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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 RTCWEB Working Group C. S. Perkins 3 Internet-Draft University of Glasgow 4 Intended status: Standards Track M. Westerlund 5 Expires: October 01, 2015 Ericsson 6 J. Ott 7 Aalto University 8 March 30, 2015 10 Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 11 draft-ietf-rtcweb-rtp-usage-23 13 Abstract 15 The Web Real-Time Communication (WebRTC) framework provides support 16 for direct interactive rich communication using audio, video, text, 17 collaboration, games, etc. between two peers' web-browsers. This 18 memo describes the media transport aspects of the WebRTC framework. 19 It specifies how the Real-time Transport Protocol (RTP) is used in 20 the WebRTC context, and gives requirements for which RTP features, 21 profiles, and extensions need to be supported. 23 Status of This Memo 25 This Internet-Draft is submitted in full conformance with the 26 provisions of BCP 78 and BCP 79. 28 Internet-Drafts are working documents of the Internet Engineering 29 Task Force (IETF). Note that other groups may also distribute 30 working documents as Internet-Drafts. The list of current Internet- 31 Drafts is at http://datatracker.ietf.org/drafts/current/. 33 Internet-Drafts are draft documents valid for a maximum of six months 34 and may be updated, replaced, or obsoleted by other documents at any 35 time. It is inappropriate to use Internet-Drafts as reference 36 material or to cite them other than as "work in progress." 38 This Internet-Draft will expire on October 01, 2015. 40 Copyright Notice 42 Copyright (c) 2015 IETF Trust and the persons identified as the 43 document authors. All rights reserved. 45 This document is subject to BCP 78 and the IETF Trust's Legal 46 Provisions Relating to IETF Documents 47 (http://trustee.ietf.org/license-info) in effect on the date of 48 publication of this document. Please review these documents 49 carefully, as they describe your rights and restrictions with respect 50 to this document. Code Components extracted from this document must 51 include Simplified BSD License text as described in Section 4.e of 52 the Trust Legal Provisions and are provided without warranty as 53 described in the Simplified BSD License. 55 Table of Contents 57 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 58 2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4 59 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 60 4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5 61 4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 5 62 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7 63 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8 64 4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 10 65 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10 66 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 11 67 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 11 68 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 12 69 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 12 70 4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 13 71 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 13 72 5.1. Conferencing Extensions and Topologies . . . . . . . . . 13 73 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 15 74 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 15 75 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 16 76 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 16 77 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 16 78 5.1.6. Temporary Maximum Media Stream Bit Rate Request 79 (TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 16 80 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 17 81 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 17 82 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 17 83 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 18 84 5.2.4. Media Stream Identification . . . . . . . . . . . . . 18 85 5.2.5. Coordination of Video Orientation . . . . . . . . . . 18 86 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 19 87 6.1. Negative Acknowledgements and RTP Retransmission . . . . 19 88 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 20 89 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 20 90 7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 21 91 7.2. Congestion Control Interoperability and Legacy Systems . 22 92 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 23 93 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 23 94 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 24 95 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 25 96 12. RTP Implementation Considerations . . . . . . . . . . . . . . 27 97 12.1. Configuration and Use of RTP Sessions . . . . . . . . . 28 98 12.1.1. Use of Multiple Media Sources Within an RTP Session 28 99 12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 29 100 12.1.3. Differentiated Treatment of RTP Packet Streams . . . 33 101 12.2. Media Source, RTP Packet Streams, and Participant 102 Identification . . . . . . . . . . . . . . . . . . . . . 35 103 12.2.1. Media Source Identification . . . . . . . . . . . . 35 104 12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 36 105 12.2.3. Media Synchronisation Context . . . . . . . . . . . 37 106 13. Security Considerations . . . . . . . . . . . . . . . . . . . 38 107 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 39 108 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39 109 16. References . . . . . . . . . . . . . . . . . . . . . . . . . 40 110 16.1. Normative References . . . . . . . . . . . . . . . . . . 40 111 16.2. Informative References . . . . . . . . . . . . . . . . . 43 112 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 45 114 1. Introduction 116 The Real-time Transport Protocol (RTP) [RFC3550] provides a framework 117 for delivery of audio and video teleconferencing data and other real- 118 time media applications. Previous work has defined the RTP protocol, 119 along with numerous profiles, payload formats, and other extensions. 120 When combined with appropriate signalling, these form the basis for 121 many teleconferencing systems. 123 The Web Real-Time communication (WebRTC) framework provides the 124 protocol building blocks to support direct, interactive, real-time 125 communication using audio, video, collaboration, games, etc., between 126 two peers' web-browsers. This memo describes how the RTP framework 127 is to be used in the WebRTC context. It proposes a baseline set of 128 RTP features that are to be implemented by all WebRTC Endpoints, 129 along with suggested extensions for enhanced functionality. 131 This memo specifies a protocol intended for use within the WebRTC 132 framework, but is not restricted to that context. An overview of the 133 WebRTC framework is given in [I-D.ietf-rtcweb-overview]. 135 The structure of this memo is as follows. Section 2 outlines our 136 rationale in preparing this memo and choosing these RTP features. 137 Section 3 defines terminology. Requirements for core RTP protocols 138 are described in Section 4 and suggested RTP extensions are described 139 in Section 5. Section 6 outlines mechanisms that can increase 140 robustness to network problems, while Section 7 describes congestion 141 control and rate adaptation mechanisms. The discussion of mandated 142 RTP mechanisms concludes in Section 8 with a review of performance 143 monitoring and network management tools. Section 9 gives some 144 guidelines for future incorporation of other RTP and RTP Control 145 Protocol (RTCP) extensions into this framework. Section 10 describes 146 requirements placed on the signalling channel. Section 11 discusses 147 the relationship between features of the RTP framework and the WebRTC 148 application programming interface (API), and Section 12 discusses RTP 149 implementation considerations. The memo concludes with security 150 considerations (Section 13) and IANA considerations (Section 14). 152 2. Rationale 154 The RTP framework comprises the RTP data transfer protocol, the RTP 155 control protocol, and numerous RTP payload formats, profiles, and 156 extensions. This range of add-ons has allowed RTP to meet various 157 needs that were not envisaged by the original protocol designers, and 158 to support many new media encodings, but raises the question of what 159 extensions are to be supported by new implementations. The 160 development of the WebRTC framework provides an opportunity to review 161 the available RTP features and extensions, and to define a common 162 baseline RTP feature set for all WebRTC Endpoints. This builds on 163 the past 20 years development of RTP to mandate the use of extensions 164 that have shown widespread utility, while still remaining compatible 165 with the wide installed base of RTP implementations where possible. 167 RTP and RTCP extensions that are not discussed in this document can 168 be implemented by WebRTC Endpoints if they are beneficial for new use 169 cases. However, they are not necessary to address the WebRTC use 170 cases and requirements identified in 171 [I-D.ietf-rtcweb-use-cases-and-requirements]. 173 While the baseline set of RTP features and extensions defined in this 174 memo is targeted at the requirements of the WebRTC framework, it is 175 expected to be broadly useful for other conferencing-related uses of 176 RTP. In particular, it is likely that this set of RTP features and 177 extensions will be appropriate for other desktop or mobile video 178 conferencing systems, or for room-based high-quality telepresence 179 applications. 181 3. Terminology 183 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 184 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 185 document are to be interpreted as described in [RFC2119]. The RFC 186 2119 interpretation of these key words applies only when written in 187 ALL CAPS. Lower- or mixed-case uses of these key words are not to be 188 interpreted as carrying special significance in this memo. 190 We define the following additional terms: 192 WebRTC MediaStream: The MediaStream concept defined by the W3C in 193 the WebRTC API [W3C.WD-mediacapture-streams-20130903]. 195 Transport-layer Flow: A uni-directional flow of transport packets 196 that are identified by having a particular 5-tuple of source IP 197 address, source port, destination IP address, destination port, 198 and transport protocol used. 200 Bi-directional Transport-layer Flow: A bi-directional transport- 201 layer flow is a transport-layer flow that is symmetric. That is, 202 the transport-layer flow in the reverse direction has a 5-tuple 203 where the source and destination address and ports are swapped 204 compared to the forward path transport-layer flow, and the 205 transport protocol is the same. 207 This document uses the terminology from 208 [I-D.ietf-avtext-rtp-grouping-taxonomy] and 209 [I-D.ietf-rtcweb-overview]. Other terms are used according to their 210 definitions from the RTP Specification [RFC3550]. Especially note 211 the following frequently used terms: RTP Packet Stream, RTP Session, 212 and End-point. 214 4. WebRTC Use of RTP: Core Protocols 216 The following sections describe the core features of RTP and RTCP 217 that need to be implemented, along with the mandated RTP profiles. 218 Also described are the core extensions providing essential features 219 that all WebRTC Endpoints need to implement to function effectively 220 on today's networks. 222 4.1. RTP and RTCP 224 The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be 225 implemented as the media transport protocol for WebRTC. RTP itself 226 comprises two parts: the RTP data transfer protocol, and the RTP 227 control protocol (RTCP). RTCP is a fundamental and integral part of 228 RTP, and MUST be implemented and used in all WebRTC Endpoints. 230 The following RTP and RTCP features are sometimes omitted in limited 231 functionality implementations of RTP, but are REQUIRED in all WebRTC 232 Endpoints: 234 o Support for use of multiple simultaneous SSRC values in a single 235 RTP session, including support for RTP end-points that send many 236 SSRC values simultaneously, following [RFC3550] and 237 [I-D.ietf-avtcore-rtp-multi-stream]. The RTCP optimisations for 238 multi-SSRC sessions defined in 239 [I-D.ietf-avtcore-rtp-multi-stream-optimisation] MAY be supported; 240 if supported the usage MUST be signalled. 242 o Random choice of SSRC on joining a session; collision detection 243 and resolution for SSRC values (see also Section 4.8). 245 o Support for reception of RTP data packets containing CSRC lists, 246 as generated by RTP mixers, and RTCP packets relating to CSRCs. 248 o Sending correct synchronisation information in the RTCP Sender 249 Reports, to allow receivers to implement lip-synchronisation; see 250 Section 5.2.1 regarding support for the rapid RTP synchronisation 251 extensions. 253 o Support for multiple synchronisation contexts. Participants that 254 send multiple simultaneous RTP packet streams SHOULD do so as part 255 of a single synchronisation context, using a single RTCP CNAME for 256 all streams and allowing receivers to play the streams out in a 257 synchronised manner. For compatibility with potential future 258 versions of this specification, or for interoperability with non- 259 WebRTC devices through a gateway, receivers MUST support multiple 260 synchronisation contexts, indicated by the use of multiple RTCP 261 CNAMEs in an RTP session. This specification requires the usage 262 of a single CNAME when sending RTP Packet Streams in some 263 circumstances, see Section 4.9. 265 o Support for sending and receiving RTCP SR, RR, SDES, and BYE 266 packet types, with OPTIONAL support for other RTCP packet types 267 unless mandated by other parts of this specification. Note that 268 additional RTCP Packet types are used by the RTP/SAVPF Profile 269 (Section 4.2) and the other RTCP extensions (Section 5). WebRTC 270 endpoints that implement the SDP bundle negotiation extension will 271 use the SDP grouping framework 'mid' attribute to identify media 272 streams. Such endpoints MUST implement the RTCP SDES MID item 273 described in [I-D.ietf-mmusic-sdp-bundle-negotiation]. 275 o Support for multiple end-points in a single RTP session, and for 276 scaling the RTCP transmission interval according to the number of 277 participants in the session; support for randomised RTCP 278 transmission intervals to avoid synchronisation of RTCP reports; 279 support for RTCP timer reconsideration (Section 6.3.6 of 280 [RFC3550]) and reverse reconsideration (Section 6.3.4 of 281 [RFC3550]). 283 o Support for configuring the RTCP bandwidth as a fraction of the 284 media bandwidth, and for configuring the fraction of the RTCP 285 bandwidth allocated to senders, e.g., using the SDP "b=" line 286 [RFC4566][RFC3556]. 288 o Support for the reduced minimum RTCP reporting interval described 289 in Section 6.2 of [RFC3550] is REQUIRED. When using the reduced 290 minimum RTCP reporting interval, the fixed (non-reduced) minimum 291 interval MUST be used when calculating the participant timeout 292 interval (see Sections 6.2 and 6.3.5 of [RFC3550]). The delay 293 before sending the initial compound RTCP packet can be set to zero 294 (see Section 6.2 of [RFC3550] as updated by 295 [I-D.ietf-avtcore-rtp-multi-stream]). 297 o Support for discontinuous transmission. RTP allows endpoints to 298 pause and resume transmission at any time. When resuming, the RTP 299 sequence number will increase by one, as usual, while the increase 300 in the RTP timestamp value will depend on the duration of the 301 pause. Discontinuous transmission is most commonly used with some 302 audio payload formats, but is not audio specific, and can be used 303 with any RTP payload format. 305 o Ignore unknown RTCP packet types and RTP header extensions. This 306 to ensure robust handling of future extensions, middlebox 307 behaviours, etc., that can result in not signalled RTCP packet 308 types or RTP header extensions being received. If a compound RTCP 309 packet is received that contains a mixture of known and unknown 310 RTCP packet types, the known packets types need to be processed as 311 usual, with only the unknown packet types being discarded. 313 It is known that a significant number of legacy RTP implementations, 314 especially those targeted at VoIP-only systems, do not support all of 315 the above features, and in some cases do not support RTCP at all. 316 Implementers are advised to consider the requirements for graceful 317 degradation when interoperating with legacy implementations. 319 Other implementation considerations are discussed in Section 12. 321 4.2. Choice of the RTP Profile 323 The complete specification of RTP for a particular application domain 324 requires the choice of an RTP Profile. For WebRTC use, the Extended 325 Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF) [RFC5124], as 326 extended by [RFC7007], MUST be implemented. The RTP/SAVPF profile is 327 the combination of basic RTP/AVP profile [RFC3551], the RTP profile 328 for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP 329 profile (RTP/SAVP) [RFC3711]. 331 The RTCP-based feedback extensions [RFC4585] are needed for the 332 improved RTCP timer model. This allows more flexible transmission of 333 RTCP packets in response to events, rather than strictly according to 334 bandwidth, and is vital for being able to report congestion signals 335 as well as media events. These extensions also allow saving RTCP 336 bandwidth, and an end-point will commonly only use the full RTCP 337 bandwidth allocation if there are many events that require feedback. 338 The timer rules are also needed to make use of the RTP conferencing 339 extensions discussed in Section 5.1. 341 Note: The enhanced RTCP timer model defined in the RTP/AVPF 342 profile is backwards compatible with legacy systems that implement 343 only the RTP/AVP or RTP/SAVP profile, given some constraints on 344 parameter configuration such as the RTCP bandwidth value and "trr- 345 int" (the most important factor for interworking with RTP/(S)AVP 346 end-points via a gateway is to set the trr-int parameter to a 347 value representing 4 seconds, see Section 6.1 in 348 [I-D.ietf-avtcore-rtp-multi-stream]). 350 The secure RTP (SRTP) profile extensions [RFC3711] are needed to 351 provide media encryption, integrity protection, replay protection and 352 a limited form of source authentication. WebRTC Endpoints MUST NOT 353 send packets using the basic RTP/AVP profile or the RTP/AVPF profile; 354 they MUST employ the full RTP/SAVPF profile to protect all RTP and 355 RTCP packets that are generated (i.e., implementations MUST use SRTP 356 and SRTCP). The RTP/SAVPF profile MUST be configured using the 357 cipher suites, DTLS-SRTP protection profiles, keying mechanisms, and 358 other parameters described in [I-D.ietf-rtcweb-security-arch]. 360 4.3. Choice of RTP Payload Formats 362 Mandatory to implement audio codecs and RTP payload formats for 363 WebRTC endpoints are defined in [I-D.ietf-rtcweb-audio]. Mandatory 364 to implement video codecs and RTP payload formats for WebRTC 365 endpoints are defined in [I-D.ietf-rtcweb-video]. WebRTC endpoints 366 MAY additionally implement any other codec for which an RTP payload 367 format and associated signalling has been defined. 369 WebRTC Endpoints cannot assume that the other participants in an RTP 370 session understand any RTP payload format, no matter how common. The 371 mapping between RTP payload type numbers and specific configurations 372 of particular RTP payload formats MUST be agreed before those payload 373 types/formats can be used. In an SDP context, this can be done using 374 the "a=rtpmap:" and "a=fmtp:" attributes associated with an "m=" 375 line, along with any other SDP attributes needed to configure the RTP 376 payload format. 378 End-points can signal support for multiple RTP payload formats, or 379 multiple configurations of a single RTP payload format, as long as 380 each unique RTP payload format configuration uses a different RTP 381 payload type number. As outlined in Section 4.8, the RTP payload 382 type number is sometimes used to associate an RTP packet stream with 383 a signalling context. This association is possible provided unique 384 RTP payload type numbers are used in each context. For example, an 385 RTP packet stream can be associated with an SDP "m=" line by 386 comparing the RTP payload type numbers used by the RTP packet stream 387 with payload types signalled in the "a=rtpmap:" lines in the media 388 sections of the SDP. This leads to the following considerations: 390 If RTP packet streams are being associated with signalling 391 contexts based on the RTP payload type, then the assignment of RTP 392 payload type numbers MUST be unique across signalling contexts. 394 If the same RTP payload format configuration is used in multiple 395 contexts, then a different RTP payload type number has to be 396 assigned in each context to ensure uniqueness. 398 If the RTP payload type number is not being used to associate RTP 399 packet streams with a signalling context, then the same RTP 400 payload type number can be used to indicate the exact same RTP 401 payload format configuration in multiple contexts. 403 A single RTP payload type number MUST NOT be assigned to different 404 RTP payload formats, or different configurations of the same RTP 405 payload format, within a single RTP session (note that the "m=" lines 406 in an SDP bundle group [I-D.ietf-mmusic-sdp-bundle-negotiation] form 407 a single RTP session). 409 An end-point that has signalled support for multiple RTP payload 410 formats MUST be able to accept data in any of those payload formats 411 at any time, unless it has previously signalled limitations on its 412 decoding capability. This requirement is constrained if several 413 types of media (e.g., audio and video) are sent in the same RTP 414 session. In such a case, a source (SSRC) is restricted to switching 415 only between the RTP payload formats signalled for the type of media 416 that is being sent by that source; see Section 4.4. To support rapid 417 rate adaptation by changing codec, RTP does not require advance 418 signalling for changes between RTP payload formats used by a single 419 SSRC that were signalled during session set-up. 421 If performing changes between two RTP payload types that use 422 different RTP clock rates, an RTP sender MUST follow the 423 recommendations in Section 4.1 of [RFC7160]. RTP receivers MUST 424 follow the recommendations in Section 4.3 of [RFC7160] in order to 425 support sources that switch between clock rates in an RTP session 426 (these recommendations for receivers are backwards compatible with 427 the case where senders use only a single clock rate). 429 4.4. Use of RTP Sessions 431 An association amongst a set of end-points communicating using RTP is 432 known as an RTP session [RFC3550]. An end-point can be involved in 433 several RTP sessions at the same time. In a multimedia session, each 434 type of media has typically been carried in a separate RTP session 435 (e.g., using one RTP session for the audio, and a separate RTP 436 session using a different transport-layer flow for the video). 437 WebRTC Endpoints are REQUIRED to implement support for multimedia 438 sessions in this way, separating each RTP session using different 439 transport-layer flows for compatibility with legacy systems (this is 440 sometimes called session multiplexing). 442 In modern day networks, however, with the widespread use of network 443 address/port translators (NAT/NAPT) and firewalls, it is desirable to 444 reduce the number of transport-layer flows used by RTP applications. 445 This can be done by sending all the RTP packet streams in a single 446 RTP session, which will comprise a single transport-layer flow (this 447 will prevent the use of some quality-of-service mechanisms, as 448 discussed in Section 12.1.3). Implementations are therefore also 449 REQUIRED to support transport of all RTP packet streams, independent 450 of media type, in a single RTP session using a single transport layer 451 flow, according to [I-D.ietf-avtcore-multi-media-rtp-session] (this 452 is sometimes called SSRC multiplexing). If multiple types of media 453 are to be used in a single RTP session, all participants in that RTP 454 session MUST agree to this usage. In an SDP context, 455 [I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to signal such a 456 bundle of RTP packet streams forming a single RTP session. 458 Further discussion about the suitability of different RTP session 459 structures and multiplexing methods to different scenarios can be 460 found in [I-D.ietf-avtcore-multiplex-guidelines]. 462 4.5. RTP and RTCP Multiplexing 464 Historically, RTP and RTCP have been run on separate transport layer 465 flows (e.g., two UDP ports for each RTP session, one port for RTP and 466 one port for RTCP). With the increased use of Network Address/Port 467 Translation (NAT/NAPT) this has become problematic, since maintaining 468 multiple NAT bindings can be costly. It also complicates firewall 469 administration, since multiple ports need to be opened to allow RTP 470 traffic. To reduce these costs and session set-up times, 471 implementations are REQUIRED to support multiplexing RTP data packets 472 and RTCP control packets on a single transport-layer flow [RFC5761]. 473 Such RTP and RTCP multiplexing MUST be negotiated in the signalling 474 channel before it is used. If SDP is used for signalling, this 475 negotiation MUST use the attributes defined in [RFC5761]. For 476 backwards compatibility, implementations are also REQUIRED to support 477 RTP and RTCP sent on separate transport-layer flows. 479 Note that the use of RTP and RTCP multiplexed onto a single 480 transport-layer flow ensures that there is occasional traffic sent on 481 that port, even if there is no active media traffic. This can be 482 useful to keep NAT bindings alive [RFC6263]. 484 4.6. Reduced Size RTCP 486 RTCP packets are usually sent as compound RTCP packets, and [RFC3550] 487 requires that those compound packets start with an Sender Report (SR) 488 or Receiver Report (RR) packet. When using frequent RTCP feedback 489 messages under the RTP/AVPF Profile [RFC4585] these statistics are 490 not needed in every packet, and unnecessarily increase the mean RTCP 491 packet size. This can limit the frequency at which RTCP packets can 492 be sent within the RTCP bandwidth share. 494 To avoid this problem, [RFC5506] specifies how to reduce the mean 495 RTCP message size and allow for more frequent feedback. Frequent 496 feedback, in turn, is essential to make real-time applications 497 quickly aware of changing network conditions, and to allow them to 498 adapt their transmission and encoding behaviour. Implementations 499 MUST support sending and receiving non-compound RTCP feedback packets 500 [RFC5506]. Use of non-compound RTCP packets MUST be negotiated using 501 the signalling channel. If SDP is used for signalling, this 502 negotiation MUST use the attributes defined in [RFC5506]. For 503 backwards compatibility, implementations are also REQUIRED to support 504 the use of compound RTCP feedback packets if the remote end-point 505 does not agree to the use of non-compound RTCP in the signalling 506 exchange. 508 4.7. Symmetric RTP/RTCP 510 To ease traversal of NAT and firewall devices, implementations are 511 REQUIRED to implement and use Symmetric RTP [RFC4961]. The reason 512 for using symmetric RTP is primarily to avoid issues with NATs and 513 Firewalls by ensuring that the send and receive RTP packet streams, 514 as well as RTCP, are actually bi-directional transport-layer flows. 515 This will keep alive the NAT and firewall pinholes, and help indicate 516 consent that the receive direction is a transport-layer flow the 517 intended recipient actually wants. In addition, it saves resources, 518 specifically ports at the end-points, but also in the network as NAT 519 mappings or firewall state is not unnecessary bloated. The amount of 520 per flow QoS state kept in the network is also reduced. 522 4.8. Choice of RTP Synchronisation Source (SSRC) 524 Implementations are REQUIRED to support signalled RTP synchronisation 525 source (SSRC) identifiers. If SDP is used, this MUST be done using 526 the "a=ssrc:" SDP attribute defined in Section 4.1 and Section 5 of 527 [RFC5576] and the "previous-ssrc" source attribute defined in 528 Section 6.2 of [RFC5576]; other per-SSRC attributes defined in 529 [RFC5576] MAY be supported. 531 While support for signalled SSRC identifiers is mandated, their use 532 in an RTP session is OPTIONAL. Implementations MUST be prepared to 533 accept RTP and RTCP packets using SSRCs that have not been explicitly 534 signalled ahead of time. Implementations MUST support random SSRC 535 assignment, and MUST support SSRC collision detection and resolution, 536 according to [RFC3550]. When using signalled SSRC values, collision 537 detection MUST be performed as described in Section 5 of [RFC5576]. 539 It is often desirable to associate an RTP packet stream with a non- 540 RTP context. For users of the WebRTC API a mapping between SSRCs and 541 MediaStreamTracks are provided per Section 11. For gateways or other 542 usages it is possible to associate an RTP packet stream with an "m=" 543 line in a session description formatted using SDP. If SSRCs are 544 signalled this is straightforward (in SDP the "a=ssrc:" line will be 545 at the media level, allowing a direct association with an "m=" line). 546 If SSRCs are not signalled, the RTP payload type numbers used in an 547 RTP packet stream are often sufficient to associate that packet 548 stream with a signalling context (e.g., if RTP payload type numbers 549 are assigned as described in Section 4.3 of this memo, the RTP 550 payload types used by an RTP packet stream can be compared with 551 values in SDP "a=rtpmap:" lines, which are at the media level in SDP, 552 and so map to an "m=" line). 554 4.9. Generation of the RTCP Canonical Name (CNAME) 556 The RTCP Canonical Name (CNAME) provides a persistent transport-level 557 identifier for an RTP end-point. While the Synchronisation Source 558 (SSRC) identifier for an RTP end-point can change if a collision is 559 detected, or when the RTP application is restarted, its RTCP CNAME is 560 meant to stay unchanged for the duration of a RTCPeerConnection 561 [W3C.WD-webrtc-20130910], so that RTP end-points can be uniquely 562 identified and associated with their RTP packet streams within a set 563 of related RTP sessions. 565 Each RTP end-point MUST have at least one RTCP CNAME, and that RTCP 566 CNAME MUST be unique within the RTCPeerConnection. RTCP CNAMEs 567 identify a particular synchronisation context, i.e., all SSRCs 568 associated with a single RTCP CNAME share a common reference clock. 569 If an end-point has SSRCs that are associated with several 570 unsynchronised reference clocks, and hence different synchronisation 571 contexts, it will need to use multiple RTCP CNAMEs, one for each 572 synchronisation context. 574 Taking the discussion in Section 11 into account, a WebRTC Endpoint 575 MUST NOT use more than one RTCP CNAME in the RTP sessions belonging 576 to single RTCPeerConnection (that is, an RTCPeerConnection forms a 577 synchronisation context). RTP middleboxes MAY generate RTP packet 578 streams associated with more than one RTCP CNAME, to allow them to 579 avoid having to resynchronize media from multiple different end- 580 points part of a multi-party RTP session. 582 The RTP specification [RFC3550] includes guidelines for choosing a 583 unique RTP CNAME, but these are not sufficient in the presence of NAT 584 devices. In addition, long-term persistent identifiers can be 585 problematic from a privacy viewpoint (Section 13). Accordingly, a 586 WebRTC Endpoint MUST generate a new, unique, short-term persistent 587 RTCP CNAME for each RTCPeerConnection, following [RFC7022], with a 588 single exception; if explicitly requested at creation an 589 RTCPeerConnection MAY use the same CNAME as as an existing 590 RTCPeerConnection within their common same-origin context. 592 An WebRTC Endpoint MUST support reception of any CNAME that matches 593 the syntax limitations specified by the RTP specification [RFC3550] 594 and cannot assume that any CNAME will be chosen according to the form 595 suggested above. 597 4.10. Handling of Leap Seconds 599 The guidelines regarding handling of leap seconds to limit their 600 impact on RTP media play-out and synchronization given in [RFC7164] 601 SHOULD be followed. 603 5. WebRTC Use of RTP: Extensions 605 There are a number of RTP extensions that are either needed to obtain 606 full functionality, or extremely useful to improve on the baseline 607 performance, in the WebRTC context. One set of these extensions is 608 related to conferencing, while others are more generic in nature. 609 The following subsections describe the various RTP extensions 610 mandated or suggested for use within WebRTC. 612 5.1. Conferencing Extensions and Topologies 614 RTP is a protocol that inherently supports group communication. 615 Groups can be implemented by having each endpoint send its RTP packet 616 streams to an RTP middlebox that redistributes the traffic, by using 617 a mesh of unicast RTP packet streams between endpoints, or by using 618 an IP multicast group to distribute the RTP packet streams. These 619 topologies can be implemented in a number of ways as discussed in 620 [I-D.ietf-avtcore-rtp-topologies-update]. 622 While the use of IP multicast groups is popular in IPTV systems, the 623 topologies based on RTP middleboxes are dominant in interactive video 624 conferencing environments. Topologies based on a mesh of unicast 625 transport-layer flows to create a common RTP session have not seen 626 widespread deployment to date. Accordingly, WebRTC Endpoints are not 627 expected to support topologies based on IP multicast groups or to 628 support mesh-based topologies, such as a point-to-multipoint mesh 629 configured as a single RTP session (Topo-Mesh in the terminology of 630 [I-D.ietf-avtcore-rtp-topologies-update]). However, a point-to- 631 multipoint mesh constructed using several RTP sessions, implemented 632 in WebRTC using independent RTCPeerConnections 633 [W3C.WD-webrtc-20130910], can be expected to be used in WebRTC, and 634 needs to be supported. 636 WebRTC Endpoints implemented according to this memo are expected to 637 support all the topologies described in 638 [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send 639 and receive unicast RTP packet streams to and from some peer device, 640 provided that peer can participate in performing congestion control 641 on the RTP packet streams. The peer device could be another RTP 642 endpoint, or it could be an RTP middlebox that redistributes the RTP 643 packet streams to other RTP endpoints. This limitation means that 644 some of the RTP middlebox-based topologies are not suitable for use 645 in WebRTC. Specifically: 647 o Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used, 648 since they make the use of RTCP for congestion control and quality 649 of service reports problematic (see Section 3.8 of 650 [I-D.ietf-avtcore-rtp-topologies-update]). 652 o The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology 653 SHOULD NOT be used because its safe use requires a congestion 654 control algorithm or RTP circuit breaker that handles point to 655 multipoint, which has not yet been standardised. 657 The following topology can be used, however it has some issues worth 658 noting: 660 o Content modifying MCUs with RTCP termination (Topo-RTCP- 661 terminating-MCU) MAY be used. Note that in this RTP Topology, RTP 662 loop detection and identification of active senders is the 663 responsibility of the WebRTC application; since the clients are 664 isolated from each other at the RTP layer, RTP cannot assist with 665 these functions (see section 3.9 of 666 [I-D.ietf-avtcore-rtp-topologies-update]). 668 The RTP extensions described in Section 5.1.1 to Section 5.1.6 are 669 designed to be used with centralised conferencing, where an RTP 670 middlebox (e.g., a conference bridge) receives a participant's RTP 671 packet streams and distributes them to the other participants. These 672 extensions are not necessary for interoperability; an RTP end-point 673 that does not implement these extensions will work correctly, but 674 might offer poor performance. Support for the listed extensions will 675 greatly improve the quality of experience and, to provide a 676 reasonable baseline quality, some of these extensions are mandatory 677 to be supported by WebRTC Endpoints. 679 The RTCP conferencing extensions are defined in Extended RTP Profile 680 for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/ 681 AVPF) [RFC4585] and the memo on Codec Control Messages (CCM) in RTP/ 682 AVPF [RFC5104]; they are fully usable by the Secure variant of this 683 profile (RTP/SAVPF) [RFC5124]. 685 5.1.1. Full Intra Request (FIR) 687 The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1 688 of the Codec Control Messages [RFC5104]. It is used to make the 689 mixer request a new Intra picture from a participant in the session. 690 This is used when switching between sources to ensure that the 691 receivers can decode the video or other predictive media encoding 692 with long prediction chains. WebRTC Endpoints that are sending media 693 MUST understand and react to FIR feedback messages they receive, 694 since this greatly improves the user experience when using 695 centralised mixer-based conferencing. Support for sending FIR 696 messages is OPTIONAL. 698 5.1.2. Picture Loss Indication (PLI) 700 The Picture Loss Indication message is defined in Section 6.3.1 of 701 the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the 702 sending encoder that it lost the decoder context and would like to 703 have it repaired somehow. This is semantically different from the 704 Full Intra Request above as there could be multiple ways to fulfil 705 the request. WebRTC Endpoints that are sending media MUST understand 706 and react to PLI feedback messages as a loss tolerance mechanism. 707 Receivers MAY send PLI messages. 709 5.1.3. Slice Loss Indication (SLI) 711 The Slice Loss Indication message is defined in Section 6.3.2 of the 712 RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the 713 encoder that it has detected the loss or corruption of one or more 714 consecutive macro blocks, and would like to have these repaired 715 somehow. It is RECOMMENDED that receivers generate SLI feedback 716 messages if slices are lost when using a codec that supports the 717 concept of macro blocks. A sender that receives an SLI feedback 718 message SHOULD attempt to repair the lost slice(s). 720 5.1.4. Reference Picture Selection Indication (RPSI) 722 Reference Picture Selection Indication (RPSI) messages are defined in 723 Section 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video encoding 724 standards allow the use of older reference pictures than the most 725 recent one for predictive coding. If such a codec is in use, and if 726 the encoder has learnt that encoder-decoder synchronisation has been 727 lost, then a known as correct reference picture can be used as a base 728 for future coding. The RPSI message allows this to be signalled. 729 Receivers that detect that encoder-decoder synchronisation has been 730 lost SHOULD generate an RPSI feedback message if codec being used 731 supports reference picture selection. A RTP packet stream sender 732 that receives such an RPSI message SHOULD act on that messages to 733 change the reference picture, if it is possible to do so within the 734 available bandwidth constraints, and with the codec being used. 736 5.1.5. Temporal-Spatial Trade-off Request (TSTR) 738 The temporal-spatial trade-off request and notification are defined 739 in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used 740 to ask the video encoder to change the trade-off it makes between 741 temporal and spatial resolution, for example to prefer high spatial 742 image quality but low frame rate. Support for TSTR requests and 743 notifications is OPTIONAL. 745 5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR) 747 The TMMBR feedback message is defined in Sections 3.5.4 and 4.2.1 of 748 the Codec Control Messages [RFC5104]. This request and its 749 notification message are used by a media receiver to inform the 750 sending party that there is a current limitation on the amount of 751 bandwidth available to this receiver. This can be various reasons 752 for this: for example, an RTP mixer can use this message to limit the 753 media rate of the sender being forwarded by the mixer (without doing 754 media transcoding) to fit the bottlenecks existing towards the other 755 session participants. WebRTC Endpoints that are sending media are 756 REQUIRED to implement support for TMMBR messages, and MUST follow 757 bandwidth limitations set by a TMMBR message received for their SSRC. 758 The sending of TMMBR requests is OPTIONAL. 760 5.2. Header Extensions 762 The RTP specification [RFC3550] provides the capability to include 763 RTP header extensions containing in-band data, but the format and 764 semantics of the extensions are poorly specified. The use of header 765 extensions is OPTIONAL in WebRTC, but if they are used, they MUST be 766 formatted and signalled following the general mechanism for RTP 767 header extensions defined in [RFC5285], since this gives well-defined 768 semantics to RTP header extensions. 770 As noted in [RFC5285], the requirement from the RTP specification 771 that header extensions are "designed so that the header extension may 772 be ignored" [RFC3550] stands. To be specific, header extensions MUST 773 only be used for data that can safely be ignored by the recipient 774 without affecting interoperability, and MUST NOT be used when the 775 presence of the extension has changed the form or nature of the rest 776 of the packet in a way that is not compatible with the way the stream 777 is signalled (e.g., as defined by the payload type). Valid examples 778 of RTP header extensions might include metadata that is additional to 779 the usual RTP information, but that can safely be ignored without 780 compromising interoperability. 782 5.2.1. Rapid Synchronisation 784 Many RTP sessions require synchronisation between audio, video, and 785 other content. This synchronisation is performed by receivers, using 786 information contained in RTCP SR packets, as described in the RTP 787 specification [RFC3550]. This basic mechanism can be slow, however, 788 so it is RECOMMENDED that the rapid RTP synchronisation extensions 789 described in [RFC6051] be implemented in addition to RTCP SR-based 790 synchronisation. 792 This header extension uses the [RFC5285] generic header extension 793 framework, and so needs to be negotiated before it can be used. 795 5.2.2. Client-to-Mixer Audio Level 797 The Client to Mixer Audio Level extension [RFC6464] is an RTP header 798 extension used by an endpoint to inform a mixer about the level of 799 audio activity in the packet to which the header is attached. This 800 enables an RTP middlebox to make mixing or selection decisions 801 without decoding or detailed inspection of the payload, reducing the 802 complexity in some types of mixers. It can also save decoding 803 resources in receivers, which can choose to decode only the most 804 relevant RTP packet streams based on audio activity levels. 806 The Client-to-Mixer Audio Level [RFC6464] header extension MUST be 807 implemented. It is REQUIRED that implementations are capable of 808 encrypting the header extension according to [RFC6904] since the 809 information contained in these header extensions can be considered 810 sensitive. The use of this encryption is RECOMMENDED, however usage 811 of the encryption can be explicitly disabled through API or 812 signalling. 814 This header extension uses the [RFC5285] generic header extension 815 framework, and so needs to be negotiated before it can be used. 817 5.2.3. Mixer-to-Client Audio Level 819 The Mixer to Client Audio Level header extension [RFC6465] provides 820 an endpoint with the audio level of the different sources mixed into 821 a common source stream by a RTP mixer. This enables a user interface 822 to indicate the relative activity level of each session participant, 823 rather than just being included or not based on the CSRC field. This 824 is a pure optimisation of non critical functions, and is hence 825 OPTIONAL to implement. If this header extension is implemented, it 826 is REQUIRED that implementations are capable of encrypting the header 827 extension according to [RFC6904] since the information contained in 828 these header extensions can be considered sensitive. It is further 829 RECOMMENDED that this encryption is used, unless the encryption has 830 been explicitly disabled through API or signalling. 832 This header extension uses the [RFC5285] generic header extension 833 framework, and so needs to be negotiated before it can be used. 835 5.2.4. Media Stream Identification 837 WebRTC endpoints that implement the SDP bundle negotiation extension 838 will use the SDP grouping framework 'mid' attribute to identify media 839 streams. Such endpoints MUST implement the RTP MID header extension 840 described in [I-D.ietf-mmusic-sdp-bundle-negotiation]. 842 This header extension uses the [RFC5285] generic header extension 843 framework, and so needs to be negotiated before it can be used. 845 5.2.5. Coordination of Video Orientation 847 WebRTC endpoints that send or receive video MUST implement the 848 coordination of video orientation (CVO) RTP header extension as 849 described in Section 4 of [I-D.ietf-rtcweb-video]. 851 This header extension uses the [RFC5285] generic header extension 852 framework, and so needs to be negotiated before it can be used. 854 6. WebRTC Use of RTP: Improving Transport Robustness 856 There are tools that can make RTP packet streams robust against 857 packet loss and reduce the impact of loss on media quality. However, 858 they generally add some overhead compared to a non-robust stream. 859 The overhead needs to be considered, and the aggregate bit-rate MUST 860 be rate controlled to avoid causing network congestion (see 861 Section 7). As a result, improving robustness might require a lower 862 base encoding quality, but has the potential to deliver that quality 863 with fewer errors. The mechanisms described in the following sub- 864 sections can be used to improve tolerance to packet loss. 866 6.1. Negative Acknowledgements and RTP Retransmission 868 As a consequence of supporting the RTP/SAVPF profile, implementations 869 can send negative acknowledgements (NACKs) for RTP data packets 870 [RFC4585]. This feedback can be used to inform a sender of the loss 871 of particular RTP packets, subject to the capacity limitations of the 872 RTCP feedback channel. A sender can use this information to optimise 873 the user experience by adapting the media encoding to compensate for 874 known lost packets. 876 RTP packet stream senders are REQUIRED to understand the Generic NACK 877 message defined in Section 6.2.1 of [RFC4585], but MAY choose to 878 ignore some or all of this feedback (following Section 4.2 of 879 [RFC4585]). Receivers MAY send NACKs for missing RTP packets. 880 Guidelines on when to send NACKs are provided in [RFC4585]. It is 881 not expected that a receiver will send a NACK for every lost RTP 882 packet, rather it needs to consider the cost of sending NACK 883 feedback, and the importance of the lost packet, to make an informed 884 decision on whether it is worth telling the sender about a packet 885 loss event. 887 The RTP Retransmission Payload Format [RFC4588] offers the ability to 888 retransmit lost packets based on NACK feedback. Retransmission needs 889 to be used with care in interactive real-time applications to ensure 890 that the retransmitted packet arrives in time to be useful, but can 891 be effective in environments with relatively low network RTT (an RTP 892 sender can estimate the RTT to the receivers using the information in 893 RTCP SR and RR packets, as described at the end of Section 6.4.1 of 894 [RFC3550]). The use of retransmissions can also increase the forward 895 RTP bandwidth, and can potentially caused increased packet loss if 896 the original packet loss was caused by network congestion. Note, 897 however, that retransmission of an important lost packet to repair 898 decoder state can have lower cost than sending a full intra frame. 899 It is not appropriate to blindly retransmit RTP packets in response 900 to a NACK. The importance of lost packets and the likelihood of them 901 arriving in time to be useful needs to be considered before RTP 902 retransmission is used. 904 Receivers are REQUIRED to implement support for RTP retransmission 905 packets [RFC4588] sent using SSRC multiplexing, and MAY also support 906 RTP retransmission packets sent using session multiplexing. Senders 907 MAY send RTP retransmission packets in response to NACKs if support 908 for the RTP retransmission payload format has been negotiated, and if 909 the sender believes it is useful to send a retransmission of the 910 packet(s) referenced in the NACK. Senders do not need to retransmit 911 every NACKed packet. 913 6.2. Forward Error Correction (FEC) 915 The use of Forward Error Correction (FEC) can provide an effective 916 protection against some degree of packet loss, at the cost of steady 917 bandwidth overhead. There are several FEC schemes that are defined 918 for use with RTP. Some of these schemes are specific to a particular 919 RTP payload format, others operate across RTP packets and can be used 920 with any payload format. It needs to be noted that using redundant 921 encoding or FEC will lead to increased play out delay, which needs to 922 be considered when choosing FEC schemes and their parameters. 924 WebRTC endpoints MUST follow the recommendations for FEC use given in 925 [I-D.ietf-rtcweb-fec]. WebRTC endpoints MAY support other types of 926 FEC, but these MUST be negotiated before they are used. 928 7. WebRTC Use of RTP: Rate Control and Media Adaptation 930 WebRTC will be used in heterogeneous network environments using a 931 variety set of link technologies, including both wired and wireless 932 links, to interconnect potentially large groups of users around the 933 world. As a result, the network paths between users can have widely 934 varying one-way delays, available bit-rates, load levels, and traffic 935 mixtures. Individual end-points can send one or more RTP packet 936 streams to each participant, and there can be several participants. 937 Each of these RTP packet streams can contain different types of 938 media, and the type of media, bit rate, and number of RTP packet 939 streams as well as transport-layer flows can be highly asymmetric. 940 Non-RTP traffic can share the network paths with RTP transport-layer 941 flows. Since the network environment is not predictable or stable, 942 WebRTC Endpoints MUST ensure that the RTP traffic they generate can 943 adapt to match changes in the available network capacity. 945 The quality of experience for users of WebRTC is very dependent on 946 effective adaptation of the media to the limitations of the network. 947 End-points have to be designed so they do not transmit significantly 948 more data than the network path can support, except for very short 949 time periods, otherwise high levels of network packet loss or delay 950 spikes will occur, causing media quality degradation. The limiting 951 factor on the capacity of the network path might be the link 952 bandwidth, or it might be competition with other traffic on the link 953 (this can be non-WebRTC traffic, traffic due to other WebRTC flows, 954 or even competition with other WebRTC flows in the same session). 956 An effective media congestion control algorithm is therefore an 957 essential part of the WebRTC framework. However, at the time of this 958 writing, there is no standard congestion control algorithm that can 959 be used for interactive media applications such as WebRTC's flows. 960 Some requirements for congestion control algorithms for 961 RTCPeerConnections are discussed in [I-D.ietf-rmcat-cc-requirements]. 962 A future version of this memo will mandate the use of a congestion 963 control algorithm that satisfies these requirements. 965 7.1. Boundary Conditions and Circuit Breakers 967 WebRTC Endpoints MUST implement the RTP circuit breaker algorithm 968 that is described in [I-D.ietf-avtcore-rtp-circuit-breakers]. The 969 RTP circuit breaker is designed to enable applications to recognise 970 and react to situations of extreme network congestion. However, 971 since the RTP circuit breaker might not be triggered until congestion 972 becomes extreme, it cannot be considered a substitute for congestion 973 control, and applications MUST also implement congestion control to 974 allow them to adapt to changes in network capacity. Any future RTP 975 congestion control algorithms are expected to operate within the 976 envelope allowed by the circuit breaker. 978 The session establishment signalling will also necessarily establish 979 boundaries to which the media bit-rate will conform. The choice of 980 media codecs provides upper- and lower-bounds on the supported bit- 981 rates that the application can utilise to provide useful quality, and 982 the packetisation choices that exist. In addition, the signalling 983 channel can establish maximum media bit-rate boundaries using, for 984 example, the SDP "b=AS:" or "b=CT:" lines and the RTP/AVPF Temporary 985 Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of 986 this memo). Signalled bandwidth limitations, such as SDP "b=AS:" or 987 "b=CT:" lines received from the peer, MUST be followed when sending 988 RTP packet streams. A WebRTC Endpoint receiving media SHOULD signal 989 its bandwidth limitations, these limitations have to be based on 990 known bandwidth limitations, for example the capacity of the edge 991 links. 993 7.2. Congestion Control Interoperability and Legacy Systems 995 All endpoints that wish to interwork with WebRTC MUST implement RTCP 996 and provide congestion feedback via the defined RTCP reporting 997 mechanisms. 999 When interworking with legacy implementations that support RTCP using 1000 the RTP/AVP profile [RFC3551], congestion feedback is provided in 1001 RTCP RR packets every few seconds. Implementations that have to 1002 interwork with such end-points MUST ensure that they keep within the 1003 RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers] 1004 constraints to limit the congestion they can cause. 1006 If a legacy end-point supports RTP/AVPF, this enables negotiation of 1007 important parameters for frequent reporting, such as the "trr-int" 1008 parameter, and the possibility that the end-point supports some 1009 useful feedback format for congestion control purpose such as TMMBR 1010 [RFC5104]. Implementations that have to interwork with such end- 1011 points MUST ensure that they stay within the RTP circuit breaker 1012 [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the 1013 congestion they can cause, but might find that they can achieve 1014 better congestion response depending on the amount of feedback that 1015 is available. 1017 With proprietary congestion control algorithms issues can arise when 1018 different algorithms and implementations interact in a communication 1019 session. If the different implementations have made different 1020 choices in regards to the type of adaptation, for example one sender 1021 based, and one receiver based, then one could end up in situation 1022 where one direction is dual controlled, when the other direction is 1023 not controlled. This memo cannot mandate behaviour for proprietary 1024 congestion control algorithms, but implementations that use such 1025 algorithms ought to be aware of this issue, and try to ensure that 1026 effective congestion control is negotiated for media flowing in both 1027 directions. If the IETF were to standardise both sender- and 1028 receiver-based congestion control algorithms for WebRTC traffic in 1029 the future, the issues of interoperability, control, and ensuring 1030 that both directions of media flow are congestion controlled would 1031 also need to be considered. 1033 8. WebRTC Use of RTP: Performance Monitoring 1035 As described in Section 4.1, implementations are REQUIRED to generate 1036 RTCP Sender Report (SR) and Reception Report (RR) packets relating to 1037 the RTP packet streams they send and receive. These RTCP reports can 1038 be used for performance monitoring purposes, since they include basic 1039 packet loss and jitter statistics. 1041 A large number of additional performance metrics are supported by the 1042 RTCP Extended Reports (XR) framework [RFC3611][RFC6792]. At the time 1043 of this writing, it is not clear what extended metrics are suitable 1044 for use in WebRTC, so there is no requirement that implementations 1045 generate RTCP XR packets. However, implementations that can use 1046 detailed performance monitoring data MAY generate RTCP XR packets as 1047 appropriate; the use of such packets SHOULD be signalled in advance. 1049 9. WebRTC Use of RTP: Future Extensions 1051 It is possible that the core set of RTP protocols and RTP extensions 1052 specified in this memo will prove insufficient for the future needs 1053 of WebRTC. In this case, future updates to this memo MUST be made 1054 following the Guidelines for Writers of RTP Payload Format 1055 Specifications [RFC2736], How to Write an RTP Payload Format 1056 [I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP 1057 Control Protocol [RFC5968], and SHOULD take into account any future 1058 guidelines for extending RTP and related protocols that have been 1059 developed. 1061 Authors of future extensions are urged to consider the wide range of 1062 environments in which RTP is used when recommending extensions, since 1063 extensions that are applicable in some scenarios can be problematic 1064 in others. Where possible, the WebRTC framework will adopt RTP 1065 extensions that are of general utility, to enable easy implementation 1066 of a gateway to other applications using RTP, rather than adopt 1067 mechanisms that are narrowly targeted at specific WebRTC use cases. 1069 10. Signalling Considerations 1071 RTP is built with the assumption that an external signalling channel 1072 exists, and can be used to configure RTP sessions and their features. 1073 The basic configuration of an RTP session consists of the following 1074 parameters: 1076 RTP Profile: The name of the RTP profile to be used in session. The 1077 RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate 1078 on basic level, as can their secure variants RTP/SAVP [RFC3711] 1079 and RTP/SAVPF [RFC5124]. The secure variants of the profiles do 1080 not directly interoperate with the non-secure variants, due to the 1081 presence of additional header fields for authentication in SRTP 1082 packets and cryptographic transformation of the payload. WebRTC 1083 requires the use of the RTP/SAVPF profile, and this MUST be 1084 signalled. Interworking functions might transform this into the 1085 RTP/SAVP profile for a legacy use case, by indicating to the 1086 WebRTC Endpoint that the RTP/SAVPF is used and configuring a trr- 1087 int value of 4 seconds. 1089 Transport Information: Source and destination IP address(s) and 1090 ports for RTP and RTCP MUST be signalled for each RTP session. In 1091 WebRTC these transport addresses will be provided by ICE [RFC5245] 1092 that signals candidates and arrives at nominated candidate address 1093 pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such 1094 that a single port, i.e. transport-layer flow, is used for RTP 1095 and RTCP flows, this MUST be signalled (see Section 4.5). 1097 RTP Payload Types, media formats, and format parameters: The mapping 1098 between media type names (and hence the RTP payload formats to be 1099 used), and the RTP payload type numbers MUST be signalled. Each 1100 media type MAY also have a number of media type parameters that 1101 MUST also be signalled to configure the codec and RTP payload 1102 format (the "a=fmtp:" line from SDP). Section 4.3 of this memo 1103 discusses requirements for uniqueness of payload types. 1105 RTP Extensions: The use of any additional RTP header extensions and 1106 RTCP packet types, including any necessary parameters, MUST be 1107 signalled. This signalling is to ensure that a WebRTC Endpoint's 1108 behaviour, especially when sending, of any extensions is 1109 predictable and consistent. For robustness, and for compatibility 1110 with non-WebRTC systems that might be connected to a WebRTC 1111 session via a gateway, implementations are REQUIRED to ignore 1112 unknown RTCP packets and RTP header extensions (see also 1113 Section 4.1). 1115 RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the 1116 end-points will be necessary. This SHALL be done as described in 1117 "Session Description Protocol (SDP) Bandwidth Modifiers for RTP 1118 Control Protocol (RTCP) Bandwidth" [RFC3556] if using SDP, or 1119 something semantically equivalent. This also ensures that the 1120 end-points have a common view of the RTCP bandwidth. A common 1121 RTCP bandwidth is important as a too different view of the 1122 bandwidths can lead to failure to interoperate. 1124 These parameters are often expressed in SDP messages conveyed within 1125 an offer/answer exchange. RTP does not depend on SDP or on the offer 1126 /answer model, but does require all the necessary parameters to be 1127 agreed upon, and provided to the RTP implementation. Note that in 1128 WebRTC it will depend on the signalling model and API how these 1129 parameters need to be configured but they will be need to either be 1130 set in the API or explicitly signalled between the peers. 1132 11. WebRTC API Considerations 1134 The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and 1135 Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses 1136 the concept of a MediaStream that consists of zero or more 1137 MediaStreamTracks. A MediaStreamTrack is an individual stream of 1138 media from any type of media source like a microphone or a camera, 1139 but also conceptual sources, like a audio mix or a video composition, 1140 are possible. The MediaStreamTracks within a MediaStream need to be 1141 possible to play out synchronised. 1143 A MediaStreamTrack's realisation in RTP in the context of an 1144 RTCPeerConnection consists of a source packet stream identified with 1145 an SSRC within an RTP session part of the RTCPeerConnection. The 1146 MediaStreamTrack can also result in additional packet streams, and 1147 thus SSRCs, in the same RTP session. These can be dependent packet 1148 streams from scalable encoding of the source stream associated with 1149 the MediaStreamTrack, if such a media encoder is used. They can also 1150 be redundancy packet streams, these are created when applying Forward 1151 Error Correction (Section 6.2) or RTP retransmission (Section 6.1) to 1152 the source packet stream. 1154 It is important to note that the same media source can be feeding 1155 multiple MediaStreamTracks. As different sets of constraints or 1156 other parameters can be applied to the MediaStreamTrack, each 1157 MediaStreamTrack instance added to a RTCPeerConnection SHALL result 1158 in an independent source packet stream, with its own set of 1159 associated packet streams, and thus different SSRC(s). It will 1160 depend on applied constraints and parameters if the source stream and 1161 the encoding configuration will be identical between different 1162 MediaStreamTracks sharing the same media source. If the encoding 1163 parameters and constraints are the same, an implementation could 1164 choose to use only one encoded stream to create the different RTP 1165 packet streams. Note that such optimisations would need to take into 1166 account that the constraints for one of the MediaStreamTracks can at 1167 any moment change, meaning that the encoding configurations might no 1168 longer be identical and two different encoder instances would then be 1169 needed. 1171 The same MediaStreamTrack can also be included in multiple 1172 MediaStreams, thus multiple sets of MediaStreams can implicitly need 1173 to use the same synchronisation base. To ensure that this works in 1174 all cases, and does not force an end-point to to disrupt the media by 1175 changing synchronisation base and CNAME during delivery of any 1176 ongoing packet streams, all MediaStreamTracks and their associated 1177 SSRCs originating from the same end-point need to be sent using the 1178 same CNAME within one RTCPeerConnection. This is motivating the 1179 discussion in Section 4.9 to only use a single CNAME. 1181 The requirement on using the same CNAME for all SSRCs that 1182 originate from the same end-point, does not require a middlebox 1183 that forwards traffic from multiple end-points to only use a 1184 single CNAME. 1186 Different CNAMEs normally need to be used for different 1187 RTCPeerConnection instances, as specified in Section 4.9. Having two 1188 communication sessions with the same CNAME could enable tracking of a 1189 user or device across different services (see Section 4.4.1 of 1190 [I-D.ietf-rtcweb-security] for details). A web application can 1191 request that the CNAMEs used in different RTCPeerConnections (within 1192 a same-orign context) be the same, this allows for synchronization of 1193 the endpoint's RTP packet streams across the different 1194 RTCPeerConnections. 1196 Note: this doesn't result in a tracking issue, since the creation 1197 of matching CNAMEs depends on existing tracking. 1199 The above will currently force a WebRTC Endpoint that receives a 1200 MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing 1201 on any RTCPeerConnection to perform resynchronisation of the stream. 1203 This, as the sending party needs to change the CNAME to the one it 1204 uses, which implies that the sender has to use a local system clock 1205 as timebase for the synchronisation. Thus, the relative relation 1206 between the timebase of the incoming stream and the system sending 1207 out needs to defined. This relation also needs monitoring for clock 1208 drift and likely adjustments of the synchronisation. The sending 1209 entity is also responsible for congestion control for its sent 1210 streams. In cases of packet loss the loss of incoming data also 1211 needs to be handled. This leads to the observation that the method 1212 that is least likely to cause issues or interruptions in the outgoing 1213 source packet stream is a model of full decoding, including repair 1214 etc., followed by encoding of the media again into the outgoing 1215 packet stream. Optimisations of this method is clearly possible and 1216 implementation specific. 1218 A WebRTC Endpoint MUST support receiving multiple MediaStreamTracks, 1219 where each of different MediaStreamTracks (and their sets of 1220 associated packet streams) uses different CNAMEs. However, 1221 MediaStreamTracks that are received with different CNAMEs have no 1222 defined synchronisation. 1224 Note: The motivation for supporting reception of multiple CNAMEs 1225 is to allow for forward compatibility with any future changes that 1226 enables more efficient stream handling when end-points relay/ 1227 forward streams. It also ensures that end-points can interoperate 1228 with certain types of multi-stream middleboxes or end-points that 1229 are not WebRTC. 1231 The binding between the WebRTC MediaStreams, MediaStreamTracks and 1232 the SSRC is done as specified in "Cross Session Stream Identification 1233 in the Session Description Protocol" [I-D.ietf-mmusic-msid]. This 1234 document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to 1235 map unknown source packet stream SSRCs to MediaStreamTracks and 1236 MediaStreams. This later is relevant to handle some cases of legacy 1237 interop. Commonly the RTP Payload Type of any incoming packets will 1238 reveal if the packet stream is a source stream or a redundancy or 1239 dependent packet stream. The association to the correct source 1240 packet stream depends on the payload format in use for the packet 1241 stream. 1243 Finally this specification puts a requirement on the WebRTC API to 1244 realize a method for determining the CSRC list (Section 4.1) as well 1245 as the Mixer-to-Client audio levels (Section 5.2.3) (when supported) 1246 and the basic requirements for this is further discussed in 1247 Section 12.2.1. 1249 12. RTP Implementation Considerations 1250 The following discussion provides some guidance on the implementation 1251 of the RTP features described in this memo. The focus is on a WebRTC 1252 Endpoint implementation perspective, and while some mention is made 1253 of the behaviour of middleboxes, that is not the focus of this memo. 1255 12.1. Configuration and Use of RTP Sessions 1257 A WebRTC Endpoint will be a simultaneous participant in one or more 1258 RTP sessions. Each RTP session can convey multiple media sources, 1259 and can include media data from multiple end-points. In the 1260 following, some ways in which WebRTC Endpoints can configure and use 1261 RTP sessions is outlined. 1263 12.1.1. Use of Multiple Media Sources Within an RTP Session 1265 RTP is a group communication protocol, and every RTP session can 1266 potentially contain multiple RTP packet streams. There are several 1267 reasons why this might be desirable: 1269 Multiple media types: Outside of WebRTC, it is common to use one RTP 1270 session for each type of media sources (e.g., one RTP session for 1271 audio sources and one for video sources, each sent over different 1272 transport layer flows). However, to reduce the number of UDP 1273 ports used, the default in WebRTC is to send all types of media in 1274 a single RTP session, as described in Section 4.4, using RTP and 1275 RTCP multiplexing (Section 4.5) to further reduce the number of 1276 UDP ports needed. This RTP session then uses only one bi- 1277 directional transport-layer flow, but will contain multiple RTP 1278 packet streams, each containing a different type of media. A 1279 common example might be an end-point with a camera and microphone 1280 that sends two RTP packet streams, one video and one audio, into a 1281 single RTP session. 1283 Multiple Capture Devices: A WebRTC Endpoint might have multiple 1284 cameras, microphones, or other media capture devices, and so might 1285 want to generate several RTP packet streams of the same media 1286 type. Alternatively, it might want to send media from a single 1287 capture device in several different formats or quality settings at 1288 once. Both can result in a single end-point sending multiple RTP 1289 packet streams of the same media type into a single RTP session at 1290 the same time. 1292 Associated Repair Data: An end-point might send a RTP packet stream 1293 that is somehow associated with another stream. For example, it 1294 might send an RTP packet stream that contains FEC or 1295 retransmission data relating to another stream. Some RTP payload 1296 formats send this sort of associated repair data as part of the 1297 source packet stream, while others send it as a separate packet 1298 stream. 1300 Layered or Multiple Description Coding: An end-point can use a 1301 layered media codec, for example H.264 SVC, or a multiple 1302 description codec, that generates multiple RTP packet streams, 1303 each with a distinct RTP SSRC, within a single RTP session. 1305 RTP Mixers, Translators, and Other Middleboxes: An RTP session, in 1306 the WebRTC context, is a point-to-point association between an 1307 end-point and some other peer device, where those devices share a 1308 common SSRC space. The peer device might be another WebRTC 1309 Endpoint, or it might be an RTP mixer, translator, or some other 1310 form of media processing middlebox. In the latter cases, the 1311 middlebox might send mixed or relayed RTP streams from several 1312 participants, that the WebRTC Endpoint will need to render. Thus, 1313 even though a WebRTC Endpoint might only be a member of a single 1314 RTP session, the peer device might be extending that RTP session 1315 to incorporate other end-points. WebRTC is a group communication 1316 environment and end-points need to be capable of receiving, 1317 decoding, and playing out multiple RTP packet streams at once, 1318 even in a single RTP session. 1320 12.1.2. Use of Multiple RTP Sessions 1322 In addition to sending and receiving multiple RTP packet streams 1323 within a single RTP session, a WebRTC Endpoint might participate in 1324 multiple RTP sessions. There are several reasons why a WebRTC 1325 Endpoint might choose to do this: 1327 To interoperate with legacy devices: The common practice in the non- 1328 WebRTC world is to send different types of media in separate RTP 1329 sessions, for example using one RTP session for audio and another 1330 RTP session, on a separate transport layer flow, for video. All 1331 WebRTC Endpoints need to support the option of sending different 1332 types of media on different RTP sessions, so they can interwork 1333 with such legacy devices. This is discussed further in 1334 Section 4.4. 1336 To provide enhanced quality of service: Some network-based quality 1337 of service mechanisms operate on the granularity of transport 1338 layer flows. If it is desired to use these mechanisms to provide 1339 differentiated quality of service for some RTP packet streams, 1340 then those RTP packet streams need to be sent in a separate RTP 1341 session using a different transport-layer flow, and with 1342 appropriate quality of service marking. This is discussed further 1343 in Section 12.1.3. 1345 To separate media with different purposes: An end-point might want 1346 to send RTP packet streams that have different purposes on 1347 different RTP sessions, to make it easy for the peer device to 1348 distinguish them. For example, some centralised multiparty 1349 conferencing systems display the active speaker in high 1350 resolution, but show low resolution "thumbnails" of other 1351 participants. Such systems might configure the end-points to send 1352 simulcast high- and low-resolution versions of their video using 1353 separate RTP sessions, to simplify the operation of the RTP 1354 middlebox. In the WebRTC context this is currently possible by 1355 establishing multiple WebRTC MediaStreamTracks that have the same 1356 media source in one (or more) RTCPeerConnection. Each 1357 MediaStreamTrack is then configured to deliver a particular media 1358 quality and thus media bit-rate, and will produce an independently 1359 encoded version with the codec parameters agreed specifically in 1360 the context of that RTCPeerConnection. The RTP middlebox can 1361 distinguish packets corresponding to the low- and high-resolution 1362 streams by inspecting their SSRC, RTP payload type, or some other 1363 information contained in RTP payload, RTP header extension or RTCP 1364 packets, but it can be easier to distinguish the RTP packet 1365 streams if they arrive on separate RTP sessions on separate 1366 transport-layer flows. 1368 To directly connect with multiple peers: A multi-party conference 1369 does not need to use an RTP middlebox. Rather, a multi-unicast 1370 mesh can be created, comprising several distinct RTP sessions, 1371 with each participant sending RTP traffic over a separate RTP 1372 session (that is, using an independent RTCPeerConnection object) 1373 to every other participant, as shown in Figure 1. This topology 1374 has the benefit of not requiring an RTP middlebox node that is 1375 trusted to access and manipulate the media data. The downside is 1376 that it increases the used bandwidth at each sender by requiring 1377 one copy of the RTP packet streams for each participant that are 1378 part of the same session beyond the sender itself. 1380 +---+ +---+ 1381 | A |<--->| B | 1382 +---+ +---+ 1383 ^ ^ 1384 \ / 1385 \ / 1386 v v 1387 +---+ 1388 | C | 1389 +---+ 1391 Figure 1: Multi-unicast using several RTP sessions 1393 The multi-unicast topology could also be implemented as a single 1394 RTP session, spanning multiple peer-to-peer transport layer 1395 connections, or as several pairwise RTP sessions, one between each 1396 pair of peers. To maintain a coherent mapping between the 1397 relation between RTP sessions and RTCPeerConnection objects it is 1398 recommend that this is implemented as several individual RTP 1399 sessions. The only downside is that end-point A will not learn of 1400 the quality of any transmission happening between B and C, since 1401 it will not see RTCP reports for the RTP session between B and C, 1402 whereas it would it all three participants were part of a single 1403 RTP session. Experience with the Mbone tools (experimental RTP- 1404 based multicast conferencing tools from the late 1990s) has showed 1405 that RTCP reception quality reports for third parties can be 1406 presented to users in a way that helps them understand asymmetric 1407 network problems, and the approach of using separate RTP sessions 1408 prevents this. However, an advantage of using separate RTP 1409 sessions is that it enables using different media bit-rates and 1410 RTP session configurations between the different peers, thus not 1411 forcing B to endure the same quality reductions if there are 1412 limitations in the transport from A to C as C will. It is 1413 believed that these advantages outweigh the limitations in 1414 debugging power. 1416 To indirectly connect with multiple peers: A common scenario in 1417 multi-party conferencing is to create indirect connections to 1418 multiple peers, using an RTP mixer, translator, or some other type 1419 of RTP middlebox. Figure 2 outlines a simple topology that might 1420 be used in a four-person centralised conference. The middlebox 1421 acts to optimise the transmission of RTP packet streams from 1422 certain perspectives, either by only sending some of the received 1423 RTP packet stream to any given receiver, or by providing a 1424 combined RTP packet stream out of a set of contributing streams. 1426 +---+ +-------------+ +---+ 1427 | A |<---->| |<---->| B | 1428 +---+ | RTP mixer, | +---+ 1429 | translator, | 1430 | or other | 1431 +---+ | middlebox | +---+ 1432 | C |<---->| |<---->| D | 1433 +---+ +-------------+ +---+ 1435 Figure 2: RTP mixer with only unicast paths 1437 There are various methods of implementation for the middlebox. If 1438 implemented as a standard RTP mixer or translator, a single RTP 1439 session will extend across the middlebox and encompass all the 1440 end-points in one multi-party session. Other types of middlebox 1441 might use separate RTP sessions between each end-point and the 1442 middlebox. A common aspect is that these RTP middleboxes can use 1443 a number of tools to control the media encoding provided by a 1444 WebRTC Endpoint. This includes functions like requesting the 1445 breaking of the encoding chain and have the encoder produce a so 1446 called Intra frame. Another is limiting the bit-rate of a given 1447 stream to better suit the mixer view of the multiple down-streams. 1448 Others are controlling the most suitable frame-rate, picture 1449 resolution, the trade-off between frame-rate and spatial quality. 1450 The middlebox has the responsibility to correctly perform 1451 congestion control, source identification, manage synchronisation 1452 while providing the application with suitable media optimisations. 1453 The middlebox also has to be a trusted node when it comes to 1454 security, since it manipulates either the RTP header or the media 1455 itself (or both) received from one end-point, before sending it on 1456 towards the end-point(s), thus they need to be able to decrypt and 1457 then re-encrypt the RTP packet stream before sending it out. 1459 RTP Mixers can create a situation where an end-point experiences a 1460 situation in-between a session with only two end-points and 1461 multiple RTP sessions. Mixers are expected to not forward RTCP 1462 reports regarding RTP packet streams across themselves. This is 1463 due to the difference in the RTP packet streams provided to the 1464 different end-points. The original media source lacks information 1465 about a mixer's manipulations prior to sending it the different 1466 receivers. This scenario also results in that an end-point's 1467 feedback or requests goes to the mixer. When the mixer can't act 1468 on this by itself, it is forced to go to the original media source 1469 to fulfil the receivers request. This will not necessarily be 1470 explicitly visible any RTP and RTCP traffic, but the interactions 1471 and the time to complete them will indicate such dependencies. 1473 Providing source authentication in multi-party scenarios is a 1474 challenge. In the mixer-based topologies, end-points source 1475 authentication is based on, firstly, verifying that media comes 1476 from the mixer by cryptographic verification and, secondly, trust 1477 in the mixer to correctly identify any source towards the end- 1478 point. In RTP sessions where multiple end-points are directly 1479 visible to an end-point, all end-points will have knowledge about 1480 each others' master keys, and can thus inject packets claimed to 1481 come from another end-point in the session. Any node performing 1482 relay can perform non-cryptographic mitigation by preventing 1483 forwarding of packets that have SSRC fields that came from other 1484 end-points before. For cryptographic verification of the source, 1485 SRTP would require additional security mechanisms, for example 1486 TESLA for SRTP [RFC4383], that are not part of the base WebRTC 1487 standards. 1489 To forward media between multiple peers: It is sometimes desirable 1490 for an end-point that receives an RTP packet stream to be able to 1491 forward that RTP packet stream to a third party. The are some 1492 obvious security and privacy implications in supporting this, but 1493 also potential uses. This is supported in the W3C API by taking 1494 the received and decoded media and using it as media source that 1495 is re-encoding and transmitted as a new stream. 1497 At the RTP layer, media forwarding acts as a back-to-back RTP 1498 receiver and RTP sender. The receiving side terminates the RTP 1499 session and decodes the media, while the sender side re-encodes 1500 and transmits the media using an entirely separate RTP session. 1501 The original sender will only see a single receiver of the media, 1502 and will not be able to tell that forwarding is happening based on 1503 RTP-layer information since the RTP session that is used to send 1504 the forwarded media is not connected to the RTP session on which 1505 the media was received by the node doing the forwarding. 1507 The end-point that is performing the forwarding is responsible for 1508 producing an RTP packet stream suitable for onwards transmission. 1509 The outgoing RTP session that is used to send the forwarded media 1510 is entirely separate to the RTP session on which the media was 1511 received. This will require media transcoding for congestion 1512 control purpose to produce a suitable bit-rate for the outgoing 1513 RTP session, reducing media quality and forcing the forwarding 1514 end-point to spend the resource on the transcoding. The media 1515 transcoding does result in a separation of the two different legs 1516 removing almost all dependencies, and allowing the forwarding end- 1517 point to optimise its media transcoding operation. The cost is 1518 greatly increased computational complexity on the forwarding node. 1519 Receivers of the forwarded stream will see the forwarding device 1520 as the sender of the stream, and will not be able to tell from the 1521 RTP layer that they are receiving a forwarded stream rather than 1522 an entirely new RTP packet stream generated by the forwarding 1523 device. 1525 12.1.3. Differentiated Treatment of RTP Packet Streams 1527 There are use cases for differentiated treatment of RTP packet 1528 streams. Such differentiation can happen at several places in the 1529 system. First of all is the prioritization within the end-point 1530 sending the media, which controls, both which RTP packet streams that 1531 will be sent, and their allocation of bit-rate out of the current 1532 available aggregate as determined by the congestion control. 1534 It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will 1535 allow the application to indicate relative priorities for different 1536 MediaStreamTracks. These priorities can then be used to influence 1537 the local RTP processing, especially when it comes to congestion 1538 control response in how to divide the available bandwidth between the 1539 RTP packet streams. Any changes in relative priority will also need 1540 to be considered for RTP packet streams that are associated with the 1541 main RTP packet streams, such as redundant streams for RTP 1542 retransmission and FEC. The importance of such redundant RTP packet 1543 streams is dependent on the media type and codec used, in regards to 1544 how robust that codec is to packet loss. However, a default policy 1545 might to be to use the same priority for redundant RTP packet stream 1546 as for the source RTP packet stream. 1548 Secondly, the network can prioritize transport-layer flows and sub- 1549 flows, including RTP packet streams. Typically, differential 1550 treatment includes two steps, the first being identifying whether an 1551 IP packet belongs to a class that has to be treated differently, the 1552 second consisting of the actual mechanism to prioritize packets. 1553 This is done according to three methods: 1555 DiffServ: The end-point marks a packet with a DiffServ code point to 1556 indicate to the network that the packet belongs to a particular 1557 class. 1559 Flow based: Packets that need to be given a particular treatment are 1560 identified using a combination of IP and port address. 1562 Deep Packet Inspection: A network classifier (DPI) inspects the 1563 packet and tries to determine if the packet represents a 1564 particular application and type that is to be prioritized. 1566 Flow-based differentiation will provide the same treatment to all 1567 packets within a transport-layer flow, i.e., relative prioritization 1568 is not possible. Moreover, if the resources are limited it might not 1569 be possible to provide differential treatment compared to best-effort 1570 for all the RTP packet streams used in a WebRTC session. When flow- 1571 based differentiation is available, the WebRTC Endpoint needs to know 1572 about it so that it can provide the separation of the RTP packet 1573 streams onto different UDP flows to enable a more granular usage of 1574 flow based differentiation. That way at least providing different 1575 prioritization of audio and video if desired by application. 1577 DiffServ assumes that either the end-point or a classifier can mark 1578 the packets with an appropriate DSCP so that the packets are treated 1579 according to that marking. If the end-point is to mark the traffic 1580 two requirements arise in the WebRTC context: 1) The WebRTC Endpoint 1581 has to know which DSCP to use and that it can use them on some set of 1582 RTP packet streams. 2) The information needs to be propagated to the 1583 operating system when transmitting the packet. Details of this 1584 process are outside the scope of this memo and are further discussed 1585 in "DSCP and other packet markings for RTCWeb QoS" 1586 [I-D.ietf-tsvwg-rtcweb-qos]. 1588 For packet based marking schemes it might be possible to mark 1589 individual RTP packets differently based on the relative priority of 1590 the RTP payload. For example video codecs that have I, P, and B 1591 pictures could prioritise any payloads carrying only B frames less, 1592 as these are less damaging to loose. However, depending on the QoS 1593 mechanism and what markings that are applied, this can result in not 1594 only different packet drop probabilities but also packet reordering, 1595 see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion. As a default 1596 policy all RTP packets related to a RTP packet stream ought to be 1597 provided with the same prioritization; per-packet prioritization is 1598 outside the scope of this memo, but might be specified elsewhere in 1599 future. 1601 It is also important to consider how RTCP packets associated with a 1602 particular RTP packet stream need to be marked. RTCP compound 1603 packets with Sender Reports (SR), ought to be marked with the same 1604 priority as the RTP packet stream itself, so the RTCP-based round- 1605 trip time (RTT) measurements are done using the same transport-layer 1606 flow priority as the RTP packet stream experiences. RTCP compound 1607 packets containing RR packet ought to be sent with the priority used 1608 by the majority of the RTP packet streams reported on. RTCP packets 1609 containing time-critical feedback packets can use higher priority to 1610 improve the timeliness and likelihood of delivery of such feedback. 1612 12.2. Media Source, RTP Packet Streams, and Participant Identification 1614 12.2.1. Media Source Identification 1616 Each RTP packet stream is identified by a unique synchronisation 1617 source (SSRC) identifier. The SSRC identifier is carried in each of 1618 the RTP packets comprising a RTP packet stream, and is also used to 1619 identify that stream in the corresponding RTCP reports. The SSRC is 1620 chosen as discussed in Section 4.8. The first stage in 1621 demultiplexing RTP and RTCP packets received on a single transport 1622 layer flow at a WebRTC Endpoint is to separate the RTP packet streams 1623 based on their SSRC value; once that is done, additional 1624 demultiplexing steps can determine how and where to render the media. 1626 RTP allows a mixer, or other RTP-layer middlebox, to combine encoded 1627 streams from multiple media sources to form a new encoded stream from 1628 a new media source (the mixer). The RTP packets in that new RTP 1629 packet stream can include a Contributing Source (CSRC) list, 1630 indicating which original SSRCs contributed to the combined source 1631 stream. As described in Section 4.1, implementations need to support 1632 reception of RTP data packets containing a CSRC list and RTCP packets 1633 that relate to sources present in the CSRC list. The CSRC list can 1634 change on a packet-by-packet basis, depending on the mixing operation 1635 being performed. Knowledge of what media sources contributed to a 1636 particular RTP packet can be important if the user interface 1637 indicates which participants are active in the session. Changes in 1638 the CSRC list included in packets needs to be exposed to the WebRTC 1639 application using some API, if the application is to be able to track 1640 changes in session participation. It is desirable to map CSRC values 1641 back into WebRTC MediaStream identities as they cross this API, to 1642 avoid exposing the SSRC/CSRC name space to WebRTC applications. 1644 If the mixer-to-client audio level extension [RFC6465] is being used 1645 in the session (see Section 5.2.3), the information in the CSRC list 1646 is augmented by audio level information for each contributing source. 1647 It is desirable to expose this information to the WebRTC application 1648 using some API, after mapping the CSRC values to WebRTC MediaStream 1649 identities, so it can be exposed in the user interface. 1651 12.2.2. SSRC Collision Detection 1653 The RTP standard requires RTP implementations to have support for 1654 detecting and handling SSRC collisions, i.e., resolve the conflict 1655 when two different end-points use the same SSRC value (see section 1656 8.2 of [RFC3550]). This requirement also applies to WebRTC 1657 Endpoints. There are several scenarios where SSRC collisions can 1658 occur: 1660 o In a point-to-point session where each SSRC is associated with 1661 either of the two end-points and where the main media carrying 1662 SSRC identifier will be announced in the signalling channel, a 1663 collision is less likely to occur due to the information about 1664 used SSRCs. If SDP is used, this information is provided by 1665 Source-Specific SDP Attributes [RFC5576]. Still, collisions can 1666 occur if both end-points start using a new SSRC identifier prior 1667 to having signalled it to the peer and received acknowledgement on 1668 the signalling message. The Source-Specific SDP Attributes 1669 [RFC5576] contains a mechanism to signal how the end-point 1670 resolved the SSRC collision. 1672 o SSRC values that have not been signalled could also appear in an 1673 RTP session. This is more likely than it appears, since some RTP 1674 functions use extra SSRCs to provide their functionality. For 1675 example, retransmission data might be transmitted using a separate 1676 RTP packet stream that requires its own SSRC, separate to the SSRC 1677 of the source RTP packet stream [RFC4588]. In those cases, an 1678 end-point can create a new SSRC that strictly doesn't need to be 1679 announced over the signalling channel to function correctly on 1680 both RTP and RTCPeerConnection level. 1682 o Multiple end-points in a multiparty conference can create new 1683 sources and signal those towards the RTP middlebox. In cases 1684 where the SSRC/CSRC are propagated between the different end- 1685 points from the RTP middlebox collisions can occur. 1687 o An RTP middlebox could connect an end-point's RTCPeerConnection to 1688 another RTCPeerConnection from the same end-point, thus forming a 1689 loop where the end-point will receive its own traffic. While it 1690 is clearly considered a bug, it is important that the end-point is 1691 able to recognise and handle the case when it occurs. This case 1692 becomes even more problematic when media mixers, and so on, are 1693 involved, where the stream received is a different stream but 1694 still contains this client's input. 1696 These SSRC/CSRC collisions can only be handled on RTP level as long 1697 as the same RTP session is extended across multiple 1698 RTCPeerConnections by a RTP middlebox. To resolve the more generic 1699 case where multiple RTCPeerConnections are interconnected, 1700 identification of the media source(s) part of a MediaStreamTrack 1701 being propagated across multiple interconnected RTCPeerConnection 1702 needs to be preserved across these interconnections. 1704 12.2.3. Media Synchronisation Context 1706 When an end-point sends media from more than one media source, it 1707 needs to consider if (and which of) these media sources are to be 1708 synchronized. In RTP/RTCP, synchronisation is provided by having a 1709 set of RTP packet streams be indicated as coming from the same 1710 synchronisation context and logical end-point by using the same RTCP 1711 CNAME identifier. 1713 The next provision is that the internal clocks of all media sources, 1714 i.e., what drives the RTP timestamp, can be correlated to a system 1715 clock that is provided in RTCP Sender Reports encoded in an NTP 1716 format. By correlating all RTP timestamps to a common system clock 1717 for all sources, the timing relation of the different RTP packet 1718 streams, also across multiple RTP sessions can be derived at the 1719 receiver and, if desired, the streams can be synchronized. The 1720 requirement is for the media sender to provide the correlation 1721 information; it is up to the receiver to use it or not. 1723 13. Security Considerations 1725 The overall security architecture for WebRTC is described in 1726 [I-D.ietf-rtcweb-security-arch], and security considerations for the 1727 WebRTC framework are described in [I-D.ietf-rtcweb-security]. These 1728 considerations also apply to this memo. 1730 The security considerations of the RTP specification, the RTP/SAVPF 1731 profile, and the various RTP/RTCP extensions and RTP payload formats 1732 that form the complete protocol suite described in this memo apply. 1733 It is not believed there are any new security considerations 1734 resulting from the combination of these various protocol extensions. 1736 The Extended Secure RTP Profile for Real-time Transport Control 1737 Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides 1738 handling of fundamental issues by offering confidentiality, integrity 1739 and partial source authentication. A mandatory to implement media 1740 security solution is created by combing this secured RTP profile and 1741 DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of 1742 [I-D.ietf-rtcweb-security-arch]. 1744 RTCP packets convey a Canonical Name (CNAME) identifier that is used 1745 to associate RTP packet streams that need to be synchronised across 1746 related RTP sessions. Inappropriate choice of CNAME values can be a 1747 privacy concern, since long-term persistent CNAME identifiers can be 1748 used to track users across multiple WebRTC calls. Section 4.9 of 1749 this memo provides guidelines for generation of untraceable CNAME 1750 values that alleviate this risk. 1752 Some potential denial of service attacks exist if the RTCP reporting 1753 interval is configured to an inappropriate value. This could be done 1754 by configuring the RTCP bandwidth fraction to an excessively large or 1755 small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some 1756 similar mechanism, or by choosing an excessively large or small value 1757 for the RTP/AVPF minimal receiver report interval (if using SDP, this 1758 is the "a=rtcp-fb:... trr-int" parameter) [RFC4585]. The risks are 1759 as follows: 1761 1. the RTCP bandwidth could be configured to make the regular 1762 reporting interval so large that effective congestion control 1763 cannot be maintained, potentially leading to denial of service 1764 due to congestion caused by the media traffic; 1766 2. the RTCP interval could be configured to a very small value, 1767 causing endpoints to generate high rate RTCP traffic, potentially 1768 leading to denial of service due to the non-congestion controlled 1769 RTCP traffic; and 1771 3. RTCP parameters could be configured differently for each 1772 endpoint, with some of the endpoints using a large reporting 1773 interval and some using a smaller interval, leading to denial of 1774 service due to premature participant timeouts due to mismatched 1775 timeout periods which are based on the reporting interval (this 1776 is a particular concern if endpoints use a small but non-zero 1777 value for the RTP/AVPF minimal receiver report interval (trr-int) 1778 [RFC4585], as discussed in Section 6.1 of 1779 [I-D.ietf-avtcore-rtp-multi-stream]). 1781 Premature participant timeout can be avoided by using the fixed (non- 1782 reduced) minimum interval when calculating the participant timeout 1783 (see Section 4.1 of this memo and Section 6.1 of 1784 [I-D.ietf-avtcore-rtp-multi-stream]). To address the other concerns, 1785 endpoints SHOULD ignore parameters that configure the RTCP reporting 1786 interval to be significantly longer than the default five second 1787 interval specified in [RFC3550] (unless the media data rate is so low 1788 that the longer reporting interval roughly corresponds to 5% of the 1789 media data rate), or that configure the RTCP reporting interval small 1790 enough that the RTCP bandwidth would exceed the media bandwidth. 1792 The guidelines in [RFC6562] apply when using variable bit rate (VBR) 1793 audio codecs such as Opus (see Section 4.3 for discussion of mandated 1794 audio codecs). The guidelines in [RFC6562] also apply, but are of 1795 lesser importance, when using the client-to-mixer audio level header 1796 extensions (Section 5.2.2) or the mixer-to-client audio level header 1797 extensions (Section 5.2.3). The use of the encryption of the header 1798 extensions are RECOMMENDED, unless there are known reasons, like RTP 1799 middleboxes or third party monitoring that will greatly benefit from 1800 the information, and this has been expressed using API or signalling. 1801 If further evidence are produced to show that information leakage is 1802 significant from audio level indications, then use of encryption 1803 needs to be mandated at that time. 1805 14. IANA Considerations 1807 This memo makes no request of IANA. 1809 Note to RFC Editor: this section is to be removed on publication as 1810 an RFC. 1812 15. Acknowledgements 1814 The authors would like to thank Bernard Aboba, Harald Alvestrand, 1815 Cary Bran, Ben Campbell, Charles Eckel, Alex Eleftheriadis, Christian 1816 Groves, Cullen Jennings, Olle Johansson, Suhas Nandakumar, Dan 1817 Romascanu, Jim Spring, Martin Thomson, and the other members of the 1818 IETF RTCWEB working group for their valuable feedback. 1820 16. References 1822 16.1. Normative References 1824 [I-D.ietf-avtcore-multi-media-rtp-session] 1825 Westerlund, M., Perkins, C., and J. Lennox, "Sending 1826 Multiple Types of Media in a Single RTP Session", draft- 1827 ietf-avtcore-multi-media-rtp-session-07 (work in 1828 progress), March 2015. 1830 [I-D.ietf-avtcore-rtp-circuit-breakers] 1831 Perkins, C. and V. Singh, "Multimedia Congestion Control: 1832 Circuit Breakers for Unicast RTP Sessions", draft-ietf- 1833 avtcore-rtp-circuit-breakers-10 (work in progress), March 1834 2015. 1836 [I-D.ietf-avtcore-rtp-multi-stream-optimisation] 1837 Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, 1838 "Sending Multiple Media Streams in a Single RTP Session: 1839 Grouping RTCP Reception Statistics and Other Feedback ", 1840 draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work 1841 in progress), July 2013. 1843 [I-D.ietf-avtcore-rtp-multi-stream] 1844 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, 1845 "Sending Multiple Media Streams in a Single RTP Session", 1846 draft-ietf-avtcore-rtp-multi-stream-07 (work in progress), 1847 March 2015. 1849 [I-D.ietf-mmusic-sdp-bundle-negotiation] 1850 Holmberg, C., Alvestrand, H., and C. Jennings, 1851 "Negotiating Media Multiplexing Using the Session 1852 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 1853 negotiation-19 (work in progress), March 2015. 1855 [I-D.ietf-rtcweb-audio] 1856 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 1857 Requirements", draft-ietf-rtcweb-audio-07 (work in 1858 progress), October 2014. 1860 [I-D.ietf-rtcweb-fec] 1861 Uberti, J., "WebRTC Forward Error Correction 1862 Requirements", draft-ietf-rtcweb-fec-01 (work in 1863 progress), March 2015. 1865 [I-D.ietf-rtcweb-security-arch] 1866 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 1867 rtcweb-security-arch-11 (work in progress), March 2015. 1869 [I-D.ietf-rtcweb-security] 1870 Rescorla, E., "Security Considerations for WebRTC", draft- 1871 ietf-rtcweb-security-08 (work in progress), February 2015. 1873 [I-D.ietf-rtcweb-video] 1874 Roach, A., "WebRTC Video Processing and Codec 1875 Requirements", draft-ietf-rtcweb-video-05 (work in 1876 progress), March 2015. 1878 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1879 Requirement Levels", BCP 14, RFC 2119, March 1997. 1881 [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP 1882 Payload Format Specifications", BCP 36, RFC 2736, December 1883 1999. 1885 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1886 Jacobson, "RTP: A Transport Protocol for Real-Time 1887 Applications", STD 64, RFC 3550, July 2003. 1889 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 1890 Video Conferences with Minimal Control", STD 65, RFC 3551, 1891 July 2003. 1893 [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth 1894 Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 1895 3556, July 2003. 1897 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 1898 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 1899 RFC 3711, March 2004. 1901 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1902 Description Protocol", RFC 4566, July 2006. 1904 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 1905 "Extended RTP Profile for Real-time Transport Control 1906 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 1907 2006. 1909 [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. 1910 Hakenberg, "RTP Retransmission Payload Format", RFC 4588, 1911 July 2006. 1913 [RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", 1914 BCP 131, RFC 4961, July 2007. 1916 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 1917 "Codec Control Messages in the RTP Audio-Visual Profile 1918 with Feedback (AVPF)", RFC 5104, February 2008. 1920 [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for 1921 Real-time Transport Control Protocol (RTCP)-Based Feedback 1922 (RTP/SAVPF)", RFC 5124, February 2008. 1924 [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP 1925 Header Extensions", RFC 5285, July 2008. 1927 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size 1928 Real-Time Transport Control Protocol (RTCP): Opportunities 1929 and Consequences", RFC 5506, April 2009. 1931 [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and 1932 Control Packets on a Single Port", RFC 5761, April 2010. 1934 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1935 Security (DTLS) Extension to Establish Keys for the Secure 1936 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 1938 [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP 1939 Flows", RFC 6051, November 2010. 1941 [RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time 1942 Transport Protocol (RTP) Header Extension for Client-to- 1943 Mixer Audio Level Indication", RFC 6464, December 2011. 1945 [RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time 1946 Transport Protocol (RTP) Header Extension for Mixer-to- 1947 Client Audio Level Indication", RFC 6465, December 2011. 1949 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of 1950 Variable Bit Rate Audio with Secure RTP", RFC 6562, March 1951 2012. 1953 [RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure 1954 Real-time Transport Protocol (SRTP)", RFC 6904, April 1955 2013. 1957 [RFC7007] Terriberry, T., "Update to Remove DVI4 from the 1958 Recommended Codecs for the RTP Profile for Audio and Video 1959 Conferences with Minimal Control (RTP/AVP)", RFC 7007, 1960 August 2013. 1962 [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, 1963 "Guidelines for Choosing RTP Control Protocol (RTCP) 1964 Canonical Names (CNAMEs)", RFC 7022, September 2013. 1966 [RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple 1967 Clock Rates in an RTP Session", RFC 7160, April 2014. 1969 [RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC 1970 7164, March 2014. 1972 16.2. Informative References 1974 [I-D.ietf-avtcore-multiplex-guidelines] 1975 Westerlund, M., Perkins, C., and H. Alvestrand, 1976 "Guidelines for using the Multiplexing Features of RTP to 1977 Support Multiple Media Streams", draft-ietf-avtcore- 1978 multiplex-guidelines-03 (work in progress), October 2014. 1980 [I-D.ietf-avtcore-rtp-topologies-update] 1981 Westerlund, M. and S. Wenger, "RTP Topologies", draft- 1982 ietf-avtcore-rtp-topologies-update-06 (work in progress), 1983 March 2015. 1985 [I-D.ietf-avtext-rtp-grouping-taxonomy] 1986 Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro, 1987 "A Taxonomy of Grouping Semantics and Mechanisms for Real- 1988 Time Transport Protocol (RTP) Sources", draft-ietf-avtext- 1989 rtp-grouping-taxonomy-06 (work in progress), March 2015. 1991 [I-D.ietf-mmusic-msid] 1992 Alvestrand, H., "WebRTC MediaStream Identification in the 1993 Session Description Protocol", draft-ietf-mmusic-msid-08 1994 (work in progress), February 2015. 1996 [I-D.ietf-mmusic-sdp-bundle-negotiation] 1997 Holmberg, C., Alvestrand, H., and C. Jennings, 1998 "Negotiating Media Multiplexing Using the Session 1999 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 2000 negotiation-19 (work in progress), March 2015. 2002 [I-D.ietf-payload-rtp-howto] 2003 Westerlund, M., "How to Write an RTP Payload Format", 2004 draft-ietf-payload-rtp-howto-13 (work in progress), 2005 January 2014. 2007 [I-D.ietf-rmcat-cc-requirements] 2008 Jesup, R. and Z. Sarker, "Congestion Control Requirements 2009 for Interactive Real-Time Media", draft-ietf-rmcat-cc- 2010 requirements-09 (work in progress), December 2014. 2012 [I-D.ietf-rtcweb-overview] 2013 Alvestrand, H., "Overview: Real Time Protocols for 2014 Browser-based Applications", draft-ietf-rtcweb-overview-13 2015 (work in progress), November 2014. 2017 [I-D.ietf-rtcweb-use-cases-and-requirements] 2018 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 2019 Time Communication Use-cases and Requirements", draft- 2020 ietf-rtcweb-use-cases-and-requirements-16 (work in 2021 progress), January 2015. 2023 [I-D.ietf-tsvwg-rtcweb-qos] 2024 Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. 2025 Polk, "DSCP and other packet markings for RTCWeb QoS", 2026 draft-ietf-tsvwg-rtcweb-qos-03 (work in progress), 2027 November 2014. 2029 [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control 2030 Protocol Extended Reports (RTCP XR)", RFC 3611, November 2031 2003. 2033 [RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient 2034 Stream Loss-Tolerant Authentication (TESLA) in the Secure 2035 Real-time Transport Protocol (SRTP)", RFC 4383, February 2036 2006. 2038 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 2039 (ICE): A Protocol for Network Address Translator (NAT) 2040 Traversal for Offer/Answer Protocols", RFC 5245, April 2041 2010. 2043 [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific 2044 Media Attributes in the Session Description Protocol 2045 (SDP)", RFC 5576, June 2009. 2047 [RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP 2048 Control Protocol (RTCP)", RFC 5968, September 2010. 2050 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 2051 Keeping Alive the NAT Mappings Associated with RTP / RTP 2052 Control Protocol (RTCP) Flows", RFC 6263, June 2011. 2054 [RFC6792] Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the 2055 RTP Monitoring Framework", RFC 6792, November 2012. 2057 [W3C.WD-mediacapture-streams-20130903] 2058 Burnett, D., Bergkvist, A., Jennings, C., and A. 2059 Narayanan, "Media Capture and Streams", World Wide Web 2060 Consortium WD WD-mediacapture-streams-20130903, September 2061 2013, . 2064 [W3C.WD-webrtc-20130910] 2065 Bergkvist, A., Burnett, D., Jennings, C., and A. 2066 Narayanan, "WebRTC 1.0: Real-time Communication Between 2067 Browsers", World Wide Web Consortium WD WD- 2068 webrtc-20130910, September 2013, 2069 . 2071 Authors' Addresses 2073 Colin Perkins 2074 University of Glasgow 2075 School of Computing Science 2076 Glasgow G12 8QQ 2077 United Kingdom 2079 Email: csp@csperkins.org 2080 URI: http://csperkins.org/ 2082 Magnus Westerlund 2083 Ericsson 2084 Farogatan 6 2085 SE-164 80 Kista 2086 Sweden 2088 Phone: +46 10 714 82 87 2089 Email: magnus.westerlund@ericsson.com 2091 Joerg Ott 2092 Aalto University 2093 School of Electrical Engineering 2094 Espoo 02150 2095 Finland 2097 Email: jorg.ott@aalto.fi