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Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Outdated reference: A later version (-54) exists of draft-ietf-mmusic-sdp-bundle-negotiation-52 == Outdated reference: A later version (-19) exists of draft-ietf-mmusic-sdp-mux-attributes-17 ** Obsolete normative reference: RFC 4566 (Obsoleted by RFC 8866) == Outdated reference: A later version (-12) exists of draft-ietf-avtcore-multiplex-guidelines-05 == Outdated reference: A later version (-20) exists of draft-ietf-payload-flexible-fec-scheme-07 Summary: 1 error (**), 0 flaws (~~), 5 warnings (==), 1 comment (--). 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: December 29, 2018 S. Nandakumar 6 M. Zanaty 7 Cisco 8 June 27, 2018 10 Using Simulcast in SDP and RTP Sessions 11 draft-ietf-mmusic-sdp-simulcast-13 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 December 29, 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 . . . . . . . . . . . . . . . . . . . . 10 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 . . . . . . . . . . . . . . . 16 79 5.6. Signaling Examples . . . . . . . . . . . . . . . . . . . 16 80 5.6.1. Single-Source Client . . . . . . . . . . . . . . . . 17 81 5.6.2. Multi-Source Client . . . . . . . . . . . . . . . . . 18 82 5.6.3. Simulcast and Redundancy . . . . . . . . . . . . . . 21 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 -12 and -13 . . . . . . 35 102 B.2. Modifications Between WG Version -11 and -12 . . . . . . 36 103 B.3. Modifications Between WG Version -10 and -11 . . . . . . 36 104 B.4. Modifications Between WG Version -09 and -10 . . . . . . 37 105 B.5. Modifications Between WG Version -08 and -09 . . . . . . 37 106 B.6. Modifications Between WG Version -07 and -08 . . . . . . 37 107 B.7. Modifications Between WG Version -06 and -07 . . . . . . 38 108 B.8. Modifications Between WG Version -05 and -06 . . . . . . 38 109 B.9. Modifications Between WG Version -04 and -05 . . . . . . 38 110 B.10. Modifications Between WG Version -03 and -04 . . . . . . 39 111 B.11. Modifications Between WG Version -02 and -03 . . . . . . 39 112 B.12. Modifications Between WG Version -01 and -02 . . . . . . 40 113 B.13. Modifications Between WG Version -00 and -01 . . . . . . 40 114 B.14. Modifications Between Individual Version -00 and WG 115 Version -00 . . . . . . . . . . . . . . . . . . . . . . . 40 116 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 40 118 1. Introduction 120 Most of today's multiparty video conference solutions make use of 121 centralized servers to reduce the bandwidth and CPU consumption in 122 the endpoints. Those servers receive RTP streams from each 123 participant and send some suitable set of possibly modified RTP 124 streams to the rest of the participants, which usually have 125 heterogeneous capabilities (screen size, CPU, bandwidth, codec, etc). 126 One of the biggest issues is how to perform RTP stream adaptation to 127 different participants' constraints with the minimum possible impact 128 on both video quality and server performance. 130 Simulcast is defined in this memo as the act of simultaneously 131 sending multiple different encoded streams of the same media source, 132 e.g. the same video source encoded with different video encoder types 133 or image resolutions. This can be done in several ways and for 134 different purposes. This document focuses on the case where it is 135 desirable to provide a media source as multiple encoded streams over 136 RTP [RFC3550] towards an intermediary so that the intermediary can 137 provide the wanted functionality by selecting which RTP stream(s) to 138 forward to other participants in the session, and more specifically 139 how the identification and grouping of the involved RTP streams are 140 done. 142 The intended scope of the defined mechanism is to support negotiation 143 and usage of simulcast when using SDP offer/answer and media 144 transport over RTP. The media transport topologies considered are 145 point to point RTP sessions as well as centralized multi-party RTP 146 sessions, where a media sender will provide the simulcasted streams 147 to an RTP middlebox or endpoint, and middleboxes may further 148 distribute the simulcast streams to other middleboxes or endpoints. 149 Simulcast could, as part of a distributed multi-party scenario, be 150 used point-to-point between middleboxes. Usage of multicast or 151 broadcast transport is out of scope and left for future extensions. 153 This document describes a few scenarios that motivate the use of 154 simulcast, and also defines the needed RTP/RTCP and SDP signaling for 155 it. 157 2. Definitions 159 2.1. Terminology 161 This document makes use of the terminology defined in RTP Taxonomy 162 [RFC7656], and RTP Topologies [RFC7667]. The following terms are 163 especially noted or here defined: 165 RTP Mixer: An RTP middle node, defined in [RFC7667] (Section 3.6 to 166 3.9). 168 RTP Session: An association among a group of participants 169 communicating with RTP, as defined in [RFC3550] and amended by 170 [RFC7656]. 172 RTP Stream: A stream of RTP packets containing media data, as 173 defined in [RFC7656]. 175 RTP Switch: A common short term for the terms "switching RTP mixer", 176 "source projecting middlebox", and "video switching MCU" as 177 discussed in [RFC7667]. 179 Simulcast Stream: One encoded stream or dependent stream from a set 180 of concurrently transmitted encoded streams and optional dependent 181 streams, all sharing a common media source, as defined in 182 [RFC7656]. For example, HD and thumbnail video simulcast versions 183 of a single media source sent concurrently as separate RTP 184 Streams. 186 Simulcast Format: Different formats of a simulcast stream serve the 187 same purpose as alternative RTP payload types in non-simulcast 188 SDP: to allow multiple alternative media formats for a given RTP 189 stream. As for multiple RTP payload types on the m-line in offer/ 190 answer [RFC3264], any one of the negotiated alternative formats 191 can be used in a single RTP stream at a given point in time, but 192 not more than one (based on RTP timestamp). What format is used 193 can change dynamically from one RTP packet to another. 195 2.2. Requirements Language 197 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 198 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 199 "OPTIONAL" in this document are to be interpreted as described in BCP 200 14 [RFC2119] [RFC8174] when, and only when, they appear in all 201 capitals, as shown here. 203 3. Use Cases 205 The use cases of simulcast described in this document relate to a 206 multi-party communication session where one or more central nodes are 207 used to adapt the view of the communication session towards 208 individual participants, and facilitate the media transport between 209 participants. Thus, these cases target the RTP Mixer type of 210 topology. 212 There are two principal approaches for an RTP Mixer to provide this 213 adapted view of the communication session to each receiving 214 participant: 216 o Transcoding (decoding and re-encoding) received RTP streams with 217 characteristics adapted to each receiving participant. This often 218 include mixing or composition of media sources from multiple 219 participants into a mixed media source originated by the RTP 220 Mixer. The main advantage of this approach is that it achieves 221 close to optimal adaptation to individual receiving participants. 222 The main disadvantages are that it can be very computationally 223 expensive to the RTP Mixer, typically degrades media Quality of 224 Experience (QoE) such as end-to-end delay for the receiving 225 participants, and requires RTP Mixer access to media content. 227 o Switching a subset of all received RTP streams or sub-streams to 228 each receiving participant, where the used subset is typically 229 specific to each receiving participant. The main advantages of 230 this approach are that it is computationally cheap to the RTP 231 Mixer, has very limited impact on media QoE, and does not require 232 RTP Mixer (full) access to media content. The main disadvantage 233 is that it can be difficult to combine a subset of received RTP 234 streams into a perfect fit to the resource situation of a 235 receiving participant. It is also a disadvantage that sending 236 multiple RTP streams consumes more network resources from the 237 sending participant to the RTP Mixer. 239 The use of simulcast relates to the latter approach, where it is more 240 important to reduce the load on the RTP Mixer and/or minimize QoE 241 impact than to achieve an optimal adaptation of resource usage. 243 3.1. Reaching a Diverse Set of Receivers 245 The media sources provided by a sending participant potentially need 246 to reach several receiving participants that differ in terms of 247 available resources. The receiver resources that typically differ 248 include, but are not limited to: 250 Codec: This includes codec type (such as RTP payload format MIME 251 type) and can include codec configuration. A couple of codec 252 resources that differ only in codec configuration will be 253 "different" if they are somehow not "compatible", like if they 254 differ in video codec profile, or the transport packetization 255 configuration. 257 Sampling: This relates to how the media source is sampled, in 258 spatial as well as in temporal domain. For video streams, spatial 259 sampling affects image resolution and temporal sampling affects 260 video frame rate. For audio, spatial sampling relates to the 261 number of audio channels and temporal sampling affects audio 262 bandwidth. This may be used to suit different rendering 263 capabilities or needs at the receiving endpoints. 265 Bitrate: This relates to the number of bits sent per second to 266 transmit the media source as an RTP stream, which typically also 267 affects the QoE for the receiving user. 269 Letting the sending participant create a simulcast of a few 270 differently configured RTP streams per media source can be a good 271 tradeoff when using an RTP switch as middlebox, instead of sending a 272 single RTP stream and using an RTP mixer to create individual 273 transcodings to each receiving participant. 275 This requires that the receiving participants can be categorized in 276 terms of available resources and that the sending participant can 277 choose a matching configuration for a single RTP stream per category 278 and media source. For example, a set of receiving participants 279 differ only in screen resolution; some are able to display video with 280 at most 360p resolution and some support 720p resolution. A sending 281 participant can then reach all receivers with best possible 282 resolution by creating a simulcast of RTP streams with 360p and 720p 283 resolution for each sent video media source. 285 The maximum number of simulcasted RTP streams that can be sent is 286 mainly limited by the amount of processing and uplink network 287 resources available to the sending participant. 289 3.2. Application Specific Media Source Handling 291 The application logic that controls the communication session may 292 include special handling of some media sources. It is, for example, 293 commonly the case that the media from a sending participant is not 294 sent back to itself. 296 It is also common that a currently active speaker participant is 297 shown in larger size or higher quality than other participants (the 298 sampling or bitrate aspects of Section 3.1) in a receiving client. 299 Many conferencing systems do not send the active speaker's media back 300 to the sender itself, which means there is some other participant's 301 media that instead is forwarded to the active speaker; typically the 302 previous active speaker. This way, the previously active speaker is 303 needed both in larger size (to current active speaker) and in small 304 size (to the rest of the participants), which can be solved with a 305 simulcast from the previously active speaker to the RTP switch. 307 3.3. Receiver Media Source Preferences 309 The application logic that controls the communication session may 310 allow receiving participants to state preferences on the 311 characteristics of the RTP stream they like to receive, for example 312 in terms of the aspects listed in Section 3.1. Sending a simulcast 313 of RTP streams is one way of accommodating receivers with conflicting 314 or otherwise incompatible preferences. 316 4. Overview 318 This memo defines SDP [RFC4566] signaling that covers the above 319 described simulcast use cases and functionalities. A number of 320 requirements for such signaling are elaborated in Appendix A. 322 The RID mechanism, as defined in [I-D.ietf-mmusic-rid], enables an 323 SDP offerer or answerer to specify a number of different RTP stream 324 restrictions for a rid-id by using the "a=rid" line. Examples of 325 such restrictions are maximum bitrate, maximum spatial video 326 resolution (width and height), maximum video framerate, etc. Each 327 rid-id may also be restricted to use only a subset of the RTP payload 328 types in the associated SDP media description. Those RTP payload 329 types can have their own configurations and parameters affecting what 330 can be sent or received, using the "a=fmtp" line as well as other SDP 331 attributes. 333 A new SDP media level attribute "a=simulcast" is defined. The 334 attribute describes, independently for send and receive directions, 335 the number of simulcast RTP streams as well as potential alternative 336 formats for each simulcast RTP stream. Each simulcast RTP stream, 337 including alternatives, is identified using the RID identifier (rid- 338 id), defined in [I-D.ietf-mmusic-rid]. 340 a=simulcast:send 1;2,3 recv 4 342 If the above line is included in an SDP offer, the "send" part 343 indicates the offerer's capability and proposal to send two simulcast 344 RTP streams. Each simulcast stream is described by one or more RTP 345 stream identifiers (rid-id), each group of rid-ids for a simulcast 346 stream is separated by a semicolon (";"). When a simulcast stream 347 has multiple rid-ids that are separated by a comma (","), they 348 describe alternative representations for that particular simulcast 349 RTP stream. Thus, the above "send" part is interpreted as an 350 intention to send two simulcast RTP streams. The first simulcast RTP 351 stream is identified and restricted according to rid-id 1. The 352 second simulcast RTP stream can be sent as two alternatives, 353 identified and restricted according to rid-ids 2 and 3. The "recv" 354 part of the above line indicates that the offerer desires to receive 355 a single RTP stream (no simulcast) according to rid-id 4. 357 A more complete example SDP offer media description is provided 358 below: 360 m=video 49300 RTP/AVP 97 98 99 361 a=rtpmap:97 H264/90000 362 a=rtpmap:98 H264/90000 363 a=rtpmap:99 VP8/90000 364 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 365 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 366 a=fmtp:99 max-fs=240; max-fr=30 367 a=rid:1 send pt=97;max-width=1280;max-height=720 368 a=rid:2 send pt=98;max-width=320;max-height=180 369 a=rid:3 send pt=99;max-width=320;max-height=180 370 a=rid:4 recv pt=97 371 a=simulcast:send 1;2,3 recv 4 372 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id 374 Figure 1: Example Simulcast Media Description in Offer 376 The above SDP media description can be interpreted at a high level to 377 say that the offerer is capable of sending two simulcast RTP streams, 378 one H.264 encoded stream in up to 720p resolution, and one additional 379 stream encoded as either H.264 or VP8 with a maximum resolution of 380 320x180 pixels. The offerer can receive one H.264 stream with 381 maximum 720p resolution. 383 The receiver of this SDP offer can generate an SDP answer that 384 indicates what it accepts. It uses the "a=simulcast" attribute to 385 indicate simulcast capability and specify what simulcast RTP streams 386 and alternatives to receive and/or send. An example of such 387 answering "a=simulcast" attribute, corresponding to the above offer, 388 is: 390 a=simulcast:recv 1;2 send 4 392 With this SDP answer, the answerer indicates in the "recv" part that 393 it wants to receive the two simulcast RTP streams. It has removed an 394 alternative that it doesn't support (rid-id 3). The send part 395 confirms to the offerer that it will receive one stream for this 396 media source according to rid-id 4. The corresponding, more complete 397 example SDP answer media description could look like: 399 m=video 49674 RTP/AVP 97 98 400 a=rtpmap:97 H264/90000 401 a=rtpmap:98 H264/90000 402 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 403 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 404 a=rid:1 recv pt=97;max-width=1280;max-height=720 405 a=rid:2 recv pt=98;max-width=320;max-height=180 406 a=rid:4 send pt=97 407 a=simulcast:recv 1;2 send 4 408 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id 410 Figure 2: Example Simulcast Media Description in Answer 412 It is assumed that a single SDP media description is used to describe 413 a single media source. This is aligned with the concepts defined in 414 [RFC7656] and will work in a WebRTC context, both with and without 415 BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation] grouping of media 416 descriptions. 418 To summarize, the "a=simulcast" line describes send and receive 419 direction simulcast streams separately. Each direction can in turn 420 describe one or more simulcast streams, separated by semicolon. The 421 identifiers describing simulcast streams on the "a=simulcast" line 422 are rid-id, as defined by "a=rid" lines in [I-D.ietf-mmusic-rid]. 423 Each simulcast stream can be offered as a list of alternative rid-id, 424 with each alternative separated by comma (not in the examples above). 425 A detailed specification can be found in Section 5 and more detailed 426 examples are outlined in Section 5.6. 428 5. Detailed Description 430 This section further details the overview above (Section 4). First, 431 formal syntax is provided (Section 5.1), followed by the rest of the 432 SDP attribute definition in Section 5.2. Relating Simulcast Streams 433 (Section 5.5) provides the definition of the RTP/RTCP mechanisms 434 used. The section is concluded with a number of examples. 436 5.1. Simulcast Attribute 438 This document defines a new SDP media-level "a=simulcast" attribute, 439 with value according to the following ABNF [RFC5234] syntax and its 440 update for Case-Sensitive String Support in ABNF [RFC7405]: 442 sc-value = ( sc-send [SP sc-recv] ) / ( sc-recv [SP sc-send] ) 443 sc-send = %s"send" SP sc-str-list 444 sc-recv = %s"recv" SP sc-str-list 445 sc-str-list = sc-alt-list *( ";" sc-alt-list ) 446 sc-alt-list = sc-id *( "," sc-id ) 447 sc-id-paused = "~" 448 sc-id = [sc-id-paused] rid-id 449 ; SP defined in [RFC5234] 450 ; rid-id defined in [I-D.ietf-mmusic-rid] 452 Figure 3: ABNF for Simulcast Value 454 Note to RFC Editor: Replace "I-D.ietf-mmusic-rid" in the above 455 figure with RFC number of draft-ietf-mmusic-rid before publication 456 of this document. 458 The "a=simulcast" attribute has a parameter in the form of one or two 459 simulcast stream descriptions, each consisting of a direction ("send" 460 or "recv"), followed by a list of one or more simulcast streams. 461 Each simulcast stream consists of one or more alternative simulcast 462 formats. Each simulcast format is identified by a simulcast stream 463 identifier (rid-id). The rid-id MUST have the form of an RTP stream 464 identifier, as described by RTP Payload Format Restrictions 465 [I-D.ietf-mmusic-rid]. 467 In the list of simulcast streams, each simulcast stream is separated 468 by a semicolon (";"). Each simulcast stream can in turn be offered 469 in one or more alternative formats, represented by rid-ids, separated 470 by a comma (","). Each rid-id can also be specified as initially 471 paused [RFC7728], indicated by prepending a "~" to the rid-id. The 472 reason to allow separate initial pause states for each rid-id is that 473 pause capability can be specified individually for each RTP payload 474 type referenced by an rid-id. Since pause capability specified via 475 the "a=rtcp-fb" attribute applies only to specified payload types and 476 rid-id specified by "a=rid" can refer to multiple different payload 477 types, it is unfeasible to pause streams with rid-id where any of the 478 related RTP payload type(s) do not have pause capability. 480 5.2. Simulcast Capability 482 Simulcast capability is expressed through a new media level SDP 483 attribute, "a=simulcast" (Section 5.1). The use of this attribute at 484 the session level is undefined. Implementations of this 485 specification MUST NOT use it at the session level and MUST ignore it 486 if received at the session level. Extensions to this specification 487 may define such session level usage. Each SDP media description MUST 488 contain at most one "a=simulcast" line. 490 There are separate and independent sets of simulcast streams in send 491 and receive directions. When listing multiple directions, each 492 direction MUST NOT occur more than once on the same line. 494 Simulcast streams using undefined rid-id MUST NOT be used as valid 495 simulcast streams by an RTP stream receiver. The direction for an 496 rid-id MUST be aligned with the direction specified for the 497 corresponding RTP stream identifier on the "a=rid" line. 499 The listed number of simulcast streams for a direction sets a limit 500 to the number of supported simulcast streams in that direction. The 501 order of the listed simulcast streams in the "send" direction 502 suggests a proposed order of preference, in decreasing order: the 503 rid-id listed first is the most preferred and subsequent streams have 504 progressively lower preference. The order of the listed rid-id in 505 the "recv" direction expresses which simulcast streams that are 506 preferred, with the leftmost being most preferred. This can be of 507 importance if the number of actually sent simulcast streams have to 508 be reduced for some reason. 510 rid-id that have explicit dependencies [RFC5583] 511 [I-D.ietf-mmusic-rid] to other rid-id (even in the same media 512 description) MAY be used. 514 Use of more than a single, alternative simulcast format for a 515 simulcast stream MAY be specified as part of the attribute parameters 516 by expressing the simulcast stream as a comma-separated list of 517 alternative rid-id. The order of the rid-id alternatives within a 518 simulcast stream is significant; the rid-id alternatives are listed 519 from (left) most preferred to (right) least preferred. For the use 520 of simulcast, this overrides the normal codec preference as expressed 521 by format type ordering on the "m=" line, using regular SDP rules. 522 This is to enable a separation of general codec preferences and 523 simulcast stream configuration preferences. However, the choice of 524 which alternative to use per simulcast stream is independent, and 525 there is currently no mechanism to align the choice between 526 alternative rid-ids between different simulcast streams. 528 A simulcast stream can use a codec defined such that the same RTP 529 SSRC can change RTP payload type multiple times during a session, 530 possibly even on a per-packet basis. A typical example can be a 531 speech codec that makes use of Comfort Noise [RFC3389] and/or DTMF 532 [RFC4733] formats. 534 If RTP stream pause/resume [RFC7728] is supported, any rid-id MAY be 535 prefixed by a "~" character to indicate that the corresponding 536 simulcast stream is initially paused already from start of the RTP 537 session. In this case, support for RTP stream pause/resume MUST also 538 be included under the same "m=" line where "a=simulcast" is included. 539 All RTP payload types related to such an initially paused simulcast 540 stream MUST be listed in the SDP as pause/resume capable as specified 541 by [RFC7728], e.g. by using the "*" wildcard format for "a=rtcp-fb". 543 An initially paused simulcast stream in "send" direction for the 544 endpoint sending the SDP MUST be considered equivalent to an 545 unsolicited locally paused stream, and be handled accordingly. 546 Initially paused simulcast streams are resumed as described by the 547 RTP pause/resume specification. An RTP stream receiver that wishes 548 to resume an unsolicited locally paused stream needs to know the SSRC 549 of that stream. The SSRC of an initially paused simulcast stream can 550 be obtained from an RTP stream sender RTCP Sender Report (SR) 551 including both the desired SSRC as "SSRC of sender", and the rid-id 552 value in an RtpStreamId RTCP SDES item [I-D.ietf-avtext-rid]. 554 If the endpoint sending the SDP includes an "recv" direction 555 simulcast stream that is initially paused, then the remote RTP sender 556 receiving the SDP SHOULD put its RTP stream in a unsolicited locally 557 paused state. The simulcast stream sender does not put the stream in 558 the locally paused state if there are other RTP stream receivers in 559 the session that do not mark the simulcast stream as initially 560 paused. However, in centralized conferencing the RTP sender usually 561 does not see the SDP signalling from RTP receivers and cannot make 562 this determination. The reason to require an initially paused "recv" 563 stream to be considered locally paused by the remote RTP sender, 564 instead of making it equivalent to implicitly sending a pause 565 request, is because the pausing RTP sender cannot know which 566 receiving SSRC owns the restriction when Temporary Maximum Media 567 Stream Bit Rate Request (TMMBR) and Temporary Maximum Media Stream 568 Bit Rate Notification (TMMBN) are used for pause/resume signaling 569 (Section 5.6 of [RFC7728]) since the RTP receiver's SSRC in send 570 direction is sometimes not yet known. 572 Use of the redundant audio data [RFC2198] format could be seen as a 573 form of simulcast for loss protection purposes, but is not considered 574 conflicting with the mechanisms described in this memo and MAY 575 therefore be used as any other format. In this case the "red" 576 format, rather than the carried formats, SHOULD be the one to list as 577 a simulcast stream on the "a=simulcast" line. 579 The media formats and corresponding characteristics of simulcast 580 streams SHOULD be chosen such that they are different, e.g. as 581 different SDP formats with differing "a=rtpmap" and/or "a=fmtp" 582 lines, or as differently defined RTP payload format restrictions. If 583 this difference is not required, it is RECOMMENDED to use RTP 584 duplication [RFC7104] procedures instead of simulcast. To avoid 585 complications in implementations, a single rid-id MUST NOT occur more 586 than once per "a=simulcast" line. Note that this does not eliminate 587 use of simulcast as an RTP duplication mechanism, since it is 588 possible to define multiple different rid-id that are effectively 589 equivalent. 591 5.3. Offer/Answer Use 593 Note: The inclusion of "a=simulcast" or the use of simulcast does 594 not change any of the interpretation or Offer/Answer procedures 595 for other SDP attributes, like "a=fmtp" or "a=rid". 597 5.3.1. Generating the Initial SDP Offer 599 An offerer wanting to use simulcast for a media description SHALL 600 include one "a=simulcast" attribute in that media description in the 601 offer. An offerer listing a set of receive simulcast streams and/or 602 alternative formats as rid-id in the offer MUST be prepared to 603 receive RTP streams for any of those simulcast streams and/or 604 alternative formats from the answerer. 606 5.3.2. Creating the SDP Answer 608 An answerer that does not understand the concept of simulcast will 609 also not know the attribute and will remove it in the SDP answer, as 610 defined in existing SDP Offer/Answer [RFC3264] procedures. Since SDP 611 session level simulcast is undefined in this memo, an answerer that 612 receives an offer with the "a=simulcast" attribute on SDP session 613 level SHALL remove it in the answer. An answerer that understands 614 the attribute but receives multiple "a=simulcast" attributes in the 615 same media description SHALL disable use of simulcast by removing all 616 "a=simulcast" lines for that media description in the answer. 618 An answerer that does understand the attribute and that wants to 619 support simulcast in an indicated direction SHALL reverse 620 directionality of the unidirectional direction parameters; "send" 621 becomes "recv" and vice versa, and include it in the answer. 623 An answerer that receives an offer with simulcast containing an 624 "a=simulcast" attribute listing alternative rid-id MAY keep all the 625 alternative rid-id in the answer, but it MAY also choose to remove 626 any non-desirable alternative rid-id in the answer. The answerer 627 MUST NOT add any alternative rid-id in send direction in the answer 628 that were not present in the offer receive direction. The answerer 629 MUST be prepared to receive any of the receive direction rid-id 630 alternatives and MAY send any of the send direction alternatives that 631 are part of the answer. 633 An answerer that receives an offer with simulcast that lists a number 634 of simulcast streams, MAY reduce the number of simulcast streams in 635 the answer, but MUST NOT add simulcast streams. 637 An answerer that receives an offer without RTP stream pause/resume 638 capability MUST NOT mark any simulcast streams as initially paused in 639 the answer. 641 An RTP stream pause/resume capable answerer that receives an offer 642 with RTP stream pause/resume capability MAY mark any rid-id that 643 refer to pause/resume capable formats as initially paused in the 644 answer. 646 An answerer that receives indication in an offer of an rid-id being 647 initially paused SHOULD mark that rid-id as initially paused also in 648 the answer, regardless of direction, unless it has good reason for 649 the rid-id not being initially paused. One reason to remove an 650 initial pause in the answer compared to the offer could, for example, 651 be that all receive direction simulcast streams for a media source 652 the answerer accepts in the answer would otherwise be paused. 654 5.3.3. Offerer Processing the SDP Answer 656 An offerer that receives an answer without "a=simulcast" MUST NOT use 657 simulcast towards the answerer. An offerer that receives an answer 658 with "a=simulcast" without any rid-id in a specified direction MUST 659 NOT use simulcast in that direction. 661 An offerer that receives an answer where some rid-id alternatives are 662 kept MUST be prepared to receive any of the kept send direction rid- 663 id alternatives, and MAY send any of the kept receive direction rid- 664 id alternatives. 666 An offerer that receives an answer where some of the rid-id are 667 removed compared to the offer MAY release the corresponding resources 668 (codec, transport, etc) in its receive direction and MUST NOT send 669 any RTP packets corresponding to the removed rid-id. 671 An offerer that offered some of its rid-id as initially paused and 672 that receives an answer that does not indicate RTP stream pause/ 673 resume capability, MUST NOT initially pause any simulcast streams. 675 An offerer with RTP stream pause/resume capability that receives an 676 answer where some rid-id are marked as initially paused, SHOULD 677 initially pause those RTP streams regardless if they were marked as 678 initially paused also in the offer, unless it has good reason for 679 those RTP streams not being initially paused. One such reason could, 680 for example, be that the answerer would otherwise initially not 681 receive any media of that type at all. 683 5.3.4. Modifying the Session 685 Offers inside an existing session follow the same rules as for 686 initial SDP offer, with these additions: 688 1. rid-id marked as initially paused in the offerer's send direction 689 SHALL reflect the offerer's opinion of the current pause state at 690 the time of creating the offer. This is purely informational, 691 and RTP stream pause/resume [RFC7728] signaling in the ongoing 692 session SHALL take precedence in case of any conflict or 693 ambiguity. 695 2. rid-id marked as initially paused in the offerer's receive 696 direction SHALL (as in an initial offer) reflect the offerer's 697 desired rid-id pause state. Except for the case where the 698 offerer already paused the corresponding RTP stream through RTP 699 stream pause/resume [RFC7728] signaling , this is identical to 700 the conditions at an initial offer. 702 Creation of SDP answers and processing of SDP answers inside an 703 existing session follow the same rules as described above for initial 704 SDP offer/answer. 706 Session modification restrictions in section 6.5 of RTP payload 707 format restrictions [I-D.ietf-mmusic-rid] also apply. 709 5.4. Use with Declarative SDP 711 This document does not define the use of "a=simulcast" in declarative 712 SDP, partly motivated by use of the simulcast format identification 713 [I-D.ietf-mmusic-rid] not being defined for use in declarative SDP. 714 If concrete use cases for simulcast in declarative SDP are identified 715 in the future, the authors of this memo expect that additional 716 specifications will address such use. 718 5.5. Relating Simulcast Streams 720 Simulcast RTP streams MUST be related on RTP level through 721 RtpStreamId [I-D.ietf-avtext-rid], as specified in the SDP 722 "a=simulcast" attribute (Section 5.2) parameters. This is sufficient 723 as long as there is only a single media source per SDP media 724 description. When using BUNDLE 725 [I-D.ietf-mmusic-sdp-bundle-negotiation], where multiple SDP media 726 descriptions jointly specify a single RTP session, the SDES MID 727 identification mechanism in BUNDLE allows relating RTP streams back 728 to individual media descriptions, after which the above described 729 RtpStreamId relations can be used. Use of the RTP header extension 730 [RFC8285] for both MID and RtpStreamId identifications can be 731 important to ensure rapid initial reception, required to correctly 732 interpret and process the RTP streams. Implementers of this 733 specification MUST support the RTCP source description (SDES) item 734 method and SHOULD support RTP header extension method to signal 735 RtpStreamId on RTP level. 737 NOTE: For the case where it is clear from SDP that RTP PT uniquely 738 maps to corresponding RtpStreamId, an RTP receiver can use RTP PT 739 to relate simulcast streams. This can sometimes enable decoding 740 even in advance to receiving RtpStreamId information in RTCP SDES 741 and/or RTP header extensions. 743 RTP streams MUST only use a single alternative rid-id at a time 744 (based on RTP timestamps), but MAY change format (and rid-id) on a 745 per-RTP packet basis. This corresponds to the existing (non- 746 simulcast) SDP offer/answer case when multiple formats are included 747 on the "m=" line in the SDP answer, enabling per-RTP packet change of 748 RTP payload type. 750 5.6. Signaling Examples 752 These examples describe a client to video conference service, using a 753 centralized media topology with an RTP mixer. 755 +---+ +-----------+ +---+ 756 | A |<---->| |<---->| B | 757 +---+ | | +---+ 758 | Mixer | 759 +---+ | | +---+ 760 | F |<---->| |<---->| J | 761 +---+ +-----------+ +---+ 763 Figure 4: Four-party Mixer-based Conference 765 5.6.1. Single-Source Client 767 Alice is calling in to the mixer with a simulcast-enabled client 768 capable of a single media source per media type. The client can send 769 a simulcast of 2 video resolutions and frame rates: HD 1280x720p 770 30fps and thumbnail 320x180p 15fps. This is defined below using the 771 "imageattr" [RFC6236]. In this example, only the "pt" "a=rid" 772 parameter is used, effectively achieving a 1:1 mapping between 773 RtpStreamId and media formats (RTP payload types), to describe 774 simulcast stream formats. Alice's Offer: 776 v=0 777 o=alice 2362969037 2362969040 IN IP4 192.0.2.156 778 s=Simulcast Enabled Client 779 t=0 0 780 c=IN IP4 192.0.2.156 781 m=audio 49200 RTP/AVP 0 782 a=rtpmap:0 PCMU/8000 783 m=video 49300 RTP/AVP 97 98 784 a=rtpmap:97 H264/90000 785 a=rtpmap:98 H264/90000 786 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 787 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 788 a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720] 789 a=imageattr:98 send [x=320,y=180] recv [x=320,y=180] 790 a=rid:1 send pt=97 791 a=rid:2 send pt=98 792 a=rid:3 recv pt=97 793 a=simulcast:send 1;2 recv 3 794 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id 796 Figure 5: Single-Source Simulcast Offer 798 The only thing in the SDP that indicates simulcast capability is the 799 line in the video media description containing the "simulcast" 800 attribute. The included "a=fmtp" and "a=imageattr" parameters 801 indicates that sent simulcast streams can differ in video resolution. 802 The RTP header extension for RtpStreamId is offered to avoid issues 803 with the initial binding between RTP streams (SSRCs) and the 804 RtpStreamId identifying the simulcast stream and its format. 806 The Answer from the server indicates that it too is simulcast 807 capable. Should it not have been simulcast capable, the 808 "a=simulcast" line would not have been present and communication 809 would have started with the media negotiated in the SDP. Also the 810 usage of the RtpStreamId RTP header extension is accepted. 812 v=0 813 o=server 823479283 1209384938 IN IP4 192.0.2.2 814 s=Answer to Simulcast Enabled Client 815 t=0 0 816 c=IN IP4 192.0.2.43 817 m=audio 49672 RTP/AVP 0 818 a=rtpmap:0 PCMU/8000 819 m=video 49674 RTP/AVP 97 98 820 a=rtpmap:97 H264/90000 821 a=rtpmap:98 H264/90000 822 a=fmtp:97 profile-level-id=42c01f;max-fs=3600;max-mbps=108000 823 a=fmtp:98 profile-level-id=42c00b;max-fs=240;max-mbps=3600 824 a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720] 825 a=imageattr:98 send [x=320,y=180] recv [x=320,y=180] 826 a=rid:1 recv pt=97 827 a=rid:2 recv pt=98 828 a=rid:3 send pt=97 829 a=simulcast:recv 1;2 send 3 830 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id 832 Figure 6: Single-Source Simulcast Answer 834 Since the server is the simulcast media receiver, it reverses the 835 direction of the "simulcast" and "rid" attribute parameters. 837 5.6.2. Multi-Source Client 839 Fred is calling in to the same conference as in the example above 840 with a two-camera, two-display system, thus capable of handling two 841 separate media sources in each direction, where each media source is 842 simulcast-enabled in the send direction. Fred's client is restricted 843 to a single media source per media description. 845 The first two simulcast streams for the first media source use 846 different codecs, H264-SVC [RFC6190] and H264 [RFC6184]. These two 847 simulcast streams also have a temporal dependency. Two different 848 video codecs, VP8 [RFC7741] and H264, are offered as alternatives for 849 the third simulcast stream for the first media source. Only the 850 highest fidelity simulcast stream is sent from start, the lower 851 fidelity streams being initially paused. 853 The second media source is offered with three different simulcast 854 streams. All video streams of this second media source are loss 855 protected by RTP retransmission [RFC4588]. Also here, all but the 856 highest fidelity simulcast stream are initially paused. Note that 857 the lower resolution is more prioritized than the medium resolution 858 simulcast stream. 860 Fred's client is also using BUNDLE to send all RTP streams from all 861 media descriptions in the same RTP session on a single media 862 transport. Although using many different simulcast streams in this 863 example, the use of RtpStreamId as simulcast stream identification 864 enables use of a low number of RTP payload types. Note that the use 865 of both BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation] and "a=rid" 866 [I-D.ietf-mmusic-rid] recommends using the RTP header extension 867 [RFC8285] for carrying these RTP stream identification fields, which 868 is consequently also included in the SDP. Note also that for 869 "a=rid", the corresponding RtpStreamId SDES attribute RTP header 870 extension is named rtp-stream-id [I-D.ietf-avtext-rid]. 872 v=0 873 o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d 874 s=Offer from Simulcast Enabled Multi-Source Client 875 t=0 0 876 c=IN IP6 2001:db8::c000:27d 877 a=group:BUNDLE foo bar zen 878 m=audio 49200 RTP/AVP 99 879 a=mid:foo 880 a=rtpmap:99 G722/8000 881 m=video 49600 RTP/AVPF 100 101 103 882 a=mid:bar 883 a=rtpmap:100 H264-SVC/90000 884 a=rtpmap:101 H264/90000 885 a=rtpmap:103 VP8/90000 886 a=fmtp:100 profile-level-id=42400d;max-fs=3600;max-mbps=216000; \ 887 mst-mode=NI-TC 888 a=fmtp:101 profile-level-id=42c00d;max-fs=3600;max-mbps=108000 889 a=fmtp:103 max-fs=900; max-fr=30 890 a=rid:1 send pt=100;max-width=1280;max-height=720;max-fps=60;depend=2 891 a=rid:2 send pt=101;max-width=1280;max-height=720;max-fps=30 892 a=rid:3 send pt=101;max-width=640;max-height=360 893 a=rid:4 send pt=103;max-width=640;max-height=360 894 a=depend:100 lay bar:101 895 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid 896 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id 897 a=rtcp-fb:* ccm pause nowait 898 a=simulcast:send 1;2;~4,3 899 m=video 49602 RTP/AVPF 96 104 900 a=mid:zen 901 a=rtpmap:96 VP8/90000 902 a=fmtp:96 max-fs=3600; max-fr=30 903 a=rtpmap:104 rtx/90000 904 a=fmtp:104 apt=96;rtx-time=200 905 a=rid:1 send max-fs=921600;max-fps=30 906 a=rid:2 send max-fs=614400;max-fps=15 907 a=rid:3 send max-fs=230400;max-fps=30 908 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid 909 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id 910 a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id 911 a=rtcp-fb:* ccm pause nowait 912 a=simulcast:send 1;~3;~2 914 Figure 7: Fred's Multi-Source Simulcast Offer 916 5.6.3. Simulcast and Redundancy 918 The example in this section looks at applying simulcast with audio 919 and video redundancy formats. The audio media description uses codec 920 and bitrate restrictions, combining it with RTP Payload for Redundant 921 Audio Data [RFC2198] for enhanced packet loss resilience. The video 922 media description applies both resolution and bitrate restrictions, 923 combining it with FEC in the form of Flexible FEC 924 [I-D.ietf-payload-flexible-fec-scheme] and RTP Retransmission 925 [RFC4588]. 927 The audio source is offered to be sent as two simulcast streams. The 928 first simulcast stream is encoded with Opus, restricted to 50 kbps 929 (rid-id=5), and the second simulcast stream is encoded either with 930 G.711 (rid-id=7) or with G.711 combined with LPC for redundancy (rid- 931 id=6). In this example, stand-alone LPC is not offered as an 932 possible payload type for the second simulcast stream's RID, which 933 could e.g. be motivated by not providing sufficient quality. 935 The video source is offered to be sent as two simulcast streams, both 936 with two alternative simulcast formats. Redundancy and repair are 937 offered in the form of both Flexible FEC and RTP Retransmission. The 938 Flexible FEC is not bound to any particular RTP streams and is 939 therefore possible to use across all RTP streams that are being sent 940 as part of this media description. 942 v=0 943 o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d 944 s=Offer from Simulcast Enabled Client using Redundancy 945 t=0 0 946 c=IN IP6 2001:db8::c000:27d 947 a=group:BUNDLE foo bar 948 m=audio 49200 RTP/AVP 97 98 99 100 101 102 949 a=mid:foo 950 a=rtpmap:97 G711/8000 951 a=rtpmap:98 LPC/8000 952 a=rtpmap:99 OPUS/48000/1 953 a=rtpmap:100 RED/8000/1 954 a=rtpmap:101 CN/8000 955 a=rtpmap:102 telephone-event/8000 956 a=fmtp:99 useinbandfec=1;usedtx=0 957 a=fmtp:100 97/98 958 a=fmtp:102 0-15 959 a=ptime:20 960 a=maxptime:40 961 a=rid:1 send pt=99,102;max-br=64000 962 a=rid:2 send pt=100,97,101,102 963 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid 964 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id 965 a=simulcast:send 1;2 966 m=video 49600 RTP/AVPF 103 104 105 106 107 967 a=mid:bar 968 a=rtpmap:103 H264/90000 969 a=rtpmap:104 VP8/90000 970 a=rtpmap:105 rtx/90000 971 a=rtpmap:106 rtx/90000 972 a=rtpmap:107 flexfec/90000 973 a=fmtp:103 profile-level-id=42c00d;max-fs=3600;max-mbps=108000 974 a=fmtp:104 max-fs=3600; max-fr=30 975 a=fmtp:105 apt=103;rtx-time=200 976 a=fmtp:106 apt=104;rtx-time=200 977 a=fmtp:107 repair-window=2000 978 a=rid:1 send pt=103;max-width=1280;max-height=720;max-fps=30 979 a=rid:2 send pt=104;max-width=1280;max-height=720;max-fps=30 980 a=rid:3 send pt=103;max-width=640;max-height=360;max-br=300000 981 a=rid:4 send pt=104;max-width=640;max-height=360;max-br=300000 982 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid 983 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id 984 a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id 985 a=rtcp-fb:* ccm pause nowait 986 a=simulcast:send 1,2;3,4 988 Figure 8: Simulcast and Redundancy Example 990 6. RTP Aspects 992 This section discusses what the different entities in a simulcast 993 media path can expect to happen on RTP level. This is explored from 994 source to sink by starting in an endpoint with a media source that is 995 simulcasted to an RTP middlebox. That RTP middlebox sends media 996 sources both to other RTP middleboxes (cascaded middleboxes), as well 997 as selecting some simulcast format of the media source and sending it 998 to receiving endpoints. Different types of RTP middleboxes and their 999 usage of the different simulcast formats results in several different 1000 behaviors. 1002 6.1. Outgoing from Endpoint with Media Source 1004 The most straightforward simulcast case is the RTP streams being 1005 emitted from the endpoint that originates a media source. When 1006 simulcast has been negotiated in the sending direction, the endpoint 1007 can transmit up to the number of RTP streams needed for the 1008 negotiated simulcast streams for that media source. Each RTP stream 1009 (SSRC) is identified by associating (Section 5.5) it with an 1010 RtpStreamId SDES item, transmitted in RTCP and possibly also as an 1011 RTP header extension. In cases where multiple media sources have 1012 been negotiated for the same RTP session and thus BUNDLE 1013 [I-D.ietf-mmusic-sdp-bundle-negotiation] is used, also the MID SDES 1014 item will be sent similarly to the RtpStreamId. 1016 Each RTP stream might not be continuously transmitted due to any of 1017 the following reasons; temporarily paused using Pause/Resume 1018 [RFC7728], sender side application logic temporarily pausing it, or 1019 lack of network resources to transmit this simulcast stream. 1020 However, all simulcast streams that have been negotiated have active 1021 and maintained SSRC (at least in regular RTCP reports), even if no 1022 RTP packets are currently transmitted. The relation between an RTP 1023 Stream (SSRC) and a particular simulcast stream is not expected to 1024 change, except in exceptional situations such as SSRC collisions. At 1025 SSRC changes, the usage of MID and RtpStreamId should enable the 1026 receiver to correctly identify the RTP streams even after an SSRC 1027 change. 1029 6.2. RTP Middlebox to Receiver 1031 RTP streams in a multi-party RTP session can be used in multiple 1032 different ways, when the session utilizes simulcast at least on the 1033 media source to middlebox legs. This is to a large degree due to the 1034 different RTP middlebox behaviors, but also the needs of the 1035 application. This text assumes that the RTP middlebox will select a 1036 media source and choose which simulcast stream for that media source 1037 to deliver to a specific receiver. In many cases, at most one 1038 simulcast stream per media source will be forwarded to a particular 1039 receiver at any instant in time, even if the selected simulcast 1040 stream may vary. For cases where this does not hold due to 1041 application needs, then the RTP stream aspects will fall under the 1042 middlebox to middlebox case Section 6.3. 1044 The selection of which simulcast streams to forward towards the 1045 receiver, is application specific. However, in conferencing 1046 applications, active speaker selection is common. In case the number 1047 of media sources possible to forward, N, is less than the total 1048 amount of media sources available in an multi-media session, the 1049 current and previous speakers (up to N in total) are often the ones 1050 forwarded. To avoid the need for media specific processing to 1051 determine the current speaker(s) in the RTP middlebox, the endpoint 1052 providing a media source may include meta data, such as the RTP 1053 Header Extension for Client-to-Mixer Audio Level Indication 1054 [RFC6464]. 1056 The possibilities for stream switching are media type specific, but 1057 for media types with significant interframe dependencies in the 1058 encoding, like most video coding, the switching needs to be made at 1059 suitable switching points in the media stream that breaks or 1060 otherwise deals with the dependency structure. Even if switching 1061 points can be included periodically, it is common to use mechanisms 1062 like Full Intra Requests [RFC5104] to request switching points from 1063 the endpoint performing the encoding of the media source. 1065 Inclusion of the RtpStreamId SDES item for an SSRC in the middlebox 1066 to receiver direction should only occur when use of RtpStreamId has 1067 been negotiated in that direction. It is worth noting that one can 1068 signal multiple RtpStreamIds when simulcast signalling indicates only 1069 a single simulcast stream, allowing one to use all of the 1070 RtpStreamIds as alternatives for that simulcast stream. One reason 1071 for including the RtpStreamId in the middlebox to receiver direction 1072 for an RTP stream is to let the receiver know which restrictions 1073 apply to the currently delivered RTP stream. In case the RtpStreamId 1074 is negotiated to be used, it is important to remember that the used 1075 identifiers will be specific to each signalling session. Even if the 1076 central entity can attempt to coordinate, it is likely that the 1077 RtpStreamIds need to be translated to the leg specific values. The 1078 below cases will have as base line that RtpStreamId is not used in 1079 the mixer to receiver direction. 1081 6.2.1. Media-Switching Mixer 1083 This section discusses the behavior in cases where the RTP middlebox 1084 behaves like the Media-Switching Mixer (Section 3.6.2) in RTP 1085 Topologies [RFC7667]. The fundamental aspect here is that the media 1086 sources delivered from the middlebox will be the mixer's conceptual 1087 or functional ones. For example, one media source may be the main 1088 speaker in high resolution video, while a number of other media 1089 sources are thumbnails of each participant. 1091 The above results in that the RTP stream produced by the mixer is one 1092 that switches between a number of received incoming RTP streams for 1093 different media sources and in different simulcast versions. The 1094 mixer selects the media source to be sent as one of the RTP streams, 1095 and then selects among the available simulcast streams for the most 1096 appropriate one. The selection criteria include available bandwidth 1097 on the mixer to receiver path and restrictions based on the 1098 functional usage of the RTP stream delivered to the receiver. As an 1099 example of the latter, it is unnecessary to forward a full HD video 1100 to a receiver if the display area is just a thumbnail. Thus, 1101 restrictions may exist to not allow some simulcast streams to be 1102 forwarded for some of the mixer's media sources. 1104 This will result in a single RTP stream being used for each of the 1105 RTP mixer's media sources. This RTP stream is at any point in time a 1106 selection of one particular RTP stream arriving to the mixer, where 1107 the RTP header field values are rewritten to provide a consistent, 1108 single RTP stream. If the RTP mixer doesn't receive any incoming 1109 stream matched to this media source, the SSRC will not transmit, but 1110 be kept alive using RTCP. The SSRC and thus RTP stream for the 1111 mixer's media source is expected to be long term stable. It will 1112 only be changed by signalling or other disruptive events. Note that 1113 although the above talks about a single RTP stream, there can in some 1114 cases be multiple RTP streams carrying the selected simulcast stream 1115 for the originating media source, including redundancy or other 1116 auxiliary RTP streams. 1118 The mixer may communicate the identity of the originating media 1119 source to the receiver by including the CSRC field with the 1120 originating media source's SSRC value. Note that due to the 1121 possibility that the RTP mixer switches between simulcast versions of 1122 the media source, the CSRC value may change, even if the media source 1123 is kept the same. 1125 It is important to note that any MID SDES item from the originating 1126 media source needs to be removed and not be associated with the RTP 1127 stream's SSRC. That is, there is nothing in the signalling between 1128 the mixer and the receiver that is structured around the originating 1129 media sources, only the mixer's media sources. If they would be 1130 associated with the SSRC, the receiver would likely believe that 1131 there has been an SSRC collision, and that the RTP stream is spurious 1132 as it doesn't carry the identifiers used to relate it to the correct 1133 context. However, this is not true for CSRC values, as long as they 1134 are never used as SSRC. In these cases one could provide CNAME and 1135 MID as SDES items. A receiver could use this to determine which CSRC 1136 values that are associated with the same originating media source. 1138 If RtpStreamIds are used in the scenario described by this section, 1139 it should be noted that the RtpStreamId on a particular SSRC will 1140 change based on the actual simulcast stream selected for switching. 1141 These RtpStreamId identifiers will be local to this leg's signalling 1142 context. In addition, the defined RtpStreamIds and their parameters 1143 need to cover all the media sources and simulcast streams received by 1144 the RTP mixer that can be switched into this media source, sent by 1145 the RTP mixer. 1147 6.2.2. Selective Forwarding Middlebox 1149 This section discusses the behavior in cases where the RTP middlebox 1150 behaves like the Selective Forwarding Middlebox (Section 3.7) in RTP 1151 Topologies [RFC7667]. Applications for this type of RTP middlebox 1152 results in that each originating media source will have a 1153 corresponding media source on the leg between the middlebox and the 1154 receiver. A Selective Forwarding Middlebox (SFM) could go as far as 1155 exposing all the simulcast streams for an media source, however this 1156 section will focus on having a single simulcast stream that can 1157 contain any of the simulcast formats. This section will assume that 1158 the SFM projection mechanism works on media source level, and maps 1159 one of the media source's simulcast streams onto one RTP stream from 1160 the SFM to the receiver. 1162 This usage will result in that the individual RTP stream(s) for one 1163 media source can switch between being active to paused, based on the 1164 subset of media sources the SFM wants to provide the receiver for the 1165 moment. With SFMs there exist no reasons to use CSRC to indicate the 1166 originating stream, as there is a one to one media source mapping. 1167 If the application requires knowing the simulcast version received to 1168 function well, then RtpStreamId should be negotiated on the SFM to 1169 receiver leg. Which simulcast stream that is being forwarded is not 1170 made explicit unless RtpStreamId is used on the leg. 1172 Any MID SDES items being sent by the SFM to the receiver are only 1173 those agreed between the SFM and the receiver, and no MID values from 1174 the originating side of the SFM are to be forwarded. 1176 A SFM could expose corresponding RTP streams for all the media 1177 sources and their simulcast streams, and then for any media source 1178 that is to be provided forward one selected simulcast stream. 1179 However, this is not recommended as it would unnecessarily increase 1180 the number of RTP streams and require the receiver to timely detect 1181 switching between simulcast streams. The above usage requires the 1182 same SFM functionality for switching, while avoiding the 1183 uncertainties of timely detecting that a RTP stream ends. The 1184 benefit would be that the received simulcast stream would be 1185 implicitly provided by which RTP stream would be active for a media 1186 source. However, using RtpStreamId to make this explicit also 1187 exposes which alternative format is used. The conclusion is that 1188 using one RTP stream per simulcast stream is unnecessary. The issue 1189 with timely detecting end of streams, independent if they are stopped 1190 temporarily or long term, is that there is no explicit indication 1191 that the transmission has intentionally been stopped. The RTCP based 1192 Pause and Resume mechanism [RFC7728] includes a PAUSED indication 1193 that provides the last RTP sequence number transmitted prior to the 1194 pause. Due to usage, the timeliness of this solution depends on when 1195 delivery using RTCP can occur in relation to the transmission of the 1196 last RTP packet. If no explicit information is provided at all, then 1197 detection based on non increasing RTCP SR field values and timers 1198 need to be used to determine pause in RTP packet delivery. This 1199 results in that one can usually not determine when the last RTP 1200 packet arrives (if it arrives) that this will be the last. That it 1201 was the last is something that one learns later. 1203 6.3. RTP Middlebox to RTP Middlebox 1205 This relates to the transmission of simulcast streams between RTP 1206 middleboxes or other usages where one wants to enable the delivery of 1207 multiple simultaneous simulcast streams per media source, but the 1208 transmitting entity is not the originating endpoint. For a 1209 particular direction between middlebox A and B, this looks very 1210 similar to the originating to middlebox case on a media source basis. 1211 However, in this case there is usually multiple media sources, 1212 originating from multiple endpoints. This can create situations 1213 where limitations in the number of simultaneously received media 1214 streams can arise, for example due to limitation in network 1215 bandwidth. In this case, a subset of not only the simulcast streams, 1216 but also media sources can be selected. This results in that 1217 individual RTP streams can be become paused at any point and later 1218 being resumed based on various criteria. 1220 The MIDs used between A and B are the ones agreed between these two 1221 identities in signalling. The RtpStreamId values will also be 1222 provided to ensure explicit information about which simulcast stream 1223 they are. The RTP stream to MID and RtpStreamId associations should 1224 here be long term stable. 1226 7. Network Aspects 1228 Simulcast is in this memo defined as the act of sending multiple 1229 alternative encoded streams of the same underlying media source. 1230 When transmitting multiple independent streams that originate from 1231 the same source, it could potentially be done in several different 1232 ways using RTP. A general discussion on considerations for use of 1233 the different RTP multiplexing alternatives can be found in 1234 Guidelines for Multiplexing in RTP 1235 [I-D.ietf-avtcore-multiplex-guidelines]. Discussion and 1236 clarification on how to handle multiple streams in an RTP session can 1237 be found in [RFC8108]. 1239 The network aspects that are relevant for simulcast are: 1241 Quality of Service: When using simulcast it might be of interest to 1242 prioritize a particular simulcast stream, rather than applying 1243 equal treatment to all streams. For example, lower bitrate 1244 streams may be prioritized over higher bitrate streams to minimize 1245 congestion or packet losses in the low bitrate streams. Thus, 1246 there is a benefit to use a simulcast solution with good QoS 1247 support. 1249 NAT/FW Traversal: Using multiple RTP sessions incurs more cost for 1250 NAT/FW traversal unless they can re-use the same transport flow, 1251 which can be achieved by Multiplexing Negotiation Using SDP Port 1252 Numbers [I-D.ietf-mmusic-sdp-bundle-negotiation]. 1254 7.1. Bitrate Adaptation 1256 Use of multiple simulcast streams can require a significant amount of 1257 network resources. The aggregate bandwidth for all simulcast streams 1258 for a media source (and thus SDP media description) is bounded by any 1259 SDP "b=" line applicable to that media source. It is assumed that a 1260 suitable congestion control mechanism is used by the application to 1261 ensure that it doesn't cause persistent congestion. If the amount of 1262 available network resources varies during an RTP session such that it 1263 does not match what is negotiated in SDP, the bitrate used by the 1264 different simulcast streams may have to be reduced dynamically. When 1265 a simulcasting media source uses a single media transport for all of 1266 the simulcast streams, it is likely that a joint congestion control 1267 across all simulcast streams is used for that media source. What 1268 simulcast streams to prioritize when allocating available bitrate 1269 among the simulcast streams in such adaptation SHOULD be taken from 1270 the simulcast stream order on the "a=simulcast" line and ordering of 1271 alternative simulcast formats Section 5.2. Simulcast streams that 1272 have pause/resume capability and that would be given such low bitrate 1273 by the adaptation process that they are considered not really useful 1274 can be temporarily paused until the limiting condition clears. 1276 8. Limitation 1278 The chosen approach has a limitation that relates to the use of a 1279 single RTP session for all simulcast formats of a media source, which 1280 comes from sending all simulcast streams related to a media source 1281 under the same SDP media description. 1283 It is not possible to use different simulcast streams on different 1284 media transports, limiting the possibilities to apply different QoS 1285 to different simulcast streams. When using unicast, QoS mechanisms 1286 based on individual packet marking are feasible, since they do not 1287 require separation of simulcast streams into different RTP sessions 1288 to apply different QoS. 1290 It is also not possible to separate different simulcast streams into 1291 different multicast groups to allow a multicast receiver to pick the 1292 stream it wants, rather than receive all of them. In this case, the 1293 only reasonable implementation is to use different RTP sessions for 1294 each multicast group so that reporting and other RTCP functions 1295 operate as intended. Such simulcast usage in multicast context is 1296 out of scope for the current document and would require additional 1297 specification. 1299 9. IANA Considerations 1301 This document requests to register a new media-level SDP attribute, 1302 "simulcast", in the "att-field (media level only)" registry within 1303 the SDP parameters registry, according to the procedures of [RFC4566] 1304 and [I-D.ietf-mmusic-sdp-mux-attributes]. 1306 Contact name, email: The IESG (iesg@ietf.org) 1308 Attribute name: simulcast 1310 Long-form attribute name: Simulcast stream description 1312 Charset dependent: No 1314 Attribute value: sc-value; see Section 5.1 of RFC XXXX. 1316 Purpose: Signals simulcast capability for a set of RTP streams 1318 MUX category: NORMAL 1319 Note to RFC Editor: Please replace "RFC XXXX" with the assigned 1320 number of this RFC. 1322 10. Security Considerations 1324 The simulcast capability, configuration attributes, and parameters 1325 are vulnerable to attacks in signaling. 1327 A false inclusion of the "a=simulcast" attribute may result in 1328 simultaneous transmission of multiple RTP streams that would 1329 otherwise not be generated. The impact is limited by the media 1330 description joint bandwidth, shared by all simulcast streams 1331 irrespective of their number. There may however be a large number of 1332 unwanted RTP streams that will impact the share of bandwidth 1333 allocated for the originally wanted RTP stream. 1335 A hostile removal of the "a=simulcast" attribute will result in 1336 simulcast not being used. 1338 Neither of the above will likely have any major consequences and can 1339 be mitigated by signaling that is at least integrity and source 1340 authenticated to prevent an attacker to change it. 1342 Security considerations related to the use of "a=rid" and the 1343 RtpStreamId SDES item is covered in [I-D.ietf-mmusic-rid] and 1344 [I-D.ietf-avtext-rid]. There are no additional security concerns 1345 related to their use in this specification. 1347 11. Contributors 1349 Morgan Lindqvist and Fredrik Jansson, both from Ericsson, have 1350 contributed with important material to the first versions of this 1351 document. Robert Hansen and Cullen Jennings, from Cisco, Peter 1352 Thatcher, from Google, and Adam Roach, from Mozilla, contributed 1353 significantly to subsequent versions. 1355 12. Acknowledgements 1357 The authors would like to thank Bernard Aboba, Thomas Belling, Roni 1358 Even, Adam Roach, Inaki Baz Castillo, Paul Kyzivat, and Arun 1359 Arunachalam for the feedback they provided during the development of 1360 this document. 1362 13. References 1363 13.1. Normative References 1365 [I-D.ietf-avtext-rid] 1366 Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream 1367 Identifier Source Description (SDES)", draft-ietf-avtext- 1368 rid-09 (work in progress), October 2016. 1370 [I-D.ietf-mmusic-rid] 1371 Roach, A., "RTP Payload Format Restrictions", draft-ietf- 1372 mmusic-rid-15 (work in progress), May 2018. 1374 [I-D.ietf-mmusic-sdp-bundle-negotiation] 1375 Holmberg, C., Alvestrand, H., and C. Jennings, 1376 "Negotiating Media Multiplexing Using the Session 1377 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 1378 negotiation-52 (work in progress), May 2018. 1380 [I-D.ietf-mmusic-sdp-mux-attributes] 1381 Nandakumar, S., "A Framework for SDP Attributes when 1382 Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-17 1383 (work in progress), February 2018. 1385 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1386 Requirement Levels", BCP 14, RFC 2119, 1387 DOI 10.17487/RFC2119, March 1997, 1388 . 1390 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1391 Jacobson, "RTP: A Transport Protocol for Real-Time 1392 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1393 July 2003, . 1395 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1396 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 1397 July 2006, . 1399 [RFC5234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax 1400 Specifications: ABNF", STD 68, RFC 5234, 1401 DOI 10.17487/RFC5234, January 2008, 1402 . 1404 [RFC7405] Kyzivat, P., "Case-Sensitive String Support in ABNF", 1405 RFC 7405, DOI 10.17487/RFC7405, December 2014, 1406 . 1408 [RFC7728] Burman, B., Akram, A., Even, R., and M. Westerlund, "RTP 1409 Stream Pause and Resume", RFC 7728, DOI 10.17487/RFC7728, 1410 February 2016, . 1412 [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 1413 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, 1414 May 2017, . 1416 13.2. Informative References 1418 [I-D.ietf-avtcore-multiplex-guidelines] 1419 Westerlund, M., Burman, B., Perkins, C., Alvestrand, H., 1420 Even, R., and H. Zheng, "Guidelines for using the 1421 Multiplexing Features of RTP to Support Multiple Media 1422 Streams", draft-ietf-avtcore-multiplex-guidelines-05 (work 1423 in progress), October 2017. 1425 [I-D.ietf-payload-flexible-fec-scheme] 1426 Zanaty, M., Singh, V., Begen, A., and G. Mandyam, "RTP 1427 Payload Format for Flexible Forward Error Correction 1428 (FEC)", draft-ietf-payload-flexible-fec-scheme-07 (work in 1429 progress), March 2018. 1431 [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., 1432 Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- 1433 Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, 1434 DOI 10.17487/RFC2198, September 1997, 1435 . 1437 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 1438 with Session Description Protocol (SDP)", RFC 3264, 1439 DOI 10.17487/RFC3264, June 2002, 1440 . 1442 [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for 1443 Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389, 1444 September 2002, . 1446 [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. 1447 Hakenberg, "RTP Retransmission Payload Format", RFC 4588, 1448 DOI 10.17487/RFC4588, July 2006, 1449 . 1451 [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF 1452 Digits, Telephony Tones, and Telephony Signals", RFC 4733, 1453 DOI 10.17487/RFC4733, December 2006, 1454 . 1456 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 1457 "Codec Control Messages in the RTP Audio-Visual Profile 1458 with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, 1459 February 2008, . 1461 [RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error 1462 Correction", RFC 5109, DOI 10.17487/RFC5109, December 1463 2007, . 1465 [RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding 1466 Dependency in the Session Description Protocol (SDP)", 1467 RFC 5583, DOI 10.17487/RFC5583, July 2009, 1468 . 1470 [RFC6184] Wang, Y., Even, R., Kristensen, T., and R. Jesup, "RTP 1471 Payload Format for H.264 Video", RFC 6184, 1472 DOI 10.17487/RFC6184, May 2011, 1473 . 1475 [RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, 1476 "RTP Payload Format for Scalable Video Coding", RFC 6190, 1477 DOI 10.17487/RFC6190, May 2011, 1478 . 1480 [RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image 1481 Attributes in the Session Description Protocol (SDP)", 1482 RFC 6236, DOI 10.17487/RFC6236, May 2011, 1483 . 1485 [RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time 1486 Transport Protocol (RTP) Header Extension for Client-to- 1487 Mixer Audio Level Indication", RFC 6464, 1488 DOI 10.17487/RFC6464, December 2011, 1489 . 1491 [RFC7104] Begen, A., Cai, Y., and H. Ou, "Duplication Grouping 1492 Semantics in the Session Description Protocol", RFC 7104, 1493 DOI 10.17487/RFC7104, January 2014, 1494 . 1496 [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and 1497 B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms 1498 for Real-Time Transport Protocol (RTP) Sources", RFC 7656, 1499 DOI 10.17487/RFC7656, November 2015, 1500 . 1502 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 1503 DOI 10.17487/RFC7667, November 2015, 1504 . 1506 [RFC7741] Westin, P., Lundin, H., Glover, M., Uberti, J., and F. 1507 Galligan, "RTP Payload Format for VP8 Video", RFC 7741, 1508 DOI 10.17487/RFC7741, March 2016, 1509 . 1511 [RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, 1512 "Sending Multiple RTP Streams in a Single RTP Session", 1513 RFC 8108, DOI 10.17487/RFC8108, March 2017, 1514 . 1516 [RFC8285] Singer, D., Desineni, H., and R. Even, Ed., "A General 1517 Mechanism for RTP Header Extensions", RFC 8285, 1518 DOI 10.17487/RFC8285, October 2017, 1519 . 1521 Appendix A. Requirements 1523 The following requirements are met by the defined solution to support 1524 the use cases (Section 3): 1526 REQ-1: Identification: 1528 REQ-1.1: It must be possible to identify a set of simulcasted RTP 1529 streams as originating from the same media source in SDP 1530 signaling. 1532 REQ-1.2: An RTP endpoint must be capable of identifying the 1533 simulcast stream a received RTP stream is associated with, 1534 knowing the content of the SDP signalling. 1536 REQ-2: Transport usage. The solution must work when using: 1538 REQ-2.1: Legacy SDP with separate media transports per SDP media 1539 description. 1541 REQ-2.2: Bundled [I-D.ietf-mmusic-sdp-bundle-negotiation] SDP 1542 media descriptions. 1544 REQ-3: Capability negotiation. It must be possible that: 1546 REQ-3.1: Sender can express capability of sending simulcast. 1548 REQ-3.2: Receiver can express capability of receiving simulcast. 1550 REQ-3.3: Sender can express maximum number of simulcast streams 1551 that can be provided. 1553 REQ-3.4: Receiver can express maximum number of simulcast streams 1554 that can be received. 1556 REQ-3.5: Sender can detail the characteristics of the simulcast 1557 streams that can be provided. 1559 REQ-3.6: Receiver can detail the characteristics of the simulcast 1560 streams that it prefers to receive. 1562 REQ-4: Distinguishing features. It must be possible to have 1563 different simulcast streams use different codec parameters, as can 1564 be expressed by SDP format values and RTP payload types. 1566 REQ-5: Compatibility. It must be possible to use simulcast in 1567 combination with other RTP mechanisms that generate additional RTP 1568 streams: 1570 REQ-5.1: RTP Retransmission [RFC4588]. 1572 REQ-5.2: RTP Forward Error Correction [RFC5109]. 1574 REQ-5.3: Related payload types such as audio Comfort Noise and/or 1575 DTMF. 1577 REQ-5.4: A single simulcast stream can consist of multiple RTP 1578 streams, to support codecs where a dependent stream is 1579 dependent on a set of encoded and dependent streams, each 1580 potentially carried in their own RTP stream. 1582 REQ-6: Interoperability. The solution must be possible to use in: 1584 REQ-6.1: Interworking with non-simulcast legacy clients using a 1585 single media source per media type. 1587 REQ-6.2: WebRTC environment with a single media source per SDP 1588 media description. 1590 Appendix B. Changes From Earlier Versions 1592 NOTE TO RFC EDITOR: Please remove this section prior to publication. 1594 B.1. Modifications Between WG Version -12 and -13 1596 o Examples corrected to follow RID ABNF 1598 o Example Figure 7 now comments on priority for second media source. 1600 o Clarified a SHOULD limitation. 1602 o Added urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id in 1603 examples with RTX. 1605 o ABNF now uses RFC 7405 to indicate case sensitivity 1607 o Various minor editorials and nits. 1609 B.2. Modifications Between WG Version -11 and -12 1611 o Modified Normative statement regarding RTP stream duplication in 1612 Section 5.2. 1614 o Clarified assumption about use of congestion control by 1615 applications. 1617 o Changed to use RFC 8174 boilerplate instead of RFC 2119. 1619 o Clarified explanation of syntax for simulcast attribute in 1620 Section 4. 1622 o Editorial clarification in Section 5.2 and 5.3.2. 1624 o Various minor editorials and nits. 1626 B.3. Modifications Between WG Version -10 and -11 1628 o Added new SDP example section on Simulcast and Redundancy, 1629 including both RED (RFC2198), RTP RTX (RFC4588), and FEC (draft- 1630 ietf-payload-flexible-fec-scheme). 1632 o Removed restriction that "related" payload formats in an RTP 1633 stream (such as CN and DTMF) must not have their own rid-id, since 1634 there is no reason to forbid this and corresponding clarification 1635 is made in draft-ietf-mmusic-rid. 1637 o Removed any mention of source-specific signaling and the reference 1638 to RFC5576, since draft-ietf-mmusic-rid is not defined for source- 1639 specific signaling. 1641 o Changed some SDP examples to use a=rid restrictions instead of 1642 a=imageattr. 1644 o Changed reference from the obsoleted RFC 5285 to RFC 8285. 1646 B.4. Modifications Between WG Version -09 and -10 1648 o Amended overview section with a bit more explanation on the 1649 examples, and added an rid-id alternative for one of the streams. 1651 o Removed SCID also from the Terminology section, which was 1652 forgotten in -09 when changing SCID to rid-id. 1654 B.5. Modifications Between WG Version -08 and -09 1656 o Changed SCID to rid-id, to align with ietf-draft-mmusic-rid 1657 naming. 1659 o Changed Overview to be based on examples and shortened it. 1661 o Changed semantics of initially paused rid-id in modified SDP 1662 offers from requiring it to follow actual RFC 7728 pause state to 1663 an informational offerer's opinion at the time of offer creation, 1664 not in any way overriding or amending RFC 7728 signaling. 1666 o Replaced text on ignoring all but the first of multiple 1667 "a=simulcast" lines in a media description with mandating that at 1668 most one "a=simulcast" line is included. 1670 o Clarified with a note that, for the case it is clear from the SDP 1671 that RTP PT uniquely maps to RtpStreamId, an RTP receiver can use 1672 RTP PT to relate simulcast streams. 1674 o Moved Section 4 Requirements to become Appendix A. 1676 o Editorial corrections and clarifications. 1678 B.6. Modifications Between WG Version -07 and -08 1680 o Correcting syntax of SDP examples in section 6.6.1, as found by 1681 Inaki Baz Castillo. 1683 o Changing ABNF to only define the sc-value, not the SDP attribute 1684 itself, as suggested by Paul Kyzivat. 1686 o Changing I-D reference to newly published RFC 8108. 1688 o Adding list of modifications between -06 and -07. 1690 B.7. Modifications Between WG Version -06 and -07 1692 o A scope clarification, as result of the discussion with Roni Even. 1694 o A reformulation of the identification requirements for simulcast 1695 stream. 1697 o Correcting the statement related to source specific signalling 1698 (RFC 5576) to address Roni Even's comment. 1700 o Update of the last paragraph in Section 6.2 regarding simulcast 1701 stream differences as well as forbidding multiple instances of the 1702 same SCID within a single a=simulcast line. 1704 o Removal of note in Section 6.4 as result of issue raised by Roni 1705 Even. 1707 o Use of "m=" has been changed to media description and a few other 1708 editorial improvements and clarifications. 1710 B.8. Modifications Between WG Version -05 and -06 1712 o Added section on RTP Aspects 1714 o Added a requirement (5-4) on that capability exchange must be 1715 capable of handling multi RTP stream cases. 1717 o Added extmap attribute also on first signalling example as it is a 1718 recommended to use mechanism. 1720 o Clarified the definition of the simulcast attribute and how 1721 simulcast streams relates to simulcast formats and SCIDs. 1723 o Updated References list and moved around some references between 1724 informative and normative categories. 1726 o Editorial improvements and corrections. 1728 B.9. Modifications Between WG Version -04 and -05 1730 o Aligned with recent changes in draft-ietf-mmusic-rid and draft- 1731 ietf-avtext-rid. 1733 o Modified the SDP offer/answer section to follow the generally 1734 accepted structure, also adding a brief text on modifying the 1735 session that is aligned with draft-ietf-mmusic-rid. 1737 o Improved text around simulcast stream identification (as opposed 1738 to the simulcast stream itself) to consistently use the acronym 1739 SCID and defined that in the Terminology section. 1741 o Changed references for RTP-level pause/resume and VP8 payload 1742 format that are now published as RFC. 1744 o Improved IANA registration text. 1746 o Removed unused reference to draft-ietf-payload-flexible-fec- 1747 scheme. 1749 o Editorial improvements and corrections. 1751 B.10. Modifications Between WG Version -03 and -04 1753 o Changed to only use RID identification, as was consensus during 1754 IETF 94. 1756 o ABNF improvements. 1758 o Clarified offer-answer rules for initially paused streams. 1760 o Changed references for RTP topologies and RTP taxonomy documents 1761 that are now published as RFC. 1763 o Added reference to the new RID draft in AVTEXT. 1765 o Re-structured section 6 to provide an easy reference by the 1766 updated IANA section. 1768 o Added a sub-section 7.1 with a discussion of bitrate adaptation. 1770 o Editorial improvements. 1772 B.11. Modifications Between WG Version -02 and -03 1774 o Removed text on multicast / broadcast from use cases, since it is 1775 not supported by the solution. 1777 o Removed explicit references to unified plan draft. 1779 o Added possibility to initiate simulcast streams in paused mode. 1781 o Enabled an offerer to offer multiple stream identification (pt or 1782 rid) methods and have the answerer choose which to use. 1784 o Added a preference indication also in send direction offers. 1786 o Added a section on limitations of the current proposal, including 1787 identification method specific limitations. 1789 B.12. Modifications Between WG Version -01 and -02 1791 o Relying on the new RID solution for codec constraints and 1792 configuration identification. This has resulted in changes in 1793 syntax to identify if pt or RID is used to describe the simulcast 1794 stream. 1796 o Renamed simulcast version and simulcast version alternative to 1797 simulcast stream and simulcast format respectively, and improved 1798 definitions for them. 1800 o Clarification that it is possible to switch between simulcast 1801 version alternatives, but that only a single one be used at any 1802 point in time. 1804 o Changed the definition so that ordering of simulcast formats for a 1805 specific simulcast stream do have a preference order. 1807 B.13. Modifications Between WG Version -00 and -01 1809 o No changes. Only preventing expiry. 1811 B.14. Modifications Between Individual Version -00 and WG Version -00 1813 o Added this appendix. 1815 Authors' Addresses 1817 Bo Burman 1818 Ericsson 1819 Gronlandsgatan 31 1820 SE-164 60 Stockholm 1821 Sweden 1823 Email: bo.burman@ericsson.com 1825 Magnus Westerlund 1826 Ericsson 1827 Farogatan 2 1828 SE-164 80 Stockholm 1829 Sweden 1831 Phone: +46 10 714 82 87 1832 Email: magnus.westerlund@ericsson.com 1833 Suhas Nandakumar 1834 Cisco 1835 170 West Tasman Drive 1836 San Jose, CA 95134 1837 USA 1839 Email: snandaku@cisco.com 1841 Mo Zanaty 1842 Cisco 1843 170 West Tasman Drive 1844 San Jose, CA 95134 1845 USA 1847 Email: mzanaty@cisco.com