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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: February 26, 2015 Ericsson 6 J. Ott 7 Aalto University 8 August 25, 2014 10 Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 11 draft-ietf-rtcweb-rtp-usage-17 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 February 26, 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-aware end- 127 points, 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 that can be used in the 142 WebRTC context. Section 9 gives some guidelines for future 143 incorporation of other RTP and RTP Control Protocol (RTCP) extensions 144 into this framework. Section 10 describes requirements placed on the 145 signalling channel. Section 11 discusses the relationship between 146 features of the RTP framework and the WebRTC application programming 147 interface (API), and Section 12 discusses RTP implementation 148 considerations. The memo concludes with security considerations 149 (Section 13) and IANA considerations (Section 14). 151 2. Rationale 153 The RTP framework comprises the RTP data transfer protocol, the RTP 154 control protocol, and numerous RTP payload formats, profiles, and 155 extensions. This range of add-ons has allowed RTP to meet various 156 needs that were not envisaged by the original protocol designers, and 157 to support many new media encodings, but raises the question of what 158 extensions are to be supported by new implementations. The 159 development of the WebRTC framework provides an opportunity to review 160 the available RTP features and extensions, and to define a common 161 baseline feature set for all WebRTC implementations of RTP. This 162 builds on the past 20 years development of RTP to mandate the use of 163 extensions that have shown widespread utility, while still remaining 164 compatible with the wide installed base of RTP implementations where 165 possible. 167 RTP and RTCP extensions that are not discussed in this document can 168 be implemented by WebRTC end-points if they are beneficial for new 169 use cases. However, they are not necessary to address the WebRTC use 170 cases and requirements identified in 171 [I-D.ietf-rtcweb-use-cases-and-requirements]. 173 While the baseline set of RTP features and extensions defined in this 174 memo is targeted at the requirements of the WebRTC framework, it is 175 expected to be broadly useful for other conferencing-related uses of 176 RTP. In particular, it is likely that this set of RTP features and 177 extensions will be appropriate for other desktop or mobile video 178 conferencing systems, or for room-based high-quality telepresence 179 applications. 181 3. Terminology 183 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 184 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 185 document are to be interpreted as described in [RFC2119]. The RFC 186 2119 interpretation of these key words applies only when written in 187 ALL CAPS. Lower- or mixed-case uses of these key words are not to be 188 interpreted as carrying special significance in this memo. 190 We define the following additional terms: 192 WebRTC MediaStream: The MediaStream concept defined by the W3C in 193 the WebRTC API [W3C.WD-mediacapture-streams-20130903]. 195 Transport-layer Flow: A uni-directional flow of transport packets 196 that are identified by having a particular 5-tuple of source IP 197 address, source port, destination IP address, destination port, 198 and transport protocol used. 200 Bi-directional Transport-layer Flow: A bi-directional transport- 201 layer flow is a transport-layer flow that is symmetric. That is, 202 the transport-layer flow in the reverse direction has a 5-tuple 203 where the source and destination address and ports are swapped 204 compared to the forward path transport-layer flow, and the 205 transport protocol is the same. 207 This document uses the terminology from 208 [I-D.ietf-avtext-rtp-grouping-taxonomy]. Other terms are used 209 according to their definitions from the RTP Specification [RFC3550]. 210 Especially note the following frequently used terms: RTP Packet 211 Stream, RTP Session, and End-point. 213 4. WebRTC Use of RTP: Core Protocols 215 The following sections describe the core features of RTP and RTCP 216 that need to be implemented, along with the mandated RTP profiles. 217 Also described are the core extensions providing essential features 218 that all WebRTC implementations need to implement to function 219 effectively on today's networks. 221 4.1. RTP and RTCP 223 The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be 224 implemented as the media transport protocol for WebRTC. RTP itself 225 comprises two parts: the RTP data transfer protocol, and the RTP 226 control protocol (RTCP). RTCP is a fundamental and integral part of 227 RTP, and MUST be implemented and used in all WebRTC applications. 229 The following RTP and RTCP features are sometimes omitted in limited 230 functionality implementations of RTP, but are REQUIRED in all WebRTC 231 implementations: 233 o Support for use of multiple simultaneous SSRC values in a single 234 RTP session, including support for RTP end-points that send many 235 SSRC values simultaneously, following [RFC3550] and 236 [I-D.ietf-avtcore-rtp-multi-stream]. The RTCP optimisations for 237 multi-SSRC sessions defined in 238 [I-D.ietf-avtcore-rtp-multi-stream-optimisation] MAY be supported; 239 if supported the usage MUST be signalled. 241 o Random choice of SSRC on joining a session; collision detection 242 and resolution for SSRC values (see also Section 4.8). 244 o Support for reception of RTP data packets containing CSRC lists, 245 as generated by RTP mixers, and RTCP packets relating to CSRCs. 247 o Sending correct synchronisation information in the RTCP Sender 248 Reports, to allow receivers to implement lip-synchronisation; see 249 Section 5.2.1 regarding support for the rapid RTP synchronisation 250 extensions. 252 o Support for multiple synchronisation contexts. Participants that 253 send multiple simultaneous RTP packet streams SHOULD do so as part 254 of a single synchronisation context, using a single RTCP CNAME for 255 all streams and allowing receivers to play the streams out in a 256 synchronised manner. For compatibility with potential future 257 versions of this specification, or for interoperability with non- 258 WebRTC devices through a gateway, receivers MUST support multiple 259 synchronisation contexts, indicated by the use of multiple RTCP 260 CNAMEs in an RTP session. This specification requires the usage 261 of a single CNAME when sending RTP Packet Streams in some 262 circumstances, see Section 4.9. 264 o Support for sending and receiving RTCP SR, RR, SDES, and BYE 265 packet types, with OPTIONAL support for other RTCP packet types 266 unless mandated by other parts of this specification. Note that 267 additional RTCP Packet types are used by the RTP/SAVPF Profile 268 (Section 4.2) and the other RTCP extensions (Section 5). 270 o Support for multiple end-points in a single RTP session, and for 271 scaling the RTCP transmission interval according to the number of 272 participants in the session; support for randomised RTCP 273 transmission intervals to avoid synchronisation of RTCP reports; 274 support for RTCP timer reconsideration (Section 6.3.6 of 275 [RFC3550]) and reverse reconsideration (Section 6.3.4 of 276 [RFC3550]). 278 o Support for configuring the RTCP bandwidth as a fraction of the 279 media bandwidth, and for configuring the fraction of the RTCP 280 bandwidth allocated to senders, e.g., using the SDP "b=" line 281 [RFC4566][RFC3556]. 283 o Support for the reduced minimum RTCP reporting interval described 284 in Section 6.2 of [RFC3550] is REQUIRED. When using the reduced 285 minimum RTCP reporting interval, the fixed (non-reduced) minimum 286 interval MUST be used when calculating the participant timeout 287 interval (see Sections 6.2 and 6.3.5 of [RFC3550]). The delay 288 before sending the initial compound RTCP packet can be set to zero 289 (see Section 6.2 of [RFC3550] as updated by 290 [I-D.ietf-avtcore-rtp-multi-stream]). 292 o Support for discontinuous transmission. RTP allows endpoints to 293 pause and resume transmission at any time. When resuming, the RTP 294 sequence number will increase by one, as usual, while the increase 295 in the RTP timestamp value will depend on the duration of the 296 pause. Discontinuous transmission is most commonly used with some 297 audio payload formats, but is not audio specific, and can be used 298 with any RTP payload format. 300 o Ignore unknown RTCP packet types and RTP header extensions. This 301 to ensure robust handling of future extensions, middlebox 302 behaviours, etc., that can result in not signalled RTCP packet 303 types or RTP header extensions being received. If a compound RTCP 304 packet is received that contains a mixture of known and unknown 305 RTCP packet types, the known packets types need to be processed as 306 usual, with only the unknown packet types being discarded. 308 It is known that a significant number of legacy RTP implementations, 309 especially those targeted at VoIP-only systems, do not support all of 310 the above features, and in some cases do not support RTCP at all. 311 Implementers are advised to consider the requirements for graceful 312 degradation when interoperating with legacy implementations. 314 Other implementation considerations are discussed in Section 12. 316 4.2. Choice of the RTP Profile 318 The complete specification of RTP for a particular application domain 319 requires the choice of an RTP Profile. For WebRTC use, the Extended 320 Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF) [RFC5124], as 321 extended by [RFC7007], MUST be implemented. The RTP/SAVPF profile is 322 the combination of basic RTP/AVP profile [RFC3551], the RTP profile 323 for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP 324 profile (RTP/SAVP) [RFC3711]. 326 The RTCP-based feedback extensions [RFC4585] are needed for the 327 improved RTCP timer model. This allows more flexible transmission of 328 RTCP packets in response to events, rather than strictly according to 329 bandwidth, and is vital for being able to report congestion signals 330 as well as media events. These extensions also allow saving RTCP 331 bandwidth, and an end-point will commonly only use the full RTCP 332 bandwidth allocation if there are many events that require feedback. 333 The timer rules are also needed to make use of the RTP conferencing 334 extensions discussed in Section 5.1. 336 Note: The enhanced RTCP timer model defined in the RTP/AVPF 337 profile is backwards compatible with legacy systems that implement 338 only the RTP/AVP or RTP/SAVP profile, given some constraints on 339 parameter configuration such as the RTCP bandwidth value and "trr- 340 int" (the most important factor for interworking with RTP/(S)AVP 341 end-points via a gateway is to set the trr-int parameter to a 342 value representing 4 seconds, see Section 6.1 in 343 [I-D.ietf-avtcore-rtp-multi-stream]). 345 The secure RTP (SRTP) profile extensions [RFC3711] are needed to 346 provide media encryption, integrity protection, replay protection and 347 a limited form of source authentication. WebRTC implementations MUST 348 NOT send packets using the basic RTP/AVP profile or the RTP/AVPF 349 profile; they MUST employ the full RTP/SAVPF profile to protect all 350 RTP and RTCP packets that are generated (i.e., implementations MUST 351 use SRTP and SRTCP). The RTP/SAVPF profile MUST be configured using 352 the cipher suites, DTLS-SRTP protection profiles, keying mechanisms, 353 and other parameters described in [I-D.ietf-rtcweb-security-arch]. 355 4.3. Choice of RTP Payload Formats 357 The set of mandatory to implement codecs and RTP payload formats for 358 WebRTC is not specified in this memo, instead they are defined in 359 separate specifications, such as [I-D.ietf-rtcweb-audio]. 360 Implementations can support any codec for which an RTP payload format 361 and associated signalling is defined. Implementation cannot assume 362 that the other participants in an RTP session understand any RTP 363 payload format, no matter how common; the mapping between RTP payload 364 type numbers and specific configurations of particular RTP payload 365 formats MUST be agreed before those payload types/formats can be 366 used. In an SDP context, this can be done using the "a=rtpmap:" and 367 "a=fmtp:" attributes associated with an "m=" line, along with any 368 other SDP attributes needed to configure the RTP payload format. 370 End-points can signal support for multiple RTP payload formats, or 371 multiple configurations of a single RTP payload format, as long as 372 each unique RTP payload format configuration uses a different RTP 373 payload type number. As outlined in Section 4.8, the RTP payload 374 type number is sometimes used to associate an RTP packet stream with 375 a signalling context. This association is possible provided unique 376 RTP payload type numbers are used in each context. For example, an 377 RTP packet stream can be associated with an SDP "m=" line by 378 comparing the RTP payload type numbers used by the RTP packet stream 379 with payload types signalled in the "a=rtpmap:" lines in the media 380 sections of the SDP. This leads to the following considerations: 382 If RTP packet streams are being associated with signalling 383 contexts based on the RTP payload type, then the assignment of RTP 384 payload type numbers MUST be unique across signalling contexts. 386 If the same RTP payload format configuration is used in multiple 387 contexts, then a different RTP payload type number has to be 388 assigned in each context to ensure uniqueness. 390 If the RTP payload type number is not being used to associate RTP 391 packet streams with a signalling context, then the same RTP 392 payload type number can be used to indicate the exact same RTP 393 payload format configuration in multiple contexts. 395 A single RTP payload type number MUST NOT be assigned to different 396 RTP payload formats, or different configurations of the same RTP 397 payload format, within a single RTP session (note that the "m=" lines 398 in an SDP bundle group [I-D.ietf-mmusic-sdp-bundle-negotiation] form 399 a single RTP session). 401 An end-point that has signalled support for multiple RTP payload 402 formats MUST be able to accept data in any of those payload formats 403 at any time, unless it has previously signalled limitations on its 404 decoding capability. This requirement is constrained if several 405 types of media (e.g., audio and video) are sent in the same RTP 406 session. In such a case, a source (SSRC) is restricted to switching 407 only between the RTP payload formats signalled for the type of media 408 that is being sent by that source; see Section 4.4. To support rapid 409 rate adaptation by changing codec, RTP does not require advance 410 signalling for changes between RTP payload formats used by a single 411 SSRC that were signalled during session set-up. 413 If performing changes between two RTP payload types that use 414 different RTP clock rates, an RTP sender MUST follow the 415 recommendations in Section 4.1 of [RFC7160]. RTP receivers MUST 416 follow the recommendations in Section 4.3 of [RFC7160] in order to 417 support sources that switch between clock rates in an RTP session 418 (these recommendations for receivers are backwards compatible with 419 the case where senders use only a single clock rate). 421 4.4. Use of RTP Sessions 423 An association amongst a set of end-points communicating using RTP is 424 known as an RTP session [RFC3550]. An end-point can be involved in 425 several RTP sessions at the same time. In a multimedia session, each 426 type of media has typically been carried in a separate RTP session 427 (e.g., using one RTP session for the audio, and a separate RTP 428 session using a different transport-layer flow for the video). 429 WebRTC implementations of RTP are REQUIRED to implement support for 430 multimedia sessions in this way, separating each session using 431 different transport-layer flows for compatibility with legacy 432 systems. 434 In modern day networks, however, with the widespread use of network 435 address/port translators (NAT/NAPT) and firewalls, it is desirable to 436 reduce the number of transport-layer flows used by RTP applications. 437 This can be done by sending all the RTP packet streams in a single 438 RTP session, which will comprise a single transport-layer flow (this 439 will prevent the use of some quality-of-service mechanisms, as 440 discussed in Section 12.1.3). Implementations are therefore also 441 REQUIRED to support transport of all RTP packet streams, independent 442 of media type, in a single RTP session using a single transport layer 443 flow, according to [I-D.ietf-avtcore-multi-media-rtp-session]. If 444 multiple types of media are to be used in a single RTP session, all 445 participants in that RTP session MUST agree to this usage. In an SDP 446 context, [I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to 447 signal such a bundle of RTP packet streams forming a single RTP 448 session. 450 Further discussion about the suitability of different RTP session 451 structures and multiplexing methods to different scenarios can be 452 found in [I-D.ietf-avtcore-multiplex-guidelines]. 454 4.5. RTP and RTCP Multiplexing 456 Historically, RTP and RTCP have been run on separate transport layer 457 flows (e.g., two UDP ports for each RTP session, one port for RTP and 458 one port for RTCP). With the increased use of Network Address/Port 459 Translation (NAT/NAPT) this has become problematic, since maintaining 460 multiple NAT bindings can be costly. It also complicates firewall 461 administration, since multiple ports need to be opened to allow RTP 462 traffic. To reduce these costs and session set-up times, 463 implementations are REQUIRED to support multiplexing RTP data packets 464 and RTCP control packets on a single transport-layer flow [RFC5761]. 465 Such RTP and RTCP multiplexing MUST be negotiated in the signalling 466 channel before it is used. If SDP is used for signalling, this 467 negotiation MUST use the attributes defined in [RFC5761]. For 468 backwards compatibility, implementations are also REQUIRED to support 469 RTP and RTCP sent on separate transport-layer flows. 471 Note that the use of RTP and RTCP multiplexed onto a single 472 transport-layer flow ensures that there is occasional traffic sent on 473 that port, even if there is no active media traffic. This can be 474 useful to keep NAT bindings alive [RFC6263]. 476 4.6. Reduced Size RTCP 477 RTCP packets are usually sent as compound RTCP packets, and [RFC3550] 478 requires that those compound packets start with an Sender Report (SR) 479 or Receiver Report (RR) packet. When using frequent RTCP feedback 480 messages under the RTP/AVPF Profile [RFC4585] these statistics are 481 not needed in every packet, and unnecessarily increase the mean RTCP 482 packet size. This can limit the frequency at which RTCP packets can 483 be sent within the RTCP bandwidth share. 485 To avoid this problem, [RFC5506] specifies how to reduce the mean 486 RTCP message size and allow for more frequent feedback. Frequent 487 feedback, in turn, is essential to make real-time applications 488 quickly aware of changing network conditions, and to allow them to 489 adapt their transmission and encoding behaviour. Implementations 490 MUST support sending and receiving non-compound RTCP feedback packets 491 [RFC5506]. Use of non-compound RTCP packets MUST be negotiated using 492 the signalling channel. If SDP is used for signalling, this 493 negotiation MUST use the attributes defined in [RFC5506]. For 494 backwards compatibility, implementations are also REQUIRED to support 495 the use of compound RTCP feedback packets if the remote end-point 496 does not agree to the use of non-compound RTCP in the signalling 497 exchange. 499 4.7. Symmetric RTP/RTCP 501 To ease traversal of NAT and firewall devices, implementations are 502 REQUIRED to implement and use Symmetric RTP [RFC4961]. The reason 503 for using symmetric RTP is primarily to avoid issues with NATs and 504 Firewalls by ensuring that the send and receive RTP packet streams, 505 as well as RTCP, are actually bi-directional transport-layer flows. 506 This will keep alive the NAT and firewall pinholes, and help indicate 507 consent that the receive direction is a transport-layer flow the 508 intended recipient actually wants. In addition, it saves resources, 509 specifically ports at the end-points, but also in the network as NAT 510 mappings or firewall state is not unnecessary bloated. The amount of 511 per flow QoS state kept in the network is also reduced. 513 4.8. Choice of RTP Synchronisation Source (SSRC) 515 Implementations are REQUIRED to support signalled RTP synchronisation 516 source (SSRC) identifiers. If SDP is used, this MUST be done using 517 the "a=ssrc:" SDP attribute defined in Section 4.1 and Section 5 of 518 [RFC5576] and the "previous-ssrc" source attribute defined in 519 Section 6.2 of [RFC5576]; other per-SSRC attributes defined in 520 [RFC5576] MAY be supported. 522 While support for signalled SSRC identifiers is mandated, their use 523 in an RTP session is OPTIONAL. Implementations MUST be prepared to 524 accept RTP and RTCP packets using SSRCs that have not been explicitly 525 signalled ahead of time. Implementations MUST support random SSRC 526 assignment, and MUST support SSRC collision detection and resolution, 527 according to [RFC3550]. When using signalled SSRC values, collision 528 detection MUST be performed as described in Section 5 of [RFC5576]. 530 It is often desirable to associate an RTP packet stream with a non- 531 RTP context. For users of the WebRTC API a mapping between SSRCs and 532 MediaStreamTracks are provided per Section 11. For gateways or other 533 usages it is possible to associate an RTP packet stream with an "m=" 534 line in a session description formatted using SDP. If SSRCs are 535 signalled this is straightforward (in SDP the "a=ssrc:" line will be 536 at the media level, allowing a direct association with an "m=" line). 537 If SSRCs are not signalled, the RTP payload type numbers used in an 538 RTP packet stream are often sufficient to associate that packet 539 stream with a signalling context (e.g., if RTP payload type numbers 540 are assigned as described in Section 4.3 of this memo, the RTP 541 payload types used by an RTP packet stream can be compared with 542 values in SDP "a=rtpmap:" lines, which are at the media level in SDP, 543 and so map to an "m=" line). 545 4.9. Generation of the RTCP Canonical Name (CNAME) 547 The RTCP Canonical Name (CNAME) provides a persistent transport-level 548 identifier for an RTP end-point. While the Synchronisation Source 549 (SSRC) identifier for an RTP end-point can change if a collision is 550 detected, or when the RTP application is restarted, its RTCP CNAME is 551 meant to stay unchanged for the duration of a RTCPeerConnection 552 [W3C.WD-webrtc-20130910], so that RTP end-points can be uniquely 553 identified and associated with their RTP packet streams within a set 554 of related RTP sessions. 556 Each RTP end-point MUST have at least one RTCP CNAME, and that RTCP 557 CNAME MUST be unique within the RTCPeerConnection. RTCP CNAMEs 558 identify a particular synchronisation context, i.e., all SSRCs 559 associated with a single RTCP CNAME share a common reference clock. 560 If an end-point has SSRCs that are associated with several 561 unsynchronised reference clocks, and hence different synchronisation 562 contexts, it will need to use multiple RTCP CNAMEs, one for each 563 synchronisation context. 565 Taking the discussion in Section 11 into account, a WebRTC end-point 566 MUST NOT use more than one RTCP CNAME in the RTP sessions belonging 567 to single RTCPeerConnection (that is, an RTCPeerConnection forms a 568 synchronisation context). RTP middleboxes MAY generate RTP packet 569 streams associated with more than one RTCP CNAME, to allow them to 570 avoid having to resynchronize media from multiple different end- 571 points part of a multi-party RTP session. 573 The RTP specification [RFC3550] includes guidelines for choosing a 574 unique RTP CNAME, but these are not sufficient in the presence of NAT 575 devices. In addition, long-term persistent identifiers can be 576 problematic from a privacy viewpoint (Section 13). Accordingly, a 577 WebRTC endpoint MUST generate a new, unique, short-term persistent 578 RTCP CNAME for each RTCPeerConnection, following [RFC7022], with a 579 single exception; if explicitly requested at creation an 580 RTCPeerConnection MAY use the same CNAME as as an existing 581 RTCPeerConnection within their common same-origin context. 583 An WebRTC end-point MUST support reception of any CNAME that matches 584 the syntax limitations specified by the RTP specification [RFC3550] 585 and cannot assume that any CNAME will be chosen according to the form 586 suggested above. 588 4.10. Handling of Leap Seconds 590 The guidelines regarding handling of leap seconds to limit their 591 impact on RTP media play-out and synchronization given in [RFC7164] 592 SHOULD be followed. 594 5. WebRTC Use of RTP: Extensions 596 There are a number of RTP extensions that are either needed to obtain 597 full functionality, or extremely useful to improve on the baseline 598 performance, in the WebRTC application context. One set of these 599 extensions is related to conferencing, while others are more generic 600 in nature. The following subsections describe the various RTP 601 extensions mandated or suggested for use within the WebRTC context. 603 5.1. Conferencing Extensions and Topologies 605 RTP is a protocol that inherently supports group communication. 606 Groups can be implemented by having each endpoint send its RTP packet 607 streams to an RTP middlebox that redistributes the traffic, by using 608 a mesh of unicast RTP packet streams between endpoints, or by using 609 an IP multicast group to distribute the RTP packet streams. These 610 topologies can be implemented in a number of ways as discussed in 611 [I-D.ietf-avtcore-rtp-topologies-update]. 613 While the use of IP multicast groups is popular in IPTV systems, the 614 topologies based on RTP middleboxes are dominant in interactive video 615 conferencing environments. Topologies based on a mesh of unicast 616 transport-layer flows to create a common RTP session have not seen 617 widespread deployment to date. Accordingly, WebRTC implementations 618 are not expected to support topologies based on IP multicast groups 619 or to support mesh-based topologies, such as a point-to-multipoint 620 mesh configured as a single RTP session (Topo-Mesh in the terminology 621 of [I-D.ietf-avtcore-rtp-topologies-update]). However, a point-to- 622 multipoint mesh constructed using several RTP sessions, implemented 623 in the WebRTC context using independent RTCPeerConnections 624 [W3C.WD-webrtc-20130910], can be expected to be utilised by WebRTC 625 applications and needs to be supported. 627 WebRTC implementations of RTP endpoints implemented according to this 628 memo are expected to 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 the WebRTC environment. 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 end-points. 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 senders MUST understand and 684 react to FIR feedback messages they receive, since this greatly 685 improves the user experience when using centralised mixer-based 686 conferencing. Support for sending FIR messages is OPTIONAL. 688 5.1.2. Picture Loss Indication (PLI) 690 The Picture Loss Indication message is defined in Section 6.3.1 of 691 the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the 692 sending encoder that it lost the decoder context and would like to 693 have it repaired somehow. This is semantically different from the 694 Full Intra Request above as there could be multiple ways to fulfil 695 the request. WebRTC senders MUST understand and react to PLI 696 feedback messages as a loss tolerance mechanism. Receivers MAY send 697 PLI messages. 699 5.1.3. Slice Loss Indication (SLI) 701 The Slice Loss Indication message is defined in Section 6.3.2 of the 702 RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the 703 encoder that it has detected the loss or corruption of one or more 704 consecutive macro blocks, and would like to have these repaired 705 somehow. It is RECOMMENDED that receivers generate SLI feedback 706 messages if slices are lost when using a codec that supports the 707 concept of macro blocks. A sender that receives an SLI feedback 708 message SHOULD attempt to repair the lost slice(s). 710 5.1.4. Reference Picture Selection Indication (RPSI) 712 Reference Picture Selection Indication (RPSI) messages are defined in 713 Section 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video encoding 714 standards allow the use of older reference pictures than the most 715 recent one for predictive coding. If such a codec is in use, and if 716 the encoder has learnt that encoder-decoder synchronisation has been 717 lost, then a known as correct reference picture can be used as a base 718 for future coding. The RPSI message allows this to be signalled. 719 Receivers that detect that encoder-decoder synchronisation has been 720 lost SHOULD generate an RPSI feedback message if codec being used 721 supports reference picture selection. A RTP packet stream sender 722 that receives such an RPSI message SHOULD act on that messages to 723 change the reference picture, if it is possible to do so within the 724 available bandwidth constraints, and with the codec being used. 726 5.1.5. Temporal-Spatial Trade-off Request (TSTR) 728 The temporal-spatial trade-off request and notification are defined 729 in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used 730 to ask the video encoder to change the trade-off it makes between 731 temporal and spatial resolution, for example to prefer high spatial 732 image quality but low frame rate. Support for TSTR requests and 733 notifications is OPTIONAL. 735 5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR) 737 The TMMBR feedback message is defined in Sections 3.5.4 and 4.2.1 of 738 the Codec Control Messages [RFC5104]. This request and its 739 notification message are used by a media receiver to inform the 740 sending party that there is a current limitation on the amount of 741 bandwidth available to this receiver. This can be various reasons 742 for this: for example, an RTP mixer can use this message to limit the 743 media rate of the sender being forwarded by the mixer (without doing 744 media transcoding) to fit the bottlenecks existing towards the other 745 session participants. WebRTC senders are REQUIRED to implement 746 support for TMMBR messages, and MUST follow bandwidth limitations set 747 by a TMMBR message received for their SSRC. The sending of TMMBR 748 requests is OPTIONAL. 750 5.2. Header Extensions 752 The RTP specification [RFC3550] provides the capability to include 753 RTP header extensions containing in-band data, but the format and 754 semantics of the extensions are poorly specified. The use of header 755 extensions is OPTIONAL in the WebRTC context, but if they are used, 756 they MUST be formatted and signalled following the general mechanism 757 for RTP header extensions defined in [RFC5285], since this gives 758 well-defined semantics to RTP header extensions. 760 As noted in [RFC5285], the requirement from the RTP specification 761 that header extensions are "designed so that the header extension may 762 be ignored" [RFC3550] stands. To be specific, header extensions MUST 763 only be used for data that can safely be ignored by the recipient 764 without affecting interoperability, and MUST NOT be used when the 765 presence of the extension has changed the form or nature of the rest 766 of the packet in a way that is not compatible with the way the stream 767 is signalled (e.g., as defined by the payload type). Valid examples 768 of RTP header extensions might include metadata that is additional to 769 the usual RTP information, but that can safely be ignored without 770 compromising interoperability. 772 5.2.1. Rapid Synchronisation 774 Many RTP sessions require synchronisation between audio, video, and 775 other content. This synchronisation is performed by receivers, using 776 information contained in RTCP SR packets, as described in the RTP 777 specification [RFC3550]. This basic mechanism can be slow, however, 778 so it is RECOMMENDED that the rapid RTP synchronisation extensions 779 described in [RFC6051] be implemented in addition to RTCP SR-based 780 synchronisation. The rapid synchronisation extensions use the 781 general RTP header extension mechanism [RFC5285], which requires 782 signalling, but are otherwise backwards compatible. 784 5.2.2. Client-to-Mixer Audio Level 786 The Client to Mixer Audio Level extension [RFC6464] is an RTP header 787 extension used by an endpoint to inform a mixer about the level of 788 audio activity in the packet to which the header is attached. This 789 enables an RTP middlebox to make mixing or selection decisions 790 without decoding or detailed inspection of the payload, reducing the 791 complexity in some types of mixers. It can also save decoding 792 resources in receivers, which can choose to decode only the most 793 relevant RTP packet streams based on audio activity levels. 795 The Client-to-Mixer Audio Level [RFC6464] header extension is 796 RECOMMENDED to be implemented. If this header extension is 797 implemented, it is REQUIRED that implementations are capable of 798 encrypting the header extension according to [RFC6904] since the 799 information contained in these header extensions can be considered 800 sensitive. The use of this encryption is RECOMMENDED, however usage 801 of the encryption can be explicitly disabled through API or 802 signalling. 804 5.2.3. Mixer-to-Client Audio Level 806 The Mixer to Client Audio Level header extension [RFC6465] provides 807 an endpoint with the audio level of the different sources mixed into 808 a common source stream by a RTP mixer. This enables a user interface 809 to indicate the relative activity level of each session participant, 810 rather than just being included or not based on the CSRC field. This 811 is a pure optimisation of non critical functions, and is hence 812 OPTIONAL to implement. If this header extension is implemented, it 813 is REQUIRED that implementations are capable of encrypting the header 814 extension according to [RFC6904] since the information contained in 815 these header extensions can be considered sensitive. It is further 816 RECOMMENDED that this encryption is used, unless the encryption has 817 been explicitly disabled through API or signalling. 819 6. WebRTC Use of RTP: Improving Transport Robustness 821 There are tools that can make RTP packet streams robust against 822 packet loss and reduce the impact of loss on media quality. However, 823 they generally add some overhead compared to a non-robust stream. 824 The overhead needs to be considered, and the aggregate bit-rate MUST 825 be rate controlled to avoid causing network congestion (see 826 Section 7). As a result, improving robustness might require a lower 827 base encoding quality, but has the potential to deliver that quality 828 with fewer errors. The mechanisms described in the following sub- 829 sections can be used to improve tolerance to packet loss. 831 6.1. Negative Acknowledgements and RTP Retransmission 833 As a consequence of supporting the RTP/SAVPF profile, implementations 834 can send negative acknowledgements (NACKs) for RTP data packets 835 [RFC4585]. This feedback can be used to inform a sender of the loss 836 of particular RTP packets, subject to the capacity limitations of the 837 RTCP feedback channel. A sender can use this information to optimise 838 the user experience by adapting the media encoding to compensate for 839 known lost packets. 841 RTP packet stream senders are REQUIRED to understand the Generic NACK 842 message defined in Section 6.2.1 of [RFC4585], but MAY choose to 843 ignore some or all of this feedback (following Section 4.2 of 844 [RFC4585]). Receivers MAY send NACKs for missing RTP packets. 845 Guidelines on when to send NACKs are provided in [RFC4585]. It is 846 not expected that a receiver will send a NACK for every lost RTP 847 packet, rather it needs to consider the cost of sending NACK 848 feedback, and the importance of the lost packet, to make an informed 849 decision on whether it is worth telling the sender about a packet 850 loss event. 852 The RTP Retransmission Payload Format [RFC4588] offers the ability to 853 retransmit lost packets based on NACK feedback. Retransmission needs 854 to be used with care in interactive real-time applications to ensure 855 that the retransmitted packet arrives in time to be useful, but can 856 be effective in environments with relatively low network RTT (an RTP 857 sender can estimate the RTT to the receivers using the information in 858 RTCP SR and RR packets, as described at the end of Section 6.4.1 of 859 [RFC3550]). The use of retransmissions can also increase the forward 860 RTP bandwidth, and can potentially caused increased packet loss if 861 the original packet loss was caused by network congestion. Note, 862 however, that retransmission of an important lost packet to repair 863 decoder state can have lower cost than sending a full intra frame. 864 It is not appropriate to blindly retransmit RTP packets in response 865 to a NACK. The importance of lost packets and the likelihood of them 866 arriving in time to be useful needs to be considered before RTP 867 retransmission is used. 869 Receivers are REQUIRED to implement support for RTP retransmission 870 packets [RFC4588]. Senders MAY send RTP retransmission packets in 871 response to NACKs if the RTP retransmission payload format has been 872 negotiated for the session, and if the sender believes it is useful 873 to send a retransmission of the packet(s) referenced in the NACK. An 874 RTP sender does not need to retransmit every NACKed packet. 876 6.2. Forward Error Correction (FEC) 878 The use of Forward Error Correction (FEC) can provide an effective 879 protection against some degree of packet loss, at the cost of steady 880 bandwidth overhead. There are several FEC schemes that are defined 881 for use with RTP. Some of these schemes are specific to a particular 882 RTP payload format, others operate across RTP packets and can be used 883 with any payload format. It needs to be noted that using redundant 884 encoding or FEC will lead to increased play out delay, which needs to 885 be considered when choosing the redundancy or FEC formats and their 886 respective parameters. 888 If an RTP payload format negotiated for use in a RTCPeerConnection 889 supports redundant transmission or FEC as a standard feature of that 890 payload format, then that support MAY be used in the 891 RTCPeerConnection, subject to any appropriate signalling. 893 There are several block-based FEC schemes that are designed for use 894 with RTP independent of the chosen RTP payload format. At the time 895 of this writing there is no consensus on which, if any, of these FEC 896 schemes is appropriate for use in the WebRTC context. Accordingly, 897 this memo makes no recommendation on the choice of block-based FEC 898 for WebRTC use. 900 7. WebRTC Use of RTP: Rate Control and Media Adaptation 902 WebRTC will be used in heterogeneous network environments using a 903 variety set of link technologies, including both wired and wireless 904 links, to interconnect potentially large groups of users around the 905 world. As a result, the network paths between users can have widely 906 varying one-way delays, available bit-rates, load levels, and traffic 907 mixtures. Individual end-points can send one or more RTP packet 908 streams to each participant in a WebRTC conference, and there can be 909 several participants. Each of these RTP packet streams can contain 910 different types of media, and the type of media, bit rate, and number 911 of RTP packet streams as well as transport-layer flows can be highly 912 asymmetric. Non-RTP traffic can share the network paths with RTP 913 transport-layer flows. Since the network environment is not 914 predictable or stable, WebRTC end-points MUST ensure that the RTP 915 traffic they generate can adapt to match changes in the available 916 network capacity. 918 The quality of experience for users of WebRTC implementation is very 919 dependent on effective adaptation of the media to the limitations of 920 the network. End-points have to be designed so they do not transmit 921 significantly more data than the network path can support, except for 922 very short time periods, otherwise high levels of network packet loss 923 or delay spikes will occur, causing media quality degradation. The 924 limiting factor on the capacity of the network path might be the link 925 bandwidth, or it might be competition with other traffic on the link 926 (this can be non-WebRTC traffic, traffic due to other WebRTC flows, 927 or even competition with other WebRTC flows in the same session). 929 An effective media congestion control algorithm is therefore an 930 essential part of the WebRTC framework. However, at the time of this 931 writing, there is no standard congestion control algorithm that can 932 be used for interactive media applications such as WebRTC's flows. 933 Some requirements for congestion control algorithms for 934 RTCPeerConnections are discussed in [I-D.ietf-rmcat-cc-requirements]. 935 A future version of this memo will mandate the use of a congestion 936 control algorithm that satisfies these requirements. 938 7.1. Boundary Conditions and Circuit Breakers 940 WebRTC implementations MUST implement the RTP circuit breaker 941 algorithm that is described in 942 [I-D.ietf-avtcore-rtp-circuit-breakers]. The RTP circuit breaker is 943 designed to enable applications to recognise and react to situations 944 of extreme network congestion. However, since the RTP circuit 945 breaker might not be triggered until congestion becomes extreme, it 946 cannot be considered a substitute for congestion control, and 947 applications MUST also implement congestion control to allow them to 948 adapt to changes in network capacity. Any future RTP congestion 949 control algorithms are expected to operate within the envelope 950 allowed by the circuit breaker. 952 The session establishment signalling will also necessarily establish 953 boundaries to which the media bit-rate will conform. The choice of 954 media codecs provides upper- and lower-bounds on the supported bit- 955 rates that the application can utilise to provide useful quality, and 956 the packetisation choices that exist. In addition, the signalling 957 channel can establish maximum media bit-rate boundaries using, for 958 example, the SDP "b=AS:" or "b=CT:" lines and the RTP/AVPF Temporary 959 Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of 960 this memo). Signalled bandwidth limitations, such as SDP "b=AS:" or 961 "b=CT:" lines received from the peer, MUST be followed when sending 962 RTP packet streams. A WebRTC endpoint receiving media SHOULD signal 963 its bandwidth limitations, these limitations have to be based on 964 known bandwidth limitations, for example the capacity of the edge 965 links. 967 7.2. Congestion Control Interoperability and Legacy Systems 969 There are legacy RTP implementations that do not implement RTCP, and 970 hence do not provide any congestion feedback. Congestion control 971 cannot be performed with these end-points. WebRTC implementations 972 that need to interwork with such end-points MUST limit their 973 transmission to a low rate, equivalent to a VoIP call using a low 974 bandwidth codec, that is unlikely to cause any significant 975 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 the WebRTC context, so there is no requirement that 1023 implementations generate RTCP XR packets. However, implementations 1024 that can use detailed performance monitoring data MAY generate RTCP 1025 XR packets as appropriate; the use of such packets SHOULD be 1026 signalled in advance. 1028 9. WebRTC Use of RTP: Future Extensions 1030 It is possible that the core set of RTP protocols and RTP extensions 1031 specified in this memo will prove insufficient for the future needs 1032 of WebRTC applications. In this case, future updates to this memo 1033 MUST be made following the Guidelines for Writers of RTP Payload 1034 Format Specifications [RFC2736], How to Write an RTP Payload Format 1035 [I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP 1036 Control Protocol [RFC5968], and SHOULD take into account any future 1037 guidelines for extending RTP and related protocols that have been 1038 developed. 1040 Authors of future extensions are urged to consider the wide range of 1041 environments in which RTP is used when recommending extensions, since 1042 extensions that are applicable in some scenarios can be problematic 1043 in others. Where possible, the WebRTC framework will adopt RTP 1044 extensions that are of general utility, to enable easy implementation 1045 of a gateway to other applications using RTP, rather than adopt 1046 mechanisms that are narrowly targeted at specific WebRTC use cases. 1048 10. Signalling Considerations 1050 RTP is built with the assumption that an external signalling channel 1051 exists, and can be used to configure RTP sessions and their features. 1052 The basic configuration of an RTP session consists of the following 1053 parameters: 1055 RTP Profile: The name of the RTP profile to be used in session. The 1056 RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate 1057 on basic level, as can their secure variants RTP/SAVP [RFC3711] 1058 and RTP/SAVPF [RFC5124]. The secure variants of the profiles do 1059 not directly interoperate with the non-secure variants, due to the 1060 presence of additional header fields for authentication in SRTP 1061 packets and cryptographic transformation of the payload. WebRTC 1062 requires the use of the RTP/SAVPF profile, and this MUST be 1063 signalled. Interworking functions might transform this into the 1064 RTP/SAVP profile for a legacy use case, by indicating to the 1065 WebRTC end-point that the RTP/SAVPF is used and configuring a trr- 1066 int value of 4 seconds. 1068 Transport Information: Source and destination IP address(s) and 1069 ports for RTP and RTCP MUST be signalled for each RTP session. In 1070 WebRTC these transport addresses will be provided by ICE [RFC5245] 1071 that signals candidates and arrives at nominated candidate address 1072 pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such 1073 that a single port, i.e. transport-layer flow, is used for RTP 1074 and RTCP flows, this MUST be signalled (see Section 4.5). 1076 RTP Payload Types, media formats, and format parameters: The mapping 1077 between media type names (and hence the RTP payload formats to be 1078 used), and the RTP payload type numbers MUST be signalled. Each 1079 media type MAY also have a number of media type parameters that 1080 MUST also be signalled to configure the codec and RTP payload 1081 format (the "a=fmtp:" line from SDP). Section 4.3 of this memo 1082 discusses requirements for uniqueness of payload types. 1084 RTP Extensions: The use of any additional RTP header extensions and 1085 RTCP packet types, including any necessary parameters, MUST be 1086 signalled. This signalling is to ensure that a WebRTC endpoint's 1087 behaviour, especially when sending, of any extensions is 1088 predictable and consistent. For robustness, and for compatibility 1089 with non-WebRTC systems that might be connected to a WebRTC 1090 session via a gateway, implementations are REQUIRED to ignore 1091 unknown RTCP packets and RTP header extensions (see also 1092 Section 4.1). 1094 RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the 1095 end-points will be necessary. This SHALL be done as described in 1096 "Session Description Protocol (SDP) Bandwidth Modifiers for RTP 1097 Control Protocol (RTCP) Bandwidth" [RFC3556] if using SDP, or 1098 something semantically equivalent. This also ensures that the 1099 end-points have a common view of the RTCP bandwidth. A common 1100 RTCP bandwidth is important as a too different view of the 1101 bandwidths can lead to failure to interoperate. 1103 These parameters are often expressed in SDP messages conveyed within 1104 an offer/answer exchange. RTP does not depend on SDP or on the offer 1105 /answer model, but does require all the necessary parameters to be 1106 agreed upon, and provided to the RTP implementation. Note that in 1107 the WebRTC context it will depend on the signalling model and API how 1108 these parameters need to be configured but they will be need to 1109 either be set in the API or explicitly signalled between the peers. 1111 11. WebRTC API Considerations 1113 The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and 1114 Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses 1115 the concept of a MediaStream that consists of zero or more 1116 MediaStreamTracks. A MediaStreamTrack is an individual stream of 1117 media from any type of media source like a microphone or a camera, 1118 but also conceptual sources, like a audio mix or a video composition, 1119 are possible. The MediaStreamTracks within a MediaStream need to be 1120 possible to play out synchronised. 1122 A MediaStreamTrack's realisation in RTP in the context of an 1123 RTCPeerConnection consists of a source packet stream identified with 1124 an SSRC within an RTP session part of the RTCPeerConnection. The 1125 MediaStreamTrack can also result in additional packet streams, and 1126 thus SSRCs, in the same RTP session. These can be dependent packet 1127 streams from scalable encoding of the source stream associated with 1128 the MediaStreamTrack, if such a media encoder is used. They can also 1129 be redundancy packet streams, these are created when applying Forward 1130 Error Correction (Section 6.2) or RTP retransmission (Section 6.1) to 1131 the source packet stream. 1133 It is important to note that the same media source can be feeding 1134 multiple MediaStreamTracks. As different sets of constraints or 1135 other parameters can be applied to the MediaStreamTrack, each 1136 MediaStreamTrack instance added to a RTCPeerConnection SHALL result 1137 in an independent source packet stream, with its own set of 1138 associated packet streams, and thus different SSRC(s). It will 1139 depend on applied constraints and parameters if the source stream and 1140 the encoding configuration will be identical between different 1141 MediaStreamTracks sharing the same media source. If the encoding 1142 parameters and constraints are the same, an implementation could 1143 choose to use only one encoded stream to create the different RTP 1144 packet streams. Note that such optimisations would need to take into 1145 account that the constraints for one of the MediaStreamTracks can at 1146 any moment change, meaning that the encoding configurations might no 1147 longer be identical and two different encoder instances would then be 1148 needed. 1150 The same MediaStreamTrack can also be included in multiple 1151 MediaStreams, thus multiple sets of MediaStreams can implicitly need 1152 to use the same synchronisation base. To ensure that this works in 1153 all cases, and does not force an end-point to to disrupt the media by 1154 changing synchronisation base and CNAME during delivery of any 1155 ongoing packet streams, all MediaStreamTracks and their associated 1156 SSRCs originating from the same end-point need to be sent using the 1157 same CNAME within one RTCPeerConnection. This is motivating the 1158 discussion in Section 4.9 to only use a single CNAME. 1160 The requirement on using the same CNAME for all SSRCs that 1161 originate from the same end-point, does not require a middlebox 1162 that forwards traffic from multiple end-points to only use a 1163 single CNAME. 1165 Different CNAMEs normally need to be used for different 1166 RTCPeerConnection instances, as specified in Section 4.9. Having two 1167 communication sessions with the same CNAME could enable tracking of a 1168 user or device across different services (see Section 4.4.1 of 1169 [I-D.ietf-rtcweb-security] for details). A web application can 1170 request that the CNAMEs used in different RTCPeerConnections (within 1171 a same-orign context) be the same, this allows for synchronization of 1172 the endpoint's RTP packet streams across the different 1173 RTCPeerConnections. 1175 Note: this doesn't result in a tracking issue, since the creation 1176 of matching CNAMEs depends on existing tracking. 1178 The above will currently force a WebRTC end-point that receives a 1179 MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing 1180 on any RTCPeerConnection to perform resynchronisation of the stream. 1181 This, as the sending party needs to change the CNAME to the one it 1182 uses, which implies that the sender has to use a local system clock 1183 as timebase for the synchronisation. Thus, the relative relation 1184 between the timebase of the incoming stream and the system sending 1185 out needs to defined. This relation also needs monitoring for clock 1186 drift and likely adjustments of the synchronisation. The sending 1187 entity is also responsible for congestion control for its sent 1188 streams. In cases of packet loss the loss of incoming data also 1189 needs to be handled. This leads to the observation that the method 1190 that is least likely to cause issues or interruptions in the outgoing 1191 source packet stream is a model of full decoding, including repair 1192 etc., followed by encoding of the media again into the outgoing 1193 packet stream. Optimisations of this method is clearly possible and 1194 implementation specific. 1196 A WebRTC end-point MUST support receiving multiple MediaStreamTracks, 1197 where each of different MediaStreamTracks (and their sets of 1198 associated packet streams) uses different CNAMEs. However, 1199 MediaStreamTracks that are received with different CNAMEs have no 1200 defined synchronisation. 1202 Note: The motivation for supporting reception of multiple CNAMEs 1203 is to allow for forward compatibility with any future changes that 1204 enables more efficient stream handling when end-points relay/ 1205 forward streams. It also ensures that end-points can interoperate 1206 with certain types of multi-stream middleboxes or end-points that 1207 are not WebRTC. 1209 The binding between the WebRTC MediaStreams, MediaStreamTracks and 1210 the SSRC is done as specified in "Cross Session Stream Identification 1211 in the Session Description Protocol" [I-D.ietf-mmusic-msid]. This 1212 document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to 1213 map unknown source packet stream SSRCs to MediaStreamTracks and 1214 MediaStreams. This later is relevant to handle some cases of legacy 1215 interop. Commonly the RTP Payload Type of any incoming packets will 1216 reveal if the packet stream is a source stream or a redundancy or 1217 dependent packet stream. The association to the correct source 1218 packet stream depends on the payload format in use for the packet 1219 stream. 1221 Finally this specification puts a requirement on the WebRTC API to 1222 realize a method for determining the CSRC list (Section 4.1) as well 1223 as the Mixer-to-Client audio levels (Section 5.2.3) (when supported) 1224 and the basic requirements for this is further discussed in 1225 Section 12.2.1. 1227 12. RTP Implementation Considerations 1229 The following discussion provides some guidance on the implementation 1230 of the RTP features described in this memo. The focus is on a WebRTC 1231 end-point implementation perspective, and while some mention is made 1232 of the behaviour of middleboxes, that is not the focus of this memo. 1234 12.1. Configuration and Use of RTP Sessions 1236 A WebRTC end-point will be a simultaneous participant in one or more 1237 RTP sessions. Each RTP session can convey multiple media sources, 1238 and can include media data from multiple end-points. In the 1239 following, some ways in which WebRTC end-points can configure and use 1240 RTP sessions is outlined. 1242 12.1.1. Use of Multiple Media Sources Within an RTP Session 1243 RTP is a group communication protocol, and every RTP session can 1244 potentially contain multiple RTP packet streams. There are several 1245 reasons why this might be desirable: 1247 Multiple media types: Outside of WebRTC, it is common to use one RTP 1248 session for each type of media sources (e.g., one RTP session for 1249 audio sources and one for video sources, each sent over different 1250 transport layer flows). However, to reduce the number of UDP 1251 ports used, the default in WebRTC is to send all types of media in 1252 a single RTP session, as described in Section 4.4, using RTP and 1253 RTCP multiplexing (Section 4.5) to further reduce the number of 1254 UDP ports needed. This RTP session then uses only one bi- 1255 directional transport-layer flow, but will contain multiple RTP 1256 packet streams, each containing a different type of media. A 1257 common example might be an end-point with a camera and microphone 1258 that sends two RTP packet streams, one video and one audio, into a 1259 single RTP session. 1261 Multiple Capture Devices: A WebRTC end-point might have multiple 1262 cameras, microphones, or other media capture devices, and so might 1263 want to generate several RTP packet streams of the same media 1264 type. Alternatively, it might want to send media from a single 1265 capture device in several different formats or quality settings at 1266 once. Both can result in a single end-point sending multiple RTP 1267 packet streams of the same media type into a single RTP session at 1268 the same time. 1270 Associated Repair Data: An end-point might send a RTP packet stream 1271 that is somehow associated with another stream. For example, it 1272 might send an RTP packet stream that contains FEC or 1273 retransmission data relating to another stream. Some RTP payload 1274 formats send this sort of associated repair data as part of the 1275 source packet stream, while others send it as a separate packet 1276 stream. 1278 Layered or Multiple Description Coding: An end-point can use a 1279 layered media codec, for example H.264 SVC, or a multiple 1280 description codec, that generates multiple RTP packet streams, 1281 each with a distinct RTP SSRC, within a single RTP session. 1283 RTP Mixers, Translators, and Other Middleboxes: An RTP session, in 1284 the WebRTC context, is a point-to-point association between an 1285 end-point and some other peer device, where those devices share a 1286 common SSRC space. The peer device might be another WebRTC end- 1287 point, or it might be an RTP mixer, translator, or some other form 1288 of media processing middlebox. In the latter cases, the middlebox 1289 might send mixed or relayed RTP streams from several participants, 1290 that the WebRTC end-point will need to render. Thus, even though 1291 a WebRTC end-point might only be a member of a single RTP session, 1292 the peer device might be extending that RTP session to incorporate 1293 other end-points. WebRTC is a group communication environment and 1294 end-points need to be capable of receiving, decoding, and playing 1295 out multiple RTP packet streams at once, even in a single RTP 1296 session. 1298 12.1.2. Use of Multiple RTP Sessions 1300 In addition to sending and receiving multiple RTP packet streams 1301 within a single RTP session, a WebRTC end-point might participate in 1302 multiple RTP sessions. There are several reasons why a WebRTC end- 1303 point might choose to do this: 1305 To interoperate with legacy devices: The common practice in the non- 1306 WebRTC world is to send different types of media in separate RTP 1307 sessions, for example using one RTP session for audio and another 1308 RTP session, on a separate transport layer flow, for video. All 1309 WebRTC end-points need to support the option of sending different 1310 types of media on different RTP sessions, so they can interwork 1311 with such legacy devices. This is discussed further in 1312 Section 4.4. 1314 To provide enhanced quality of service: Some network-based quality 1315 of service mechanisms operate on the granularity of transport 1316 layer flows. If it is desired to use these mechanisms to provide 1317 differentiated quality of service for some RTP packet streams, 1318 then those RTP packet streams need to be sent in a separate RTP 1319 session using a different transport-layer flow, and with 1320 appropriate quality of service marking. This is discussed further 1321 in Section 12.1.3. 1323 To separate media with different purposes: An end-point might want 1324 to send RTP packet streams that have different purposes on 1325 different RTP sessions, to make it easy for the peer device to 1326 distinguish them. For example, some centralised multiparty 1327 conferencing systems display the active speaker in high 1328 resolution, but show low resolution "thumbnails" of other 1329 participants. Such systems might configure the end-points to send 1330 simulcast high- and low-resolution versions of their video using 1331 separate RTP sessions, to simplify the operation of the RTP 1332 middlebox. In the WebRTC context this is currently possible by 1333 establishing multiple WebRTC MediaStreamTracks that have the same 1334 media source in one (or more) RTCPeerConnection. Each 1335 MediaStreamTrack is then configured to deliver a particular media 1336 quality and thus media bit-rate, and will produce an independently 1337 encoded version with the codec parameters agreed specifically in 1338 the context of that RTCPeerConnection. The RTP middlebox can 1339 distinguish packets corresponding to the low- and high-resolution 1340 streams by inspecting their SSRC, RTP payload type, or some other 1341 information contained in RTP payload, RTP header extension or RTCP 1342 packets, but it can be easier to distinguish the RTP packet 1343 streams if they arrive on separate RTP sessions on separate 1344 transport-layer flows. 1346 To directly connect with multiple peers: A multi-party conference 1347 does not need to use an RTP middlebox. Rather, a multi-unicast 1348 mesh can be created, comprising several distinct RTP sessions, 1349 with each participant sending RTP traffic over a separate RTP 1350 session (that is, using an independent RTCPeerConnection object) 1351 to every other participant, as shown in Figure 1. This topology 1352 has the benefit of not requiring an RTP middlebox node that is 1353 trusted to access and manipulate the media data. The downside is 1354 that it increases the used bandwidth at each sender by requiring 1355 one copy of the RTP packet streams for each participant that are 1356 part of the same session beyond the sender itself. 1358 +---+ +---+ 1359 | A |<--->| B | 1360 +---+ +---+ 1361 ^ ^ 1362 \ / 1363 \ / 1364 v v 1365 +---+ 1366 | C | 1367 +---+ 1369 Figure 1: Multi-unicast using several RTP sessions 1371 The multi-unicast topology could also be implemented as a single 1372 RTP session, spanning multiple peer-to-peer transport layer 1373 connections, or as several pairwise RTP sessions, one between each 1374 pair of peers. To maintain a coherent mapping between the 1375 relation between RTP sessions and RTCPeerConnection objects it is 1376 recommend that this is implemented as several individual RTP 1377 sessions. The only downside is that end-point A will not learn of 1378 the quality of any transmission happening between B and C, since 1379 it will not see RTCP reports for the RTP session between B and C, 1380 whereas it would it all three participants were part of a single 1381 RTP session. Experience with the Mbone tools (experimental RTP- 1382 based multicast conferencing tools from the late 1990s) has showed 1383 that RTCP reception quality reports for third parties can be 1384 presented to users in a way that helps them understand asymmetric 1385 network problems, and the approach of using separate RTP sessions 1386 prevents this. However, an advantage of using separate RTP 1387 sessions is that it enables using different media bit-rates and 1388 RTP session configurations between the different peers, thus not 1389 forcing B to endure the same quality reductions if there are 1390 limitations in the transport from A to C as C will. It is 1391 believed that these advantages outweigh the limitations in 1392 debugging power. 1394 To indirectly connect with multiple peers: A common scenario in 1395 multi-party conferencing is to create indirect connections to 1396 multiple peers, using an RTP mixer, translator, or some other type 1397 of RTP middlebox. Figure 2 outlines a simple topology that might 1398 be used in a four-person centralised conference. The middlebox 1399 acts to optimise the transmission of RTP packet streams from 1400 certain perspectives, either by only sending some of the received 1401 RTP packet stream to any given receiver, or by providing a 1402 combined RTP packet stream out of a set of contributing streams. 1404 +---+ +-------------+ +---+ 1405 | A |<---->| |<---->| B | 1406 +---+ | RTP mixer, | +---+ 1407 | translator, | 1408 | or other | 1409 +---+ | middlebox | +---+ 1410 | C |<---->| |<---->| D | 1411 +---+ +-------------+ +---+ 1413 Figure 2: RTP mixer with only unicast paths 1415 There are various methods of implementation for the middlebox. If 1416 implemented as a standard RTP mixer or translator, a single RTP 1417 session will extend across the middlebox and encompass all the 1418 end-points in one multi-party session. Other types of middlebox 1419 might use separate RTP sessions between each end-point and the 1420 middlebox. A common aspect is that these RTP middleboxes can use 1421 a number of tools to control the media encoding provided by a 1422 WebRTC end-point. This includes functions like requesting the 1423 breaking of the encoding chain and have the encoder produce a so 1424 called Intra frame. Another is limiting the bit-rate of a given 1425 stream to better suit the mixer view of the multiple down-streams. 1426 Others are controlling the most suitable frame-rate, picture 1427 resolution, the trade-off between frame-rate and spatial quality. 1428 The middlebox has the responsibility to correctly perform 1429 congestion control, source identification, manage synchronisation 1430 while providing the application with suitable media optimisations. 1431 The middlebox also has to be a trusted node when it comes to 1432 security, since it manipulates either the RTP header or the media 1433 itself (or both) received from one end-point, before sending it on 1434 towards the end-point(s), thus they need to be able to decrypt and 1435 then re-encrypt the RTP packet stream before sending it out. 1437 RTP Mixers can create a situation where an end-point experiences a 1438 situation in-between a session with only two end-points and 1439 multiple RTP sessions. Mixers are expected to not forward RTCP 1440 reports regarding RTP packet streams across themselves. This is 1441 due to the difference in the RTP packet streams provided to the 1442 different end-points. The original media source lacks information 1443 about a mixer's manipulations prior to sending it the different 1444 receivers. This scenario also results in that an end-point's 1445 feedback or requests goes to the mixer. When the mixer can't act 1446 on this by itself, it is forced to go to the original media source 1447 to fulfil the receivers request. This will not necessarily be 1448 explicitly visible any RTP and RTCP traffic, but the interactions 1449 and the time to complete them will indicate such dependencies. 1451 Providing source authentication in multi-party scenarios is a 1452 challenge. In the mixer-based topologies, end-points source 1453 authentication is based on, firstly, verifying that media comes 1454 from the mixer by cryptographic verification and, secondly, trust 1455 in the mixer to correctly identify any source towards the end- 1456 point. In RTP sessions where multiple end-points are directly 1457 visible to an end-point, all end-points will have knowledge about 1458 each others' master keys, and can thus inject packets claimed to 1459 come from another end-point in the session. Any node performing 1460 relay can perform non-cryptographic mitigation by preventing 1461 forwarding of packets that have SSRC fields that came from other 1462 end-points before. For cryptographic verification of the source, 1463 SRTP would require additional security mechanisms, for example 1464 TESLA for SRTP [RFC4383], that are not part of the base WebRTC 1465 standards. 1467 To forward media between multiple peers: It is sometimes desirable 1468 for an end-point that receives an RTP packet stream to be able to 1469 forward that RTP packet stream to a third party. The are some 1470 obvious security and privacy implications in supporting this, but 1471 also potential uses. This is supported in the W3C API by taking 1472 the received and decoded media and using it as media source that 1473 is re-encoding and transmitted as a new stream. 1475 At the RTP layer, media forwarding acts as a back-to-back RTP 1476 receiver and RTP sender. The receiving side terminates the RTP 1477 session and decodes the media, while the sender side re-encodes 1478 and transmits the media using an entirely separate RTP session. 1479 The original sender will only see a single receiver of the media, 1480 and will not be able to tell that forwarding is happening based on 1481 RTP-layer information since the RTP session that is used to send 1482 the forwarded media is not connected to the RTP session on which 1483 the media was received by the node doing the forwarding. 1485 The end-point that is performing the forwarding is responsible for 1486 producing an RTP packet stream suitable for onwards transmission. 1487 The outgoing RTP session that is used to send the forwarded media 1488 is entirely separate to the RTP session on which the media was 1489 received. This will require media transcoding for congestion 1490 control purpose to produce a suitable bit-rate for the outgoing 1491 RTP session, reducing media quality and forcing the forwarding 1492 end-point to spend the resource on the transcoding. The media 1493 transcoding does result in a separation of the two different legs 1494 removing almost all dependencies, and allowing the forwarding end- 1495 point to optimise its media transcoding operation. The cost is 1496 greatly increased computational complexity on the forwarding node. 1497 Receivers of the forwarded stream will see the forwarding device 1498 as the sender of the stream, and will not be able to tell from the 1499 RTP layer that they are receiving a forwarded stream rather than 1500 an entirely new RTP packet stream generated by the forwarding 1501 device. 1503 12.1.3. Differentiated Treatment of RTP Packet Streams 1505 There are use cases for differentiated treatment of RTP packet 1506 streams. Such differentiation can happen at several places in the 1507 system. First of all is the prioritization within the end-point 1508 sending the media, which controls, both which RTP packet streams that 1509 will be sent, and their allocation of bit-rate out of the current 1510 available aggregate as determined by the congestion control. 1512 It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will 1513 allow the application to indicate relative priorities for different 1514 MediaStreamTracks. These priorities can then be used to influence 1515 the local RTP processing, especially when it comes to congestion 1516 control response in how to divide the available bandwidth between the 1517 RTP packet streams. Any changes in relative priority will also need 1518 to be considered for RTP packet streams that are associated with the 1519 main RTP packet streams, such as redundant streams for RTP 1520 retransmission and FEC. The importance of such redundant RTP packet 1521 streams is dependent on the media type and codec used, in regards to 1522 how robust that codec is to packet loss. However, a default policy 1523 might to be to use the same priority for redundant RTP packet stream 1524 as for the source RTP packet stream. 1526 Secondly, the network can prioritize transport-layer flows and sub- 1527 flows, including RTP packet streams. Typically, differential 1528 treatment includes two steps, the first being identifying whether an 1529 IP packet belongs to a class that has to be treated differently, the 1530 second consisting of the actual mechanism to prioritize packets. 1531 This is done according to three methods: 1533 DiffServ: The end-point marks a packet with a DiffServ code point to 1534 indicate to the network that the packet belongs to a particular 1535 class. 1537 Flow based: Packets that need to be given a particular treatment are 1538 identified using a combination of IP and port address. 1540 Deep Packet Inspection: A network classifier (DPI) inspects the 1541 packet and tries to determine if the packet represents a 1542 particular application and type that is to be prioritized. 1544 Flow-based differentiation will provide the same treatment to all 1545 packets within a transport-layer flow, i.e., relative prioritization 1546 is not possible. Moreover, if the resources are limited it might not 1547 be possible to provide differential treatment compared to best-effort 1548 for all the RTP packet streams in a WebRTC application. When flow- 1549 based differentiation is available the WebRTC application needs to 1550 know about it so that it can provide the separation of the RTP packet 1551 streams onto different UDP flows to enable a more granular usage of 1552 flow based differentiation. That way at least providing different 1553 prioritization of audio and video if desired by application. 1555 DiffServ assumes that either the end-point or a classifier can mark 1556 the packets with an appropriate DSCP so that the packets are treated 1557 according to that marking. If the end-point is to mark the traffic 1558 two requirements arise in the WebRTC context: 1) The WebRTC 1559 application or browser has to know which DSCP to use and that it can 1560 use them on some set of RTP packet streams. 2) The information needs 1561 to be propagated to the operating system when transmitting the 1562 packet. Details of this process are outside the scope of this memo 1563 and are further discussed in "DSCP and other packet markings for 1564 RTCWeb QoS" [I-D.ietf-tsvwg-rtcweb-qos]. 1566 For packet based marking schemes it might be possible to mark 1567 individual RTP packets differently based on the relative priority of 1568 the RTP payload. For example video codecs that have I, P, and B 1569 pictures could prioritise any payloads carrying only B frames less, 1570 as these are less damaging to loose. However, depending on the QoS 1571 mechanism and what markings that are applied, this can result in not 1572 only different packet drop probabilities but also packet reordering, 1573 see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion. As a default 1574 policy all RTP packets related to a RTP packet stream ought to be 1575 provided with the same prioritization; per-packet prioritization is 1576 outside the scope of this memo, but might be specified elsewhere in 1577 future. 1579 It is also important to consider how RTCP packets associated with a 1580 particular RTP packet stream need to be marked. RTCP compound 1581 packets with Sender Reports (SR), ought to be marked with the same 1582 priority as the RTP packet stream itself, so the RTCP-based round- 1583 trip time (RTT) measurements are done using the same transport-layer 1584 flow priority as the RTP packet stream experiences. RTCP compound 1585 packets containing RR packet ought to be sent with the priority used 1586 by the majority of the RTP packet streams reported on. RTCP packets 1587 containing time-critical feedback packets can use higher priority to 1588 improve the timeliness and likelihood of delivery of such feedback. 1590 12.2. Media Source, RTP Packet Streams, and Participant Identification 1592 12.2.1. Media Source Identification 1594 Each RTP packet stream is identified by a unique synchronisation 1595 source (SSRC) identifier. The SSRC identifier is carried in each of 1596 the RTP packets comprising a RTP packet stream, and is also used to 1597 identify that stream in the corresponding RTCP reports. The SSRC is 1598 chosen as discussed in Section 4.8. The first stage in 1599 demultiplexing RTP and RTCP packets received on a single transport 1600 layer flow at a WebRTC end-point is to separate the RTP packet 1601 streams based on their SSRC value; once that is done, additional 1602 demultiplexing steps can determine how and where to render the media. 1604 RTP allows a mixer, or other RTP-layer middlebox, to combine encoded 1605 streams from multiple media sources to form a new encoded stream from 1606 a new media source (the mixer). The RTP packets in that new RTP 1607 packet stream can include a Contributing Source (CSRC) list, 1608 indicating which original SSRCs contributed to the combined source 1609 stream. As described in Section 4.1, implementations need to support 1610 reception of RTP data packets containing a CSRC list and RTCP packets 1611 that relate to sources present in the CSRC list. The CSRC list can 1612 change on a packet-by-packet basis, depending on the mixing operation 1613 being performed. Knowledge of what media sources contributed to a 1614 particular RTP packet can be important if the user interface 1615 indicates which participants are active in the session. Changes in 1616 the CSRC list included in packets needs to be exposed to the WebRTC 1617 application using some API, if the application is to be able to track 1618 changes in session participation. It is desirable to map CSRC values 1619 back into WebRTC MediaStream identities as they cross this API, to 1620 avoid exposing the SSRC/CSRC name space to JavaScript applications. 1622 If the mixer-to-client audio level extension [RFC6465] is being used 1623 in the session (see Section 5.2.3), the information in the CSRC list 1624 is augmented by audio level information for each contributing source. 1625 It is desirable to expose this information to the WebRTC application 1626 using some API, after mapping the CSRC values to WebRTC MediaStream 1627 identities, so it can be exposed in the user interface. 1629 12.2.2. SSRC Collision Detection 1631 The RTP standard requires RTP implementations to have support for 1632 detecting and handling SSRC collisions, i.e., resolve the conflict 1633 when two different end-points use the same SSRC value (see section 1634 8.2 of [RFC3550]). This requirement also applies to WebRTC end- 1635 points. There are several scenarios where SSRC collisions can 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 1761 [I-D.ietf-avtcore-rtp-multi-stream]). To address the other concerns, 1762 endpoints SHOULD ignore parameters that configure the RTCP reporting 1763 interval to be significantly longer than the default five second 1764 interval specified in [RFC3550] (unless the media data rate is so low 1765 that the longer reporting interval roughly corresponds to 5% of the 1766 media data rate), or that configure the RTCP reporting interval small 1767 enough that the RTCP bandwidth would exceed the media bandwidth. 1769 The guidelines in [RFC6562] apply when using variable bit rate (VBR) 1770 audio codecs such as Opus (see Section 4.3 for discussion of mandated 1771 audio codecs). The guidelines in [RFC6562] also apply, but are of 1772 lesser importance, when using the client-to-mixer audio level header 1773 extensions (Section 5.2.2) or the mixer-to-client audio level header 1774 extensions (Section 5.2.3). The use of the encryption of the header 1775 extensions are RECOMMENDED, unless there are known reasons, like RTP 1776 middleboxes or third party monitoring that will greatly benefit from 1777 the information, and this has been expressed using API or signalling. 1778 If further evidence are produced to show that information leakage is 1779 significant from audio level indications, then use of encryption 1780 needs to be mandated at that time. 1782 14. IANA Considerations 1784 This memo makes no request of IANA. 1786 Note to RFC Editor: this section is to be removed on publication as 1787 an RFC. 1789 15. Acknowledgements 1791 The authors would like to thank Bernard Aboba, Harald Alvestrand, 1792 Cary Bran, Ben Campbell, Charles Eckel, Alex Eleftheriadis, Christian 1793 Groves, Cullen Jennings, Olle Johansson, Suhas Nandakumar, Dan 1794 Romascanu, Jim Spring, Martin Thomson, and the other members of the 1795 IETF RTCWEB working group for their valuable feedback. 1797 16. References 1799 16.1. Normative References 1801 [I-D.ietf-avtcore-multi-media-rtp-session] 1802 Westerlund, M., Perkins, C., and J. Lennox, "Sending 1803 Multiple Types of Media in a Single RTP Session", draft- 1804 ietf-avtcore-multi-media-rtp-session-05 (work in 1805 progress), February 2014. 1807 [I-D.ietf-avtcore-rtp-circuit-breakers] 1808 Perkins, C. and V. Singh, "Multimedia Congestion Control: 1809 Circuit Breakers for Unicast RTP Sessions", draft-ietf- 1810 avtcore-rtp-circuit-breakers-06 (work in progress), July 1811 2014. 1813 [I-D.ietf-avtcore-rtp-multi-stream-optimisation] 1814 Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, 1815 "Sending Multiple Media Streams in a Single RTP Session: 1816 Grouping RTCP Reception Statistics and Other Feedback ", 1817 draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work 1818 in progress), July 2013. 1820 [I-D.ietf-avtcore-rtp-multi-stream] 1821 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, 1822 "Sending Multiple Media Streams in a Single RTP Session", 1823 draft-ietf-avtcore-rtp-multi-stream-05 (work in progress), 1824 July 2014. 1826 [I-D.ietf-rtcweb-security-arch] 1827 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 1828 rtcweb-security-arch-10 (work in progress), July 2014. 1830 [I-D.ietf-rtcweb-security] 1831 Rescorla, E., "Security Considerations for WebRTC", draft- 1832 ietf-rtcweb-security-07 (work in progress), July 2014. 1834 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1835 Requirement Levels", BCP 14, RFC 2119, March 1997. 1837 [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP 1838 Payload Format Specifications", BCP 36, RFC 2736, December 1839 1999. 1841 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1842 Jacobson, "RTP: A Transport Protocol for Real-Time 1843 Applications", STD 64, RFC 3550, July 2003. 1845 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 1846 Video Conferences with Minimal Control", STD 65, RFC 3551, 1847 July 2003. 1849 [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth 1850 Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 1851 3556, July 2003. 1853 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 1854 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 1855 RFC 3711, March 2004. 1857 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1858 Description Protocol", RFC 4566, July 2006. 1860 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 1861 "Extended RTP Profile for Real-time Transport Control 1862 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 1863 2006. 1865 [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. 1866 Hakenberg, "RTP Retransmission Payload Format", RFC 4588, 1867 July 2006. 1869 [RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", 1870 BCP 131, RFC 4961, July 2007. 1872 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 1873 "Codec Control Messages in the RTP Audio-Visual Profile 1874 with Feedback (AVPF)", RFC 5104, February 2008. 1876 [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for 1877 Real-time Transport Control Protocol (RTCP)-Based Feedback 1878 (RTP/SAVPF)", RFC 5124, February 2008. 1880 [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP 1881 Header Extensions", RFC 5285, July 2008. 1883 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size 1884 Real-Time Transport Control Protocol (RTCP): Opportunities 1885 and Consequences", RFC 5506, April 2009. 1887 [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and 1888 Control Packets on a Single Port", RFC 5761, April 2010. 1890 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1891 Security (DTLS) Extension to Establish Keys for the Secure 1892 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 1894 [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP 1895 Flows", RFC 6051, November 2010. 1897 [RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time 1898 Transport Protocol (RTP) Header Extension for Client-to- 1899 Mixer Audio Level Indication", RFC 6464, December 2011. 1901 [RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time 1902 Transport Protocol (RTP) Header Extension for Mixer-to- 1903 Client Audio Level Indication", RFC 6465, December 2011. 1905 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of 1906 Variable Bit Rate Audio with Secure RTP", RFC 6562, March 1907 2012. 1909 [RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure 1910 Real-time Transport Protocol (SRTP)", RFC 6904, April 1911 2013. 1913 [RFC7007] Terriberry, T., "Update to Remove DVI4 from the 1914 Recommended Codecs for the RTP Profile for Audio and Video 1915 Conferences with Minimal Control (RTP/AVP)", RFC 7007, 1916 August 2013. 1918 [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, 1919 "Guidelines for Choosing RTP Control Protocol (RTCP) 1920 Canonical Names (CNAMEs)", RFC 7022, September 2013. 1922 [RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple 1923 Clock Rates in an RTP Session", RFC 7160, April 2014. 1925 [RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC 1926 7164, March 2014. 1928 16.2. Informative References 1930 [I-D.ietf-avtcore-multiplex-guidelines] 1931 Westerlund, M., Perkins, C., and H. Alvestrand, 1932 "Guidelines for using the Multiplexing Features of RTP to 1933 Support Multiple Media Streams", draft-ietf-avtcore- 1934 multiplex-guidelines-02 (work in progress), January 2014. 1936 [I-D.ietf-avtcore-rtp-topologies-update] 1937 Westerlund, M. and S. Wenger, "RTP Topologies", draft- 1938 ietf-avtcore-rtp-topologies-update-04 (work in progress), 1939 August 2014. 1941 [I-D.ietf-avtext-rtp-grouping-taxonomy] 1942 Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro, 1943 "A Taxonomy of Grouping Semantics and Mechanisms for Real- 1944 Time Transport Protocol (RTP) Sources", draft-ietf-avtext- 1945 rtp-grouping-taxonomy-02 (work in progress), June 2014. 1947 [I-D.ietf-mmusic-msid] 1948 Alvestrand, H., "WebRTC MediaStream Identification in the 1949 Session Description Protocol", draft-ietf-mmusic-msid-06 1950 (work in progress), June 2014. 1952 [I-D.ietf-mmusic-sdp-bundle-negotiation] 1953 Holmberg, C., Alvestrand, H., and C. Jennings, 1954 "Negotiating Media Multiplexing Using the Session 1955 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 1956 negotiation-08 (work in progress), August 2014. 1958 [I-D.ietf-payload-rtp-howto] 1959 Westerlund, M., "How to Write an RTP Payload Format", 1960 draft-ietf-payload-rtp-howto-13 (work in progress), 1961 January 2014. 1963 [I-D.ietf-rmcat-cc-requirements] 1964 Jesup, R., "Congestion Control Requirements For RMCAT", 1965 draft-ietf-rmcat-cc-requirements-05 (work in progress), 1966 July 2014. 1968 [I-D.ietf-rtcweb-audio] 1969 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 1970 Requirements", draft-ietf-rtcweb-audio-05 (work in 1971 progress), February 2014. 1973 [I-D.ietf-rtcweb-overview] 1974 Alvestrand, H., "Overview: Real Time Protocols for 1975 Browser-based Applications", draft-ietf-rtcweb-overview-11 1976 (work in progress), August 2014. 1978 [I-D.ietf-rtcweb-use-cases-and-requirements] 1979 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 1980 Time Communication Use-cases and Requirements", draft- 1981 ietf-rtcweb-use-cases-and-requirements-14 (work in 1982 progress), February 2014. 1984 [I-D.ietf-tsvwg-rtcweb-qos] 1985 Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. 1986 Polk, "DSCP and other packet markings for RTCWeb QoS", 1987 draft-ietf-tsvwg-rtcweb-qos-02 (work in progress), June 1988 2014. 1990 [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control 1991 Protocol Extended Reports (RTCP XR)", RFC 3611, November 1992 2003. 1994 [RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient 1995 Stream Loss-Tolerant Authentication (TESLA) in the Secure 1996 Real-time Transport Protocol (SRTP)", RFC 4383, February 1997 2006. 1999 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 2000 (ICE): A Protocol for Network Address Translator (NAT) 2001 Traversal for Offer/Answer Protocols", RFC 5245, April 2002 2010. 2004 [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific 2005 Media Attributes in the Session Description Protocol 2006 (SDP)", RFC 5576, June 2009. 2008 [RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP 2009 Control Protocol (RTCP)", RFC 5968, September 2010. 2011 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 2012 Keeping Alive the NAT Mappings Associated with RTP / RTP 2013 Control Protocol (RTCP) Flows", RFC 6263, June 2011. 2015 [RFC6792] Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the 2016 RTP Monitoring Framework", RFC 6792, November 2012. 2018 [W3C.WD-mediacapture-streams-20130903] 2019 Burnett, D., Bergkvist, A., Jennings, C., and A. 2020 Narayanan, "Media Capture and Streams", World Wide Web 2021 Consortium WD WD-mediacapture-streams-20130903, September 2022 2013, . 2025 [W3C.WD-webrtc-20130910] 2026 Bergkvist, A., Burnett, D., Jennings, C., and A. 2027 Narayanan, "WebRTC 1.0: Real-time Communication Between 2028 Browsers", World Wide Web Consortium WD WD- 2029 webrtc-20130910, September 2013, 2030 . 2032 Authors' Addresses 2034 Colin Perkins 2035 University of Glasgow 2036 School of Computing Science 2037 Glasgow G12 8QQ 2038 United Kingdom 2040 Email: csp@csperkins.org 2041 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 2051 Joerg Ott 2052 Aalto University 2053 School of Electrical Engineering 2054 Espoo 02150 2055 Finland 2057 Email: jorg.ott@aalto.fi