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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group H. Alvestrand 3 Internet-Draft Google 4 Intended status: Standards Track July 6, 2015 5 Expires: January 7, 2016 7 Transports for WebRTC 8 draft-ietf-rtcweb-transports-09 10 Abstract 12 This document describes the data transport protocols used by WebRTC, 13 including the protocols used for interaction with intermediate boxes 14 such as firewalls, relays and NAT boxes. 16 Status of This Memo 18 This Internet-Draft is submitted in full conformance with the 19 provisions of BCP 78 and BCP 79. 21 Internet-Drafts are working documents of the Internet Engineering 22 Task Force (IETF). Note that other groups may also distribute 23 working documents as Internet-Drafts. The list of current Internet- 24 Drafts is at http://datatracker.ietf.org/drafts/current/. 26 Internet-Drafts are draft documents valid for a maximum of six months 27 and may be updated, replaced, or obsoleted by other documents at any 28 time. It is inappropriate to use Internet-Drafts as reference 29 material or to cite them other than as "work in progress." 31 This Internet-Draft will expire on January 7, 2016. 33 Copyright Notice 35 Copyright (c) 2015 IETF Trust and the persons identified as the 36 document authors. All rights reserved. 38 This document is subject to BCP 78 and the IETF Trust's Legal 39 Provisions Relating to IETF Documents 40 (http://trustee.ietf.org/license-info) in effect on the date of 41 publication of this document. Please review these documents 42 carefully, as they describe your rights and restrictions with respect 43 to this document. Code Components extracted from this document must 44 include Simplified BSD License text as described in Section 4.e of 45 the Trust Legal Provisions and are provided without warranty as 46 described in the Simplified BSD License. 48 Table of Contents 50 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 51 2. Requirements language . . . . . . . . . . . . . . . . . . . . 3 52 3. Transport and Middlebox specification . . . . . . . . . . . . 3 53 3.1. System-provided interfaces . . . . . . . . . . . . . . . 3 54 3.2. Ability to use IPv4 and IPv6 . . . . . . . . . . . . . . 4 55 3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4 56 3.4. Middle box related functions . . . . . . . . . . . . . . 4 57 3.5. Transport protocols implemented . . . . . . . . . . . . . 5 58 4. Media Prioritization . . . . . . . . . . . . . . . . . . . . 6 59 4.1. Usage of Quality of Service - DSCP and Multiplexing . . . 6 60 4.2. Local prioritization . . . . . . . . . . . . . . . . . . 8 61 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 62 6. Security Considerations . . . . . . . . . . . . . . . . . . . 9 63 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 9 64 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 9 65 8.1. Normative References . . . . . . . . . . . . . . . . . . 9 66 8.2. Informative References . . . . . . . . . . . . . . . . . 12 67 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 12 68 A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 12 69 A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 13 70 A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 13 71 A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 14 72 A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 14 73 A.6. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 14 74 A.7. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 14 75 A.8. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 15 76 A.9. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 15 77 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 15 79 1. Introduction 81 WebRTC is a protocol suite aimed at real time multimedia exchange 82 between browsers, and between browsers and other entities. 84 WebRTC is described in the WebRTC overview document, 85 [I-D.ietf-rtcweb-overview], which also defines terminology used in 86 this document, including the terms "WebRTC device" and "WebRTC 87 browser". 89 This document focuses on the data transport protocols that are used 90 by conforming implementations, including the protocols used for 91 interaction with intermediate boxes such as firewalls, relays and NAT 92 boxes. 94 This protocol suite intends to satisfy the security considerations 95 described in the WebRTC security documents, 96 [I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch]. 98 This document describes requirements that apply to all WebRTC 99 devices. When there are requirements that apply only to WebRTC 100 browsers, this is called out explicitly. 102 2. Requirements language 104 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 105 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 106 document are to be interpreted as described in RFC 2119 [RFC2119]. 108 3. Transport and Middlebox specification 110 3.1. System-provided interfaces 112 The protocol specifications used here assume that the following 113 protocols are available to the implementations of the WebRTC 114 protocols: 116 o UDP [RFC0768]. This is the protocol assumed by most protocol 117 elements described. 119 o TCP [RFC0793]. This is used for HTTP/WebSockets, as well as for 120 TURN/SSL and ICE-TCP. 122 For both protocols, IPv4 and IPv6 support is assumed. 124 For UDP, this specification assumes the ability to set the DSCP code 125 point of the sockets opened on a per-packet basis, in order to 126 achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos] 127 (see Section 4.1) when multiple media types are multiplexed. It does 128 not assume that the DSCP codepoints will be honored, and does assume 129 that they may be zeroed or changed, since this is a local 130 configuration issue. 132 Platforms that do not give access to these interfaces will not be 133 able to support a conforming WebRTC implementation. 135 This specification does not assume that the implementation will have 136 access to ICMP or raw IP. 138 3.2. Ability to use IPv4 and IPv6 140 Web applications running in a WebRTC browser MUST be able to utilize 141 both IPv4 and IPv6 where available - that is, when two peers have 142 only IPv4 connectivity to each other, or they have only IPv6 143 connectivity to each other, applications running in the WebRTC 144 browser MUST be able to communicate. 146 When TURN is used, and the TURN server has IPv4 or IPv6 connectivity 147 to the peer or its TURN server, candidates of the appropriate types 148 MUST be supported. The "Happy Eyeballs" specification for ICE 149 [I-D.martinsen-mmusic-ice-dualstack-fairness] SHOULD be supported. 151 3.3. Usage of temporary IPv6 addresses 153 The IPv6 default address selection specification [RFC6724] specifies 154 that temporary addresses [RFC4941] are to be preferred over permanent 155 addresses. This is a change from the rules specified by [RFC3484]. 156 For applications that select a single address, this is usually done 157 by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014]. 158 However, this rule is not completely obvious in the ICE scope. This 159 is therefore clarified as follows: 161 When a client gathers all IPv6 addresses on a host, and both 162 temporary addresses and permanent addresses of the same scope are 163 present, the client SHOULD discard the permanent addresses before 164 exposing addresses to the application or using them in ICE. This is 165 consistent with the default policy described in [RFC6724]. 167 3.4. Middle box related functions 169 The primary mechanism to deal with middle boxes is ICE, which is an 170 appropriate way to deal with NAT boxes and firewalls that accept 171 traffic from the inside, but only from the outside if it is in 172 response to inside traffic (simple stateful firewalls). 174 ICE [RFC5245] MUST be supported. The implementation MUST be a full 175 ICE implementation, not ICE-Lite. A full ICE implementation allows 176 interworking with both ICE and ICE-Lite implementations when they are 177 deployed appropriately. 179 In order to deal with situations where both parties are behind NATs 180 of the type that perform endpoint-dependent mapping (as defined in 181 [RFC5128] section 2.4), TURN [RFC5766] MUST be supported. 183 WebRTC browsers MUST support configuration of STUN and TURN servers, 184 both from browser configuration and from an application. 186 In order to deal with firewalls that block all UDP traffic, the mode 187 of TURN that uses TCP between the client and the server MUST be 188 supported, and the mode of TURN that uses TLS over TCP between the 189 client and the server MUST be supported. See [RFC5766] section 2.1 190 for details. 192 In order to deal with situations where one party is on an IPv4 193 network and the other party is on an IPv6 network, TURN extensions 194 for IPv6 [RFC6156] MUST be supported. 196 TURN TCP candidates, where the connection from the client's TURN 197 server to the peer is a TCP connection, [RFC6062] MAY be supported. 199 However, such candidates are not seen as providing any significant 200 benefit, for the following reasons. 202 First, use of TURN TCP candidates would only be relevant in cases 203 which both peers are required to use TCP to establish a 204 PeerConnection. 206 Second, that use case is supported in a different way by both sides 207 establishing UDP relay candidates using TURN over TCP to connect to 208 their respective relay servers. 210 Third, using TCP only between the endpoint and its relay may result 211 in less issues with TCP in regards to real-time constraints, e.g. due 212 to head of line blocking. 214 ICE-TCP candidates [RFC6544] MUST be supported; this may allow 215 applications to communicate to peers with public IP addresses across 216 UDP-blocking firewalls without using a TURN server. 218 If TCP connections are used, RTP framing according to [RFC4571] MUST 219 be used, both for the RTP packets and for the DTLS packets used to 220 carry data channels. 222 The ALTERNATE-SERVER mechanism specified in [RFC5389] (STUN) section 223 11 (300 Try Alternate) MUST be supported. 225 The WebRTC implementation MAY support accessing the Internet through 226 an HTTP proxy. If it does so, it MUST support the "connect" header 227 as specified in [I-D.ietf-httpbis-tunnel-protocol]. 229 3.5. Transport protocols implemented 231 For transport of media, secure RTP is used. The details of the 232 profile of RTP used are described in "RTP Usage" 234 [I-D.ietf-rtcweb-rtp-usage]. Key exchange MUST be done using DTLS- 235 SRTP, as described in [I-D.ietf-rtcweb-security-arch]. 237 For data transport over the WebRTC data channel 238 [I-D.ietf-rtcweb-data-channel], WebRTC implementations MUST support 239 SCTP over DTLS over ICE. This encapsulation is specified in 240 [I-D.ietf-tsvwg-sctp-dtls-encaps]. Negotiation of this transport in 241 SDP is defined in [I-D.ietf-mmusic-sctp-sdp]. The SCTP extension for 242 NDATA, [I-D.ietf-tsvwg-sctp-ndata], MUST be supported. 244 The setup protocol for WebRTC data channels described in 245 [I-D.ietf-rtcweb-data-protocol] MUST be supported. 247 Note: DTLS-SRTP as defined in [RFC5764] section 6.7.1 defines the 248 interaction between DTLS and ICE ( [RFC5245]). The effect of this 249 specification is that all ICE candidate pairs associated with a 250 single component are part of the same DTLS association. Thus, there 251 will only be one DTLS handshake even if there are multiple valid 252 candidate pairs. 254 WebRTC implementations MUST support multiplexing of DTLS and RTP over 255 the same port pair, as described in the DTLS-SRTP specification 256 [RFC5764], section 5.1.2. All application layer protocol payloads 257 over this DTLS connection are SCTP packets. 259 Protocol identification MUST be supplied as part of the DTLS 260 handshake, as specified in [I-D.ietf-rtcweb-alpn]. 262 4. Media Prioritization 264 The WebRTC prioritization model is that the application tells the 265 WebRTC implementation about the priority of media and data flows 266 through an API. 268 The priority associated with a media or data flow is classified as 269 "normal", "below normal", "high" or "very high". There are only four 270 priority levels at the API. 272 The priority settings affect two pieces of behavior: Packet markings 273 and packet send sequence decisions. Each is described in its own 274 section below. 276 4.1. Usage of Quality of Service - DSCP and Multiplexing 278 Implementations SHOULD attempt to set QoS on the packets sent, 279 according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos]. It is 280 appropriate to depart from this recommendation when running on 281 platforms where QoS marking is not implemented. 283 The implementation MAY turn off use of DSCP markings if it detects 284 symptoms of unexpected behaviour like priority inversion or blocking 285 of packets with certain DSCP markings. The detection of these 286 conditions is implementation dependent. (Question: Does there need 287 to be an API knob to turn off DSCP markings? If nobody argues for 288 it, this parenthesis will be removed.) 290 All packets carrying data from the SCTP association supporting the 291 data channels MUST use a single DSCP code point. 293 All packets on one TCP connection, no matter what it carries, MUST 294 use a single DSCP code point. 296 More advice on the use of DSCP code points with RTP is given in 297 [I-D.ietf-dart-dscp-rtp]. 299 There exist a number of schemes for achieving quality of service that 300 do not depend solely on DSCP code points. Some of these schemes 301 depend on classifying the traffic into flows based on 5-tuple (source 302 address, source port, protocol, destination address, destination 303 port) or 6-tuple (5-tuple + DSCP code point). Under differing 304 conditions, it may therefore make sense for a sending application to 305 choose any of the configurations: 307 o Each media stream carried on its own 5-tuple 309 o Media streams grouped by media type into 5-tuples (such as 310 carrying all audio on one 5-tuple) 312 o All media sent over a single 5-tuple, with or without 313 differentiation into 6-tuples based on DSCP code points 315 In each of the configurations mentioned, data channels may be carried 316 in its own 5-tuple, or multiplexed together with one of the media 317 flows. 319 More complex configurations, such as sending a high priority video 320 stream on one 5-tuple and sending all other video streams multiplexed 321 together over another 5-tuple, can also be envisioned. More 322 information on mapping media flows to 5-tuples can be found in 323 [I-D.ietf-rtcweb-rtp-usage]. 325 A sending implementation MUST be able to support the following 326 configurations: 328 o multiplex all media and data on a single 5-tuple (fully bundled) 329 o send each media stream on its own 5-tuple and data on its own 330 5-tuple (fully unbundled) 332 It MAY choose to support other configurations, such as bundling each 333 media type (audio, video or data) into its own 5-tuple (bundling by 334 media type). 336 Sending data over multiple 5-tuples is not supported. 338 A receiving implementation MUST be able to receive media and data in 339 all these configurations. 341 4.2. Local prioritization 343 When an WebRTC implementation has packets to send on multiple streams 344 (with each media stream and each data channel considered as one 345 "stream" for this purpose) that are congestion-controlled under the 346 same congestion controller, the WebRTC implementation SHOULD cause 347 data to be emitted in such a way that each stream at each level of 348 priority is being given approximately twice the transmission capacity 349 (measured in payload bytes) of the level below. 351 Thus, when congestion occurs, a "very high" priority flow will have 352 the ability to send 8 times as much data as a "below normal" flow if 353 both have data to send. This prioritization is independent of the 354 media type. The details of which packet to send first are 355 implementation defined. 357 For example: If there is a very high priority audio flow sending 100 358 byte packets, and a normal priority video flow sending 1000 byte 359 packets, and outgoing capacity exists for sending >5000 payload 360 bytes, it would be appropriate to send 4000 bytes (40 packets) of 361 audio and 1000 bytes (one packet) of video as the result of a single 362 pass of sending decisions. 364 Conversely, if the audio flow is marked normal priority and the video 365 flow is marked very high priority, the scheduler may decide to send 2 366 video packets (2000 bytes) and 5 audio packets (500 bytes) when 367 outgoing capacity exists for sending > 2500 payload bytes. 369 If there are two very high priority audio flows, each will be able to 370 send 4000 bytes in the same period where a normal priority video flow 371 is able to send 1000 bytes. 373 Two example implementation strategies are: 375 o When the available bandwidth is known from the congestion control 376 algorithm, configure each codec and each data channel with a 377 target send rate that is appropriate to its share of the available 378 bandwidth. 380 o When congestion control indicates that a specified number of 381 packets can be sent, send packets that are available to send using 382 a weighted round robin scheme across the connections. 384 Any combination of these, or other schemes that have the same effect, 385 is valid, as long as the distribution of transmission capacity is 386 approximately correct. 388 For media, it is usually inappropriate to use deep queues for 389 sending; it is more useful to, for instance, skip intermediate frames 390 that have no dependencies on them in order to achieve a lower 391 bitrate. For reliable data, queues are useful. 393 5. IANA Considerations 395 This document makes no request of IANA. 397 Note to RFC Editor: this section may be removed on publication as an 398 RFC. 400 6. Security Considerations 402 Security considerations are enumerated in [I-D.ietf-rtcweb-security]. 404 7. Acknowledgements 406 This document is based on earlier versions embedded in 407 [I-D.ietf-rtcweb-overview], which were the results of contributions 408 from many RTCWEB WG members. 410 Special thanks for reviews of earlier versions of this draft go to 411 Eduardo Gueiros, Magnus Westerlund, Markus Isomaki and Dan Wing; the 412 contributions from Andrew Hutton also deserve special mention. 414 8. References 416 8.1. Normative References 418 [I-D.ietf-httpbis-tunnel-protocol] 419 Hutton, A., Uberti, J., and M. Thomson, "The Tunnel- 420 Protocol HTTP Request Header Field", draft-ietf-httpbis- 421 tunnel-protocol-01 (work in progress), January 2015. 423 [I-D.ietf-mmusic-sctp-sdp] 424 Holmberg, C., Loreto, S., and G. Camarillo, "Stream 425 Control Transmission Protocol (SCTP)-Based Media Transport 426 in the Session Description Protocol (SDP)", draft-ietf- 427 mmusic-sctp-sdp-12 (work in progress), January 2015. 429 [I-D.ietf-rtcweb-alpn] 430 Thomson, M., "Application Layer Protocol Negotiation for 431 Web Real-Time Communications (WebRTC)", draft-ietf-rtcweb- 432 alpn-00 (work in progress), July 2014. 434 [I-D.ietf-rtcweb-data-channel] 435 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 436 Channels", draft-ietf-rtcweb-data-channel-13 (work in 437 progress), January 2015. 439 [I-D.ietf-rtcweb-data-protocol] 440 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 441 Establishment Protocol", draft-ietf-rtcweb-data- 442 protocol-09 (work in progress), January 2015. 444 [I-D.ietf-rtcweb-rtp-usage] 445 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 446 Communication (WebRTC): Media Transport and Use of RTP", 447 draft-ietf-rtcweb-rtp-usage-22 (work in progress), 448 February 2015. 450 [I-D.ietf-rtcweb-security] 451 Rescorla, E., "Security Considerations for WebRTC", draft- 452 ietf-rtcweb-security-07 (work in progress), July 2014. 454 [I-D.ietf-rtcweb-security-arch] 455 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 456 rtcweb-security-arch-10 (work in progress), July 2014. 458 [I-D.ietf-tsvwg-rtcweb-qos] 459 Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. 460 Polk, "DSCP and other packet markings for RTCWeb QoS", 461 draft-ietf-tsvwg-rtcweb-qos-03 (work in progress), 462 November 2014. 464 [I-D.ietf-tsvwg-sctp-dtls-encaps] 465 Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS 466 Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- 467 dtls-encaps-09 (work in progress), January 2015. 469 [I-D.ietf-tsvwg-sctp-ndata] 470 Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, 471 "Stream Schedulers and a New Data Chunk for the Stream 472 Control Transmission Protocol", draft-ietf-tsvwg-sctp- 473 ndata-02 (work in progress), January 2015. 475 [I-D.martinsen-mmusic-ice-dualstack-fairness] 476 Martinsen, P., Reddy, T., and P. Patil, "ICE IPv4/IPv6 477 Dual Stack Fairness", draft-martinsen-mmusic-ice- 478 dualstack-fairness-02 (work in progress), February 2015. 480 [RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, 481 August 1980. 483 [RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC 484 793, September 1981. 486 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 487 Requirement Levels", BCP 14, RFC 2119, March 1997. 489 [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) 490 and RTP Control Protocol (RTCP) Packets over Connection- 491 Oriented Transport", RFC 4571, July 2006. 493 [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy 494 Extensions for Stateless Address Autoconfiguration in 495 IPv6", RFC 4941, September 2007. 497 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 498 (ICE): A Protocol for Network Address Translator (NAT) 499 Traversal for Offer/Answer Protocols", RFC 5245, April 500 2010. 502 [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, 503 "Session Traversal Utilities for NAT (STUN)", RFC 5389, 504 October 2008. 506 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 507 Security (DTLS) Extension to Establish Keys for the Secure 508 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 510 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 511 Relays around NAT (TURN): Relay Extensions to Session 512 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 514 [RFC6062] Perreault, S. and J. Rosenberg, "Traversal Using Relays 515 around NAT (TURN) Extensions for TCP Allocations", RFC 516 6062, November 2010. 518 [RFC6156] Camarillo, G., Novo, O., and S. Perreault, "Traversal 519 Using Relays around NAT (TURN) Extension for IPv6", RFC 520 6156, April 2011. 522 [RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach, 523 "TCP Candidates with Interactive Connectivity 524 Establishment (ICE)", RFC 6544, March 2012. 526 [RFC6724] Thaler, D., Draves, R., Matsumoto, A., and T. Chown, 527 "Default Address Selection for Internet Protocol Version 6 528 (IPv6)", RFC 6724, September 2012. 530 8.2. Informative References 532 [I-D.ietf-dart-dscp-rtp] 533 Black, D. and P. Jones, "Differentiated Services 534 (DiffServ) and Real-time Communication", draft-ietf-dart- 535 dscp-rtp-10 (work in progress), November 2014. 537 [I-D.ietf-rtcweb-overview] 538 Alvestrand, H., "Overview: Real Time Protocols for 539 Browser-based Applications", draft-ietf-rtcweb-overview-13 540 (work in progress), November 2014. 542 [RFC3484] Draves, R., "Default Address Selection for Internet 543 Protocol version 6 (IPv6)", RFC 3484, February 2003. 545 [RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6 546 Socket API for Source Address Selection", RFC 5014, 547 September 2007. 549 [RFC5128] Srisuresh, P., Ford, B., and D. Kegel, "State of Peer-to- 550 Peer (P2P) Communication across Network Address 551 Translators (NATs)", RFC 5128, March 2008. 553 Appendix A. Change log 555 This section should be removed before publication as an RFC. 557 A.1. Changes from -00 to -01 559 o Clarified DSCP requirements, with reference to -qos- 561 o Clarified "symmetric NAT" -> "NATs which perform endpoint- 562 dependent mapping" 564 o Made support of TURN over TCP mandatory 565 o Made support of TURN over TLS a MAY, and added open question 567 o Added an informative reference to -firewalls- 569 o Called out that we don't make requirements on HTTP proxy 570 interaction (yet 572 A.2. Changes from -01 to -02 574 o Required support for 300 Alternate Server from STUN. 576 o Separated the ICE-TCP candidate requirement from the TURN-TCP 577 requirement. 579 o Added new sections on using QoS functions, and on multiplexing 580 considerations. 582 o Removed all mention of RTP profiles. Those are the business of 583 the RTP usage draft, not this one. 585 o Required support for TURN IPv6 extensions. 587 o Removed reference to the TURN URI scheme, as it was unnecessary. 589 o Made an explicit statement that multiplexing (or not) is an 590 application matter. 592 . 594 A.3. Changes from -02 to -03 596 o Added required support for draft-ietf-tsvwg-sctp-ndata 598 o Removed discussion of multiplexing, since this is present in rtp- 599 usage. 601 o Added RFC 4571 reference for framing RTP packets over TCP. 603 o Downgraded TURN TCP candidates from SHOULD to MAY, and added more 604 language discussing TCP usage. 606 o Added language on IPv6 temporary addresses. 608 o Added language describing multiplexing choices. 610 o Added a separate section detailing what it means when we say that 611 an WebRTC implementation MUST support both IPv4 and IPv6. 613 A.4. Changes from -03 to -04 615 o Added a section on prioritization, moved the DSCP section into it, 616 and added a section on local prioritization, giving a specific 617 algorithm for interpreting "priority" in local prioritization. 619 o ICE-TCP candidates was changed from MAY to MUST, in recognition of 620 the sense of the room at the London IETF. 622 A.5. Changes from -04 to -05 624 o Reworded introduction 626 o Removed all references to "WebRTC". It now uses only the term 627 RTCWEB. 629 o Addressed a number of clarity / language comments 631 o Rewrote the prioritization to cover data channels and to describe 632 multiple ways of prioritizing flows 634 o Made explicit reference to "MUST do DTLS-SRTP", and referred to 635 security-arch for details 637 A.6. Changes from -05 to -06 639 o Changed all references to "RTCWEB" to "WebRTC", except one 640 reference to the working group 642 o Added reference to the httpbis "connect" protocol (being adopted 643 by HTTPBIS) 645 o Added reference to the ALPN header (being adopted by RTCWEB) 647 o Added reference to the DART RTP document 649 o Said explicitly that SCTP for data channels has a single DSCP 650 codepoint 652 A.7. Changes from -06 to -07 654 o Updated references 656 o Removed reference to draft-hutton-rtcweb-nat-firewall- 657 considerations 659 A.8. Changes from -07 to -08 661 o Updated references 663 o Deleted "bundle each media type (audio, video or data) into its 664 own 5-tuple (bundling by media type)" from MUST support 665 configuration, since JSEP does not have a means to negotiate this 666 configuration 668 A.9. Changes from -08 to -09 670 o Added a clarifying note about DTLS-SRTP and ICE interaction. 672 Author's Address 674 Harald Alvestrand 675 Google 677 Email: harald@alvestrand.no