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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 October 22, 2014 5 Expires: April 25, 2015 7 Transports for WebRTC 8 draft-ietf-rtcweb-transports-07 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 April 25, 2015. 33 Copyright Notice 35 Copyright (c) 2014 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 . . . . . . . . . . . . . . 3 55 3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4 56 3.4. Middle box related functions . . . . . . . . . . . . . . 4 57 3.5. Transport protocols implemented . . . . . . . . . . . . . 6 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 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 15 77 1. Introduction 79 WebRTC is a protocol suite aimed at real time multimedia exchange 80 between browsers, and between browsers and other entities. 82 WebRTC is described in the WebRTC overview document, 83 [I-D.ietf-rtcweb-overview], which also defines terminology used in 84 this document. 86 This document focuses on the data transport protocols that are used 87 by conforming implementations, including the protocols used for 88 interaction with intermediate boxes such as firewalls, relays and NAT 89 boxes. 91 This protocol suite intends to satisfy the security considerations 92 described in the WebRTC security documents, 93 [I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch]. 95 This document describes requirements that apply to all WebRTC 96 devices. When there are requirements that apply only to WebRTC User 97 Agents (also called browsers) , this is called out. 99 The form "WebRTC endpoint" is used as a synonym for "WebRTC device" 100 in contexts where other text talks about endpoints. 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 WebRTC devices: 115 o UDP. This is the protocol assumed by most protocol elements 116 described. 118 o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL 119 and ICE-TCP. 121 For both protocols, IPv4 and IPv6 support is assumed. 123 For UDP, this specification assumes the ability to set the DSCP code 124 point of the sockets opened on a per-packet basis, in order to 125 achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos] 126 (see Section 4.1) when multiple media types are multiplexed. It does 127 not assume that the DSCP codepoints will be honored, and does assume 128 that they may be zeroed or changed, since this is a local 129 configuration issue. 131 Platforms that do not give access to these interfaces will not be 132 able to support a conforming WebRTC implementation. 134 This specification does not assume that the implementation will have 135 access to ICMP or raw IP. 137 3.2. Ability to use IPv4 and IPv6 139 Web applications running in a WebRTC browser MUST be able to utilize 140 both IPv4 and IPv6 where available - that is, when two peers have 141 only IPv4 connectivity to each other, or they have only IPv6 142 connectivity to each other, applications running in the WebRTC 143 browser MUST be able to communicate. 145 WebRTC devices, when attached to networks with appropriate protocol 146 support MUST also be able to communicate using IPv6 and IPv4. 148 When TURN is used, and the TURN server has IPv4 or IPv6 connectivity 149 to the peer or its TURN server, candidates of the appropriate types 150 MUST be supported. The "Happy Eyeballs" specification for ICE 151 [I-D.reddy-mmusic-ice-happy-eyeballs] SHOULD be supported. 153 3.3. Usage of temporary IPv6 addresses 155 The IPv6 default address selection specification [RFC6724] specifies 156 that temporary addresses [RFC4941] are to be preferred over permanent 157 addresses. This is a change from the rules specified by [RFC3484]. 158 For applications that select a single address, this is usually done 159 by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014]. 160 However, this rule is not completely obvious in the ICE scope. This 161 is therefore clarified as follows: 163 When a WebRTC endpoint gathers all IPv6 addresses on a host, and both 164 temporary addresses and permanent addresses of the same scope are 165 present, the client SHOULD discard the permanent addresses before 166 forming pairs. This is consistent with the default policy described 167 in [RFC6724]. 169 3.4. Middle box related functions 171 Except when called out, all requirements in this section apply to all 172 WebRTC devices. 174 The primary mechanism to deal with middle boxes is ICE, which is an 175 appropriate way to deal with NAT boxes and firewalls that accept 176 traffic from the inside, but only from the outside if it is in 177 response to inside traffic (simple stateful firewalls). 179 WebRTC endpoints MUST support ICE [RFC5245]. The implementation MUST 180 be a full ICE implementation, not ICE-Lite. A full ICE 181 implementation allows interworking with both ICE and ICE-Lite 182 implementations when they are deployed appropriately. 184 In order to deal with situations where both parties are behind NATs 185 of the type that perform endpoint-dependent mapping (as defined in 186 [RFC5128] section 2.4), WebRTC endpoints MUST support TURN [RFC5766]. 188 WebRTC browsers MUST support configuration of STUN and TURN servers, 189 both from browser configuration and from an application. 191 In order to deal with firewalls that block all UDP traffic, the mode 192 of TURN that uses TCP between the client and the server MUST be 193 supported, and the mode of TURN that uses TLS over TCP between the 194 client and the server MUST be supported. See [RFC5766] section 2.1 195 for details. 197 In order to deal with situations where one party is on an IPv4 198 network and the other party is on an IPv6 network, TURN extensions 199 for IPv6 [RFC6156] MUST be supported. 201 TURN TCP candidates, where the connection from the client's TURN 202 server to the peer is a TCP connection, [RFC6062] MAY be supported. 204 However, such candidates are not seen as providing any significant 205 benefit, for the following reasons. 207 First, use of TURN TCP candidates would only be relevant in cases 208 which both peers are required to use TCP to establish a 209 PeerConnection. 211 Second, that use case is supported in a different way by both sides 212 establishing UDP relay candidates using TURN over TCP to connect to 213 their respective relay servers. 215 Third, using TCP only between the endpoint and its relay may result 216 in less issues with TCP in regards to real-time constraints, e.g. due 217 to head of line blocking. 219 ICE-TCP candidates [RFC6544] MUST be supported; this may allow 220 applications to communicate to peers with public IP addresses across 221 UDP-blocking firewalls without using a TURN server. 223 If ICE-TCP connections are used, RTP framing according to [RFC4571] 224 MUST be used for all content that doesn't have its own framing 225 mechanism. 227 The ALTERNATE-SERVER mechanism specified in [RFC5389] (STUN) section 228 11 (300 Try Alternate) MUST be supported. 230 In order to deal with the scenario in which the media must traverse a 231 HTTP Proxy, WebRTC browser MUST support the HTTP CONNECT request 232 (Section 4.3.6 of [RFC7231]). WebRTC devices SHOULD support this 233 request. 235 The HTTP Proxy may require authentication and therefore, if HTTP 236 CONNECT request is supported, proxy authentication as described in 237 Section 4.3.6 of [RFC7231] and [RFC7235] MUST also be supported. 239 In addition, the HTTP CONNECT MUST include an indication of the 240 protocol being used with the HTTP CONNECT initiated tunnel as 241 described in [I-D.ietf-httpbis-tunnel-protocol] 243 3.5. Transport protocols implemented 245 For transport of media, secure RTP is used. The details of the 246 profile of RTP used are described in "RTP Usage" 247 [I-D.ietf-rtcweb-rtp-usage]. Key exchange MUST be done using DTLS- 248 SRTP, as described in [I-D.ietf-rtcweb-security-arch]. 250 For data transport over the WebRTC data channel 251 [I-D.ietf-rtcweb-data-channel], WebRTC endpoints MUST support SCTP 252 over DTLS over ICE. This encapsulation is specified in 253 [I-D.ietf-tsvwg-sctp-dtls-encaps]. Negotiation of this transport in 254 SDP is defined in [I-D.ietf-mmusic-sctp-sdp]. The SCTP extension for 255 NDATA, [I-D.ietf-tsvwg-sctp-ndata], MUST be supported. 257 The setup protocol for WebRTC data channels is described in 258 [I-D.jesup-rtcweb-data-protocol]. 260 WebRTC devices MUST support multiplexing of DTLS and RTP over the 261 same port pair, as described in the DTLS_SRTP specification 262 [RFC5764], section 5.1.2. All application layer protocol payloads 263 over this DTLS connection are SCTP packets. 265 Protocol identification MUST be supplied as part of the DTLS 266 handshake, as specified in [I-D.thomson-rtcweb-alpn]. 268 4. Media Prioritization 270 The WebRTC prioritization model is that the application tells the 271 WebRTC browser about the priority of media and data flows through an 272 API. 274 The priority associated with a media or data flow is classified as 275 "normal", "below normal", "high" or "very high". There are only four 276 priority levels at the API. 278 The priority settings affect two pieces of behavior: Packet markings 279 and packet send sequence decisions. Each is described in its own 280 section below. 282 4.1. Usage of Quality of Service - DSCP and Multiplexing 284 WebRTC endpoints SHOULD attempt to set QoS on the packets sent, 285 according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos]. It is 286 appropriate to depart from this recommendation when running on 287 platforms where QoS marking is not implemented. 289 The WebRTC endpoint MAY turn off use of DSCP markings if it detects 290 symptoms of unexpected behaviour like priority inversion or blocking 291 of packets with certain DSCP markings. The detection of these 292 conditions is implementation dependent. (Question: Does there need 293 to be an API knob to turn off DSCP markings?) 295 All packets carrying data from the SCTP association supporting the 296 data channels MUST use a single DSCP code point. 298 All packets on one TCP connection, no matter what it carries, MUST 299 use a single DSCP code point. 301 More advice on the use of DSCP code points with RTP is given in 302 [I-D.ietf-dart-dscp-rtp]. 304 There exist a number of schemes for achieving quality of service that 305 do not depend solely on DSCP code points. Some of these schemes 306 depend on classifying the traffic into flows based on 5-tuple (source 307 address, source port, protocol, destination address, destination 308 port) or 6-tuple (5-tuple + DSCP code point). Under differing 309 conditions, it may therefore make sense for a sending application to 310 choose any of the configurations: 312 o Each media stream carried on its own 5-tuple 314 o Media streams grouped by media type into 5-tuples (such as 315 carrying all audio on one 5-tuple) 317 o All media sent over a single 5-tuple, with or without 318 differentiation into 6-tuples based on DSCP code points 320 In each of the configurations mentioned, data channels may be carried 321 in its own 5-tuple, or multiplexed together with one of the media 322 flows. 324 More complex configurations, such as sending a high priority video 325 stream on one 5-tuple and sending all other video streams multiplexed 326 together over another 5-tuple, can also be envisioned. More 327 information on mapping media flows to 5-tuples can be found in 328 [I-D.ietf-rtcweb-rtp-usage]. 330 A sending WebRTC endpoint MUST be able to support the following 331 configurations: 333 o multiplex all media and data on a single 5-tuple (fully bundled) 334 o send each media stream on its own 5-tuple and data on its own 335 5-tuple (fully unbundled) 337 o bundle each media type (audio, video or data) into its own 5-tuple 338 (bundling by media type) 340 It MAY choose to support other configurations. 342 Sending data over multiple 5-tuples is not supported. 344 A receiving WebRTC endpoint MUST be able to receive media and data in 345 all these configurations. 347 4.2. Local prioritization 349 When an WebRTC endpoint has packets to send on multiple streams (with 350 each media stream and each data channel considered as one "stream" 351 for this purpose) that are congestion-controlled under the same 352 congestion controller, the WebRTC endpoint SHOULD cause data to be 353 emitted in such a way that each stream at each level of priority is 354 being given approximately twice the transmission capacity (measured 355 in payload bytes) of the level below. 357 Thus, when congestion occurs, a "very high" priority flow will have 358 the ability to send 8 times as much data as a "below normal" flow if 359 both have data to send. This prioritization is independent of the 360 media type. The details of which packet to send first are 361 implementation defined. 363 For example: If there is a very high priority audio flow sending 100 364 byte packets, and a normal priority video flow sending 1000 byte 365 packets, and outgoing capacity exists for sending >5000 payload 366 bytes, it would be appropriate to send 4000 bytes (40 packets) of 367 audio and 1000 bytes (one packet) of video as the result of a single 368 pass of sending decisions. 370 Conversely, if the audio flow is marked normal priority and the video 371 flow is marked very high priority, the scheduler may decide to send 2 372 video packets (2000 bytes) and 5 audio packets (500 bytes) when 373 outgoing capacity exists for sending > 2500 payload bytes. 375 If there are two very high priority audio flows, each will be able to 376 send 4000 bytes in the same period where a normal priority video flow 377 is able to send 1000 bytes. 379 Two example implementation strategies are: 381 o When the available bandwidth is known from the congestion control 382 algorithm, configure each codec and each data channel with a 383 target send rate that is appropriate to its share of the available 384 bandwidth. 386 o When congestion control indicates that a specified number of 387 packets can be sent, send packets that are available to send using 388 a weighted round robin scheme across the connections. 390 Any combination of these, or other schemes that have the same effect, 391 is valid, as long as the distribution of transmission capacity is 392 approximately correct. 394 For media, it is usually inappropriate to use deep queues for 395 sending; it is more useful to, for instance, skip intermediate frames 396 that have no dependencies on them in order to achieve a lower 397 bitrate. For reliable data, queues are useful. 399 5. IANA Considerations 401 This document makes no request of IANA. 403 Note to RFC Editor: this section may be removed on publication as an 404 RFC. 406 6. Security Considerations 408 Security considerations are enumerated in [I-D.ietf-rtcweb-security]. 410 7. Acknowledgements 412 This document is based on earlier versions embedded in 413 [I-D.ietf-rtcweb-overview], which were the results of contributions 414 from many RTCWEB WG members. 416 Special thanks for reviews of earlier versions of this draft go to 417 Eduardo Gueiros, Magnus Westerlund, Markus Isomaki and Dan Wing; the 418 contributions from Andrew Hutton also deserve special mention. 420 8. References 422 8.1. Normative References 424 [I-D.ietf-httpbis-tunnel-protocol] 425 Hutton, A., Uberti, J., and M. Thomson, "The Tunnel- 426 Protocol HTTP Request Header Field", draft-ietf-httpbis- 427 tunnel-protocol-00 (work in progress), August 2014. 429 [I-D.ietf-mmusic-sctp-sdp] 430 Loreto, S. and G. Camarillo, "Stream Control Transmission 431 Protocol (SCTP)-Based Media Transport in the Session 432 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-07 433 (work in progress), July 2014. 435 [I-D.ietf-rtcweb-data-channel] 436 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 437 Channels", draft-ietf-rtcweb-data-channel-11 (work in 438 progress), July 2014. 440 [I-D.ietf-rtcweb-rtp-usage] 441 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 442 Communication (WebRTC): Media Transport and Use of RTP", 443 draft-ietf-rtcweb-rtp-usage-15 (work in progress), May 444 2014. 446 [I-D.ietf-rtcweb-security] 447 Rescorla, E., "Security Considerations for WebRTC", draft- 448 ietf-rtcweb-security-07 (work in progress), July 2014. 450 [I-D.ietf-rtcweb-security-arch] 451 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 452 rtcweb-security-arch-10 (work in progress), July 2014. 454 [I-D.ietf-tsvwg-rtcweb-qos] 455 Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. 456 Polk, "DSCP and other packet markings for RTCWeb QoS", 457 draft-ietf-tsvwg-rtcweb-qos-02 (work in progress), June 458 2014. 460 [I-D.ietf-tsvwg-sctp-dtls-encaps] 461 Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS 462 Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- 463 dtls-encaps-05 (work in progress), July 2014. 465 [I-D.ietf-tsvwg-sctp-ndata] 466 Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, 467 "Stream Schedulers and a New Data Chunk for the Stream 468 Control Transmission Protocol", draft-ietf-tsvwg-sctp- 469 ndata-01 (work in progress), July 2014. 471 [I-D.reddy-mmusic-ice-happy-eyeballs] 472 Reddy, T., Patil, P., and P. Martinsen, "Happy Eyeballs 473 Extension for ICE", draft-reddy-mmusic-ice-happy- 474 eyeballs-07 (work in progress), June 2014. 476 [I-D.thomson-rtcweb-alpn] 477 Thomson, M., "Application Layer Protocol Negotiation for 478 Web Real-Time Communications (WebRTC)", draft-thomson- 479 rtcweb-alpn-00 (work in progress), April 2014. 481 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 482 Requirement Levels", BCP 14, RFC 2119, March 1997. 484 [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) 485 and RTP Control Protocol (RTCP) Packets over Connection- 486 Oriented Transport", RFC 4571, July 2006. 488 [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy 489 Extensions for Stateless Address Autoconfiguration in 490 IPv6", RFC 4941, September 2007. 492 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 493 (ICE): A Protocol for Network Address Translator (NAT) 494 Traversal for Offer/Answer Protocols", RFC 5245, April 495 2010. 497 [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, 498 "Session Traversal Utilities for NAT (STUN)", RFC 5389, 499 October 2008. 501 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 502 Security (DTLS) Extension to Establish Keys for the Secure 503 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 505 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 506 Relays around NAT (TURN): Relay Extensions to Session 507 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 509 [RFC6062] Perreault, S. and J. Rosenberg, "Traversal Using Relays 510 around NAT (TURN) Extensions for TCP Allocations", RFC 511 6062, November 2010. 513 [RFC6156] Camarillo, G., Novo, O., and S. Perreault, "Traversal 514 Using Relays around NAT (TURN) Extension for IPv6", RFC 515 6156, April 2011. 517 [RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach, 518 "TCP Candidates with Interactive Connectivity 519 Establishment (ICE)", RFC 6544, March 2012. 521 [RFC6724] Thaler, D., Draves, R., Matsumoto, A., and T. Chown, 522 "Default Address Selection for Internet Protocol Version 6 523 (IPv6)", RFC 6724, September 2012. 525 [RFC7231] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol 526 (HTTP/1.1): Semantics and Content", RFC 7231, June 2014. 528 [RFC7235] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol 529 (HTTP/1.1): Authentication", RFC 7235, June 2014. 531 8.2. Informative References 533 [I-D.ietf-dart-dscp-rtp] 534 Black, D. and P. Jones, "Differentiated Services 535 (DiffServ) and Real-time Communication", draft-ietf-dart- 536 dscp-rtp-08 (work in progress), October 2014. 538 [I-D.ietf-rtcweb-overview] 539 Alvestrand, H., "Overview: Real Time Protocols for 540 Browser-based Applications", draft-ietf-rtcweb-overview-10 541 (work in progress), June 2014. 543 [I-D.jesup-rtcweb-data-protocol] 544 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 545 Protocol", draft-jesup-rtcweb-data-protocol-04 (work in 546 progress), February 2013. 548 [RFC3484] Draves, R., "Default Address Selection for Internet 549 Protocol version 6 (IPv6)", RFC 3484, February 2003. 551 [RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6 552 Socket API for Source Address Selection", RFC 5014, 553 September 2007. 555 [RFC5128] Srisuresh, P., Ford, B., and D. Kegel, "State of Peer-to- 556 Peer (P2P) Communication across Network Address 557 Translators (NATs)", RFC 5128, March 2008. 559 Appendix A. Change log 561 This section should be removed before publication as an RFC. 563 A.1. Changes from -00 to -01 565 o Clarified DSCP requirements, with reference to -qos- 567 o Clarified "symmetric NAT" -> "NATs which perform endpoint- 568 dependent mapping" 570 o Made support of TURN over TCP mandatory 572 o Made support of TURN over TLS a MAY, and added open question 573 o Added an informative reference to -firewalls- 575 o Called out that we don't make requirements on HTTP proxy 576 interaction (yet 578 A.2. Changes from -01 to -02 580 o Required support for 300 Alternate Server from STUN. 582 o Separated the ICE-TCP candidate requirement from the TURN-TCP 583 requirement. 585 o Added new sections on using QoS functions, and on multiplexing 586 considerations. 588 o Removed all mention of RTP profiles. Those are the business of 589 the RTP usage draft, not this one. 591 o Required support for TURN IPv6 extensions. 593 o Removed reference to the TURN URI scheme, as it was unnecessary. 595 o Made an explicit statement that multiplexing (or not) is an 596 application matter. 598 . 600 A.3. Changes from -02 to -03 602 o Added required support for draft-ietf-tsvwg-sctp-ndata 604 o Removed discussion of multiplexing, since this is present in rtp- 605 usage. 607 o Added RFC 4571 reference for framing RTP packets over TCP. 609 o Downgraded TURN TCP candidates from SHOULD to MAY, and added more 610 language discussing TCP usage. 612 o Added language on IPv6 temporary addresses. 614 o Added language describing multiplexing choices. 616 o Added a separate section detailing what it means when we say that 617 an WebRTC implementation MUST support both IPv4 and IPv6. 619 A.4. Changes from -03 to -04 621 o Added a section on prioritization, moved the DSCP section into it, 622 and added a section on local prioritization, giving a specific 623 algorithm for interpreting "priority" in local prioritization. 625 o ICE-TCP candidates was changed from MAY to MUST, in recognition of 626 the sense of the room at the London IETF. 628 A.5. Changes from -04 to -05 630 o Reworded introduction 632 o Removed all references to "WebRTC". It now uses only the term 633 RTCWEB. 635 o Addressed a number of clarity / language comments 637 o Rewrote the prioritization to cover data channels and to describe 638 multiple ways of prioritizing flows 640 o Made explicit reference to "MUST do DTLS-SRTP", and referred to 641 security-arch for details 643 A.6. Changes from -05 to -06 645 o Changed all references to "RTCWEB" to "WebRTC", except one 646 reference to the working group 648 o Added reference to the httpbis "connect" protocol (being adopted 649 by HTTPBIS) 651 o Added reference to the ALPN header (being adopted by RTCWEB) 653 o Added reference to the DART RTP document 655 o Said explicitly that SCTP for data channels has a single DSCP 656 codepoint 658 A.7. Changes from -06 to -07 660 o Updated terminology in accordance with -overview. Got rid of all 661 occurences of "WebRTC implementation". 663 o Modified description of ICE-TCP encapsulation in accordance with 664 list discussion. 666 o Added HTTP CONNECT requirement in accordance with list discussion. 668 Author's Address 670 Harald Alvestrand 671 Google 673 Email: harald@alvestrand.no