in time, even if the selected simulcast 1026 stream may vary. For cases where this does not hold due to 1027 application needs, then the RTP stream aspects will fall under the 1028 middlebox to middlebox case Section 6.3. 1030 The selection of which simulcast streams to forward towards the 1031 receiver, is application specific. However, in conferencing 1032 applications, active speaker selection is common. In case the number 1033 of media sources possible to forward, N, is less than the total 1034 amount of media sources available in an multi-media session, the 1035 current and previous speakers (up to N in total) are often the ones 1036 forwarded. To avoid the need for media specific processing to 1037 determine the current speaker(s) in the RTP middlebox, the endpoint 1038 providing a media source may include meta data, such as the RTP 1039 Header Extension for Client-to-Mixer Audio Level Indication 1040 [RFC6464]. 1042 The possibilities for stream switching are media type specific, but 1043 for media types with significant interframe dependencies in the 1044 encoding, like most video coding, the switching needs to be made at 1045 suitable switching points in the media stream that breaks or 1046 otherwise deals with the dependency structure. Even if switching 1047 points can be included periodically, it is common to use mechanisms 1048 like Full Intra Requests [RFC5104] to request switching points from 1049 the endpoint performing the encoding of the media source. 1051 Inclusion of the RtpStreamId SDES item for an SSRC in the middlebox 1052 to receiver direction should only occur when use of RtpStreamId has 1053 been negotiated in that direction. It is worth noting that one can 1054 signal multiple RtpStreamIds when simulcast signalling indicates only 1055 a single simulcast stream, allowing one to use all of the 1056 RtpStreamIds as alternatives for that simulcast stream. One reason 1057 for including the RtpStreamId in the middlebox to receiver direction 1058 for an RTP stream is to let the receiver know which restrictions 1059 apply to the currently delivered RTP stream. In case the RtpStreamId 1060 is negotiated to be used, it is important to remember that the used 1061 identifiers will be specific to each signalling session. Even if the 1062 central entity can attempt to coordinate, it is likely that the 1063 RtpStreamIds need to be translated to the leg specific values. The 1064 below cases will have as base line that RtpStreamId is not used in 1065 the mixer to receiver direction. 1067 6.2.1. Media-Switching Mixer 1069 This section discusses the behavior in cases where the RTP middlebox 1070 behaves like the Media-Switching Mixer (Section 3.6.2) in RTP 1071 Topologies [RFC7667]. The fundamental aspect here is that the media 1072 sources delivered from the middlebox will be the mixer's conceptual 1073 or functional ones. For example, one media source may be the main 1074 speaker in high resolution video, while a number of other media 1075 sources are thumbnails of each participant. 1077 The above results in that the RTP stream produced by the mixer is one 1078 that switches between a number of received incoming RTP streams for 1079 different media sources and in different simulcast versions. The 1080 mixer selects the media source to be sent as one of the RTP streams, 1081 and then selects among the available simulcast streams for the most 1082 appropriate one. The selection criteria include available bandwidth 1083 on the mixer to receiver path and restrictions based on the 1084 functional usage of the RTP stream delivered to the receiver. As an 1085 example of the latter, it is unnecessary to forward a full HD video 1086 to a receiver if the display area is just a thumbnail. Thus, 1087 restrictions may exist to not allow some simulcast streams to be 1088 forwarded for some of the mixer's media sources. 1090 This will result in a single RTP stream being used for each of the 1091 RTP mixer's media sources. This RTP stream is at any point in time a 1092 selection of one particular RTP stream arriving to the mixer, where 1093 the RTP header field values are rewritten to provide a consistent, 1094 single RTP stream. If the RTP mixer doesn't receive any incoming 1095 stream matched to this media source, the SSRC will not transmit, but 1096 be kept alive using RTCP. The SSRC and thus RTP stream for the 1097 mixer's media source is expected to be long term stable. It will 1098 only be changed by signalling or other disruptive events. Note that 1099 although the above talks about a single RTP stream, there can in some 1100 cases be multiple RTP streams carrying the selected simulcast stream 1101 for the originating media source, including redundancy or other 1102 auxiliary RTP streams. 1104 The mixer may communicate the identity of the originating media 1105 source to the receiver by including the CSRC field with the 1106 originating media source's SSRC value. Note that due to the 1107 possibility that the RTP mixer switches between simulcast versions of 1108 the media source, the CSRC value may change, even if the media source 1109 is kept the same. 1111 It is important to note that any MID SDES item from the originating 1112 media source needs to be removed and not be associated with the RTP 1113 stream's SSRC. That is, there is nothing in the signalling between 1114 the mixer and the receiver that is structured around the originating 1115 media sources, only the mixer's media sources. If they would be 1116 associated with the SSRC, the receiver would likely believe that 1117 there has been an SSRC collision, and that the RTP stream is spurious 1118 as it doesn't carry the identifiers used to relate it to the correct 1119 context. However, this is not true for CSRC values, as long as they 1120 are never used as SSRC. In these cases one could provide CNAME and 1121 MID as SDES items. A receiver could use this to determine which CSRC 1122 values that are associated with the same originating media source. 1124 If RtpStreamIds are used in the scenario described by this section, 1125 it should be noted that the RtpStreamId on a particular SSRC will 1126 change based on the actual simulcast stream selected for switching. 1127 These RtpStreamId identifiers will be local to this leg's signalling 1128 context. In addition, the defined RtpStreamIds and their parameters 1129 need to cover all the media sources and simulcast streams received by 1130 the RTP mixer that can be switched into this media source, sent by 1131 the RTP mixer. 1133 6.2.2. Selective Forwarding Middlebox 1135 This section discusses the behavior in cases where the RTP middlebox 1136 behaves like the Selective Forwarding Middlebox (Section 3.7) in RTP 1137 Topologies [RFC7667]. Applications for this type of RTP middlebox 1138 results in that each originating media source will have a 1139 corresponding media source on the leg between the middlebox and the 1140 receiver. A Selective Forwarding Middlebox (SFM) could go as far as 1141 exposing all the simulcast streams for an media source, however this 1142 section will focus on having a single simulcast stream that can 1143 contain any of the simulcast formats. This section will assume that 1144 the SFM projection mechanism works on media source level, and maps 1145 one of the media source's simulcast streams onto one RTP stream from 1146 the SFM to the receiver. 1148 This usage will result in that the individual RTP stream(s) for one 1149 media source can switch between being active to paused, based on the 1150 subset of media sources the SFM wants to provide the receiver for the 1151 moment. With SFMs there exist no reasons to use CSRC to indicate the 1152 originating stream, as there is a one to one media source mapping. 1153 If the application requires knowing the simulcast version received to 1154 function well, then RtpStreamId should be negotiated on the SFM to 1155 receiver leg. Which simulcast stream that is being forwarded is not 1156 made explicit unless RtpStreamId is used on the leg. 1158 Any MID SDES items being sent by the SFM to the receiver are only 1159 those agreed between the SFM and the receiver, and no MID values from 1160 the originating side of the SFM are to be forwarded. 1162 A SFM could expose corresponding RTP streams for all the media 1163 sources and their simulcast streams, and then for any media source 1164 that is to be provided forward one selected simulcast stream. 1165 However, this is not recommended as it would unnecessarily increase 1166 the number of RTP streams and require the receiver to timely detect 1167 switching between simulcast streams. The above usage requires the 1168 same SFM functionality for switching, while avoiding the 1169 uncertainties of timely detecting that a RTP stream ends. The 1170 benefit would be that the received simulcast stream would be 1171 implicitly provided by which RTP stream would be active for a media 1172 source. However, using RtpStreamId to make this explicit also 1173 exposes which alternative format is used. The conclusion is that 1174 using one RTP stream per simulcast stream is unnecessary. The issue 1175 with timely detecting end of streams, independent if they are stopped 1176 temporarily or long term, is that there is no explicit indication 1177 that the transmission has intentionally been stopped. The RTCP based 1178 Pause and Resume mechanism [RFC7728] includes a PAUSED indication 1179 that provides the last RTP sequence number transmitted prior to the 1180 pause. Due to usage, the timeliness of this solution depends on when 1181 delivery using RTCP can occur in relation to the transmission of the 1182 last RTP packet. If no explicit information is provided at all, then 1183 detection based on non increasing RTCP SR field values and timers 1184 need to be used to determine pause in RTP packet delivery. This 1185 results in that one can usually not determine when the last RTP 1186 packet arrives (if it arrives) that this will be the last. That it 1187 was the last is something that one learns later. 1189 6.3. RTP Middlebox to RTP Middlebox 1191 This relates to the transmission of simulcast streams between RTP 1192 middleboxes or other usages where one wants to enable the delivery of 1193 multiple simultaneous simulcast streams per media source, but the 1194 transmitting entity is not the originating endpoint. For a 1195 particular direction between middlebox A and B, this looks very 1196 similar to the originating to middlebox case on a media source basis. 1197 However, in this case there is usually multiple media sources, 1198 originating from multiple endpoints. This can create situations 1199 where limitations in the number of simultaneously received media 1200 streams can arise, for example due to limitation in network 1201 bandwidth. In this case, a subset of not only the simulcast streams, 1202 but also media sources can be selected. This results in that 1203 individual RTP streams can be become paused at any point and later 1204 being resumed based on various criteria. 1206 The MIDs used between A and B are the ones agreed between these two 1207 identities in signalling. The RtpStreamId values will also be 1208 provided to ensure explicit information about which simulcast stream 1209 they are. The RTP stream to MID and RtpStreamId associations should 1210 here be long term stable. 1212 7. Network Aspects 1214 Simulcast is in this memo defined as the act of sending multiple 1215 alternative encoded streams of the same underlying media source. 1216 When transmitting multiple independent streams that originate from 1217 the same source, it could potentially be done in several different 1218 ways using RTP. A general discussion on considerations for use of 1219 the different RTP multiplexing alternatives can be found in 1220 Guidelines for Multiplexing in RTP 1221 [I-D.ietf-avtcore-multiplex-guidelines]. Discussion and 1222 clarification on how to handle multiple streams in an RTP session can 1223 be found in [RFC8108]. 1225 The network aspects that are relevant for simulcast are: 1227 Quality of Service: When using simulcast it might be of interest to 1228 prioritize a particular simulcast stream, rather than applying 1229 equal treatment to all streams. For example, lower bitrate 1230 streams may be prioritized over higher bitrate streams to minimize 1231 congestion or packet losses in the low bitrate streams. Thus, 1232 there is a benefit to use a simulcast solution with good QoS 1233 support. 1235 NAT/FW Traversal: Using multiple RTP sessions incurs more cost for 1236 NAT/FW traversal unless they can re-use the same transport flow, 1237 which can be achieved by Multiplexing Negotiation Using SDP Port 1238 Numbers [I-D.ietf-mmusic-sdp-bundle-negotiation]. 1240 7.1. Bitrate Adaptation 1242 Use of multiple simulcast streams can require a significant amount of 1243 network resources. The aggregate bandwidth for all simulcast streams 1244 for a media source (and thus SDP media description) is bounded by any 1245 SDP "b=" line applicable to that media source. It is assumed that a 1246 suitable congestion control mechanism is used by the application to 1247 ensure that it doesn't cause persistent congestion. If the amount of 1248 available network resources varies during an RTP session such that it 1249 does not match what is negotiated in SDP, the bitrate used by the 1250 different simulcast streams may have to be reduced dynamically. When 1251 a simulcasting media source uses a single media transport for all of 1252 the simulcast streams, it is likely that a joint congestion control 1253 across all simulcast streams is used for that media source. What 1254 simulcast streams to prioritize when allocating available bitrate 1255 among the simulcast streams in such adaptation SHOULD be taken from 1256 the simulcast stream order on the "a=simulcast" line and ordering of 1257 alternative simulcast formats Section 5.2. Simulcast streams that 1258 have pause/resume capability and that would be given such low bitrate 1259 by the adaptation process that they are considered not really useful 1260 can be temporarily paused until the limiting condition clears. 1262 8. Limitation 1264 The chosen approach has a limitation that relates to the use of a 1265 single RTP session for all simulcast formats of a media source, which 1266 comes from sending all simulcast streams related to a media source 1267 under the same SDP media description. 1269 It is not possible to use different simulcast streams on different 1270 media transports, limiting the possibilities to apply different QoS 1271 to different simulcast streams. When using unicast, QoS mechanisms 1272 based on individual packet marking are feasible, since they do not 1273 require separation of simulcast streams into different RTP sessions 1274 to apply different QoS. 1276 It is also not possible to separate different simulcast streams into 1277 different multicast groups to allow a multicast receiver to pick the 1278 stream it wants, rather than receive all of them. In this case, the 1279 only reasonable implementation is to use different RTP sessions for 1280 each multicast group so that reporting and other RTCP functions 1281 operate as intended. Such simulcast usage in multicast context is 1282 out of scope for the current document and would require additional 1283 specification. 1285 9. IANA Considerations 1287 This document requests to register a new media-level SDP attribute, 1288 "simulcast", in the "att-field (media level only)" registry within 1289 the SDP parameters registry, according to the procedures of [RFC4566] 1290 and [I-D.ietf-mmusic-sdp-mux-attributes]. 1292 Contact name, email: The IESG (iesg@ietf.org) 1294 Attribute name: simulcast 1296 Long-form attribute name: Simulcast stream description 1298 Charset dependent: No 1300 Attribute value: sc-value; see Section 5.1 of RFC XXXX. 1302 Purpose: Signals simulcast capability for a set of RTP streams 1304 MUX category: NORMAL 1305 Note to RFC Editor: Please replace "RFC XXXX" with the assigned 1306 number of this RFC. 1308 10. Security Considerations 1310 The simulcast capability, configuration attributes, and parameters 1311 are vulnerable to attacks in signaling. 1313 A false inclusion of the "a=simulcast" attribute may result in 1314 simultaneous transmission of multiple RTP streams that would 1315 otherwise not be generated. The impact is limited by the media 1316 description joint bandwidth, shared by all simulcast streams 1317 irrespective of their number. There may however be a large number of 1318 unwanted RTP streams that will impact the share of bandwidth 1319 allocated for the originally wanted RTP stream. 1321 A hostile removal of the "a=simulcast" attribute will result in 1322 simulcast not being used. 1324 Neither of the above will likely have any major consequences and can 1325 be mitigated by signaling that is at least integrity and source 1326 authenticated to prevent an attacker to change it. 1328 Security considerations related to the use of "a=rid" and the 1329 RtpStreamId SDES item is covered in [I-D.ietf-mmusic-rid] and 1330 [I-D.ietf-avtext-rid]. There are no additional security concerns 1331 related to their use in this specification. 1333 11. Contributors 1335 Morgan Lindqvist and Fredrik Jansson, both from Ericsson, have 1336 contributed with important material to the first versions of this 1337 document. Robert Hansen and Cullen Jennings, from Cisco, Peter 1338 Thatcher, from Google, and Adam Roach, from Mozilla, contributed 1339 significantly to subsequent versions. 1341 12. Acknowledgements 1343 The authors would like to thank Bernard Aboba, Thomas Belling, Roni 1344 Even, Adam Roach, Inaki Baz Castillo, Paul Kyzivat, and Arun 1345 Arunachalam for the feedback they provided during the development of 1346 this document. 1348 13. References 1349 13.1. Normative References 1351 [I-D.ietf-avtext-rid] 1352 Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream 1353 Identifier Source Description (SDES)", draft-ietf-avtext- 1354 rid-09 (work in progress), October 2016. 1356 [I-D.ietf-mmusic-rid] 1357 Roach, A., "RTP Payload Format Restrictions", draft-ietf- 1358 mmusic-rid-14 (work in progress), February 2018. 1360 [I-D.ietf-mmusic-sdp-bundle-negotiation] 1361 Holmberg, C., Alvestrand, H., and C. Jennings, 1362 "Negotiating Media Multiplexing Using the Session 1363 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 1364 negotiation-49 (work in progress), March 2018. 1366 [I-D.ietf-mmusic-sdp-mux-attributes] 1367 Nandakumar, S., "A Framework for SDP Attributes when 1368 Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-17 1369 (work in progress), February 2018. 1371 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1372 Requirement Levels", BCP 14, RFC 2119, 1373 DOI 10.17487/RFC2119, March 1997, 1374 . 1376 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1377 Jacobson, "RTP: A Transport Protocol for Real-Time 1378 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1379 July 2003, . 1381 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1382 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 1383 July 2006, . 1385 [RFC5234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax 1386 Specifications: ABNF", STD 68, RFC 5234, 1387 DOI 10.17487/RFC5234, January 2008, 1388 . 1390 [RFC7728] Burman, B., Akram, A., Even, R., and M. Westerlund, "RTP 1391 Stream Pause and Resume", RFC 7728, DOI 10.17487/RFC7728, 1392 February 2016, . 1394 [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 1395 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, 1396 May 2017, . 1398 13.2. Informative References 1400 [I-D.ietf-avtcore-multiplex-guidelines] 1401 Westerlund, M., Burman, B., Perkins, C., Alvestrand, H., 1402 Even, R., and H. Zheng, "Guidelines for using the 1403 Multiplexing Features of RTP to Support Multiple Media 1404 Streams", draft-ietf-avtcore-multiplex-guidelines-05 (work 1405 in progress), October 2017. 1407 [I-D.ietf-payload-flexible-fec-scheme] 1408 Zanaty, M., Singh, V., Begen, A., and G. Mandyam, "RTP 1409 Payload Format for Flexible Forward Error Correction 1410 (FEC)", draft-ietf-payload-flexible-fec-scheme-07 (work in 1411 progress), March 2018. 1413 [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., 1414 Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- 1415 Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, 1416 DOI 10.17487/RFC2198, September 1997, 1417 . 1419 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 1420 with Session Description Protocol (SDP)", RFC 3264, 1421 DOI 10.17487/RFC3264, June 2002, 1422 . 1424 [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for 1425 Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389, 1426 September 2002, . 1428 [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. 1429 Hakenberg, "RTP Retransmission Payload Format", RFC 4588, 1430 DOI 10.17487/RFC4588, July 2006, 1431 . 1433 [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF 1434 Digits, Telephony Tones, and Telephony Signals", RFC 4733, 1435 DOI 10.17487/RFC4733, December 2006, 1436 . 1438 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 1439 "Codec Control Messages in the RTP Audio-Visual Profile 1440 with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, 1441 February 2008, . 1443 [RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error 1444 Correction", RFC 5109, DOI 10.17487/RFC5109, December 1445 2007, . 1447 [RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding 1448 Dependency in the Session Description Protocol (SDP)", 1449 RFC 5583, DOI 10.17487/RFC5583, July 2009, 1450 . 1452 [RFC6184] Wang, Y., Even, R., Kristensen, T., and R. Jesup, "RTP 1453 Payload Format for H.264 Video", RFC 6184, 1454 DOI 10.17487/RFC6184, May 2011, 1455 . 1457 [RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, 1458 "RTP Payload Format for Scalable Video Coding", RFC 6190, 1459 DOI 10.17487/RFC6190, May 2011, 1460 . 1462 [RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image 1463 Attributes in the Session Description Protocol (SDP)", 1464 RFC 6236, DOI 10.17487/RFC6236, May 2011, 1465 . 1467 [RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time 1468 Transport Protocol (RTP) Header Extension for Client-to- 1469 Mixer Audio Level Indication", RFC 6464, 1470 DOI 10.17487/RFC6464, December 2011, 1471 . 1473 [RFC7104] Begen, A., Cai, Y., and H. Ou, "Duplication Grouping 1474 Semantics in the Session Description Protocol", RFC 7104, 1475 DOI 10.17487/RFC7104, January 2014, 1476 . 1478 [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and 1479 B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms 1480 for Real-Time Transport Protocol (RTP) Sources", RFC 7656, 1481 DOI 10.17487/RFC7656, November 2015, 1482 . 1484 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 1485 DOI 10.17487/RFC7667, November 2015, 1486 . 1488 [RFC7741] Westin, P., Lundin, H., Glover, M., Uberti, J., and F. 1489 Galligan, "RTP Payload Format for VP8 Video", RFC 7741, 1490 DOI 10.17487/RFC7741, March 2016, 1491 . 1493 [RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, 1494 "Sending Multiple RTP Streams in a Single RTP Session", 1495 RFC 8108, DOI 10.17487/RFC8108, March 2017, 1496 . 1498 [RFC8285] Singer, D., Desineni, H., and R. Even, Ed., "A General 1499 Mechanism for RTP Header Extensions", RFC 8285, 1500 DOI 10.17487/RFC8285, October 2017, 1501 . 1503 Appendix A. Requirements 1505 The following requirements are met by the defined solution to support 1506 the use cases (Section 3): 1508 REQ-1: Identification: 1510 REQ-1.1: It must be possible to identify a set of simulcasted RTP 1511 streams as originating from the same media source in SDP 1512 signaling. 1514 REQ-1.2: An RTP endpoint must be capable of identifying the 1515 simulcast stream a received RTP stream is associated with, 1516 knowing the content of the SDP signalling. 1518 REQ-2: Transport usage. The solution must work when using: 1520 REQ-2.1: Legacy SDP with separate media transports per SDP media 1521 description. 1523 REQ-2.2: Bundled [I-D.ietf-mmusic-sdp-bundle-negotiation] SDP 1524 media descriptions. 1526 REQ-3: Capability negotiation. It must be possible that: 1528 REQ-3.1: Sender can express capability of sending simulcast. 1530 REQ-3.2: Receiver can express capability of receiving simulcast. 1532 REQ-3.3: Sender can express maximum number of simulcast streams 1533 that can be provided. 1535 REQ-3.4: Receiver can express maximum number of simulcast streams 1536 that can be received. 1538 REQ-3.5: Sender can detail the characteristics of the simulcast 1539 streams that can be provided. 1541 REQ-3.6: Receiver can detail the characteristics of the simulcast 1542 streams that it prefers to receive. 1544 REQ-4: Distinguishing features. It must be possible to have 1545 different simulcast streams use different codec parameters, as can 1546 be expressed by SDP format values and RTP payload types. 1548 REQ-5: Compatibility. It must be possible to use simulcast in 1549 combination with other RTP mechanisms that generate additional RTP 1550 streams: 1552 REQ-5.1: RTP Retransmission [RFC4588]. 1554 REQ-5.2: RTP Forward Error Correction [RFC5109]. 1556 REQ-5.3: Related payload types such as audio Comfort Noise and/or 1557 DTMF. 1559 REQ-5.4: A single simulcast stream can consist of multiple RTP 1560 streams, to support codecs where a dependent stream is 1561 dependent on a set of encoded and dependent streams, each 1562 potentially carried in their own RTP stream. 1564 REQ-6: Interoperability. The solution must be possible to use in: 1566 REQ-6.1: Interworking with non-simulcast legacy clients using a 1567 single media source per media type. 1569 REQ-6.2: WebRTC environment with a single media source per SDP 1570 media description. 1572 Appendix B. Changes From Earlier Versions 1574 NOTE TO RFC EDITOR: Please remove this section prior to publication. 1576 B.1. Modifications Between WG Version -11 and -12 1578 o Modified Normative statement regarding RTP stream duplication in 1579 Section 5.2. 1581 o Clarified assumption about use of congestion control by 1582 applications. 1584 o Changed to use RFC 8174 boilerplate instead of RFC 2119. 1586 o Clarified explanation of syntax for simulcast attribute in 1587 Section 4. 1589 o Editorial clarification in Section 5.2 and 5.3.2. 1591 o Various minor editorials and nits. 1593 B.2. Modifications Between WG Version -10 and -11 1595 o Added new SDP example section on Simulcast and Redundancy, 1596 including both RED (RFC2198), RTP RTX (RFC4588), and FEC (draft- 1597 ietf-payload-flexible-fec-scheme). 1599 o Removed restriction that "related" payload formats in an RTP 1600 stream (such as CN and DTMF) must not have their own rid-id, since 1601 there is no reason to forbid this and corresponding clarification 1602 is made in draft-ietf-mmusic-rid. 1604 o Removed any mention of source-specific signaling and the reference 1605 to RFC5576, since draft-ietf-mmusic-rid is not defined for source- 1606 specific signaling. 1608 o Changed some SDP examples to use a=rid restrictions instead of 1609 a=imageattr. 1611 o Changed reference from the obsoleted RFC 5285 to RFC 8285. 1613 B.3. Modifications Between WG Version -09 and -10 1615 o Amended overview section with a bit more explanation on the 1616 examples, and added an rid-id alternative for one of the streams. 1618 o Removed SCID also from the Terminology section, which was 1619 forgotten in -09 when changing SCID to rid-id. 1621 B.4. Modifications Between WG Version -08 and -09 1623 o Changed SCID to rid-id, to align with ietf-draft-mmusic-rid 1624 naming. 1626 o Changed Overview to be based on examples and shortened it. 1628 o Changed semantics of initially paused rid-id in modified SDP 1629 offers from requiring it to follow actual RFC 7728 pause state to 1630 an informational offerer's opinion at the time of offer creation, 1631 not in any way overriding or amending RFC 7728 signaling. 1633 o Replaced text on ignoring all but the first of multiple 1634 "a=simulcast" lines in a media description with mandating that at 1635 most one "a=simulcast" line is included. 1637 o Clarified with a note that, for the case it is clear from the SDP 1638 that RTP PT uniquely maps to RtpStreamId, an RTP receiver can use 1639 RTP PT to relate simulcast streams. 1641 o Moved Section 4 Requirements to become Appendix A. 1643 o Editorial corrections and clarifications. 1645 B.5. Modifications Between WG Version -07 and -08 1647 o Correcting syntax of SDP examples in section 6.6.1, as found by 1648 Inaki Baz Castillo. 1650 o Changing ABNF to only define the sc-value, not the SDP attribute 1651 itself, as suggested by Paul Kyzivat. 1653 o Changing I-D reference to newly published RFC 8108. 1655 o Adding list of modifications between -06 and -07. 1657 B.6. Modifications Between WG Version -06 and -07 1659 o A scope clarification, as result of the discussion with Roni Even. 1661 o A reformulation of the identification requirements for simulcast 1662 stream. 1664 o Correcting the statement related to source specific signalling 1665 (RFC 5576) to address Roni Even's comment. 1667 o Update of the last paragraph in Section 6.2 regarding simulcast 1668 stream differences as well as forbidding multiple instances of the 1669 same SCID within a single a=simulcast line. 1671 o Removal of note in Section 6.4 as result of issue raised by Roni 1672 Even. 1674 o Use of "m=" has been changed to media description and a few other 1675 editorial improvements and clarifications. 1677 B.7. Modifications Between WG Version -05 and -06 1679 o Added section on RTP Aspects 1681 o Added a requirement (5-4) on that capability exchange must be 1682 capable of handling multi RTP stream cases. 1684 o Added extmap attribute also on first signalling example as it is a 1685 recommended to use mechanism. 1687 o Clarified the definition of the simulcast attribute and how 1688 simulcast streams relates to simulcast formats and SCIDs. 1690 o Updated References list and moved around some references between 1691 informative and normative categories. 1693 o Editorial improvements and corrections. 1695 B.8. Modifications Between WG Version -04 and -05 1697 o Aligned with recent changes in draft-ietf-mmusic-rid and draft- 1698 ietf-avtext-rid. 1700 o Modified the SDP offer/answer section to follow the generally 1701 accepted structure, also adding a brief text on modifying the 1702 session that is aligned with draft-ietf-mmusic-rid. 1704 o Improved text around simulcast stream identification (as opposed 1705 to the simulcast stream itself) to consistently use the acronym 1706 SCID and defined that in the Terminology section. 1708 o Changed references for RTP-level pause/resume and VP8 payload 1709 format that are now published as RFC. 1711 o Improved IANA registration text. 1713 o Removed unused reference to draft-ietf-payload-flexible-fec- 1714 scheme. 1716 o Editorial improvements and corrections. 1718 B.9. Modifications Between WG Version -03 and -04 1720 o Changed to only use RID identification, as was consensus during 1721 IETF 94. 1723 o ABNF improvements. 1725 o Clarified offer-answer rules for initially paused streams. 1727 o Changed references for RTP topologies and RTP taxonomy documents 1728 that are now published as RFC. 1730 o Added reference to the new RID draft in AVTEXT. 1732 o Re-structured section 6 to provide an easy reference by the 1733 updated IANA section. 1735 o Added a sub-section 7.1 with a discussion of bitrate adaptation. 1737 o Editorial improvements. 1739 B.10. Modifications Between WG Version -02 and -03 1741 o Removed text on multicast / broadcast from use cases, since it is 1742 not supported by the solution. 1744 o Removed explicit references to unified plan draft. 1746 o Added possibility to initiate simulcast streams in paused mode. 1748 o Enabled an offerer to offer multiple stream identification (pt or 1749 rid) methods and have the answerer choose which to use. 1751 o Added a preference indication also in send direction offers. 1753 o Added a section on limitations of the current proposal, including 1754 identification method specific limitations. 1756 B.11. Modifications Between WG Version -01 and -02 1758 o Relying on the new RID solution for codec constraints and 1759 configuration identification. This has resulted in changes in 1760 syntax to identify if pt or RID is used to describe the simulcast 1761 stream. 1763 o Renamed simulcast version and simulcast version alternative to 1764 simulcast stream and simulcast format respectively, and improved 1765 definitions for them. 1767 o Clarification that it is possible to switch between simulcast 1768 version alternatives, but that only a single one be used at any 1769 point in time. 1771 o Changed the definition so that ordering of simulcast formats for a 1772 specific simulcast stream do have a preference order. 1774 B.12. Modifications Between WG Version -00 and -01 1776 o No changes. Only preventing expiry. 1778 B.13. Modifications Between Individual Version -00 and WG Version -00 1780 o Added this appendix. 1782 Authors' Addresses 1784 Bo Burman 1785 Ericsson 1786 Gronlandsgatan 31 1787 SE-164 60 Stockholm 1788 Sweden 1790 Email: bo.burman@ericsson.com 1792 Magnus Westerlund 1793 Ericsson 1794 Farogatan 2 1795 SE-164 80 Stockholm 1796 Sweden 1798 Phone: +46 10 714 82 87 1799 Email: magnus.westerlund@ericsson.com 1801 Suhas Nandakumar 1802 Cisco 1803 170 West Tasman Drive 1804 San Jose, CA 95134 1805 USA 1807 Email: snandaku@cisco.com 1809 Mo Zanaty 1810 Cisco 1811 170 West Tasman Drive 1812 San Jose, CA 95134 1813 USA 1815 Email: mzanaty@cisco.com