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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: January 4, 2018 S. Nandakumar 6 M. Zanaty 7 Cisco 8 July 3, 2017 10 Using Simulcast in SDP and RTP Sessions 11 draft-ietf-mmusic-sdp-simulcast-09 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 http://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 January 4, 2018. 43 Copyright Notice 45 Copyright (c) 2017 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 (http://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 . . . . . . . . . . . . . . . . . . . 9 71 5.2. Simulcast Capability . . . . . . . . . . . . . . . . . . 10 72 5.3. Offer/Answer Use . . . . . . . . . . . . . . . . . . . . 12 73 5.3.1. Generating the Initial SDP Offer . . . . . . . . . . 12 74 5.3.2. Creating the SDP Answer . . . . . . . . . . . . . . . 12 75 5.3.3. Offerer Processing the SDP Answer . . . . . . . . . . 13 76 5.3.4. Modifying the Session . . . . . . . . . . . . . . . . 14 77 5.4. Use with Declarative SDP . . . . . . . . . . . . . . . . 15 78 5.5. Relating Simulcast Streams . . . . . . . . . . . . . . . 15 79 5.6. Signaling Examples . . . . . . . . . . . . . . . . . . . 15 80 5.6.1. Single-Source Client . . . . . . . . . . . . . . . . 16 81 5.6.2. Multi-Source Client . . . . . . . . . . . . . . . . . 17 82 6. RTP Aspects . . . . . . . . . . . . . . . . . . . . . . . . . 20 83 6.1. Outgoing from Endpoint with Media Source . . . . . . . . 20 84 6.2. RTP Middlebox to Receiver . . . . . . . . . . . . . . . . 20 85 6.2.1. Media-Switching Mixer . . . . . . . . . . . . . . . . 22 86 6.2.2. Selective Forwarding Middlebox . . . . . . . . . . . 23 87 6.3. RTP Middlebox to RTP Middlebox . . . . . . . . . . . . . 24 88 7. Network Aspects . . . . . . . . . . . . . . . . . . . . . . . 25 89 7.1. Bitrate Adaptation . . . . . . . . . . . . . . . . . . . 25 90 8. Limitation . . . . . . . . . . . . . . . . . . . . . . . . . 26 91 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 26 92 10. Security Considerations . . . . . . . . . . . . . . . . . . . 27 93 11. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 27 94 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 27 95 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 27 96 13.1. Normative References . . . . . . . . . . . . . . . . . . 27 97 13.2. Informative References . . . . . . . . . . . . . . . . . 28 98 Appendix A. Requirements . . . . . . . . . . . . . . . . . . . . 31 99 Appendix B. Changes From Earlier Versions . . . . . . . . . . . 32 100 B.1. Modifications Between WG Version -08 and -09 . . . . . . 32 101 B.2. Modifications Between WG Version -07 and -08 . . . . . . 33 102 B.3. Modifications Between WG Version -06 and -07 . . . . . . 33 103 B.4. Modifications Between WG Version -05 and -06 . . . . . . 33 104 B.5. Modifications Between WG Version -04 and -05 . . . . . . 34 105 B.6. Modifications Between WG Version -03 and -04 . . . . . . 34 106 B.7. Modifications Between WG Version -02 and -03 . . . . . . 35 107 B.8. Modifications Between WG Version -01 and -02 . . . . . . 35 108 B.9. Modifications Between WG Version -00 and -01 . . . . . . 35 109 B.10. Modifications Between Individual Version -00 and WG 110 Version -00 . . . . . . . . . . . . . . . . . . . . . . . 35 111 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 35 113 1. Introduction 115 Most of today's multiparty video conference solutions make use of 116 centralized servers to reduce the bandwidth and CPU consumption in 117 the endpoints. Those servers receive RTP streams from each 118 participant and send some suitable set of possibly modified RTP 119 streams to the rest of the participants, which usually have 120 heterogeneous capabilities (screen size, CPU, bandwidth, codec, etc). 121 One of the biggest issues is how to perform RTP stream adaptation to 122 different participants' constraints with the minimum possible impact 123 on both video quality and server performance. 125 Simulcast is defined in this memo as the act of simultaneously 126 sending multiple different encoded streams of the same media source, 127 e.g. the same video source encoded with different video encoder types 128 or image resolutions. This can be done in several ways and for 129 different purposes. This document focuses on the case where it is 130 desirable to provide a media source as multiple encoded streams over 131 RTP [RFC3550] towards an intermediary so that the intermediary can 132 provide the wanted functionality by selecting which RTP stream(s) to 133 forward to other participants in the session, and more specifically 134 how the identification and grouping of the involved RTP streams are 135 done. 137 The intended scope of the defined mechanism is to support negotiation 138 and usage of simulcast when using SDP offer/answer and media 139 transport over RTP. The media transport topologies considered are 140 point to point RTP sessions as well as centralized multi-party RTP 141 sessions, where a media sender will provide the simulcasted streams 142 to an RTP middlebox or endpoint, and middleboxes may further 143 distribute the simulcast streams to other middleboxes or endpoints. 145 Usage of multicast or broadcast transport is out of scope and left 146 for future extension. 148 This document describes a few scenarios where it is motivated to use 149 simulcast, and also defines the needed RTP/RTCP and SDP signaling for 150 it. 152 2. Definitions 154 2.1. Terminology 156 This document makes use of the terminology defined in RTP Taxonomy 157 [RFC7656], and RTP Topologies [RFC7667]. The following terms are 158 especially noted or here defined: 160 RTP Mixer: An RTP middle node, defined in [RFC7667] (Section 3.6 to 161 3.9). 163 RTP Session: An association among a group of participants 164 communicating with RTP, as defined in [RFC3550] and amended by 165 [RFC7656]. 167 RTP Stream: A stream of RTP packets containing media data, as 168 defined in [RFC7656]. 170 RTP Switch: A common short term for the terms "switching RTP mixer", 171 "source projecting middlebox", and "video switching MCU" as 172 discussed in [RFC7667]. 174 Simulcast Stream: One encoded stream or dependent stream from a set 175 of concurrently transmitted encoded streams and optional dependent 176 streams, all sharing a common media source, as defined in 177 [RFC7656]. For example, HD and thumbnail video simulcast versions 178 of a single media source sent concurrently as separate RTP 179 Streams. 181 Simulcast Format: Different formats of a simulcast stream serve the 182 same purpose as alternative RTP payload types in non-simulcast 183 SDP: to allow multiple alternative media formats for a given RTP 184 stream. As for multiple RTP payload types on the m-line in offer/ 185 answer [RFC3264], any one of the negotiated alternative formats 186 can be used in a single RTP stream at a given point in time, but 187 not more than one (based on RTP timestamp). What format is used 188 can change dynamically from one RTP packet to another. 190 Simulcast Stream Identifier (SCID): The identification value used to 191 refer to an individual simulcast format, identical to the "rid-id" 192 identification value for an RTP Payload Format Restriction 194 [I-D.ietf-mmusic-rid] and the corresponding content of 195 "RtpStreamId" RTCP SDES Item [I-D.ietf-avtext-rid]. 197 2.2. Requirements Language 199 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 200 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 201 document are to be interpreted as described in RFC 2119 [RFC2119]. 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 principle 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. 237 The use of simulcast relates to the latter approach, where it is more 238 important to reduce the load on the RTP Mixer and/or minimize QoE 239 impact than to achieve an optimal adaptation of resource usage. 241 3.1. Reaching a Diverse Set of Receivers 243 The media sources provided by a sending participant potentially need 244 to reach several receiving participants that differ in terms of 245 available resources. The receiver resources that typically differ 246 include, but are not limited to: 248 Codec: This includes codec type (such as SDP MIME type) and can 249 include codec configuration. A couple of codec resources that 250 differ only in codec configuration will be "different" if they are 251 somehow not "compatible", like if they differ in video codec 252 profile, or the transport packetization configuration. 254 Sampling: This relates to how the media source is sampled, in 255 spatial as well as in temporal domain. For video streams, spatial 256 sampling affects image resolution and temporal sampling affects 257 video frame rate. For audio, spatial sampling relates to the 258 number of audio channels and temporal sampling affects audio 259 bandwidth. This may be used to suit different rendering 260 capabilities or needs at the receiving endpoints. 262 Bitrate: This relates to the amount of bits sent per second to 263 transmit the media source as an RTP stream, which typically also 264 affects the Quality of Experience (QoE) for the receiving user. 266 Letting the sending participant create a simulcast of a few 267 differently configured RTP streams per media source can be a good 268 tradeoff when using an RTP switch as middlebox, instead of sending a 269 single RTP stream and using an RTP mixer to create individual 270 transcodings to each receiving participant. 272 This requires that the receiving participants can be categorized in 273 terms of available resources and that the sending participant can 274 choose a matching configuration for a single RTP stream per category 275 and media source. For example, a set of receiving participants 276 differ only in screen resolution; some are able to display video with 277 at most 360p resolution and some support 720p resolution. A sending 278 participant can then reach all receivers with best possible 279 resolution by creating a simulcast of RTP streams with 360p and 720p 280 resolution for each sent video media source. 282 The maximum number of simulcasted RTP streams that can be sent is 283 mainly limited by the amount of processing and uplink network 284 resources available to the sending participant. 286 3.2. Application Specific Media Source Handling 288 The application logic that controls the communication session may 289 include special handling of some media sources. It is, for example, 290 commonly the case that the media from a sending participant is not 291 sent back to itself. 293 It is also common that a currently active speaker participant is 294 shown in larger size or higher quality than other participants (the 295 sampling or bitrate aspects of Section 3.1). Not sending the active 296 speaker media back to itself means there is some other participant's 297 media that instead has to receive special handling towards the active 298 speaker; typically the previous active speaker. This way, the 299 previously active speaker is needed both in larger size (to current 300 active speaker) and in small size (to the rest of the participants), 301 which can be solved with a simulcast from the previously active 302 speaker to the RTP switch. 304 3.3. Receiver Media Source Preferences 306 The application logic that controls the communication session may 307 allow receiving participants to apply preferences to the 308 characteristics of the RTP stream they receive, for example in terms 309 of the aspects listed in Section 3.1. Sending a simulcast of RTP 310 streams is one way of accommodating receivers with conflicting or 311 otherwise incompatible preferences. 313 4. Overview 315 This memo defines SDP [RFC4566] signaling that covers the above 316 described simulcast use cases and functionalities. A number of 317 requirements for such signaling are elaborated in Appendix A. 319 A new SDP media level attribute "a=simulcast" is defined: 321 m=video 49300 RTP/AVP 97 98 322 a=rtpmap:97 H264/90000 323 a=rtpmap:98 H264/90000 324 a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000 325 a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600 326 a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720] 327 a=imageattr:98 send [x=320,y=180] recv [x=320,y=180] 328 a=rid:1 send pt=97 329 a=rid:2 send pt=98 330 a=rid:3 recv pt=97 331 a=simulcast:send 1;2 recv 3 332 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 334 Figure 1: Example Simulcast Media Description in Offer 336 The corresponding SDP answer media description example extract could 337 look like: 339 m=video 49674 RTP/AVP 97 98 340 a=rtpmap:97 H264/90000 341 a=rtpmap:98 H264/90000 342 a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000 343 a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600 344 a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720] 345 a=imageattr:98 send [x=320,y=180] recv [x=320,y=180] 346 a=rid:1 recv pt=97 347 a=rid:2 recv pt=98 348 a=rid:3 send pt=97 349 a=simulcast:recv 1;2 send 3 350 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 352 Figure 2: Example Simulcast Media Description in Answer 354 The above are only SDP media description extracts, not a complete 355 SDP. The only difference to non-simulcast SDP media descriptions is 356 the added "a=simulcast" line. It is assumed that a single SDP media 357 description is used to describe a single media source. This is 358 aligned with the concepts defined in [RFC7656] and will work in a 359 WebRTC context, both with and without BUNDLE 360 [I-D.ietf-mmusic-sdp-bundle-negotiation] grouping of media 361 descriptions. 363 The "a=simulcast" line describes send and receive direction simulcast 364 streams separately. Each direction can in turn describe one or more 365 simulcast streams, separated by semicolon. The identifiers 366 describing simulcast streams on the "a=simulcast" line are rid-id, as 367 defined by "a=rid" lines in [I-D.ietf-mmusic-rid]. Each simulcast 368 stream can be offered as a list of alternative rid-id, with each 369 alternative separated by comma (not in the examples above). A 370 detailed specification can be found in Section 5 and more detailed 371 examples are outlined in Section 5.6. 373 5. Detailed Description 375 This section further details the overview above (Section 4). First, 376 formal syntax is provided (Section 5.1), followed by the rest of the 377 SDP attribute definition in Section 5.2. Relating Simulcast Streams 378 (Section 5.5) provides the definition of the RTP/RTCP mechanisms 379 used. The section is concluded with a number of examples. 381 5.1. Simulcast Attribute 383 This document defines a new SDP media-level "a=simulcast" attribute, 384 with value according to the following ABNF [RFC5234] syntax: 386 sc-value = ( sc-send [SP sc-recv] ) / ( sc-recv [SP sc-send] ) 387 sc-send = "send" SP sc-str-list 388 sc-recv = "recv" SP sc-str-list 389 sc-str-list = sc-alt-list *( ";" sc-alt-list ) 390 sc-alt-list = sc-id *( "," sc-id ) 391 sc-id-paused = "~" 392 sc-id = [sc-id-paused] rid-id 393 ; SP defined in [RFC5234] 394 ; rid-id defined in [I-D.ietf-mmusic-rid] 396 Figure 3: ABNF for Simulcast Value 398 The "a=simulcast" attribute has a parameter in the form of one or two 399 simulcast stream descriptions, each consisting of a direction ("send" 400 or "recv"), followed by a list of one or more simulcast streams. 401 Each simulcast stream consists of one or more alternative simulcast 402 formats. Each simulcast format is identified by a simulcast stream 403 identifier (rid-id). The rid-id MUST have the form of an RTP stream 404 identifier, as described by RTP Payload Format Restrictions 405 [I-D.ietf-mmusic-rid]. 407 In the list of simulcast streams, each simulcast stream is separated 408 by a semicolon (";"). Each simulcast stream can in turn be offered 409 in one or more alternative formats, represented by rid-ids, separated 410 by a comma (","). Each rid-id can also be specified as initially 411 paused [RFC7728], indicated by prepending a "~" to the rid-id. The 412 reason to allow separate initial pause states for each rid-id is that 413 pause capability can be specified individually for each RTP payload 414 type referenced by an rid-id. Since pause capability specified via 415 the "a=rtcp-fb" attribute and rid-id specified by "a=rid" can refer 416 to common payload types, it is unfeasible to pause streams with rid- 417 id where any of the related RTP payload type(s) do not have pause 418 capability. 420 It is possible to use source-specific signaling [RFC5576] with 421 "a=simulcast", but it is only in certain cases possible to learn from 422 that signaling which SSRC will belong to a particular simulcast 423 stream. 425 5.2. Simulcast Capability 427 Simulcast capability is expressed through a new media level SDP 428 attribute, "a=simulcast" (Section 5.1). The meaning of the attribute 429 on SDP session level is undefined, MUST NOT be used by 430 implementations of this specification and MUST be ignored if received 431 on session level. Extensions to this specification MAY define such 432 session level usage. Each SDP media description MUST contain at most 433 one "a=simulcast" line. 435 There are separate and independent sets of simulcast streams in send 436 and receive directions. When listing multiple directions, each 437 direction MUST NOT occur more than once on the same line. 439 Simulcast streams using undefined rid-id MUST NOT be used as valid 440 simulcast streams by an RTP stream receiver. The direction for an 441 rid-id MUST be aligned with the direction specified for the 442 corresponding RTP stream identifier on the "a=rid" line. 444 The listed number of simulcast streams for a direction sets a limit 445 to the number of supported simulcast streams in that direction. The 446 order of the listed simulcast streams in the "send" direction 447 suggests a proposed order of preference, in decreasing order: the 448 rid-id listed first is the most preferred and subsequent streams have 449 progressively lower preference. The order of the listed rid-id in 450 the "recv" direction expresses which simulcast streams that are 451 preferred, with the leftmost being most preferred. This can be of 452 importance if the number of actually sent simulcast streams have to 453 be reduced for some reason. 455 rid-id that have explicit dependencies [RFC5583] 456 [I-D.ietf-mmusic-rid] to other rid-id (even in the same media 457 description) MAY be used. 459 Use of more than a single, alternative simulcast format for a 460 simulcast stream MAY be specified as part of the attribute parameters 461 by expressing the simulcast stream as a comma-separated list of 462 alternative rid-id. In this case, it is not possible to align what 463 alternative rid-id that are used across different simulcast streams, 464 like requiring all simulcast streams to use rid-id alternatives 465 referring to the same codec format. The order of the rid-id 466 alternatives within a simulcast stream is significant; the rid-id 467 alternatives are listed from (left) most preferred to (right) least 468 preferred. For the use of simulcast, this overrides the normal codec 469 preference as expressed by format type ordering on the "m=" line, 470 using regular SDP rules. This is to enable a separation of general 471 codec preferences and simulcast stream configuration preferences. 473 A simulcast stream can use a codec defined such that the same RTP 474 SSRC can change RTP payload type multiple times during a session, 475 possibly even on a per-packet basis. A typical example can be a 476 speech codec that makes use of Comfort Noise [RFC3389] and/or DTMF 477 [RFC4733] formats. In those cases, such "related" formats MUST NOT 478 be defined as having their own rid-id listed explicitly in the 479 attribute parameters, since they are not strictly simulcast streams 480 of the media source, but rather a specific way of generating the RTP 481 stream of a single simulcast stream with varying RTP payload type. 483 If RTP stream pause/resume [RFC7728] is supported, any rid-id MAY be 484 prefixed by a "~" character to indicate that the corresponding 485 simulcast stream is initially paused already from start of the RTP 486 session. In this case, support for RTP stream pause/resume MUST also 487 be included under the same "m=" line where "a=simulcast" is included. 488 All RTP payload types related to such initially paused simulcast 489 stream MUST be listed in the SDP as pause/resume capable as specified 490 by [RFC7728], e.g. by using the "*" wildcard format for "a=rtcp-fb". 492 An initially paused simulcast stream in "send" direction for the part 493 sending the SDP MUST be considered equivalent to an unsolicited 494 locally paused stream, and be handled accordingly. Initially paused 495 simulcast streams are resumed as described by the RTP pause/resume 496 specification. An RTP stream receiver that wishes to resume an 497 unsolicited locally paused stream needs to know the SSRC of that 498 stream. The SSRC of an initially paused simulcast stream can be 499 obtained from an RTP stream sender RTCP Sender Report (SR) including 500 both the desired SSRC as "SSRC of sender", and the rid-id value in an 501 RtpStreamId RTCP SDES item [I-D.ietf-avtext-rid]. 503 Including an initially paused simulcast stream in "recv" direction 504 for the part sending the SDP, sent towards an RTP sender, SHOULD 505 cause the remote RTP sender to put the stream as unsolicited locally 506 paused, unless there are other RTP stream receivers that do not mark 507 the simulcast stream as initially paused. The reason to require an 508 initially paused "recv" stream to be considered locally paused by the 509 remote RTP sender, instead of making it equivalent to implicitly 510 sending a pause request, is because the pausing RTP sender cannot 511 know which receiving SSRC owns the restriction when TMMBR/TMMBN are 512 used for pause/resume signaling (Section 5.6 of [RFC7728]) since the 513 RTP receiver's SSRC in send direction is sometimes not yet known. 515 Use of the redundant audio data [RFC2198] format could be seen as a 516 form of simulcast for loss protection purposes, but is not considered 517 conflicting with the mechanisms described in this memo and MAY 518 therefore be used as any other format. In this case the "red" 519 format, rather than the carried formats, SHOULD be the one to list as 520 a simulcast stream on the "a=simulcast" line. 522 The media formats and corresponding characteristics of simulcast 523 streams SHOULD be chosen such that they are different, e.g. as 524 different SDP formats with differing "a=rtpmap" and/or "a=fmtp" 525 lines, or as differently defined RTP payload format restrictions. If 526 this difference is not required, RTP duplication [RFC7104] procedures 527 SHOULD be considered instead of simulcast. To avoid complications in 528 implementations, a single rid-id MUST NOT occur more than once per 529 "a=simulcast" line. Note that this does not eliminate use of 530 simulcast as an RTP duplication mechanism, since it is possible to 531 define multiple different rid-id that are effectively equivalent. 533 5.3. Offer/Answer Use 535 Note: The inclusion of "a=simulcast" or the use of simulcast does 536 not change any of the interpretation or Offer/Answer procedures 537 for other SDP attributes, like "a=fmtp" or "a=rid". 539 5.3.1. Generating the Initial SDP Offer 541 An offerer wanting to use simulcast for a media description SHALL 542 include one "a=simulcast" attribute in that media description in the 543 offer. An offerer listing a set of receive simulcast streams and/or 544 alternative formats as rid-id in the offer MUST be prepared to 545 receive RTP streams for any of those simulcast streams and/or 546 alternative formats from the answerer. 548 5.3.2. Creating the SDP Answer 550 An answerer that does not understand the concept of simulcast will 551 also not know the attribute and will remove it in the SDP answer, as 552 defined in existing SDP Offer/Answer [RFC3264] procedures. Since SDP 553 session level simulcast is undefined in this memo, an answerer that 554 receives an offer with the "a=simulcast" attribute on SDP session 555 level SHALL remove it in the answer. An answerer that understands 556 the attribute but receives multiple "a=simulcast" attributes in the 557 same media description SHALL disable use of simulcast by removing all 558 "a=simulcast" lines for that media description in the answer. 560 An answerer that does understand the attribute and that wants to 561 support simulcast in an indicated direction SHALL reverse 562 directionality of the unidirectional direction parameters; "send" 563 becomes "recv" and vice versa, and include it in the answer. 565 An answerer that receives an offer with simulcast containing an 566 "a=simulcast" attribute listing alternative rid-id MAY keep all the 567 alternative rid-id in the answer, but it MAY also choose to remove 568 any non-desirable alternative rid-id in the answer. The answerer 569 MUST NOT add any alternative rid-id in send direction in the answer 570 that were not present in the offer receive direction. The answerer 571 MUST be prepared to receive any of the receive direction rid-id 572 alternatives, and MAY send any of the send direction alternatives 573 that are kept in the answer. 575 An answerer that receives an offer with simulcast that lists a number 576 of simulcast streams, MAY reduce the number of simulcast streams in 577 the answer, but MUST NOT add simulcast streams. 579 An answerer that receives an offer without RTP stream pause/resume 580 capability MUST NOT mark any simulcast streams as initially paused in 581 the answer. 583 An RTP stream pause/resume capable answerer that receives an offer 584 with RTP stream pause/resume capability MAY mark any rid-id that 585 refer to pause/resume capable formats as initially paused in the 586 answer. 588 An answerer that receives indication in an offer of an rid-id being 589 initially paused SHOULD mark that rid-id as initially paused also in 590 the answer, regardless of direction, unless it has good reason for 591 the rid-id not being initially paused. One reason to remove an 592 initial pause in the answer compared to the offer could, for example, 593 be that all receive direction simulcast streams for a media source 594 the answerer accepts in the answer would otherwise be paused. 596 5.3.3. Offerer Processing the SDP Answer 598 An offerer that receives an answer without "a=simulcast" MUST NOT use 599 simulcast towards the answerer. An offerer that receives an answer 600 with "a=simulcast" without any rid-id in a specified direction MUST 601 NOT use simulcast in that direction. 603 An offerer that receives an answer where some rid-id alternatives are 604 kept MUST be prepared to receive any of the kept send direction rid- 605 id alternatives, and MAY send any of the kept receive direction rid- 606 id alternatives. 608 An offerer that receives an answer where some of the rid-id are 609 removed compared to the offer MAY release the corresponding resources 610 (codec, transport, etc) in its receive direction and MUST NOT send 611 any RTP packets corresponding to the removed rid-id. 613 An offerer that offered some of its rid-id as initially paused and 614 that receives an answer that does not indicate RTP stream pause/ 615 resume capability, MUST NOT initially pause any simulcast streams. 617 An offerer with RTP stream pause/resume capability that receives an 618 answer where some rid-id are marked as initially paused, SHOULD 619 initially pause those RTP streams regardless if they were marked as 620 initially paused also in the offer, unless it has good reason for 621 those RTP streams not being initially paused. One such reason could, 622 for example, be that the answerer would otherwise initially not 623 receive any media of that type at all. 625 5.3.4. Modifying the Session 627 Offers inside an existing session follow the same rules as for 628 initial SDP offer, with these additions: 630 1. rid-id marked as initially paused in the offerer's send direction 631 SHALL reflect the offerer's opinion of the current pause state at 632 the time of creating the offer. This is purely informational, 633 and RTP stream pause/resume [RFC7728] signaling in the ongoing 634 session SHALL take precedence in case of any conflict or 635 ambiguity. 637 2. rid-id marked as initally paused in the offerer's receive 638 direction SHALL (as in an initial offer) reflect the offerer's 639 desired rid-id pause state. Except for the case where the 640 offerer already paused the corresponding RTP stream through RTP 641 stream pause/resume [RFC7728] signaling , this is identical to 642 the conditions at an initial offer. 644 Creation of SDP answers and processing of SDP answers inside an 645 existing session follow the same rules as described above for initial 646 SDP offer/answer. 648 Session modification restrictions in section 6.5 of RTP payload 649 format restrictions [I-D.ietf-mmusic-rid] also apply. 651 5.4. Use with Declarative SDP 653 This document does not define the use of "a=simulcast" in declarative 654 SDP, partly motivated by use of the simulcast format identification 655 [I-D.ietf-mmusic-rid] not being defined for use in declarative SDP. 656 If concrete use cases for simulcast in declarative SDP are identified 657 in the future, the authors of this memo expect that additional 658 specifications will address such use. 660 5.5. Relating Simulcast Streams 662 Simulcast RTP streams MUST be related on RTP level through 663 RtpStreamId [I-D.ietf-avtext-rid], as specified in the SDP 664 "a=simulcast" attribute (Section 5.2) parameters. This is sufficient 665 as long as there is only a single media source per SDP media 666 description. When using BUNDLE 667 [I-D.ietf-mmusic-sdp-bundle-negotiation], where multiple SDP media 668 descriptions jointly specify a single RTP session, the SDES MID 669 identification mechanism in BUNDLE allows relating RTP streams back 670 to individual media descriptions, after which the above described 671 RtpStreamId relations can be used. Use of the RTP header extension 672 [RFC5285] for both MID and RtpStreamId identifications can be 673 important to ensure rapid initial reception, required to correctly 674 interpret and process the RTP streams. Implementers of this 675 specification MUST support the RTCP source description (SDES) item 676 method and SHOULD support RTP header extension method to signal 677 RtpStreamId on RTP level. 679 NOTE: For the case where it is clear from SDP that RTP PT uniquely 680 maps to corresponding RtpStreamId, an RTP receiver can use RTP PT 681 to relate simulcast streams. This can sometimes enable decoding 682 even in advance to receiving RtpStreamId information in RTCP SDES 683 and/or RTP header extensions. 685 RTP streams MUST only use a single alternative rid-id at a time 686 (based on RTP timestamps), but MAY change format (and rid-id) on a 687 per-RTP packet basis. This corresponds to the existing (non- 688 simulcast) SDP offer/answer case when multiple formats are included 689 on the "m=" line in the SDP answer, enabling per-RTP packet change of 690 RTP payload type. 692 5.6. Signaling Examples 694 These examples describe a client to video conference service, using a 695 centralized media topology with an RTP mixer. 697 +---+ +-----------+ +---+ 698 | A |<---->| |<---->| B | 699 +---+ | | +---+ 700 | Mixer | 701 +---+ | | +---+ 702 | F |<---->| |<---->| J | 703 +---+ +-----------+ +---+ 705 Figure 4: Four-party Mixer-based Conference 707 5.6.1. Single-Source Client 709 Alice is calling in to the mixer with a simulcast-enabled client 710 capable of a single media source per media type. The client can send 711 a simulcast of 2 video resolutions and frame rates: HD 1280x720p 712 30fps and thumbnail 320x180p 15fps. This is defined below using the 713 "imageattr" [RFC6236]. In this example, only the "pt" "a=rid" 714 parameter is used, effectively achieving a 1:1 mapping between 715 RtpStreamId and media formats (RTP payload types), to describe 716 simulcast stream formats. Alice's Offer: 718 v=0 719 o=alice 2362969037 2362969040 IN IP4 192.0.2.156 720 s=Simulcast Enabled Client 721 t=0 0 722 c=IN IP4 192.0.2.156 723 m=audio 49200 RTP/AVP 0 724 a=rtpmap:0 PCMU/8000 725 m=video 49300 RTP/AVP 97 98 726 a=rtpmap:97 H264/90000 727 a=rtpmap:98 H264/90000 728 a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000 729 a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600 730 a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720] 731 a=imageattr:98 send [x=320,y=180] recv [x=320,y=180] 732 a=rid:1 send pt=97 733 a=rid:2 send pt=98 734 a=rid:3 recv pt=97 735 a=simulcast:send 1;2 recv 3 736 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 738 Figure 5: Single-Source Simulcast Offer 740 The only thing in the SDP that indicates simulcast capability is the 741 line in the video media description containing the "simulcast" 742 attribute. The included "a=fmtp" and "a=imageattr" parameters 743 indicates that sent simulcast streams can differ in video resolution. 745 The RTP header extension for RtpStreamId is offered to avoid issues 746 with the initial binding between RTP streams (SSRCs) and the 747 RtpStreamId identifying the simulcast stream and its format. 749 The Answer from the server indicates that it too is simulcast 750 capable. Should it not have been simulcast capable, the 751 "a=simulcast" line would not have been present and communication 752 would have started with the media negotiated in the SDP. Also the 753 usage of the RtpStreamId RTP header extension is accepted. 755 v=0 756 o=server 823479283 1209384938 IN IP4 192.0.2.2 757 s=Answer to Simulcast Enabled Client 758 t=0 0 759 c=IN IP4 192.0.2.43 760 m=audio 49672 RTP/AVP 0 761 a=rtpmap:0 PCMU/8000 762 m=video 49674 RTP/AVP 97 98 763 a=rtpmap:97 H264/90000 764 a=rtpmap:98 H264/90000 765 a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000 766 a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600 767 a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720] 768 a=imageattr:98 send [x=320,y=180] recv [x=320,y=180] 769 a=rid:1 recv pt=97 770 a=rid:2 recv pt=98 771 a=rid:3 send pt=97 772 a=simulcast:recv 1;2 send 3 773 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 775 Figure 6: Single-Source Simulcast Answer 777 Since the server is the simulcast media receiver, it reverses the 778 direction of the "simulcast" and "rid" attribute parameters. 780 5.6.2. Multi-Source Client 782 Fred is calling in to the same conference as in the example above 783 with a two-camera, two-display system, thus capable of handling two 784 separate media sources in each direction, where each media source is 785 simulcast-enabled in the send direction. Fred's client is restricted 786 to a single media source per media description. 788 The first two simulcast streams for the first media source use 789 different codecs, H264-SVC [RFC6190] and H264 [RFC6184]. These two 790 simulcast streams also have a temporal dependency. Two different 791 video codecs, VP8 [RFC7741] and H264, are offered as alternatives for 792 the third simulcast stream for the first media source. Only the 793 highest fidelity simulcast stream is sent from start, the lower 794 fidelity streams being initially paused. 796 The second media source is offered with three different simulcast 797 streams. All video streams of this second media source are loss 798 protected by RTP retransmission [RFC4588]. Also here, all but the 799 highest fidelity simulcast stream are initially paused. 801 Fred's client is also using BUNDLE to send all RTP streams from all 802 media descriptions in the same RTP session on a single media 803 transport. Although using many different simulcast streams in this 804 example, the use of RtpStreamId as simulcast stream identification 805 enables use of a low number of RTP payload types. Note that the use 806 of both BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation] and "a=rid" 807 [I-D.ietf-mmusic-rid] recommends using the RTP header extension 808 [RFC5285] for carrying these RTP stream identification fields, which 809 is consequently also included in the SDP. Note also that for 810 "a=rid", the corresponding SDES attribute is named RtpStreamId 811 [I-D.ietf-avtext-rid]. 813 v=0 814 o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d 815 s=Offer from Simulcast Enabled Multi-Source Client 816 t=0 0 817 c=IN IP6 2001:db8::c000:27d 818 a=group:BUNDLE foo bar zen 820 m=audio 49200 RTP/AVP 99 821 a=mid:foo 822 a=rtpmap:99 G722/8000 824 m=video 49600 RTP/AVPF 100 101 103 825 a=mid:bar 826 a=rtpmap:100 H264-SVC/90000 827 a=rtpmap:101 H264/90000 828 a=rtpmap:103 VP8/90000 829 a=fmtp:100 profile-level-id=42400d; max-fs=3600; max-mbps=108000; \ 830 mst-mode=NI-TC 831 a=fmtp:101 profile-level-id=42c00d; max-fs=3600; max-mbps=54000 832 a=fmtp:103 max-fs=900; max-fr=30 833 a=rid:1 send pt=100;max-width=1280;max-height=720;max-fps=60;depend=2 834 a=rid:2 send pt=101;max-width=1280;max-height=720;max-fps=30 835 a=rid:3 send pt=101;max-width=640;max-height=360 836 a=rid:4 send pt=103;max-width=640;max-height=360 837 a=depend:100 lay bar:101 838 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid 839 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 840 a=rtcp-fb:* ccm pause nowait 841 a=simulcast:send 1;2;~4,3 843 m=video 49602 RTP/AVPF 96 104 844 a=mid:zen 845 a=rtpmap:96 VP8/90000 846 a=fmtp:96 max-fs=3600; max-fr=30 847 a=rtpmap:104 rtx/90000 848 a=fmtp:104 apt=96;rtx-time=200 849 a=rid:1 send pt=96;max-fs=921600;max-fps=30 850 a=rid:2 send pt=96;max-fs=614400;max-fps=15 851 a=rid:3 send pt=96;max-fs=230400;max-fps=30 852 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid 853 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId 854 a=rtcp-fb:* ccm pause nowait 855 a=simulcast:send 1;~2;~3 857 Figure 7: Fred's Multi-Source Simulcast Offer 859 Note: Empty lines in the SDP above are added only for readability 860 and would not be present in an actual SDP. 862 6. RTP Aspects 864 This section discusses what the different entities in a simulcast 865 media path can expect to happen on RTP level. This is explored from 866 source to sink by starting in an endpoint with a media source that is 867 simulcasted to an RTP middlebox. That RTP middlebox sends media 868 sources both to other RTP middleboxes (cascaded middleboxes), as well 869 as selecting some simulcast format of the media source and sending it 870 to receiving endpoints. Different types of RTP middleboxes and their 871 usage of the different simulcast formats results in several different 872 behaviors. 874 6.1. Outgoing from Endpoint with Media Source 876 The most straightforward simulcast case is the RTP streams being 877 emitted from the endpoint that originates a media source. When 878 simulcast has been negotiated in the sending direction, the endpoint 879 can transmit up to the number of RTP streams needed for the 880 negotiated simulcast streams for that media source. Each RTP stream 881 (SSRC) is identified by associating (Section 5.5) it with an 882 RtpStreamId SDES item, transmitted in RTCP and possibly also as an 883 RTP header extension. In cases where multiple media sources have 884 been negotiated for the same RTP session and thus BUNDLE 885 [I-D.ietf-mmusic-sdp-bundle-negotiation] is used, also the MID SDES 886 item will be sent similarly to the RtpStreamId. 888 Each RTP stream may not be continuously transmitted due to any of the 889 following reasons; temporarily paused using Pause/Resume [RFC7728], 890 sender side application logic temporarily pausing it, or lack of 891 network resources to transmit this simulcast stream. However, all 892 simulcast streams that have been negotiated have active and 893 maintained SSRC (at least in regular RTCP reports), even if no RTP 894 packets are currently transmitted. The relation between an RTP 895 Stream (SSRC) and a particular simulcast stream is not expected to 896 change, except in exceptional situations such as SSRC collisions. At 897 SSRC changes, the usage of MID and RtpStreamId should enable the 898 receiver to correctly identify the RTP streams even after an SSRC 899 change. 901 6.2. RTP Middlebox to Receiver 903 RTP streams in a multi-party RTP session can be used in multiple 904 different ways, when the session utilizes simulcast at least on the 905 media source to middlebox legs. This is to a large degree due to the 906 different RTP middlebox behaviors, but also the needs of the 907 application. This text assumes that the RTP middlebox will select a 908 media source and choose which simulcast stream for that media source 909 to deliver to a specific receiver. In many cases, at most one 910 simulcast stream per media source will be forwarded to a particular 911 receiver at any instant in time, even if the selected simulcast 912 stream may vary. For cases where this does not hold due to 913 application needs, then the RTP stream aspects will fall under the 914 middlebox to middlebox case Section 6.3. 916 The selection of which simulcast streams to forward towards the 917 receiver, is application specific. However, in conferencing 918 applications, active speaker selection is common. In case the number 919 of media sources possible to forward, N, is less than the total 920 amount of media sources available in an multi-media session, the 921 current and previous speakers (up to N in total) are often the ones 922 forwarded. To avoid the need for media specific processing to 923 determine the current speaker(s) in the RTP middlebox, the endpoint 924 providing a media source may include meta data, such as the RTP 925 Header Extension for Client-to-Mixer Audio Level Indication 926 [RFC6464]. 928 The possibilities for stream switching are media type specific, but 929 for media types with significant interframe dependencies in the 930 encoding, like most video coding, the switching needs to be made at 931 suitable switching points in the media stream that breaks or 932 otherwise deals with the dependency structure. Even if switching 933 points can be included periodically, it is common to use mechanisms 934 like Full Intra Requests [RFC5104] to request switching points from 935 the endpoint performing the encoding of the media source. 937 Inclusion of the RtpStreamId SDES item for an SSRC in the middlebox 938 to receiver direction should only occur when use of RtpStreamId has 939 been negotiated in that direction. It is worth noting that one can 940 signal multiple RtpStreamIds when simulcast signalling indicates only 941 a single simulcast stream, allowing one to use all of the 942 RtpStreamIds as alternatives for that simulcast stream. One reason 943 for including the RtpStreamId in the middlebox to receiver direction 944 for an RTP stream is to let the receiver know which restrictions 945 apply to the currently delivered RTP stream. In case the RtpStreamId 946 is negotiated to be used, it is important to remember that the used 947 identifiers will be specific to each signalling session. Even if the 948 central entity can attempt to coordinate, it is likely that the 949 RtpStreamIds need to be translated to the leg specific values. The 950 below cases will have as base line that RtpStreamId is not used in 951 the mixer to receiver direction. 953 6.2.1. Media-Switching Mixer 955 This section discusses the behavior in cases where the RTP middlebox 956 behaves like the Media-Switching Mixer (Section 3.6.2) in RTP 957 Topologies [RFC7667]. The fundamental aspect here is that the media 958 sources delivered from the middlebox will be the mixer's conceptual 959 or functional ones. For example, one media source may be the main 960 speaker in high resolution video, while a number of other media 961 sources are thumbnails of each participant. 963 The above results in that the RTP stream produced by the mixer is one 964 that switches between a number of received incoming RTP streams for 965 different media sources and in different simulcast versions. The 966 mixer selects the media source to be sent as one of the RTP streams, 967 and then selects among the available simulcast streams for the most 968 appropriate one. The selection criteria include available bandwidth 969 on the mixer to receiver path and restrictions based on the 970 functional usage of the RTP stream delivered to the receiver. As an 971 example of the latter, it is unnecessary to forward a full HD video 972 to a receiver if the display area is just a thumbnail. Thus, 973 restrictions may exist to not allow some simulcast streams to be 974 forwarded for some of the mixer's media sources. 976 This will result in a single RTP stream being used for each of the 977 RTP mixer's media sources. This RTP stream is at any point in time a 978 selection of one particular RTP stream arriving to the mixer, where 979 the RTP header field values are rewritten to provide a consistent, 980 single RTP stream. If the RTP mixer doesn't receive any incoming 981 stream matched to this media source, the SSRC will not transmit, but 982 be kept alive using RTCP. The SSRC and thus RTP stream for the 983 mixer's media source is expected to be long term stable. It will 984 only be changed by signalling or other disruptive events. Note that 985 although the above talks about a single RTP stream, there can in some 986 cases be multiple RTP streams carrying the selected simulcast stream 987 for the originating media source, including redundancy or other 988 auxiliary RTP streams. 990 The mixer may communicate the identity of the originating media 991 source to the receiver by including the CSRC field with the 992 originating media source's SSRC value. Note that due to the 993 possibility that the RTP mixer switches between simulcast versions of 994 the media source, the CSRC value may change, even if the media source 995 is kept the same. 997 It is important to note that any MID SDES item from the originating 998 media source needs to be removed and not be associated with the RTP 999 stream's SSRC. That is, there is nothing in the signalling between 1000 the mixer and the receiver that is structured around the originating 1001 media sources, only the mixer's media sources. If they would be 1002 associated with the SSRC, the receiver would likely believe that 1003 there has been an SSRC collision, and that the RTP stream is spurious 1004 as it doesn't carry the identifiers used to relate it to the correct 1005 context. However, this is not true for CSRC values, as long as they 1006 are never used as SSRC. In these cases one could provide CNAME and 1007 MID as SDES items. A receiver could use this to determine which CSRC 1008 values that are associated with the same originating media source. 1010 If RtpStreamIds are used in the scenario described by this section, 1011 it should be noted that the RtpStreamId on a particular SSRC will 1012 change based on the actual simulcast stream selected for switching. 1013 These RtpStreamId identifiers will be local to this leg's signalling 1014 context. In addition, the defined RtpStreamIds and their parameters 1015 need to cover all the media sources and simulcast streams received by 1016 the RTP mixer that can be switched into this media source, sent by 1017 the RTP mixer. 1019 6.2.2. Selective Forwarding Middlebox 1021 This section discusses the behavior in cases where the RTP middlebox 1022 behaves like the Selective Forwarding Middlebox (Section 3.7) in RTP 1023 Topologies [RFC7667]. Applications for this type of RTP middlebox 1024 results in that each originating media source will have a 1025 corresponding media source on the leg between the middlebox and the 1026 receiver. A Selective Forwarding Middlebox (SFM) could go as far as 1027 exposing all the simulcast streams for an media source, however this 1028 section will focus on having a single simulcast stream that can 1029 contain any of the simulcast formats. This section will assume that 1030 the SFM projection mechanism works on media source level, and maps 1031 one of the media source's simulcast streams onto one RTP stream from 1032 the SFM to the receiver. 1034 This usage will result in that the individual RTP stream(s) for one 1035 media source can switch between being active to paused, based on the 1036 subset of media sources the SFM wants to provide the receiver for the 1037 moment. With SFMs there exist no reasons to use CSRC to indicate the 1038 originating stream, as there is a one to one media source mapping. 1039 If the application requires knowing the simulcast version received to 1040 function well, then RtpStreamId should be negotiated on the SFM to 1041 receiver leg. Which simulcast stream that is being forwarded is not 1042 made explicit unless RtpStreamId is used on the leg. 1044 Any MID SDES items being sent by the SFM to the receiver are only 1045 those agreed between the SFM and the receiver, and no MID values from 1046 the originating side of the SFM are to be forwarded. 1048 A SFM could expose corresponding RTP streams for all the media 1049 sources and their simulcast streams, and then for any media source 1050 that is to be provided forward one selected simulcast stream. 1051 However, this is not recommended as it would unnecessarily increase 1052 the number of RTP streams and require the receiver to timely detect 1053 switching between simulcast streams. The above usage requires the 1054 same SFM functionality for switching, while avoiding the 1055 uncertainties of timely detecting that a RTP stream ends. The 1056 benefit would be that the received simulcast stream would be 1057 implicitly provided by which RTP stream would be active for a media 1058 source. However, using RtpStreamId to make this explicit also 1059 exposes which alternative format is used. The conclusion is that 1060 using one RTP stream per simulcast stream is unnecessary. The issue 1061 with timely detecting end of streams, independent if they are stopped 1062 temporarily or long term, is that there is no explicit indication 1063 that the transmission has intentionally been stopped. The RTCP based 1064 Pause and Resume mechanism [RFC7728] includes a PAUSED indication 1065 that provides the last RTP sequence number transmitted prior to the 1066 pause. Due to usage, the timeliness of this solution depends on when 1067 delivery using RTCP can occur in relation to the transmission of the 1068 last RTP packet. If no explicit information is provided at all, then 1069 detection based on non increasing RTCP SR field values and timers 1070 need to be used to determine pause in RTP packet delivery. This 1071 results in that one can usually not determine when the last RTP 1072 packet arrives (if it arrives) that this will be the last. That it 1073 was the last is something that one learns later. 1075 6.3. RTP Middlebox to RTP Middlebox 1077 This relates to the transmission of simulcast streams between RTP 1078 middleboxes or other usages where one wants to enable the delivery of 1079 multiple simultaneous simulcast streams per media source, but the 1080 transmitting entity is not the originating endpoint. For a 1081 particular direction between middlebox A and B, this looks very 1082 similar to the originating to middlebox case on a media source basis. 1083 However, in this case there is usually multiple media sources, 1084 originating from multiple endpoints. This can create situations 1085 where limitations in the number of simultaneously received media 1086 streams can arise, for example due to limitation in network 1087 bandwidth. In this case, a subset of not only the simulcast streams, 1088 but also media sources can be selected. This results in that 1089 individual RTP streams can be become paused at any point and later 1090 being resumed based on various criteria. 1092 The MIDs used between A and B are the ones agreed between these two 1093 identities in signalling. The RtpStreamId values will also be 1094 provided to ensure explicit information about which simulcast stream 1095 they are. The RTP stream to MID and RtpStreamId associations should 1096 here be long term stable. 1098 7. Network Aspects 1100 Simulcast is in this memo defined as the act of sending multiple 1101 alternative encoded streams of the same underlying media source. 1102 When transmitting multiple independent streams that originate from 1103 the same source, it could potentially be done in several different 1104 ways using RTP. A general discussion on considerations for use of 1105 the different RTP multiplexing alternatives can be found in 1106 Guidelines for Multiplexing in RTP 1107 [I-D.ietf-avtcore-multiplex-guidelines]. Discussion and 1108 clarification on how to handle multiple streams in an RTP session can 1109 be found in [RFC8108]. 1111 The network aspects that are relevant for simulcast are: 1113 Quality of Service: When using simulcast it might be of interest to 1114 prioritize a particular simulcast stream, rather than applying 1115 equal treatment to all streams. For example, lower bit-rate 1116 streams may be prioritized over higher bit-rate streams to 1117 minimize congestion or packet losses in the low bit-rate streams. 1118 Thus, there is a benefit to use a simulcast solution with good QoS 1119 support. 1121 NAT/FW Traversal: Using multiple RTP sessions incurs more cost for 1122 NAT/FW traversal unless they can re-use the same transport flow, 1123 which can be achieved by Multiplexing Negotiation Using SDP Port 1124 Numbers [I-D.ietf-mmusic-sdp-bundle-negotiation]. 1126 7.1. Bitrate Adaptation 1128 Use of multiple simulcast streams can require a significant amount of 1129 network resources. If the amount of available network resources 1130 varies during an RTP session such that it does not match what is 1131 negotiated in SDP, the bitrate used by the different simulcast 1132 streams may have to be reduced dynamically. What simulcast streams 1133 to prioritize when allocating available bitrate among the simulcast 1134 streams in such adaptation SHOULD be taken from the simulcast stream 1135 order on the "a=simulcast" line and ordering of alternative simulcast 1136 formats Section 5.2. Simulcast streams that have pause/resume 1137 capability and that would be given such low bitrate by the adaptation 1138 process that they are considered not really useful can be temporarily 1139 paused until the limiting condition clears. 1141 8. Limitation 1143 The chosen approach has a limitation that relates to the use of a 1144 single RTP session for all simulcast formats of a media source, which 1145 comes from sending all simulcast streams related to a media source 1146 under the same SDP media description. 1148 It is not possible to use different simulcast streams on different 1149 media transports, limiting the possibilities to apply different QoS 1150 to different simulcast streams. When using unicast, QoS mechanisms 1151 based on individual packet marking are feasible, since they do not 1152 require separation of simulcast streams into different RTP sessions 1153 to apply different QoS. 1155 It is also not possible to separate different simulcast streams into 1156 different multicast groups to allow a multicast receiver to pick the 1157 stream it wants, rather than receive all of them. In this case, the 1158 only reasonable implementation is to use different RTP sessions for 1159 each multicast group so that reporting and other RTCP functions 1160 operate as intended. Such simulcast usage in multicast context is 1161 out of scope for the current document and would require additional 1162 specification. 1164 9. IANA Considerations 1166 This document requests to register a new media-level SDP attribute, 1167 "simulcast", in the "att-field (media level only)" registry within 1168 the SDP parameters registry, according to the procedures of [RFC4566] 1169 and [I-D.ietf-mmusic-sdp-mux-attributes]. 1171 Contact name, email: IETF, contacted via mmusic@ietf.org, or a 1172 successor address designated by IESG 1174 Attribute name: simulcast 1176 Long-form attribute name: Simulcast stream description 1178 Charset dependent: No 1180 Attribute value: sc-value; see Section 5.1 of RFC XXXX. 1182 Purpose: Signals simulcast capability for a set of RTP streams 1184 MUX category: NORMAL 1186 Note to RFC Editor: Please replace "RFC XXXX" with the assigned 1187 number of this RFC. 1189 10. Security Considerations 1191 The simulcast capability, configuration attributes, and parameters 1192 are vulnerable to attacks in signaling. 1194 A false inclusion of the "a=simulcast" attribute may result in 1195 simultaneous transmission of multiple RTP streams that would 1196 otherwise not be generated. The impact is limited by the media 1197 description joint bandwidth, shared by all simulcast streams 1198 irrespective of their number. There may however be a large number of 1199 unwanted RTP streams that will impact the share of bandwidth 1200 allocated for the originally wanted RTP stream. 1202 A hostile removal of the "a=simulcast" attribute will result in 1203 simulcast not being used. 1205 Neither of the above will likely have any major consequences and can 1206 be mitigated by signaling that is at least integrity and source 1207 authenticated to prevent an attacker to change it. 1209 Security considerations related to the use of "a=rid" and the 1210 RtpStreamId SDES item is covered in [I-D.ietf-mmusic-rid] and 1211 [I-D.ietf-avtext-rid]. There are no additional security concerns 1212 related to their use in this specification. 1214 11. Contributors 1216 Morgan Lindqvist and Fredrik Jansson, both from Ericsson, have 1217 contributed with important material to the first versions of this 1218 document. Robert Hansen and Cullen Jennings, from Cisco, Peter 1219 Thatcher, from Google, and Adam Roach, from Mozilla, contributed 1220 significantly to subsequent versions. 1222 12. Acknowledgements 1224 The authors would like to thank Bernard Aboba, Thomas Belling, Roni 1225 Even, Adam Roach, Inaki Baz Castillo, Paul Kyzivat, and Arun 1226 Arunachalam for the feedback they provided during the development of 1227 this document. 1229 13. References 1231 13.1. Normative References 1233 [I-D.ietf-avtext-rid] 1234 Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream 1235 Identifier Source Description (SDES)", draft-ietf-avtext- 1236 rid-09 (work in progress), October 2016. 1238 [I-D.ietf-mmusic-rid] 1239 Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B., 1240 Roach, A., and B. Campen, "RTP Payload Format 1241 Restrictions", draft-ietf-mmusic-rid-10 (work in 1242 progress), March 2017. 1244 [I-D.ietf-mmusic-sdp-bundle-negotiation] 1245 Holmberg, C., Alvestrand, H., and C. Jennings, 1246 "Negotiating Media Multiplexing Using the Session 1247 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 1248 negotiation-38 (work in progress), April 2017. 1250 [I-D.ietf-mmusic-sdp-mux-attributes] 1251 Nandakumar, S., "A Framework for SDP Attributes when 1252 Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-16 1253 (work in progress), December 2016. 1255 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1256 Requirement Levels", BCP 14, RFC 2119, 1257 DOI 10.17487/RFC2119, March 1997, 1258 . 1260 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1261 Jacobson, "RTP: A Transport Protocol for Real-Time 1262 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 1263 July 2003, . 1265 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1266 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 1267 July 2006, . 1269 [RFC5234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax 1270 Specifications: ABNF", STD 68, RFC 5234, 1271 DOI 10.17487/RFC5234, January 2008, 1272 . 1274 [RFC7728] Burman, B., Akram, A., Even, R., and M. Westerlund, "RTP 1275 Stream Pause and Resume", RFC 7728, DOI 10.17487/RFC7728, 1276 February 2016, . 1278 13.2. Informative References 1280 [I-D.ietf-avtcore-multiplex-guidelines] 1281 Westerlund, M., Perkins, C., and H. Alvestrand, 1282 "Guidelines for using the Multiplexing Features of RTP to 1283 Support Multiple Media Streams", draft-ietf-avtcore- 1284 multiplex-guidelines-03 (work in progress), October 2014. 1286 [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., 1287 Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- 1288 Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, 1289 DOI 10.17487/RFC2198, September 1997, 1290 . 1292 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 1293 with Session Description Protocol (SDP)", RFC 3264, 1294 DOI 10.17487/RFC3264, June 2002, 1295 . 1297 [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for 1298 Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389, 1299 September 2002, . 1301 [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. 1302 Hakenberg, "RTP Retransmission Payload Format", RFC 4588, 1303 DOI 10.17487/RFC4588, July 2006, 1304 . 1306 [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF 1307 Digits, Telephony Tones, and Telephony Signals", RFC 4733, 1308 DOI 10.17487/RFC4733, December 2006, 1309 . 1311 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 1312 "Codec Control Messages in the RTP Audio-Visual Profile 1313 with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, 1314 February 2008, . 1316 [RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error 1317 Correction", RFC 5109, DOI 10.17487/RFC5109, December 1318 2007, . 1320 [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP 1321 Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July 1322 2008, . 1324 [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific 1325 Media Attributes in the Session Description Protocol 1326 (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009, 1327 . 1329 [RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding 1330 Dependency in the Session Description Protocol (SDP)", 1331 RFC 5583, DOI 10.17487/RFC5583, July 2009, 1332 . 1334 [RFC6184] Wang, Y., Even, R., Kristensen, T., and R. Jesup, "RTP 1335 Payload Format for H.264 Video", RFC 6184, 1336 DOI 10.17487/RFC6184, May 2011, 1337 . 1339 [RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, 1340 "RTP Payload Format for Scalable Video Coding", RFC 6190, 1341 DOI 10.17487/RFC6190, May 2011, 1342 . 1344 [RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image 1345 Attributes in the Session Description Protocol (SDP)", 1346 RFC 6236, DOI 10.17487/RFC6236, May 2011, 1347 . 1349 [RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time 1350 Transport Protocol (RTP) Header Extension for Client-to- 1351 Mixer Audio Level Indication", RFC 6464, 1352 DOI 10.17487/RFC6464, December 2011, 1353 . 1355 [RFC7104] Begen, A., Cai, Y., and H. Ou, "Duplication Grouping 1356 Semantics in the Session Description Protocol", RFC 7104, 1357 DOI 10.17487/RFC7104, January 2014, 1358 . 1360 [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and 1361 B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms 1362 for Real-Time Transport Protocol (RTP) Sources", RFC 7656, 1363 DOI 10.17487/RFC7656, November 2015, 1364 . 1366 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 1367 DOI 10.17487/RFC7667, November 2015, 1368 . 1370 [RFC7741] Westin, P., Lundin, H., Glover, M., Uberti, J., and F. 1371 Galligan, "RTP Payload Format for VP8 Video", RFC 7741, 1372 DOI 10.17487/RFC7741, March 2016, 1373 . 1375 [RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, 1376 "Sending Multiple RTP Streams in a Single RTP Session", 1377 RFC 8108, DOI 10.17487/RFC8108, March 2017, 1378 . 1380 Appendix A. Requirements 1382 The following requirements have to be met to support the use cases 1383 (Section 3): 1385 REQ-1: Identification: 1387 REQ-1.1: It must be possible to identify a set of simulcasted RTP 1388 streams as originating from the same media source in SDP 1389 signaling. 1391 REQ-1.2: An RTP endpoint must be capable of identifying the 1392 simulcast stream a received RTP stream is associated with, 1393 knowing the content of the SDP signalling. 1395 REQ-2: Transport usage. The solution must work when using: 1397 REQ-2.1: Legacy SDP with separate media transports per SDP media 1398 description. 1400 REQ-2.2: Bundled [I-D.ietf-mmusic-sdp-bundle-negotiation] SDP 1401 media descriptions. 1403 REQ-3: Capability negotiation. It must be possible that: 1405 REQ-3.1: Sender can express capability of sending simulcast. 1407 REQ-3.2: Receiver can express capability of receiving simulcast. 1409 REQ-3.3: Sender can express maximum number of simulcast streams 1410 that can be provided. 1412 REQ-3.4: Receiver can express maximum number of simulcast streams 1413 that can be received. 1415 REQ-3.5: Sender can detail the characteristics of the simulcast 1416 streams that can be provided. 1418 REQ-3.6: Receiver can detail the characteristics of the simulcast 1419 streams that it prefers to receive. 1421 REQ-4: Distinguishing features. It must be possible to have 1422 different simulcast streams use different codec parameters, as can 1423 be expressed by SDP format values and RTP payload types. 1425 REQ-5: Compatibility. It must be possible to use simulcast in 1426 combination with other RTP mechanisms that generate additional RTP 1427 streams: 1429 REQ-5.1: RTP Retransmission [RFC4588]. 1431 REQ-5.2: RTP Forward Error Correction [RFC5109]. 1433 REQ-5.3: Related payload types such as audio Comfort Noise and/or 1434 DTMF. 1436 REQ-5.4: A single simulcast stream can consist of multiple RTP 1437 streams, to support codecs where a dependent stream is 1438 dependent on a set of encoded and dependent streams, each 1439 potentially carried in their own RTP stream. 1441 REQ-6: Interoperability. The solution must be possible to use in: 1443 REQ-6.1: Interworking with non-simulcast legacy clients using a 1444 single media source per media type. 1446 REQ-6.2: WebRTC environment with a single media source per SDP 1447 media description. 1449 Appendix B. Changes From Earlier Versions 1451 NOTE TO RFC EDITOR: Please remove this section prior to publication. 1453 B.1. Modifications Between WG Version -08 and -09 1455 o Changed SCID to rid-id, to align with ietf-draft-mmusic-rid 1456 naming. 1458 o Changed Overview to be based on examples and shortened it. 1460 o Changed semantics of initially paused rid-id in modified SDP 1461 offers from requiring it to follow actual RFC 7728 pause state to 1462 an informational offerer's opinion at the time of offer creation, 1463 not in any way overriding or amending RFC 7728 signaling. 1465 o Replaced text on ignoring all but the first of multiple 1466 "a=simulcast" lines in a media description with mandating that at 1467 most one "a=simulcast" line is included. 1469 o Clarified with a note that, for the case it is clear from the SDP 1470 that RTP PT uniquely maps to RtpStreamId, an RTP receiver can use 1471 RTP PT to relate simulcast streams. 1473 o Moved Section 4 Requirements to become Appendix A. 1475 o Editorial corrections and clarifications. 1477 B.2. Modifications Between WG Version -07 and -08 1479 o Correcting syntax of SDP examples in section 6.6.1, as found by 1480 Inaki Baz Castillo. 1482 o Changing ABNF to only define the sc-value, not the SDP attribute 1483 itself, as suggested by Paul Kyzivat. 1485 o Changing I-D reference to newly published RFC 8108. 1487 o Adding list of modifications between -06 and -07. 1489 B.3. Modifications Between WG Version -06 and -07 1491 o A scope clarification, as result of the discussion with Roni Even. 1493 o A reformulation of the identification requirements for simulcast 1494 stream. 1496 o Correcting the statement related to source specific signalling 1497 (RFC 5576) to address Roni Even's comment. 1499 o Update of the last paragraph in Section 6.2 regarding simulcast 1500 stream differences as well as forbidding multiple instances of the 1501 same SCID within a single a=simulcast line. 1503 o Removal of note in Section 6.4 as result of issue raised by Roni 1504 Even. 1506 o Use of "m=" has been changed to media description and a few other 1507 editorial improvements and clarifications. 1509 B.4. Modifications Between WG Version -05 and -06 1511 o Added section on RTP Aspects 1513 o Added a requirement (5-4) on that capability exchange must be 1514 capable of handling multi RTP stream cases. 1516 o Added extmap attribute also on first signalling example as it is a 1517 recommended to use mechanism. 1519 o Clarified the definition of the simulcast attribute and how 1520 simulcast streams relates to simulcast formats and SCIDs. 1522 o Updated References list and moved around some references between 1523 informative and normative categories. 1525 o Editorial improvements and corrections. 1527 B.5. Modifications Between WG Version -04 and -05 1529 o Aligned with recent changes in draft-ietf-mmusic-rid and draft- 1530 ietf-avtext-rid. 1532 o Modified the SDP offer/answer section to follow the generally 1533 accepted structure, also adding a brief text on modifying the 1534 session that is aligned with draft-ietf-mmusic-rid. 1536 o Improved text around simulcast stream identification (as opposed 1537 to the simulcast stream itself) to consistently use the acronym 1538 SCID and defined that in the Terminology section. 1540 o Changed references for RTP-level pause/resume and VP8 payload 1541 format that are now published as RFC. 1543 o Improved IANA registration text. 1545 o Removed unused reference to draft-ietf-payload-flexible-fec- 1546 scheme. 1548 o Editorial improvements and corrections. 1550 B.6. Modifications Between WG Version -03 and -04 1552 o Changed to only use RID identification, as was consensus during 1553 IETF 94. 1555 o ABNF improvements. 1557 o Clarified offer-answer rules for initially paused streams. 1559 o Changed references for RTP topologies and RTP taxonomy documents 1560 that are now published as RFC. 1562 o Added reference to the new RID draft in AVTEXT. 1564 o Re-structured section 6 to provide an easy reference by the 1565 updated IANA section. 1567 o Added a sub-section 7.1 with a discussion of bitrate adaptation. 1569 o Editorial improvements. 1571 B.7. Modifications Between WG Version -02 and -03 1573 o Removed text on multicast / broadcast from use cases, since it is 1574 not supported by the solution. 1576 o Removed explicit references to unified plan draft. 1578 o Added possibility to initiate simulcast streams in paused mode. 1580 o Enabled an offerer to offer multiple stream identification (pt or 1581 rid) methods and have the answerer choose which to use. 1583 o Added a preference indication also in send direction offers. 1585 o Added a section on limitations of the current proposal, including 1586 identification method specific limitations. 1588 B.8. Modifications Between WG Version -01 and -02 1590 o Relying on the new RID solution for codec constraints and 1591 configuration identification. This has resulted in changes in 1592 syntax to identify if pt or RID is used to describe the simulcast 1593 stream. 1595 o Renamed simulcast version and simulcast version alternative to 1596 simulcast stream and simulcast format respectively, and improved 1597 definitions for them. 1599 o Clarification that it is possible to switch between simulcast 1600 version alternatives, but that only a single one be used at any 1601 point in time. 1603 o Changed the definition so that ordering of simulcast formats for a 1604 specific simulcast stream do have a preference order. 1606 B.9. Modifications Between WG Version -00 and -01 1608 o No changes. Only preventing expiry. 1610 B.10. Modifications Between Individual Version -00 and WG Version -00 1612 o Added this appendix. 1614 Authors' Addresses 1615 Bo Burman 1616 Ericsson 1617 Gronlandsgatan 31 1618 SE-164 60 Stockholm 1619 Sweden 1621 Email: bo.burman@ericsson.com 1623 Magnus Westerlund 1624 Ericsson 1625 Farogatan 2 1626 SE-164 80 Stockholm 1627 Sweden 1629 Phone: +46 10 714 82 87 1630 Email: magnus.westerlund@ericsson.com 1632 Suhas Nandakumar 1633 Cisco 1634 170 West Tasman Drive 1635 San Jose, CA 95134 1636 USA 1638 Email: snandaku@cisco.com 1640 Mo Zanaty 1641 Cisco 1642 170 West Tasman Drive 1643 San Jose, CA 95134 1644 USA 1646 Email: mzanaty@cisco.com