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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: April 30, 2015 Ericsson 6 J. Ott 7 Aalto University 8 October 27, 2014 10 Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 11 draft-ietf-rtcweb-rtp-usage-19 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 April 30, 2015. 40 Copyright Notice 42 Copyright (c) 2014 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 . . . . . . . . . . . . . . . . . . . 9 65 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10 66 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10 67 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 11 68 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 11 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) . . . . . . . . . . . . . 15 76 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 15 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 . . . . . . . . . . . . . . . . . . . . 16 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 . . . . . . . . . . . . . 17 84 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 18 85 6.1. Negative Acknowledgements and RTP Retransmission . . . . 18 86 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 19 87 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 19 88 7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 20 89 7.2. Congestion Control Interoperability and Legacy Systems . 21 90 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 22 91 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 22 92 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 22 93 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 24 94 12. RTP Implementation Considerations . . . . . . . . . . . . . . 26 95 12.1. Configuration and Use of RTP Sessions . . . . . . . . . 26 96 12.1.1. Use of Multiple Media Sources Within an RTP Session 26 97 12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 28 98 12.1.3. Differentiated Treatment of RTP Packet Streams . . . 32 99 12.2. Media Source, RTP Packet Streams, and Participant 100 Identification . . . . . . . . . . . . . . . . . . . . . 34 101 12.2.1. Media Source Identification . . . . . . . . . . . . 34 102 12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 35 103 12.2.3. Media Synchronisation Context . . . . . . . . . . . 36 104 13. Security Considerations . . . . . . . . . . . . . . . . . . . 36 105 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 38 106 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 38 107 16. References . . . . . . . . . . . . . . . . . . . . . . . . . 38 108 16.1. Normative References . . . . . . . . . . . . . . . . . . 38 109 16.2. Informative References . . . . . . . . . . . . . . . . . 41 110 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 43 112 1. Introduction 114 The Real-time Transport Protocol (RTP) [RFC3550] provides a framework 115 for delivery of audio and video teleconferencing data and other real- 116 time media applications. Previous work has defined the RTP protocol, 117 along with numerous profiles, payload formats, and other extensions. 118 When combined with appropriate signalling, these form the basis for 119 many teleconferencing systems. 121 The Web Real-Time communication (WebRTC) framework provides the 122 protocol building blocks to support direct, interactive, real-time 123 communication using audio, video, collaboration, games, etc., between 124 two peers' web-browsers. This memo describes how the RTP framework 125 is to be used in the WebRTC context. It proposes a baseline set of 126 RTP features that are to be implemented by all WebRTC Endpoints, 127 along with suggested extensions for enhanced functionality. 129 This memo specifies a protocol intended for use within the WebRTC 130 framework, but is not restricted to that context. An overview of the 131 WebRTC framework is given in [I-D.ietf-rtcweb-overview]. 133 The structure of this memo is as follows. Section 2 outlines our 134 rationale in preparing this memo and choosing these RTP features. 135 Section 3 defines terminology. Requirements for core RTP protocols 136 are described in Section 4 and suggested RTP extensions are described 137 in Section 5. Section 6 outlines mechanisms that can increase 138 robustness to network problems, while Section 7 describes congestion 139 control and rate adaptation mechanisms. The discussion of mandated 140 RTP mechanisms concludes in Section 8 with a review of performance 141 monitoring and network management tools. Section 9 gives some 142 guidelines for future incorporation of other RTP and RTP Control 143 Protocol (RTCP) extensions into this framework. Section 10 describes 144 requirements placed on the signalling channel. Section 11 discusses 145 the relationship between features of the RTP framework and the WebRTC 146 application programming interface (API), and Section 12 discusses RTP 147 implementation considerations. The memo concludes with security 148 considerations (Section 13) and IANA considerations (Section 14). 150 2. Rationale 152 The RTP framework comprises the RTP data transfer protocol, the RTP 153 control protocol, and numerous RTP payload formats, profiles, and 154 extensions. This range of add-ons has allowed RTP to meet various 155 needs that were not envisaged by the original protocol designers, and 156 to support many new media encodings, but raises the question of what 157 extensions are to be supported by new implementations. The 158 development of the WebRTC framework provides an opportunity to review 159 the available RTP features and extensions, and to define a common 160 baseline RTP feature set for all WebRTC Endpoints. This builds on 161 the past 20 years development of RTP to mandate the use of extensions 162 that have shown widespread utility, while still remaining compatible 163 with the wide installed base of RTP implementations where possible. 165 RTP and RTCP extensions that are not discussed in this document can 166 be implemented by WebRTC Endpoints if they are beneficial for new use 167 cases. However, they are not necessary to address the WebRTC use 168 cases and requirements identified in 169 [I-D.ietf-rtcweb-use-cases-and-requirements]. 171 While the baseline set of RTP features and extensions defined in this 172 memo is targeted at the requirements of the WebRTC framework, it is 173 expected to be broadly useful for other conferencing-related uses of 174 RTP. In particular, it is likely that this set of RTP features and 175 extensions will be appropriate for other desktop or mobile video 176 conferencing systems, or for room-based high-quality telepresence 177 applications. 179 3. Terminology 181 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 182 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 183 document are to be interpreted as described in [RFC2119]. The RFC 184 2119 interpretation of these key words applies only when written in 185 ALL CAPS. Lower- or mixed-case uses of these key words are not to be 186 interpreted as carrying special significance in this memo. 188 We define the following additional terms: 190 WebRTC MediaStream: The MediaStream concept defined by the W3C in 191 the WebRTC API [W3C.WD-mediacapture-streams-20130903]. 193 Transport-layer Flow: A uni-directional flow of transport packets 194 that are identified by having a particular 5-tuple of source IP 195 address, source port, destination IP address, destination port, 196 and transport protocol used. 198 Bi-directional Transport-layer Flow: A bi-directional transport- 199 layer flow is a transport-layer flow that is symmetric. That is, 200 the transport-layer flow in the reverse direction has a 5-tuple 201 where the source and destination address and ports are swapped 202 compared to the forward path transport-layer flow, and the 203 transport protocol is the same. 205 This document uses the terminology from 206 [I-D.ietf-avtext-rtp-grouping-taxonomy] and 207 [I-D.ietf-rtcweb-overview]. Other terms are used according to their 208 definitions from the RTP Specification [RFC3550]. Especially note 209 the following frequently used terms: RTP Packet Stream, RTP Session, 210 and End-point. 212 4. WebRTC Use of RTP: Core Protocols 214 The following sections describe the core features of RTP and RTCP 215 that need to be implemented, along with the mandated RTP profiles. 216 Also described are the core extensions providing essential features 217 that all WebRTC Endpoints need to implement to function effectively 218 on today's networks. 220 4.1. RTP and RTCP 222 The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be 223 implemented as the media transport protocol for WebRTC. RTP itself 224 comprises two parts: the RTP data transfer protocol, and the RTP 225 control protocol (RTCP). RTCP is a fundamental and integral part of 226 RTP, and MUST be implemented and used in all WebRTC Endpoints. 228 The following RTP and RTCP features are sometimes omitted in limited 229 functionality implementations of RTP, but are REQUIRED in all WebRTC 230 Endpoints: 232 o Support for use of multiple simultaneous SSRC values in a single 233 RTP session, including support for RTP end-points that send many 234 SSRC values simultaneously, following [RFC3550] and 235 [I-D.ietf-avtcore-rtp-multi-stream]. The RTCP optimisations for 236 multi-SSRC sessions defined in 237 [I-D.ietf-avtcore-rtp-multi-stream-optimisation] MAY be supported; 238 if supported the usage MUST be signalled. 240 o Random choice of SSRC on joining a session; collision detection 241 and resolution for SSRC values (see also Section 4.8). 243 o Support for reception of RTP data packets containing CSRC lists, 244 as generated by RTP mixers, and RTCP packets relating to CSRCs. 246 o Sending correct synchronisation information in the RTCP Sender 247 Reports, to allow receivers to implement lip-synchronisation; see 248 Section 5.2.1 regarding support for the rapid RTP synchronisation 249 extensions. 251 o Support for multiple synchronisation contexts. Participants that 252 send multiple simultaneous RTP packet streams SHOULD do so as part 253 of a single synchronisation context, using a single RTCP CNAME for 254 all streams and allowing receivers to play the streams out in a 255 synchronised manner. For compatibility with potential future 256 versions of this specification, or for interoperability with non- 257 WebRTC devices through a gateway, receivers MUST support multiple 258 synchronisation contexts, indicated by the use of multiple RTCP 259 CNAMEs in an RTP session. This specification requires the usage 260 of a single CNAME when sending RTP Packet Streams in some 261 circumstances, see Section 4.9. 263 o Support for sending and receiving RTCP SR, RR, SDES, and BYE 264 packet types, with OPTIONAL support for other RTCP packet types 265 unless mandated by other parts of this specification. Note that 266 additional RTCP Packet types are used by the RTP/SAVPF Profile 267 (Section 4.2) and the other RTCP extensions (Section 5). 269 o Support for multiple end-points in a single RTP session, and for 270 scaling the RTCP transmission interval according to the number of 271 participants in the session; support for randomised RTCP 272 transmission intervals to avoid synchronisation of RTCP reports; 273 support for RTCP timer reconsideration (Section 6.3.6 of 274 [RFC3550]) and reverse reconsideration (Section 6.3.4 of 275 [RFC3550]). 277 o Support for configuring the RTCP bandwidth as a fraction of the 278 media bandwidth, and for configuring the fraction of the RTCP 279 bandwidth allocated to senders, e.g., using the SDP "b=" line 280 [RFC4566][RFC3556]. 282 o Support for the reduced minimum RTCP reporting interval described 283 in Section 6.2 of [RFC3550] is REQUIRED. When using the reduced 284 minimum RTCP reporting interval, the fixed (non-reduced) minimum 285 interval MUST be used when calculating the participant timeout 286 interval (see Sections 6.2 and 6.3.5 of [RFC3550]). The delay 287 before sending the initial compound RTCP packet can be set to zero 288 (see Section 6.2 of [RFC3550] as updated by 289 [I-D.ietf-avtcore-rtp-multi-stream]). 291 o Support for discontinuous transmission. RTP allows endpoints to 292 pause and resume transmission at any time. When resuming, the RTP 293 sequence number will increase by one, as usual, while the increase 294 in the RTP timestamp value will depend on the duration of the 295 pause. Discontinuous transmission is most commonly used with some 296 audio payload formats, but is not audio specific, and can be used 297 with any RTP payload format. 299 o Ignore unknown RTCP packet types and RTP header extensions. This 300 to ensure robust handling of future extensions, middlebox 301 behaviours, etc., that can result in not signalled RTCP packet 302 types or RTP header extensions being received. If a compound RTCP 303 packet is received that contains a mixture of known and unknown 304 RTCP packet types, the known packets types need to be processed as 305 usual, with only the unknown packet types being discarded. 307 It is known that a significant number of legacy RTP implementations, 308 especially those targeted at VoIP-only systems, do not support all of 309 the above features, and in some cases do not support RTCP at all. 310 Implementers are advised to consider the requirements for graceful 311 degradation when interoperating with legacy implementations. 313 Other implementation considerations are discussed in Section 12. 315 4.2. Choice of the RTP Profile 317 The complete specification of RTP for a particular application domain 318 requires the choice of an RTP Profile. For WebRTC use, the Extended 319 Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF) [RFC5124], as 320 extended by [RFC7007], MUST be implemented. The RTP/SAVPF profile is 321 the combination of basic RTP/AVP profile [RFC3551], the RTP profile 322 for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP 323 profile (RTP/SAVP) [RFC3711]. 325 The RTCP-based feedback extensions [RFC4585] are needed for the 326 improved RTCP timer model. This allows more flexible transmission of 327 RTCP packets in response to events, rather than strictly according to 328 bandwidth, and is vital for being able to report congestion signals 329 as well as media events. These extensions also allow saving RTCP 330 bandwidth, and an end-point will commonly only use the full RTCP 331 bandwidth allocation if there are many events that require feedback. 332 The timer rules are also needed to make use of the RTP conferencing 333 extensions discussed in Section 5.1. 335 Note: The enhanced RTCP timer model defined in the RTP/AVPF 336 profile is backwards compatible with legacy systems that implement 337 only the RTP/AVP or RTP/SAVP profile, given some constraints on 338 parameter configuration such as the RTCP bandwidth value and "trr- 339 int" (the most important factor for interworking with RTP/(S)AVP 340 end-points via a gateway is to set the trr-int parameter to a 341 value representing 4 seconds, see Section 6.1 in 342 [I-D.ietf-avtcore-rtp-multi-stream]). 344 The secure RTP (SRTP) profile extensions [RFC3711] are needed to 345 provide media encryption, integrity protection, replay protection and 346 a limited form of source authentication. WebRTC Endpoints MUST NOT 347 send packets using the basic RTP/AVP profile or the RTP/AVPF profile; 348 they MUST employ the full RTP/SAVPF profile to protect all RTP and 349 RTCP packets that are generated (i.e., implementations MUST use SRTP 350 and SRTCP). The RTP/SAVPF profile MUST be configured using the 351 cipher suites, DTLS-SRTP protection profiles, keying mechanisms, and 352 other parameters described in [I-D.ietf-rtcweb-security-arch]. 354 4.3. Choice of RTP Payload Formats 356 The set of mandatory to implement codecs and RTP payload formats for 357 WebRTC is not specified in this memo, instead they are defined in 358 separate specifications, such as [I-D.ietf-rtcweb-audio]. 359 Implementations can support any codec for which an RTP payload format 360 and associated signalling is defined. Implementation cannot assume 361 that the other participants in an RTP session understand any RTP 362 payload format, no matter how common; the mapping between RTP payload 363 type numbers and specific configurations of particular RTP payload 364 formats MUST be agreed before those payload types/formats can be 365 used. In an SDP context, this can be done using the "a=rtpmap:" and 366 "a=fmtp:" attributes associated with an "m=" line, along with any 367 other SDP attributes needed to configure the RTP payload format. 369 End-points can signal support for multiple RTP payload formats, or 370 multiple configurations of a single RTP payload format, as long as 371 each unique RTP payload format configuration uses a different RTP 372 payload type number. As outlined in Section 4.8, the RTP payload 373 type number is sometimes used to associate an RTP packet stream with 374 a signalling context. This association is possible provided unique 375 RTP payload type numbers are used in each context. For example, an 376 RTP packet stream can be associated with an SDP "m=" line by 377 comparing the RTP payload type numbers used by the RTP packet stream 378 with payload types signalled in the "a=rtpmap:" lines in the media 379 sections of the SDP. This leads to the following considerations: 381 If RTP packet streams are being associated with signalling 382 contexts based on the RTP payload type, then the assignment of RTP 383 payload type numbers MUST be unique across signalling contexts. 385 If the same RTP payload format configuration is used in multiple 386 contexts, then a different RTP payload type number has to be 387 assigned in each context to ensure uniqueness. 389 If the RTP payload type number is not being used to associate RTP 390 packet streams with a signalling context, then the same RTP 391 payload type number can be used to indicate the exact same RTP 392 payload format configuration in multiple contexts. 394 A single RTP payload type number MUST NOT be assigned to different 395 RTP payload formats, or different configurations of the same RTP 396 payload format, within a single RTP session (note that the "m=" lines 397 in an SDP bundle group [I-D.ietf-mmusic-sdp-bundle-negotiation] form 398 a single RTP session). 400 An end-point that has signalled support for multiple RTP payload 401 formats MUST be able to accept data in any of those payload formats 402 at any time, unless it has previously signalled limitations on its 403 decoding capability. This requirement is constrained if several 404 types of media (e.g., audio and video) are sent in the same RTP 405 session. In such a case, a source (SSRC) is restricted to switching 406 only between the RTP payload formats signalled for the type of media 407 that is being sent by that source; see Section 4.4. To support rapid 408 rate adaptation by changing codec, RTP does not require advance 409 signalling for changes between RTP payload formats used by a single 410 SSRC that were signalled during session set-up. 412 If performing changes between two RTP payload types that use 413 different RTP clock rates, an RTP sender MUST follow the 414 recommendations in Section 4.1 of [RFC7160]. RTP receivers MUST 415 follow the recommendations in Section 4.3 of [RFC7160] in order to 416 support sources that switch between clock rates in an RTP session 417 (these recommendations for receivers are backwards compatible with 418 the case where senders use only a single clock rate). 420 4.4. Use of RTP Sessions 422 An association amongst a set of end-points communicating using RTP is 423 known as an RTP session [RFC3550]. An end-point can be involved in 424 several RTP sessions at the same time. In a multimedia session, each 425 type of media has typically been carried in a separate RTP session 426 (e.g., using one RTP session for the audio, and a separate RTP 427 session using a different transport-layer flow for the video). 428 WebRTC Endpoints are REQUIRED to implement support for multimedia 429 sessions in this way, separating each RTP session using different 430 transport-layer flows for compatibility with legacy systems (this is 431 sometimes called session multiplexing). 433 In modern day networks, however, with the widespread use of network 434 address/port translators (NAT/NAPT) and firewalls, it is desirable to 435 reduce the number of transport-layer flows used by RTP applications. 436 This can be done by sending all the RTP packet streams in a single 437 RTP session, which will comprise a single transport-layer flow (this 438 will prevent the use of some quality-of-service mechanisms, as 439 discussed in Section 12.1.3). Implementations are therefore also 440 REQUIRED to support transport of all RTP packet streams, independent 441 of media type, in a single RTP session using a single transport layer 442 flow, according to [I-D.ietf-avtcore-multi-media-rtp-session] (this 443 is sometimes called SSRC multiplexing). If multiple types of media 444 are to be used in a single RTP session, all participants in that RTP 445 session MUST agree to this usage. In an SDP context, 446 [I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to signal such a 447 bundle of RTP packet streams forming a single RTP session. 449 Further discussion about the suitability of different RTP session 450 structures and multiplexing methods to different scenarios can be 451 found in [I-D.ietf-avtcore-multiplex-guidelines]. 453 4.5. RTP and RTCP Multiplexing 455 Historically, RTP and RTCP have been run on separate transport layer 456 flows (e.g., two UDP ports for each RTP session, one port for RTP and 457 one port for RTCP). With the increased use of Network Address/Port 458 Translation (NAT/NAPT) this has become problematic, since maintaining 459 multiple NAT bindings can be costly. It also complicates firewall 460 administration, since multiple ports need to be opened to allow RTP 461 traffic. To reduce these costs and session set-up times, 462 implementations are REQUIRED to support multiplexing RTP data packets 463 and RTCP control packets on a single transport-layer flow [RFC5761]. 464 Such RTP and RTCP multiplexing MUST be negotiated in the signalling 465 channel before it is used. If SDP is used for signalling, this 466 negotiation MUST use the attributes defined in [RFC5761]. For 467 backwards compatibility, implementations are also REQUIRED to support 468 RTP and RTCP sent on separate transport-layer flows. 470 Note that the use of RTP and RTCP multiplexed onto a single 471 transport-layer flow ensures that there is occasional traffic sent on 472 that port, even if there is no active media traffic. This can be 473 useful to keep NAT bindings alive [RFC6263]. 475 4.6. Reduced Size RTCP 476 RTCP packets are usually sent as compound RTCP packets, and [RFC3550] 477 requires that those compound packets start with an Sender Report (SR) 478 or Receiver Report (RR) packet. When using frequent RTCP feedback 479 messages under the RTP/AVPF Profile [RFC4585] these statistics are 480 not needed in every packet, and unnecessarily increase the mean RTCP 481 packet size. This can limit the frequency at which RTCP packets can 482 be sent within the RTCP bandwidth share. 484 To avoid this problem, [RFC5506] specifies how to reduce the mean 485 RTCP message size and allow for more frequent feedback. Frequent 486 feedback, in turn, is essential to make real-time applications 487 quickly aware of changing network conditions, and to allow them to 488 adapt their transmission and encoding behaviour. Implementations 489 MUST support sending and receiving non-compound RTCP feedback packets 490 [RFC5506]. Use of non-compound RTCP packets MUST be negotiated using 491 the signalling channel. If SDP is used for signalling, this 492 negotiation MUST use the attributes defined in [RFC5506]. For 493 backwards compatibility, implementations are also REQUIRED to support 494 the use of compound RTCP feedback packets if the remote end-point 495 does not agree to the use of non-compound RTCP in the signalling 496 exchange. 498 4.7. Symmetric RTP/RTCP 500 To ease traversal of NAT and firewall devices, implementations are 501 REQUIRED to implement and use Symmetric RTP [RFC4961]. The reason 502 for using symmetric RTP is primarily to avoid issues with NATs and 503 Firewalls by ensuring that the send and receive RTP packet streams, 504 as well as RTCP, are actually bi-directional transport-layer flows. 505 This will keep alive the NAT and firewall pinholes, and help indicate 506 consent that the receive direction is a transport-layer flow the 507 intended recipient actually wants. In addition, it saves resources, 508 specifically ports at the end-points, but also in the network as NAT 509 mappings or firewall state is not unnecessary bloated. The amount of 510 per flow QoS state kept in the network is also reduced. 512 4.8. Choice of RTP Synchronisation Source (SSRC) 514 Implementations are REQUIRED to support signalled RTP synchronisation 515 source (SSRC) identifiers. If SDP is used, this MUST be done using 516 the "a=ssrc:" SDP attribute defined in Section 4.1 and Section 5 of 517 [RFC5576] and the "previous-ssrc" source attribute defined in 518 Section 6.2 of [RFC5576]; other per-SSRC attributes defined in 519 [RFC5576] MAY be supported. 521 While support for signalled SSRC identifiers is mandated, their use 522 in an RTP session is OPTIONAL. Implementations MUST be prepared to 523 accept RTP and RTCP packets using SSRCs that have not been explicitly 524 signalled ahead of time. Implementations MUST support random SSRC 525 assignment, and MUST support SSRC collision detection and resolution, 526 according to [RFC3550]. When using signalled SSRC values, collision 527 detection MUST be performed as described in Section 5 of [RFC5576]. 529 It is often desirable to associate an RTP packet stream with a non- 530 RTP context. For users of the WebRTC API a mapping between SSRCs and 531 MediaStreamTracks are provided per Section 11. For gateways or other 532 usages it is possible to associate an RTP packet stream with an "m=" 533 line in a session description formatted using SDP. If SSRCs are 534 signalled this is straightforward (in SDP the "a=ssrc:" line will be 535 at the media level, allowing a direct association with an "m=" line). 536 If SSRCs are not signalled, the RTP payload type numbers used in an 537 RTP packet stream are often sufficient to associate that packet 538 stream with a signalling context (e.g., if RTP payload type numbers 539 are assigned as described in Section 4.3 of this memo, the RTP 540 payload types used by an RTP packet stream can be compared with 541 values in SDP "a=rtpmap:" lines, which are at the media level in SDP, 542 and so map to an "m=" line). 544 4.9. Generation of the RTCP Canonical Name (CNAME) 546 The RTCP Canonical Name (CNAME) provides a persistent transport-level 547 identifier for an RTP end-point. While the Synchronisation Source 548 (SSRC) identifier for an RTP end-point can change if a collision is 549 detected, or when the RTP application is restarted, its RTCP CNAME is 550 meant to stay unchanged for the duration of a RTCPeerConnection 551 [W3C.WD-webrtc-20130910], so that RTP end-points can be uniquely 552 identified and associated with their RTP packet streams within a set 553 of related RTP sessions. 555 Each RTP end-point MUST have at least one RTCP CNAME, and that RTCP 556 CNAME MUST be unique within the RTCPeerConnection. RTCP CNAMEs 557 identify a particular synchronisation context, i.e., all SSRCs 558 associated with a single RTCP CNAME share a common reference clock. 559 If an end-point has SSRCs that are associated with several 560 unsynchronised reference clocks, and hence different synchronisation 561 contexts, it will need to use multiple RTCP CNAMEs, one for each 562 synchronisation context. 564 Taking the discussion in Section 11 into account, a WebRTC Endpoint 565 MUST NOT use more than one RTCP CNAME in the RTP sessions belonging 566 to single RTCPeerConnection (that is, an RTCPeerConnection forms a 567 synchronisation context). RTP middleboxes MAY generate RTP packet 568 streams associated with more than one RTCP CNAME, to allow them to 569 avoid having to resynchronize media from multiple different end- 570 points part of a multi-party RTP session. 572 The RTP specification [RFC3550] includes guidelines for choosing a 573 unique RTP CNAME, but these are not sufficient in the presence of NAT 574 devices. In addition, long-term persistent identifiers can be 575 problematic from a privacy viewpoint (Section 13). Accordingly, a 576 WebRTC Endpoint MUST generate a new, unique, short-term persistent 577 RTCP CNAME for each RTCPeerConnection, following [RFC7022], with a 578 single exception; if explicitly requested at creation an 579 RTCPeerConnection MAY use the same CNAME as as an existing 580 RTCPeerConnection within their common same-origin context. 582 An WebRTC Endpoint MUST support reception of any CNAME that matches 583 the syntax limitations specified by the RTP specification [RFC3550] 584 and cannot assume that any CNAME will be chosen according to the form 585 suggested above. 587 4.10. Handling of Leap Seconds 589 The guidelines regarding handling of leap seconds to limit their 590 impact on RTP media play-out and synchronization given in [RFC7164] 591 SHOULD be followed. 593 5. WebRTC Use of RTP: Extensions 595 There are a number of RTP extensions that are either needed to obtain 596 full functionality, or extremely useful to improve on the baseline 597 performance, in the WebRTC context. One set of these extensions is 598 related to conferencing, while others are more generic in nature. 599 The following subsections describe the various RTP extensions 600 mandated or suggested for use within WebRTC. 602 5.1. Conferencing Extensions and Topologies 604 RTP is a protocol that inherently supports group communication. 605 Groups can be implemented by having each endpoint send its RTP packet 606 streams to an RTP middlebox that redistributes the traffic, by using 607 a mesh of unicast RTP packet streams between endpoints, or by using 608 an IP multicast group to distribute the RTP packet streams. These 609 topologies can be implemented in a number of ways as discussed in 610 [I-D.ietf-avtcore-rtp-topologies-update]. 612 While the use of IP multicast groups is popular in IPTV systems, the 613 topologies based on RTP middleboxes are dominant in interactive video 614 conferencing environments. Topologies based on a mesh of unicast 615 transport-layer flows to create a common RTP session have not seen 616 widespread deployment to date. Accordingly, WebRTC Endpoints are not 617 expected to support topologies based on IP multicast groups or to 618 support mesh-based topologies, such as a point-to-multipoint mesh 619 configured as a single RTP session (Topo-Mesh in the terminology of 621 [I-D.ietf-avtcore-rtp-topologies-update]). However, a point-to- 622 multipoint mesh constructed using several RTP sessions, implemented 623 in WebRTC using independent RTCPeerConnections 624 [W3C.WD-webrtc-20130910], can be expected to be used in WebRTC, and 625 needs to be supported. 627 WebRTC Endpoints implemented according to this memo are expected to 628 support all the topologies described in 629 [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send 630 and receive unicast RTP packet streams to and from some peer device, 631 provided that peer can participate in performing congestion control 632 on the RTP packet streams. The peer device could be another RTP 633 endpoint, or it could be an RTP middlebox that redistributes the RTP 634 packet streams to other RTP endpoints. This limitation means that 635 some of the RTP middlebox-based topologies are not suitable for use 636 in WebRTC. Specifically: 638 o Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used, 639 since they make the use of RTCP for congestion control and quality 640 of service reports problematic (see Section 3.8 of 641 [I-D.ietf-avtcore-rtp-topologies-update]). 643 o The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology 644 SHOULD NOT be used because its safe use requires a congestion 645 control algorithm or RTP circuit breaker that handles point to 646 multipoint, which has not yet been standardised. 648 The following topology can be used, however it has some issues worth 649 noting: 651 o Content modifying MCUs with RTCP termination (Topo-RTCP- 652 terminating-MCU) MAY be used. Note that in this RTP Topology, RTP 653 loop detection and identification of active senders is the 654 responsibility of the WebRTC application; since the clients are 655 isolated from each other at the RTP layer, RTP cannot assist with 656 these functions (see section 3.9 of 657 [I-D.ietf-avtcore-rtp-topologies-update]). 659 The RTP extensions described in Section 5.1.1 to Section 5.1.6 are 660 designed to be used with centralised conferencing, where an RTP 661 middlebox (e.g., a conference bridge) receives a participant's RTP 662 packet streams and distributes them to the other participants. These 663 extensions are not necessary for interoperability; an RTP end-point 664 that does not implement these extensions will work correctly, but 665 might offer poor performance. Support for the listed extensions will 666 greatly improve the quality of experience and, to provide a 667 reasonable baseline quality, some of these extensions are mandatory 668 to be supported by WebRTC Endpoints. 670 The RTCP conferencing extensions are defined in Extended RTP Profile 671 for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/ 672 AVPF) [RFC4585] and the memo on Codec Control Messages (CCM) in RTP/ 673 AVPF [RFC5104]; they are fully usable by the Secure variant of this 674 profile (RTP/SAVPF) [RFC5124]. 676 5.1.1. Full Intra Request (FIR) 678 The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1 679 of the Codec Control Messages [RFC5104]. It is used to make the 680 mixer request a new Intra picture from a participant in the session. 681 This is used when switching between sources to ensure that the 682 receivers can decode the video or other predictive media encoding 683 with long prediction chains. WebRTC Endpoints that are sending media 684 MUST understand and react to FIR feedback messages they receive, 685 since this greatly improves the user experience when using 686 centralised mixer-based conferencing. Support for sending FIR 687 messages is OPTIONAL. 689 5.1.2. Picture Loss Indication (PLI) 691 The Picture Loss Indication message is defined in Section 6.3.1 of 692 the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the 693 sending encoder that it lost the decoder context and would like to 694 have it repaired somehow. This is semantically different from the 695 Full Intra Request above as there could be multiple ways to fulfil 696 the request. WebRTC Endpoints that are sending media MUST understand 697 and react to PLI feedback messages as a loss tolerance mechanism. 698 Receivers MAY send PLI messages. 700 5.1.3. Slice Loss Indication (SLI) 702 The Slice Loss Indication message is defined in Section 6.3.2 of the 703 RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the 704 encoder that it has detected the loss or corruption of one or more 705 consecutive macro blocks, and would like to have these repaired 706 somehow. It is RECOMMENDED that receivers generate SLI feedback 707 messages if slices are lost when using a codec that supports the 708 concept of macro blocks. A sender that receives an SLI feedback 709 message SHOULD attempt to repair the lost slice(s). 711 5.1.4. Reference Picture Selection Indication (RPSI) 713 Reference Picture Selection Indication (RPSI) messages are defined in 714 Section 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video encoding 715 standards allow the use of older reference pictures than the most 716 recent one for predictive coding. If such a codec is in use, and if 717 the encoder has learnt that encoder-decoder synchronisation has been 718 lost, then a known as correct reference picture can be used as a base 719 for future coding. The RPSI message allows this to be signalled. 720 Receivers that detect that encoder-decoder synchronisation has been 721 lost SHOULD generate an RPSI feedback message if codec being used 722 supports reference picture selection. A RTP packet stream sender 723 that receives such an RPSI message SHOULD act on that messages to 724 change the reference picture, if it is possible to do so within the 725 available bandwidth constraints, and with the codec being used. 727 5.1.5. Temporal-Spatial Trade-off Request (TSTR) 729 The temporal-spatial trade-off request and notification are defined 730 in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used 731 to ask the video encoder to change the trade-off it makes between 732 temporal and spatial resolution, for example to prefer high spatial 733 image quality but low frame rate. Support for TSTR requests and 734 notifications is OPTIONAL. 736 5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR) 738 The TMMBR feedback message is defined in Sections 3.5.4 and 4.2.1 of 739 the Codec Control Messages [RFC5104]. This request and its 740 notification message are used by a media receiver to inform the 741 sending party that there is a current limitation on the amount of 742 bandwidth available to this receiver. This can be various reasons 743 for this: for example, an RTP mixer can use this message to limit the 744 media rate of the sender being forwarded by the mixer (without doing 745 media transcoding) to fit the bottlenecks existing towards the other 746 session participants. WebRTC Endpoints that are sending media are 747 REQUIRED to implement support for TMMBR messages, and MUST follow 748 bandwidth limitations set by a TMMBR message received for their SSRC. 749 The sending of TMMBR requests is OPTIONAL. 751 5.2. Header Extensions 753 The RTP specification [RFC3550] provides the capability to include 754 RTP header extensions containing in-band data, but the format and 755 semantics of the extensions are poorly specified. The use of header 756 extensions is OPTIONAL in WebRTC, but if they are used, they MUST be 757 formatted and signalled following the general mechanism for RTP 758 header extensions defined in [RFC5285], since this gives well-defined 759 semantics to RTP header extensions. 761 As noted in [RFC5285], the requirement from the RTP specification 762 that header extensions are "designed so that the header extension may 763 be ignored" [RFC3550] stands. To be specific, header extensions MUST 764 only be used for data that can safely be ignored by the recipient 765 without affecting interoperability, and MUST NOT be used when the 766 presence of the extension has changed the form or nature of the rest 767 of the packet in a way that is not compatible with the way the stream 768 is signalled (e.g., as defined by the payload type). Valid examples 769 of RTP header extensions might include metadata that is additional to 770 the usual RTP information, but that can safely be ignored without 771 compromising interoperability. 773 5.2.1. Rapid Synchronisation 775 Many RTP sessions require synchronisation between audio, video, and 776 other content. This synchronisation is performed by receivers, using 777 information contained in RTCP SR packets, as described in the RTP 778 specification [RFC3550]. This basic mechanism can be slow, however, 779 so it is RECOMMENDED that the rapid RTP synchronisation extensions 780 described in [RFC6051] be implemented in addition to RTCP SR-based 781 synchronisation. The rapid synchronisation extensions use the 782 general RTP header extension mechanism [RFC5285], which requires 783 signalling, but are otherwise backwards compatible. 785 5.2.2. Client-to-Mixer Audio Level 787 The Client to Mixer Audio Level extension [RFC6464] is an RTP header 788 extension used by an endpoint to inform a mixer about the level of 789 audio activity in the packet to which the header is attached. This 790 enables an RTP middlebox to make mixing or selection decisions 791 without decoding or detailed inspection of the payload, reducing the 792 complexity in some types of mixers. It can also save decoding 793 resources in receivers, which can choose to decode only the most 794 relevant RTP packet streams based on audio activity levels. 796 The Client-to-Mixer Audio Level [RFC6464] header extension is 797 RECOMMENDED to be implemented. If this header extension is 798 implemented, it is REQUIRED that implementations are capable of 799 encrypting the header extension according to [RFC6904] since the 800 information contained in these header extensions can be considered 801 sensitive. The use of this encryption is RECOMMENDED, however usage 802 of the encryption can be explicitly disabled through API or 803 signalling. 805 5.2.3. Mixer-to-Client Audio Level 807 The Mixer to Client Audio Level header extension [RFC6465] provides 808 an endpoint with the audio level of the different sources mixed into 809 a common source stream by a RTP mixer. This enables a user interface 810 to indicate the relative activity level of each session participant, 811 rather than just being included or not based on the CSRC field. This 812 is a pure optimisation of non critical functions, and is hence 813 OPTIONAL to implement. If this header extension is implemented, it 814 is REQUIRED that implementations are capable of encrypting the header 815 extension according to [RFC6904] since the information contained in 816 these header extensions can be considered sensitive. It is further 817 RECOMMENDED that this encryption is used, unless the encryption has 818 been explicitly disabled through API or signalling. 820 6. WebRTC Use of RTP: Improving Transport Robustness 822 There are tools that can make RTP packet streams robust against 823 packet loss and reduce the impact of loss on media quality. However, 824 they generally add some overhead compared to a non-robust stream. 825 The overhead needs to be considered, and the aggregate bit-rate MUST 826 be rate controlled to avoid causing network congestion (see 827 Section 7). As a result, improving robustness might require a lower 828 base encoding quality, but has the potential to deliver that quality 829 with fewer errors. The mechanisms described in the following sub- 830 sections can be used to improve tolerance to packet loss. 832 6.1. Negative Acknowledgements and RTP Retransmission 834 As a consequence of supporting the RTP/SAVPF profile, implementations 835 can send negative acknowledgements (NACKs) for RTP data packets 836 [RFC4585]. This feedback can be used to inform a sender of the loss 837 of particular RTP packets, subject to the capacity limitations of the 838 RTCP feedback channel. A sender can use this information to optimise 839 the user experience by adapting the media encoding to compensate for 840 known lost packets. 842 RTP packet stream senders are REQUIRED to understand the Generic NACK 843 message defined in Section 6.2.1 of [RFC4585], but MAY choose to 844 ignore some or all of this feedback (following Section 4.2 of 845 [RFC4585]). Receivers MAY send NACKs for missing RTP packets. 846 Guidelines on when to send NACKs are provided in [RFC4585]. It is 847 not expected that a receiver will send a NACK for every lost RTP 848 packet, rather it needs to consider the cost of sending NACK 849 feedback, and the importance of the lost packet, to make an informed 850 decision on whether it is worth telling the sender about a packet 851 loss event. 853 The RTP Retransmission Payload Format [RFC4588] offers the ability to 854 retransmit lost packets based on NACK feedback. Retransmission needs 855 to be used with care in interactive real-time applications to ensure 856 that the retransmitted packet arrives in time to be useful, but can 857 be effective in environments with relatively low network RTT (an RTP 858 sender can estimate the RTT to the receivers using the information in 859 RTCP SR and RR packets, as described at the end of Section 6.4.1 of 860 [RFC3550]). The use of retransmissions can also increase the forward 861 RTP bandwidth, and can potentially caused increased packet loss if 862 the original packet loss was caused by network congestion. Note, 863 however, that retransmission of an important lost packet to repair 864 decoder state can have lower cost than sending a full intra frame. 865 It is not appropriate to blindly retransmit RTP packets in response 866 to a NACK. The importance of lost packets and the likelihood of them 867 arriving in time to be useful needs to be considered before RTP 868 retransmission is used. 870 Receivers are REQUIRED to implement support for RTP retransmission 871 packets [RFC4588] (both session multiplexing and SSRC multiplexing 872 need to be supported; see Section 4.4). Senders MAY send RTP 873 retransmission packets in response to NACKs if the RTP retransmission 874 payload format has been negotiated for the session, and if the sender 875 believes it is useful to send a retransmission of the packet(s) 876 referenced in the NACK. An RTP sender does not need to retransmit 877 every NACKed packet. 879 6.2. Forward Error Correction (FEC) 881 The use of Forward Error Correction (FEC) can provide an effective 882 protection against some degree of packet loss, at the cost of steady 883 bandwidth overhead. There are several FEC schemes that are defined 884 for use with RTP. Some of these schemes are specific to a particular 885 RTP payload format, others operate across RTP packets and can be used 886 with any payload format. It needs to be noted that using redundant 887 encoding or FEC will lead to increased play out delay, which needs to 888 be considered when choosing the redundancy or FEC formats and their 889 respective parameters. 891 If an RTP payload format negotiated for use in a RTCPeerConnection 892 supports redundant transmission or FEC as a standard feature of that 893 payload format, then that support MAY be used in the 894 RTCPeerConnection, subject to any appropriate signalling. 896 There are several block-based FEC schemes that are designed for use 897 with RTP independent of the chosen RTP payload format. At the time 898 of this writing there is no consensus on which, if any, of these FEC 899 schemes is appropriate for use in WebRTC. Accordingly, this memo 900 makes no recommendation on the choice of block-based FEC for WebRTC 901 use. 903 7. WebRTC Use of RTP: Rate Control and Media Adaptation 905 WebRTC will be used in heterogeneous network environments using a 906 variety set of link technologies, including both wired and wireless 907 links, to interconnect potentially large groups of users around the 908 world. As a result, the network paths between users can have widely 909 varying one-way delays, available bit-rates, load levels, and traffic 910 mixtures. Individual end-points can send one or more RTP packet 911 streams to each participant, and there can be several participants. 912 Each of these RTP packet streams can contain different types of 913 media, and the type of media, bit rate, and number of RTP packet 914 streams as well as transport-layer flows can be highly asymmetric. 915 Non-RTP traffic can share the network paths with RTP transport-layer 916 flows. Since the network environment is not predictable or stable, 917 WebRTC Endpoints MUST ensure that the RTP traffic they generate can 918 adapt to match changes in the available network capacity. 920 The quality of experience for users of WebRTC is very dependent on 921 effective adaptation of the media to the limitations of the network. 922 End-points have to be designed so they do not transmit significantly 923 more data than the network path can support, except for very short 924 time periods, otherwise high levels of network packet loss or delay 925 spikes will occur, causing media quality degradation. The limiting 926 factor on the capacity of the network path might be the link 927 bandwidth, or it might be competition with other traffic on the link 928 (this can be non-WebRTC traffic, traffic due to other WebRTC flows, 929 or even competition with other WebRTC flows in the same session). 931 An effective media congestion control algorithm is therefore an 932 essential part of the WebRTC framework. However, at the time of this 933 writing, there is no standard congestion control algorithm that can 934 be used for interactive media applications such as WebRTC's flows. 935 Some requirements for congestion control algorithms for 936 RTCPeerConnections are discussed in [I-D.ietf-rmcat-cc-requirements]. 937 A future version of this memo will mandate the use of a congestion 938 control algorithm that satisfies these requirements. 940 7.1. Boundary Conditions and Circuit Breakers 942 WebRTC Endpoints MUST implement the RTP circuit breaker algorithm 943 that is described in [I-D.ietf-avtcore-rtp-circuit-breakers]. The 944 RTP circuit breaker is designed to enable applications to recognise 945 and react to situations of extreme network congestion. However, 946 since the RTP circuit breaker might not be triggered until congestion 947 becomes extreme, it cannot be considered a substitute for congestion 948 control, and applications MUST also implement congestion control to 949 allow them to adapt to changes in network capacity. Any future RTP 950 congestion control algorithms are expected to operate within the 951 envelope allowed by the circuit breaker. 953 The session establishment signalling will also necessarily establish 954 boundaries to which the media bit-rate will conform. The choice of 955 media codecs provides upper- and lower-bounds on the supported bit- 956 rates that the application can utilise to provide useful quality, and 957 the packetisation choices that exist. In addition, the signalling 958 channel can establish maximum media bit-rate boundaries using, for 959 example, the SDP "b=AS:" or "b=CT:" lines and the RTP/AVPF Temporary 960 Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of 961 this memo). Signalled bandwidth limitations, such as SDP "b=AS:" or 962 "b=CT:" lines received from the peer, MUST be followed when sending 963 RTP packet streams. A WebRTC Endpoint receiving media SHOULD signal 964 its bandwidth limitations, these limitations have to be based on 965 known bandwidth limitations, for example the capacity of the edge 966 links. 968 7.2. Congestion Control Interoperability and Legacy Systems 970 There are legacy RTP implementations that do not implement RTCP, and 971 hence do not provide any congestion feedback. Congestion control 972 cannot be performed with these end-points. WebRTC Endpoints that 973 need to interwork with such end-points MUST limit their transmission 974 to a low rate, equivalent to a VoIP call using a low bandwidth codec, 975 that is unlikely to cause any significant congestion. 977 When interworking with legacy implementations that support RTCP using 978 the RTP/AVP profile [RFC3551], congestion feedback is provided in 979 RTCP RR packets every few seconds. Implementations that have to 980 interwork with such end-points MUST ensure that they keep within the 981 RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers] 982 constraints to limit the congestion they can cause. 984 If a legacy end-point supports RTP/AVPF, this enables negotiation of 985 important parameters for frequent reporting, such as the "trr-int" 986 parameter, and the possibility that the end-point supports some 987 useful feedback format for congestion control purpose such as TMMBR 988 [RFC5104]. Implementations that have to interwork with such end- 989 points MUST ensure that they stay within the RTP circuit breaker 990 [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the 991 congestion they can cause, but might find that they can achieve 992 better congestion response depending on the amount of feedback that 993 is available. 995 With proprietary congestion control algorithms issues can arise when 996 different algorithms and implementations interact in a communication 997 session. If the different implementations have made different 998 choices in regards to the type of adaptation, for example one sender 999 based, and one receiver based, then one could end up in situation 1000 where one direction is dual controlled, when the other direction is 1001 not controlled. This memo cannot mandate behaviour for proprietary 1002 congestion control algorithms, but implementations that use such 1003 algorithms ought to be aware of this issue, and try to ensure that 1004 effective congestion control is negotiated for media flowing in both 1005 directions. If the IETF were to standardise both sender- and 1006 receiver-based congestion control algorithms for WebRTC traffic in 1007 the future, the issues of interoperability, control, and ensuring 1008 that both directions of media flow are congestion controlled would 1009 also need to be considered. 1011 8. WebRTC Use of RTP: Performance Monitoring 1013 As described in Section 4.1, implementations are REQUIRED to generate 1014 RTCP Sender Report (SR) and Reception Report (RR) packets relating to 1015 the RTP packet streams they send and receive. These RTCP reports can 1016 be used for performance monitoring purposes, since they include basic 1017 packet loss and jitter statistics. 1019 A large number of additional performance metrics are supported by the 1020 RTCP Extended Reports (XR) framework [RFC3611][RFC6792]. At the time 1021 of this writing, it is not clear what extended metrics are suitable 1022 for use in WebRTC, so there is no requirement that implementations 1023 generate RTCP XR packets. However, implementations that can use 1024 detailed performance monitoring data MAY generate RTCP XR packets as 1025 appropriate; the use of such packets SHOULD be signalled in advance. 1027 9. WebRTC Use of RTP: Future Extensions 1029 It is possible that the core set of RTP protocols and RTP extensions 1030 specified in this memo will prove insufficient for the future needs 1031 of WebRTC. In this case, future updates to this memo MUST be made 1032 following the Guidelines for Writers of RTP Payload Format 1033 Specifications [RFC2736], How to Write an RTP Payload Format 1034 [I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP 1035 Control Protocol [RFC5968], and SHOULD take into account any future 1036 guidelines for extending RTP and related protocols that have been 1037 developed. 1039 Authors of future extensions are urged to consider the wide range of 1040 environments in which RTP is used when recommending extensions, since 1041 extensions that are applicable in some scenarios can be problematic 1042 in others. Where possible, the WebRTC framework will adopt RTP 1043 extensions that are of general utility, to enable easy implementation 1044 of a gateway to other applications using RTP, rather than adopt 1045 mechanisms that are narrowly targeted at specific WebRTC use cases. 1047 10. Signalling Considerations 1049 RTP is built with the assumption that an external signalling channel 1050 exists, and can be used to configure RTP sessions and their features. 1051 The basic configuration of an RTP session consists of the following 1052 parameters: 1054 RTP Profile: The name of the RTP profile to be used in session. The 1055 RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate 1056 on basic level, as can their secure variants RTP/SAVP [RFC3711] 1057 and RTP/SAVPF [RFC5124]. The secure variants of the profiles do 1058 not directly interoperate with the non-secure variants, due to the 1059 presence of additional header fields for authentication in SRTP 1060 packets and cryptographic transformation of the payload. WebRTC 1061 requires the use of the RTP/SAVPF profile, and this MUST be 1062 signalled. Interworking functions might transform this into the 1063 RTP/SAVP profile for a legacy use case, by indicating to the 1064 WebRTC Endpoint that the RTP/SAVPF is used and configuring a trr- 1065 int value of 4 seconds. 1067 Transport Information: Source and destination IP address(s) and 1068 ports for RTP and RTCP MUST be signalled for each RTP session. In 1069 WebRTC these transport addresses will be provided by ICE [RFC5245] 1070 that signals candidates and arrives at nominated candidate address 1071 pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such 1072 that a single port, i.e. transport-layer flow, is used for RTP 1073 and RTCP flows, this MUST be signalled (see Section 4.5). 1075 RTP Payload Types, media formats, and format parameters: The mapping 1076 between media type names (and hence the RTP payload formats to be 1077 used), and the RTP payload type numbers MUST be signalled. Each 1078 media type MAY also have a number of media type parameters that 1079 MUST also be signalled to configure the codec and RTP payload 1080 format (the "a=fmtp:" line from SDP). Section 4.3 of this memo 1081 discusses requirements for uniqueness of payload types. 1083 RTP Extensions: The use of any additional RTP header extensions and 1084 RTCP packet types, including any necessary parameters, MUST be 1085 signalled. This signalling is to ensure that a WebRTC Endpoint's 1086 behaviour, especially when sending, of any extensions is 1087 predictable and consistent. For robustness, and for compatibility 1088 with non-WebRTC systems that might be connected to a WebRTC 1089 session via a gateway, implementations are REQUIRED to ignore 1090 unknown RTCP packets and RTP header extensions (see also 1091 Section 4.1). 1093 RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the 1094 end-points will be necessary. This SHALL be done as described in 1095 "Session Description Protocol (SDP) Bandwidth Modifiers for RTP 1096 Control Protocol (RTCP) Bandwidth" [RFC3556] if using SDP, or 1097 something semantically equivalent. This also ensures that the 1098 end-points have a common view of the RTCP bandwidth. A common 1099 RTCP bandwidth is important as a too different view of the 1100 bandwidths can lead to failure to interoperate. 1102 These parameters are often expressed in SDP messages conveyed within 1103 an offer/answer exchange. RTP does not depend on SDP or on the offer 1104 /answer model, but does require all the necessary parameters to be 1105 agreed upon, and provided to the RTP implementation. Note that in 1106 WebRTC it will depend on the signalling model and API how these 1107 parameters need to be configured but they will be need to either be 1108 set in the API or explicitly signalled between the peers. 1110 11. WebRTC API Considerations 1112 The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and 1113 Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses 1114 the concept of a MediaStream that consists of zero or more 1115 MediaStreamTracks. A MediaStreamTrack is an individual stream of 1116 media from any type of media source like a microphone or a camera, 1117 but also conceptual sources, like a audio mix or a video composition, 1118 are possible. The MediaStreamTracks within a MediaStream need to be 1119 possible to play out synchronised. 1121 A MediaStreamTrack's realisation in RTP in the context of an 1122 RTCPeerConnection consists of a source packet stream identified with 1123 an SSRC within an RTP session part of the RTCPeerConnection. The 1124 MediaStreamTrack can also result in additional packet streams, and 1125 thus SSRCs, in the same RTP session. These can be dependent packet 1126 streams from scalable encoding of the source stream associated with 1127 the MediaStreamTrack, if such a media encoder is used. They can also 1128 be redundancy packet streams, these are created when applying Forward 1129 Error Correction (Section 6.2) or RTP retransmission (Section 6.1) to 1130 the source packet stream. 1132 It is important to note that the same media source can be feeding 1133 multiple MediaStreamTracks. As different sets of constraints or 1134 other parameters can be applied to the MediaStreamTrack, each 1135 MediaStreamTrack instance added to a RTCPeerConnection SHALL result 1136 in an independent source packet stream, with its own set of 1137 associated packet streams, and thus different SSRC(s). It will 1138 depend on applied constraints and parameters if the source stream and 1139 the encoding configuration will be identical between different 1140 MediaStreamTracks sharing the same media source. If the encoding 1141 parameters and constraints are the same, an implementation could 1142 choose to use only one encoded stream to create the different RTP 1143 packet streams. Note that such optimisations would need to take into 1144 account that the constraints for one of the MediaStreamTracks can at 1145 any moment change, meaning that the encoding configurations might no 1146 longer be identical and two different encoder instances would then be 1147 needed. 1149 The same MediaStreamTrack can also be included in multiple 1150 MediaStreams, thus multiple sets of MediaStreams can implicitly need 1151 to use the same synchronisation base. To ensure that this works in 1152 all cases, and does not force an end-point to to disrupt the media by 1153 changing synchronisation base and CNAME during delivery of any 1154 ongoing packet streams, all MediaStreamTracks and their associated 1155 SSRCs originating from the same end-point need to be sent using the 1156 same CNAME within one RTCPeerConnection. This is motivating the 1157 discussion in Section 4.9 to only use a single CNAME. 1159 The requirement on using the same CNAME for all SSRCs that 1160 originate from the same end-point, does not require a middlebox 1161 that forwards traffic from multiple end-points to only use a 1162 single CNAME. 1164 Different CNAMEs normally need to be used for different 1165 RTCPeerConnection instances, as specified in Section 4.9. Having two 1166 communication sessions with the same CNAME could enable tracking of a 1167 user or device across different services (see Section 4.4.1 of 1168 [I-D.ietf-rtcweb-security] for details). A web application can 1169 request that the CNAMEs used in different RTCPeerConnections (within 1170 a same-orign context) be the same, this allows for synchronization of 1171 the endpoint's RTP packet streams across the different 1172 RTCPeerConnections. 1174 Note: this doesn't result in a tracking issue, since the creation 1175 of matching CNAMEs depends on existing tracking. 1177 The above will currently force a WebRTC Endpoint that receives a 1178 MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing 1179 on any RTCPeerConnection to perform resynchronisation of the stream. 1180 This, as the sending party needs to change the CNAME to the one it 1181 uses, which implies that the sender has to use a local system clock 1182 as timebase for the synchronisation. Thus, the relative relation 1183 between the timebase of the incoming stream and the system sending 1184 out needs to defined. This relation also needs monitoring for clock 1185 drift and likely adjustments of the synchronisation. The sending 1186 entity is also responsible for congestion control for its sent 1187 streams. In cases of packet loss the loss of incoming data also 1188 needs to be handled. This leads to the observation that the method 1189 that is least likely to cause issues or interruptions in the outgoing 1190 source packet stream is a model of full decoding, including repair 1191 etc., followed by encoding of the media again into the outgoing 1192 packet stream. Optimisations of this method is clearly possible and 1193 implementation specific. 1195 A WebRTC Endpoint MUST support receiving multiple MediaStreamTracks, 1196 where each of different MediaStreamTracks (and their sets of 1197 associated packet streams) uses different CNAMEs. However, 1198 MediaStreamTracks that are received with different CNAMEs have no 1199 defined synchronisation. 1201 Note: The motivation for supporting reception of multiple CNAMEs 1202 is to allow for forward compatibility with any future changes that 1203 enables more efficient stream handling when end-points relay/ 1204 forward streams. It also ensures that end-points can interoperate 1205 with certain types of multi-stream middleboxes or end-points that 1206 are not WebRTC. 1208 The binding between the WebRTC MediaStreams, MediaStreamTracks and 1209 the SSRC is done as specified in "Cross Session Stream Identification 1210 in the Session Description Protocol" [I-D.ietf-mmusic-msid]. This 1211 document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to 1212 map unknown source packet stream SSRCs to MediaStreamTracks and 1213 MediaStreams. This later is relevant to handle some cases of legacy 1214 interop. Commonly the RTP Payload Type of any incoming packets will 1215 reveal if the packet stream is a source stream or a redundancy or 1216 dependent packet stream. The association to the correct source 1217 packet stream depends on the payload format in use for the packet 1218 stream. 1220 Finally this specification puts a requirement on the WebRTC API to 1221 realize a method for determining the CSRC list (Section 4.1) as well 1222 as the Mixer-to-Client audio levels (Section 5.2.3) (when supported) 1223 and the basic requirements for this is further discussed in 1224 Section 12.2.1. 1226 12. RTP Implementation Considerations 1228 The following discussion provides some guidance on the implementation 1229 of the RTP features described in this memo. The focus is on a WebRTC 1230 Endpoint implementation perspective, and while some mention is made 1231 of the behaviour of middleboxes, that is not the focus of this memo. 1233 12.1. Configuration and Use of RTP Sessions 1235 A WebRTC Endpoint will be a simultaneous participant in one or more 1236 RTP sessions. Each RTP session can convey multiple media sources, 1237 and can include media data from multiple end-points. In the 1238 following, some ways in which WebRTC Endpoints can configure and use 1239 RTP sessions is outlined. 1241 12.1.1. Use of Multiple Media Sources Within an RTP Session 1242 RTP is a group communication protocol, and every RTP session can 1243 potentially contain multiple RTP packet streams. There are several 1244 reasons why this might be desirable: 1246 Multiple media types: Outside of WebRTC, it is common to use one RTP 1247 session for each type of media sources (e.g., one RTP session for 1248 audio sources and one for video sources, each sent over different 1249 transport layer flows). However, to reduce the number of UDP 1250 ports used, the default in WebRTC is to send all types of media in 1251 a single RTP session, as described in Section 4.4, using RTP and 1252 RTCP multiplexing (Section 4.5) to further reduce the number of 1253 UDP ports needed. This RTP session then uses only one bi- 1254 directional transport-layer flow, but will contain multiple RTP 1255 packet streams, each containing a different type of media. A 1256 common example might be an end-point with a camera and microphone 1257 that sends two RTP packet streams, one video and one audio, into a 1258 single RTP session. 1260 Multiple Capture Devices: A WebRTC Endpoint might have multiple 1261 cameras, microphones, or other media capture devices, and so might 1262 want to generate several RTP packet streams of the same media 1263 type. Alternatively, it might want to send media from a single 1264 capture device in several different formats or quality settings at 1265 once. Both can result in a single end-point sending multiple RTP 1266 packet streams of the same media type into a single RTP session at 1267 the same time. 1269 Associated Repair Data: An end-point might send a RTP packet stream 1270 that is somehow associated with another stream. For example, it 1271 might send an RTP packet stream that contains FEC or 1272 retransmission data relating to another stream. Some RTP payload 1273 formats send this sort of associated repair data as part of the 1274 source packet stream, while others send it as a separate packet 1275 stream. 1277 Layered or Multiple Description Coding: An end-point can use a 1278 layered media codec, for example H.264 SVC, or a multiple 1279 description codec, that generates multiple RTP packet streams, 1280 each with a distinct RTP SSRC, within a single RTP session. 1282 RTP Mixers, Translators, and Other Middleboxes: An RTP session, in 1283 the WebRTC context, is a point-to-point association between an 1284 end-point and some other peer device, where those devices share a 1285 common SSRC space. The peer device might be another WebRTC 1286 Endpoint, or it might be an RTP mixer, translator, or some other 1287 form of media processing middlebox. In the latter cases, the 1288 middlebox might send mixed or relayed RTP streams from several 1289 participants, that the WebRTC Endpoint will need to render. Thus, 1290 even though a WebRTC Endpoint might only be a member of a single 1291 RTP session, the peer device might be extending that RTP session 1292 to incorporate other end-points. WebRTC is a group communication 1293 environment and end-points need to be capable of receiving, 1294 decoding, and playing out multiple RTP packet streams at once, 1295 even in a single RTP session. 1297 12.1.2. Use of Multiple RTP Sessions 1299 In addition to sending and receiving multiple RTP packet streams 1300 within a single RTP session, a WebRTC Endpoint might participate in 1301 multiple RTP sessions. There are several reasons why a WebRTC 1302 Endpoint might choose to do this: 1304 To interoperate with legacy devices: The common practice in the non- 1305 WebRTC world is to send different types of media in separate RTP 1306 sessions, for example using one RTP session for audio and another 1307 RTP session, on a separate transport layer flow, for video. All 1308 WebRTC Endpoints need to support the option of sending different 1309 types of media on different RTP sessions, so they can interwork 1310 with such legacy devices. This is discussed further in 1311 Section 4.4. 1313 To provide enhanced quality of service: Some network-based quality 1314 of service mechanisms operate on the granularity of transport 1315 layer flows. If it is desired to use these mechanisms to provide 1316 differentiated quality of service for some RTP packet streams, 1317 then those RTP packet streams need to be sent in a separate RTP 1318 session using a different transport-layer flow, and with 1319 appropriate quality of service marking. This is discussed further 1320 in Section 12.1.3. 1322 To separate media with different purposes: An end-point might want 1323 to send RTP packet streams that have different purposes on 1324 different RTP sessions, to make it easy for the peer device to 1325 distinguish them. For example, some centralised multiparty 1326 conferencing systems display the active speaker in high 1327 resolution, but show low resolution "thumbnails" of other 1328 participants. Such systems might configure the end-points to send 1329 simulcast high- and low-resolution versions of their video using 1330 separate RTP sessions, to simplify the operation of the RTP 1331 middlebox. In the WebRTC context this is currently possible by 1332 establishing multiple WebRTC MediaStreamTracks that have the same 1333 media source in one (or more) RTCPeerConnection. Each 1334 MediaStreamTrack is then configured to deliver a particular media 1335 quality and thus media bit-rate, and will produce an independently 1336 encoded version with the codec parameters agreed specifically in 1337 the context of that RTCPeerConnection. The RTP middlebox can 1338 distinguish packets corresponding to the low- and high-resolution 1339 streams by inspecting their SSRC, RTP payload type, or some other 1340 information contained in RTP payload, RTP header extension or RTCP 1341 packets, but it can be easier to distinguish the RTP packet 1342 streams if they arrive on separate RTP sessions on separate 1343 transport-layer flows. 1345 To directly connect with multiple peers: A multi-party conference 1346 does not need to use an RTP middlebox. Rather, a multi-unicast 1347 mesh can be created, comprising several distinct RTP sessions, 1348 with each participant sending RTP traffic over a separate RTP 1349 session (that is, using an independent RTCPeerConnection object) 1350 to every other participant, as shown in Figure 1. This topology 1351 has the benefit of not requiring an RTP middlebox node that is 1352 trusted to access and manipulate the media data. The downside is 1353 that it increases the used bandwidth at each sender by requiring 1354 one copy of the RTP packet streams for each participant that are 1355 part of the same session beyond the sender itself. 1357 +---+ +---+ 1358 | A |<--->| B | 1359 +---+ +---+ 1360 ^ ^ 1361 \ / 1362 \ / 1363 v v 1364 +---+ 1365 | C | 1366 +---+ 1368 Figure 1: Multi-unicast using several RTP sessions 1370 The multi-unicast topology could also be implemented as a single 1371 RTP session, spanning multiple peer-to-peer transport layer 1372 connections, or as several pairwise RTP sessions, one between each 1373 pair of peers. To maintain a coherent mapping between the 1374 relation between RTP sessions and RTCPeerConnection objects it is 1375 recommend that this is implemented as several individual RTP 1376 sessions. The only downside is that end-point A will not learn of 1377 the quality of any transmission happening between B and C, since 1378 it will not see RTCP reports for the RTP session between B and C, 1379 whereas it would it all three participants were part of a single 1380 RTP session. Experience with the Mbone tools (experimental RTP- 1381 based multicast conferencing tools from the late 1990s) has showed 1382 that RTCP reception quality reports for third parties can be 1383 presented to users in a way that helps them understand asymmetric 1384 network problems, and the approach of using separate RTP sessions 1385 prevents this. However, an advantage of using separate RTP 1386 sessions is that it enables using different media bit-rates and 1387 RTP session configurations between the different peers, thus not 1388 forcing B to endure the same quality reductions if there are 1389 limitations in the transport from A to C as C will. It is 1390 believed that these advantages outweigh the limitations in 1391 debugging power. 1393 To indirectly connect with multiple peers: A common scenario in 1394 multi-party conferencing is to create indirect connections to 1395 multiple peers, using an RTP mixer, translator, or some other type 1396 of RTP middlebox. Figure 2 outlines a simple topology that might 1397 be used in a four-person centralised conference. The middlebox 1398 acts to optimise the transmission of RTP packet streams from 1399 certain perspectives, either by only sending some of the received 1400 RTP packet stream to any given receiver, or by providing a 1401 combined RTP packet stream out of a set of contributing streams. 1403 +---+ +-------------+ +---+ 1404 | A |<---->| |<---->| B | 1405 +---+ | RTP mixer, | +---+ 1406 | translator, | 1407 | or other | 1408 +---+ | middlebox | +---+ 1409 | C |<---->| |<---->| D | 1410 +---+ +-------------+ +---+ 1412 Figure 2: RTP mixer with only unicast paths 1414 There are various methods of implementation for the middlebox. If 1415 implemented as a standard RTP mixer or translator, a single RTP 1416 session will extend across the middlebox and encompass all the 1417 end-points in one multi-party session. Other types of middlebox 1418 might use separate RTP sessions between each end-point and the 1419 middlebox. A common aspect is that these RTP middleboxes can use 1420 a number of tools to control the media encoding provided by a 1421 WebRTC Endpoint. This includes functions like requesting the 1422 breaking of the encoding chain and have the encoder produce a so 1423 called Intra frame. Another is limiting the bit-rate of a given 1424 stream to better suit the mixer view of the multiple down-streams. 1425 Others are controlling the most suitable frame-rate, picture 1426 resolution, the trade-off between frame-rate and spatial quality. 1427 The middlebox has the responsibility to correctly perform 1428 congestion control, source identification, manage synchronisation 1429 while providing the application with suitable media optimisations. 1430 The middlebox also has to be a trusted node when it comes to 1431 security, since it manipulates either the RTP header or the media 1432 itself (or both) received from one end-point, before sending it on 1433 towards the end-point(s), thus they need to be able to decrypt and 1434 then re-encrypt the RTP packet stream before sending it out. 1436 RTP Mixers can create a situation where an end-point experiences a 1437 situation in-between a session with only two end-points and 1438 multiple RTP sessions. Mixers are expected to not forward RTCP 1439 reports regarding RTP packet streams across themselves. This is 1440 due to the difference in the RTP packet streams provided to the 1441 different end-points. The original media source lacks information 1442 about a mixer's manipulations prior to sending it the different 1443 receivers. This scenario also results in that an end-point's 1444 feedback or requests goes to the mixer. When the mixer can't act 1445 on this by itself, it is forced to go to the original media source 1446 to fulfil the receivers request. This will not necessarily be 1447 explicitly visible any RTP and RTCP traffic, but the interactions 1448 and the time to complete them will indicate such dependencies. 1450 Providing source authentication in multi-party scenarios is a 1451 challenge. In the mixer-based topologies, end-points source 1452 authentication is based on, firstly, verifying that media comes 1453 from the mixer by cryptographic verification and, secondly, trust 1454 in the mixer to correctly identify any source towards the end- 1455 point. In RTP sessions where multiple end-points are directly 1456 visible to an end-point, all end-points will have knowledge about 1457 each others' master keys, and can thus inject packets claimed to 1458 come from another end-point in the session. Any node performing 1459 relay can perform non-cryptographic mitigation by preventing 1460 forwarding of packets that have SSRC fields that came from other 1461 end-points before. For cryptographic verification of the source, 1462 SRTP would require additional security mechanisms, for example 1463 TESLA for SRTP [RFC4383], that are not part of the base WebRTC 1464 standards. 1466 To forward media between multiple peers: It is sometimes desirable 1467 for an end-point that receives an RTP packet stream to be able to 1468 forward that RTP packet stream to a third party. The are some 1469 obvious security and privacy implications in supporting this, but 1470 also potential uses. This is supported in the W3C API by taking 1471 the received and decoded media and using it as media source that 1472 is re-encoding and transmitted as a new stream. 1474 At the RTP layer, media forwarding acts as a back-to-back RTP 1475 receiver and RTP sender. The receiving side terminates the RTP 1476 session and decodes the media, while the sender side re-encodes 1477 and transmits the media using an entirely separate RTP session. 1478 The original sender will only see a single receiver of the media, 1479 and will not be able to tell that forwarding is happening based on 1480 RTP-layer information since the RTP session that is used to send 1481 the forwarded media is not connected to the RTP session on which 1482 the media was received by the node doing the forwarding. 1484 The end-point that is performing the forwarding is responsible for 1485 producing an RTP packet stream suitable for onwards transmission. 1486 The outgoing RTP session that is used to send the forwarded media 1487 is entirely separate to the RTP session on which the media was 1488 received. This will require media transcoding for congestion 1489 control purpose to produce a suitable bit-rate for the outgoing 1490 RTP session, reducing media quality and forcing the forwarding 1491 end-point to spend the resource on the transcoding. The media 1492 transcoding does result in a separation of the two different legs 1493 removing almost all dependencies, and allowing the forwarding end- 1494 point to optimise its media transcoding operation. The cost is 1495 greatly increased computational complexity on the forwarding node. 1496 Receivers of the forwarded stream will see the forwarding device 1497 as the sender of the stream, and will not be able to tell from the 1498 RTP layer that they are receiving a forwarded stream rather than 1499 an entirely new RTP packet stream generated by the forwarding 1500 device. 1502 12.1.3. Differentiated Treatment of RTP Packet Streams 1504 There are use cases for differentiated treatment of RTP packet 1505 streams. Such differentiation can happen at several places in the 1506 system. First of all is the prioritization within the end-point 1507 sending the media, which controls, both which RTP packet streams that 1508 will be sent, and their allocation of bit-rate out of the current 1509 available aggregate as determined by the congestion control. 1511 It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will 1512 allow the application to indicate relative priorities for different 1513 MediaStreamTracks. These priorities can then be used to influence 1514 the local RTP processing, especially when it comes to congestion 1515 control response in how to divide the available bandwidth between the 1516 RTP packet streams. Any changes in relative priority will also need 1517 to be considered for RTP packet streams that are associated with the 1518 main RTP packet streams, such as redundant streams for RTP 1519 retransmission and FEC. The importance of such redundant RTP packet 1520 streams is dependent on the media type and codec used, in regards to 1521 how robust that codec is to packet loss. However, a default policy 1522 might to be to use the same priority for redundant RTP packet stream 1523 as for the source RTP packet stream. 1525 Secondly, the network can prioritize transport-layer flows and sub- 1526 flows, including RTP packet streams. Typically, differential 1527 treatment includes two steps, the first being identifying whether an 1528 IP packet belongs to a class that has to be treated differently, the 1529 second consisting of the actual mechanism to prioritize packets. 1530 This is done according to three methods: 1532 DiffServ: The end-point marks a packet with a DiffServ code point to 1533 indicate to the network that the packet belongs to a particular 1534 class. 1536 Flow based: Packets that need to be given a particular treatment are 1537 identified using a combination of IP and port address. 1539 Deep Packet Inspection: A network classifier (DPI) inspects the 1540 packet and tries to determine if the packet represents a 1541 particular application and type that is to be prioritized. 1543 Flow-based differentiation will provide the same treatment to all 1544 packets within a transport-layer flow, i.e., relative prioritization 1545 is not possible. Moreover, if the resources are limited it might not 1546 be possible to provide differential treatment compared to best-effort 1547 for all the RTP packet streams used in a WebRTC session. When flow- 1548 based differentiation is available, the WebRTC Endpoint needs to know 1549 about it so that it can provide the separation of the RTP packet 1550 streams onto different UDP flows to enable a more granular usage of 1551 flow based differentiation. That way at least providing different 1552 prioritization of audio and video if desired by application. 1554 DiffServ assumes that either the end-point or a classifier can mark 1555 the packets with an appropriate DSCP so that the packets are treated 1556 according to that marking. If the end-point is to mark the traffic 1557 two requirements arise in the WebRTC context: 1) The WebRTC Endpoint 1558 has to know which DSCP to use and that it can use them on some set of 1559 RTP packet streams. 2) The information needs to be propagated to the 1560 operating system when transmitting the packet. Details of this 1561 process are outside the scope of this memo and are further discussed 1562 in "DSCP and other packet markings for RTCWeb QoS" 1563 [I-D.ietf-tsvwg-rtcweb-qos]. 1565 For packet based marking schemes it might be possible to mark 1566 individual RTP packets differently based on the relative priority of 1567 the RTP payload. For example video codecs that have I, P, and B 1568 pictures could prioritise any payloads carrying only B frames less, 1569 as these are less damaging to loose. However, depending on the QoS 1570 mechanism and what markings that are applied, this can result in not 1571 only different packet drop probabilities but also packet reordering, 1572 see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion. As a default 1573 policy all RTP packets related to a RTP packet stream ought to be 1574 provided with the same prioritization; per-packet prioritization is 1575 outside the scope of this memo, but might be specified elsewhere in 1576 future. 1578 It is also important to consider how RTCP packets associated with a 1579 particular RTP packet stream need to be marked. RTCP compound 1580 packets with Sender Reports (SR), ought to be marked with the same 1581 priority as the RTP packet stream itself, so the RTCP-based round- 1582 trip time (RTT) measurements are done using the same transport-layer 1583 flow priority as the RTP packet stream experiences. RTCP compound 1584 packets containing RR packet ought to be sent with the priority used 1585 by the majority of the RTP packet streams reported on. RTCP packets 1586 containing time-critical feedback packets can use higher priority to 1587 improve the timeliness and likelihood of delivery of such feedback. 1589 12.2. Media Source, RTP Packet Streams, and Participant Identification 1591 12.2.1. Media Source Identification 1593 Each RTP packet stream is identified by a unique synchronisation 1594 source (SSRC) identifier. The SSRC identifier is carried in each of 1595 the RTP packets comprising a RTP packet stream, and is also used to 1596 identify that stream in the corresponding RTCP reports. The SSRC is 1597 chosen as discussed in Section 4.8. The first stage in 1598 demultiplexing RTP and RTCP packets received on a single transport 1599 layer flow at a WebRTC Endpoint is to separate the RTP packet streams 1600 based on their SSRC value; once that is done, additional 1601 demultiplexing steps can determine how and where to render the media. 1603 RTP allows a mixer, or other RTP-layer middlebox, to combine encoded 1604 streams from multiple media sources to form a new encoded stream from 1605 a new media source (the mixer). The RTP packets in that new RTP 1606 packet stream can include a Contributing Source (CSRC) list, 1607 indicating which original SSRCs contributed to the combined source 1608 stream. As described in Section 4.1, implementations need to support 1609 reception of RTP data packets containing a CSRC list and RTCP packets 1610 that relate to sources present in the CSRC list. The CSRC list can 1611 change on a packet-by-packet basis, depending on the mixing operation 1612 being performed. Knowledge of what media sources contributed to a 1613 particular RTP packet can be important if the user interface 1614 indicates which participants are active in the session. Changes in 1615 the CSRC list included in packets needs to be exposed to the WebRTC 1616 application using some API, if the application is to be able to track 1617 changes in session participation. It is desirable to map CSRC values 1618 back into WebRTC MediaStream identities as they cross this API, to 1619 avoid exposing the SSRC/CSRC name space to WebRTC applications. 1621 If the mixer-to-client audio level extension [RFC6465] is being used 1622 in the session (see Section 5.2.3), the information in the CSRC list 1623 is augmented by audio level information for each contributing source. 1624 It is desirable to expose this information to the WebRTC application 1625 using some API, after mapping the CSRC values to WebRTC MediaStream 1626 identities, so it can be exposed in the user interface. 1628 12.2.2. SSRC Collision Detection 1630 The RTP standard requires RTP implementations to have support for 1631 detecting and handling SSRC collisions, i.e., resolve the conflict 1632 when two different end-points use the same SSRC value (see section 1633 8.2 of [RFC3550]). This requirement also applies to WebRTC 1634 Endpoints. There are several scenarios where SSRC collisions can 1635 occur: 1637 o In a point-to-point session where each SSRC is associated with 1638 either of the two end-points and where the main media carrying 1639 SSRC identifier will be announced in the signalling channel, a 1640 collision is less likely to occur due to the information about 1641 used SSRCs. If SDP is used, this information is provided by 1642 Source-Specific SDP Attributes [RFC5576]. Still, collisions can 1643 occur if both end-points start using a new SSRC identifier prior 1644 to having signalled it to the peer and received acknowledgement on 1645 the signalling message. The Source-Specific SDP Attributes 1646 [RFC5576] contains a mechanism to signal how the end-point 1647 resolved the SSRC collision. 1649 o SSRC values that have not been signalled could also appear in an 1650 RTP session. This is more likely than it appears, since some RTP 1651 functions use extra SSRCs to provide their functionality. For 1652 example, retransmission data might be transmitted using a separate 1653 RTP packet stream that requires its own SSRC, separate to the SSRC 1654 of the source RTP packet stream [RFC4588]. In those cases, an 1655 end-point can create a new SSRC that strictly doesn't need to be 1656 announced over the signalling channel to function correctly on 1657 both RTP and RTCPeerConnection level. 1659 o Multiple end-points in a multiparty conference can create new 1660 sources and signal those towards the RTP middlebox. In cases 1661 where the SSRC/CSRC are propagated between the different end- 1662 points from the RTP middlebox collisions can occur. 1664 o An RTP middlebox could connect an end-point's RTCPeerConnection to 1665 another RTCPeerConnection from the same end-point, thus forming a 1666 loop where the end-point will receive its own traffic. While it 1667 is clearly considered a bug, it is important that the end-point is 1668 able to recognise and handle the case when it occurs. This case 1669 becomes even more problematic when media mixers, and so on, are 1670 involved, where the stream received is a different stream but 1671 still contains this client's input. 1673 These SSRC/CSRC collisions can only be handled on RTP level as long 1674 as the same RTP session is extended across multiple 1675 RTCPeerConnections by a RTP middlebox. To resolve the more generic 1676 case where multiple RTCPeerConnections are interconnected, 1677 identification of the media source(s) part of a MediaStreamTrack 1678 being propagated across multiple interconnected RTCPeerConnection 1679 needs to be preserved across these interconnections. 1681 12.2.3. Media Synchronisation Context 1683 When an end-point sends media from more than one media source, it 1684 needs to consider if (and which of) these media sources are to be 1685 synchronized. In RTP/RTCP, synchronisation is provided by having a 1686 set of RTP packet streams be indicated as coming from the same 1687 synchronisation context and logical end-point by using the same RTCP 1688 CNAME identifier. 1690 The next provision is that the internal clocks of all media sources, 1691 i.e., what drives the RTP timestamp, can be correlated to a system 1692 clock that is provided in RTCP Sender Reports encoded in an NTP 1693 format. By correlating all RTP timestamps to a common system clock 1694 for all sources, the timing relation of the different RTP packet 1695 streams, also across multiple RTP sessions can be derived at the 1696 receiver and, if desired, the streams can be synchronized. The 1697 requirement is for the media sender to provide the correlation 1698 information; it is up to the receiver to use it or not. 1700 13. Security Considerations 1702 The overall security architecture for WebRTC is described in 1703 [I-D.ietf-rtcweb-security-arch], and security considerations for the 1704 WebRTC framework are described in [I-D.ietf-rtcweb-security]. These 1705 considerations also apply to this memo. 1707 The security considerations of the RTP specification, the RTP/SAVPF 1708 profile, and the various RTP/RTCP extensions and RTP payload formats 1709 that form the complete protocol suite described in this memo apply. 1710 It is not believed there are any new security considerations 1711 resulting from the combination of these various protocol extensions. 1713 The Extended Secure RTP Profile for Real-time Transport Control 1714 Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides 1715 handling of fundamental issues by offering confidentiality, integrity 1716 and partial source authentication. A mandatory to implement media 1717 security solution is created by combing this secured RTP profile and 1718 DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of 1719 [I-D.ietf-rtcweb-security-arch]. 1721 RTCP packets convey a Canonical Name (CNAME) identifier that is used 1722 to associate RTP packet streams that need to be synchronised across 1723 related RTP sessions. Inappropriate choice of CNAME values can be a 1724 privacy concern, since long-term persistent CNAME identifiers can be 1725 used to track users across multiple WebRTC calls. Section 4.9 of 1726 this memo provides guidelines for generation of untraceable CNAME 1727 values that alleviate this risk. 1729 Some potential denial of service attacks exist if the RTCP reporting 1730 interval is configured to an inappropriate value. This could be done 1731 by configuring the RTCP bandwidth fraction to an excessively large or 1732 small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some 1733 similar mechanism, or by choosing an excessively large or small value 1734 for the RTP/AVPF minimal receiver report interval (if using SDP, this 1735 is the "a=rtcp-fb:... trr-int" parameter) [RFC4585]. The risks are 1736 as follows: 1738 1. the RTCP bandwidth could be configured to make the regular 1739 reporting interval so large that effective congestion control 1740 cannot be maintained, potentially leading to denial of service 1741 due to congestion caused by the media traffic; 1743 2. the RTCP interval could be configured to a very small value, 1744 causing endpoints to generate high rate RTCP traffic, potentially 1745 leading to denial of service due to the non-congestion controlled 1746 RTCP traffic; and 1748 3. RTCP parameters could be configured differently for each 1749 endpoint, with some of the endpoints using a large reporting 1750 interval and some using a smaller interval, leading to denial of 1751 service due to premature participant timeouts due to mismatched 1752 timeout periods which are based on the reporting interval (this 1753 is a particular concern if endpoints use a small but non-zero 1754 value for the RTP/AVPF minimal receiver report interval (trr-int) 1755 [RFC4585], as discussed in Section 6.1 of 1756 [I-D.ietf-avtcore-rtp-multi-stream]). 1758 Premature participant timeout can be avoided by using the fixed (non- 1759 reduced) minimum interval when calculating the participant timeout 1760 (see Section 4.1 of this memo and Section 6.1 of 1762 [I-D.ietf-avtcore-rtp-multi-stream]). To address the other concerns, 1763 endpoints SHOULD ignore parameters that configure the RTCP reporting 1764 interval to be significantly longer than the default five second 1765 interval specified in [RFC3550] (unless the media data rate is so low 1766 that the longer reporting interval roughly corresponds to 5% of the 1767 media data rate), or that configure the RTCP reporting interval small 1768 enough that the RTCP bandwidth would exceed the media bandwidth. 1770 The guidelines in [RFC6562] apply when using variable bit rate (VBR) 1771 audio codecs such as Opus (see Section 4.3 for discussion of mandated 1772 audio codecs). The guidelines in [RFC6562] also apply, but are of 1773 lesser importance, when using the client-to-mixer audio level header 1774 extensions (Section 5.2.2) or the mixer-to-client audio level header 1775 extensions (Section 5.2.3). The use of the encryption of the header 1776 extensions are RECOMMENDED, unless there are known reasons, like RTP 1777 middleboxes or third party monitoring that will greatly benefit from 1778 the information, and this has been expressed using API or signalling. 1779 If further evidence are produced to show that information leakage is 1780 significant from audio level indications, then use of encryption 1781 needs to be mandated at that time. 1783 14. IANA Considerations 1785 This memo makes no request of IANA. 1787 Note to RFC Editor: this section is to be removed on publication as 1788 an RFC. 1790 15. Acknowledgements 1792 The authors would like to thank Bernard Aboba, Harald Alvestrand, 1793 Cary Bran, Ben Campbell, Charles Eckel, Alex Eleftheriadis, Christian 1794 Groves, Cullen Jennings, Olle Johansson, Suhas Nandakumar, Dan 1795 Romascanu, Jim Spring, Martin Thomson, and the other members of the 1796 IETF RTCWEB working group for their valuable feedback. 1798 16. References 1800 16.1. Normative References 1802 [I-D.ietf-avtcore-multi-media-rtp-session] 1803 Westerlund, M., Perkins, C., and J. Lennox, "Sending 1804 Multiple Types of Media in a Single RTP Session", draft- 1805 ietf-avtcore-multi-media-rtp-session-06 (work in 1806 progress), October 2014. 1808 [I-D.ietf-avtcore-rtp-circuit-breakers] 1809 Perkins, C. and V. Singh, "Multimedia Congestion Control: 1810 Circuit Breakers for Unicast RTP Sessions", draft-ietf- 1811 avtcore-rtp-circuit-breakers-06 (work in progress), July 1812 2014. 1814 [I-D.ietf-avtcore-rtp-multi-stream-optimisation] 1815 Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, 1816 "Sending Multiple Media Streams in a Single RTP Session: 1817 Grouping RTCP Reception Statistics and Other Feedback ", 1818 draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work 1819 in progress), July 2013. 1821 [I-D.ietf-avtcore-rtp-multi-stream] 1822 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, 1823 "Sending Multiple Media Streams in a Single RTP Session", 1824 draft-ietf-avtcore-rtp-multi-stream-05 (work in progress), 1825 July 2014. 1827 [I-D.ietf-rtcweb-security-arch] 1828 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 1829 rtcweb-security-arch-10 (work in progress), July 2014. 1831 [I-D.ietf-rtcweb-security] 1832 Rescorla, E., "Security Considerations for WebRTC", draft- 1833 ietf-rtcweb-security-07 (work in progress), July 2014. 1835 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1836 Requirement Levels", BCP 14, RFC 2119, March 1997. 1838 [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP 1839 Payload Format Specifications", BCP 36, RFC 2736, December 1840 1999. 1842 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1843 Jacobson, "RTP: A Transport Protocol for Real-Time 1844 Applications", STD 64, RFC 3550, July 2003. 1846 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 1847 Video Conferences with Minimal Control", STD 65, RFC 3551, 1848 July 2003. 1850 [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth 1851 Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 1852 3556, July 2003. 1854 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 1855 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 1856 RFC 3711, March 2004. 1858 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1859 Description Protocol", RFC 4566, July 2006. 1861 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 1862 "Extended RTP Profile for Real-time Transport Control 1863 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 1864 2006. 1866 [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. 1867 Hakenberg, "RTP Retransmission Payload Format", RFC 4588, 1868 July 2006. 1870 [RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", 1871 BCP 131, RFC 4961, July 2007. 1873 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 1874 "Codec Control Messages in the RTP Audio-Visual Profile 1875 with Feedback (AVPF)", RFC 5104, February 2008. 1877 [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for 1878 Real-time Transport Control Protocol (RTCP)-Based Feedback 1879 (RTP/SAVPF)", RFC 5124, February 2008. 1881 [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP 1882 Header Extensions", RFC 5285, July 2008. 1884 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size 1885 Real-Time Transport Control Protocol (RTCP): Opportunities 1886 and Consequences", RFC 5506, April 2009. 1888 [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and 1889 Control Packets on a Single Port", RFC 5761, April 2010. 1891 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1892 Security (DTLS) Extension to Establish Keys for the Secure 1893 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 1895 [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP 1896 Flows", RFC 6051, November 2010. 1898 [RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time 1899 Transport Protocol (RTP) Header Extension for Client-to- 1900 Mixer Audio Level Indication", RFC 6464, December 2011. 1902 [RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time 1903 Transport Protocol (RTP) Header Extension for Mixer-to- 1904 Client Audio Level Indication", RFC 6465, December 2011. 1906 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of 1907 Variable Bit Rate Audio with Secure RTP", RFC 6562, March 1908 2012. 1910 [RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure 1911 Real-time Transport Protocol (SRTP)", RFC 6904, April 1912 2013. 1914 [RFC7007] Terriberry, T., "Update to Remove DVI4 from the 1915 Recommended Codecs for the RTP Profile for Audio and Video 1916 Conferences with Minimal Control (RTP/AVP)", RFC 7007, 1917 August 2013. 1919 [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, 1920 "Guidelines for Choosing RTP Control Protocol (RTCP) 1921 Canonical Names (CNAMEs)", RFC 7022, September 2013. 1923 [RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple 1924 Clock Rates in an RTP Session", RFC 7160, April 2014. 1926 [RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC 1927 7164, March 2014. 1929 16.2. Informative References 1931 [I-D.ietf-avtcore-multiplex-guidelines] 1932 Westerlund, M., Perkins, C., and H. Alvestrand, 1933 "Guidelines for using the Multiplexing Features of RTP to 1934 Support Multiple Media Streams", draft-ietf-avtcore- 1935 multiplex-guidelines-03 (work in progress), October 2014. 1937 [I-D.ietf-avtcore-rtp-topologies-update] 1938 Westerlund, M. and S. Wenger, "RTP Topologies", draft- 1939 ietf-avtcore-rtp-topologies-update-04 (work in progress), 1940 August 2014. 1942 [I-D.ietf-avtext-rtp-grouping-taxonomy] 1943 Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro, 1944 "A Taxonomy of Grouping Semantics and Mechanisms for Real- 1945 Time Transport Protocol (RTP) Sources", draft-ietf-avtext- 1946 rtp-grouping-taxonomy-02 (work in progress), June 2014. 1948 [I-D.ietf-mmusic-msid] 1949 Alvestrand, H., "WebRTC MediaStream Identification in the 1950 Session Description Protocol", draft-ietf-mmusic-msid-07 1951 (work in progress), October 2014. 1953 [I-D.ietf-mmusic-sdp-bundle-negotiation] 1954 Holmberg, C., Alvestrand, H., and C. Jennings, 1955 "Negotiating Media Multiplexing Using the Session 1956 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 1957 negotiation-12 (work in progress), October 2014. 1959 [I-D.ietf-payload-rtp-howto] 1960 Westerlund, M., "How to Write an RTP Payload Format", 1961 draft-ietf-payload-rtp-howto-13 (work in progress), 1962 January 2014. 1964 [I-D.ietf-rmcat-cc-requirements] 1965 Jesup, R., "Congestion Control Requirements For RMCAT", 1966 draft-ietf-rmcat-cc-requirements-06 (work in progress), 1967 October 2014. 1969 [I-D.ietf-rtcweb-audio] 1970 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 1971 Requirements", draft-ietf-rtcweb-audio-07 (work in 1972 progress), October 2014. 1974 [I-D.ietf-rtcweb-overview] 1975 Alvestrand, H., "Overview: Real Time Protocols for 1976 Browser-based Applications", draft-ietf-rtcweb-overview-12 1977 (work in progress), October 2014. 1979 [I-D.ietf-rtcweb-use-cases-and-requirements] 1980 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 1981 Time Communication Use-cases and Requirements", draft- 1982 ietf-rtcweb-use-cases-and-requirements-14 (work in 1983 progress), February 2014. 1985 [I-D.ietf-tsvwg-rtcweb-qos] 1986 Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. 1987 Polk, "DSCP and other packet markings for RTCWeb QoS", 1988 draft-ietf-tsvwg-rtcweb-qos-02 (work in progress), June 1989 2014. 1991 [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control 1992 Protocol Extended Reports (RTCP XR)", RFC 3611, November 1993 2003. 1995 [RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient 1996 Stream Loss-Tolerant Authentication (TESLA) in the Secure 1997 Real-time Transport Protocol (SRTP)", RFC 4383, February 1998 2006. 2000 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 2001 (ICE): A Protocol for Network Address Translator (NAT) 2002 Traversal for Offer/Answer Protocols", RFC 5245, April 2003 2010. 2005 [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific 2006 Media Attributes in the Session Description Protocol 2007 (SDP)", RFC 5576, June 2009. 2009 [RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP 2010 Control Protocol (RTCP)", RFC 5968, September 2010. 2012 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 2013 Keeping Alive the NAT Mappings Associated with RTP / RTP 2014 Control Protocol (RTCP) Flows", RFC 6263, June 2011. 2016 [RFC6792] Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the 2017 RTP Monitoring Framework", RFC 6792, November 2012. 2019 [W3C.WD-mediacapture-streams-20130903] 2020 Burnett, D., Bergkvist, A., Jennings, C., and A. 2021 Narayanan, "Media Capture and Streams", World Wide Web 2022 Consortium WD WD-mediacapture-streams-20130903, September 2023 2013, . 2026 [W3C.WD-webrtc-20130910] 2027 Bergkvist, A., Burnett, D., Jennings, C., and A. 2028 Narayanan, "WebRTC 1.0: Real-time Communication Between 2029 Browsers", World Wide Web Consortium WD WD- 2030 webrtc-20130910, September 2013, 2031 . 2033 Authors' Addresses 2035 Colin Perkins 2036 University of Glasgow 2037 School of Computing Science 2038 Glasgow G12 8QQ 2039 United Kingdom 2041 Email: csp@csperkins.org 2042 URI: http://csperkins.org/ 2043 Magnus Westerlund 2044 Ericsson 2045 Farogatan 6 2046 SE-164 80 Kista 2047 Sweden 2049 Phone: +46 10 714 82 87 2050 Email: magnus.westerlund@ericsson.com 2052 Joerg Ott 2053 Aalto University 2054 School of Electrical Engineering 2055 Espoo 02150 2056 Finland 2058 Email: jorg.ott@aalto.fi