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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 August 11, 2014 5 Expires: February 12, 2015 7 Transports for WebRTC 8 draft-ietf-rtcweb-transports-06 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 February 12, 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 . . . . . . . . . . . . . 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 . . . . . . . . . . . . . . . . . 11 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 . . . . . . . . . . . . . . . . . 13 72 A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 14 73 A.6. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 14 74 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 14 76 1. Introduction 78 WebRTC is a protocol suite aimed at real time multimedia exchange 79 between browsers, and between browsers and other entities. 81 WebRTC is described in the WebRTC overview document, 82 [I-D.ietf-rtcweb-overview], which also defines terminology used in 83 this document. 85 This document focuses on the data transport protocols that are used 86 by conforming implementations, including the protocols used for 87 interaction with intermediate boxes such as firewalls, relays and NAT 88 boxes. 90 This protocol suite intends to satisfy the security considerations 91 described in the WebRTC security documents, 92 [I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch]. 94 This document describes requirements that apply to all WebRTC 95 devices. When there are requirements that apply only to WebRTC 96 browsers, this is called out by using the word "browser". 98 2. Requirements language 100 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 101 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 102 document are to be interpreted as described in RFC 2119 [RFC2119]. 104 3. Transport and Middlebox specification 106 3.1. System-provided interfaces 108 The protocol specifications used here assume that the following 109 protocols are available to the implementations of the WebRTC 110 protocols: 112 o UDP. This is the protocol assumed by most protocol elements 113 described. 115 o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL 116 and ICE-TCP. 118 For both protocols, IPv4 and IPv6 support is assumed. 120 For UDP, this specification assumes the ability to set the DSCP code 121 point of the sockets opened on a per-packet basis, in order to 122 achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos] 123 (see Section 4.1) when multiple media types are multiplexed. It does 124 not assume that the DSCP codepoints will be honored, and does assume 125 that they may be zeroed or changed, since this is a local 126 configuration issue. 128 Platforms that do not give access to these interfaces will not be 129 able to support a conforming WebRTC implementation. 131 This specification does not assume that the implementation will have 132 access to ICMP or raw IP. 134 3.2. Ability to use IPv4 and IPv6 136 Web applications running in a WebRTC browser MUST be able to utilize 137 both IPv4 and IPv6 where available - that is, when two peers have 138 only IPv4 connectivity to each other, or they have only IPv6 139 connectivity to each other, applications running in the WebRTC 140 browser MUST be able to communicate. 142 When TURN is used, and the TURN server has IPv4 or IPv6 connectivity 143 to the peer or its TURN server, candidates of the appropriate types 144 MUST be supported. The "Happy Eyeballs" specification for ICE 145 [I-D.reddy-mmusic-ice-happy-eyeballs] SHOULD be supported. 147 3.3. Usage of temporary IPv6 addresses 149 The IPv6 default address selection specification [RFC6724] specifies 150 that temporary addresses [RFC4941] are to be preferred over permanent 151 addresses. This is a change from the rules specified by [RFC3484]. 152 For applications that select a single address, this is usually done 153 by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014]. 154 However, this rule is not completely obvious in the ICE scope. This 155 is therefore clarified as follows: 157 When a client gathers all IPv6 addresses on a host, and both 158 temporary addresses and permanent addresses of the same scope are 159 present, the client SHOULD discard the permanent addresses before 160 forming pairs. This is consistent with the default policy described 161 in [RFC6724]. 163 3.4. Middle box related functions 165 The primary mechanism to deal with middle boxes is ICE, which is an 166 appropriate way to deal with NAT boxes and firewalls that accept 167 traffic from the inside, but only from the outside if it is in 168 response to inside traffic (simple stateful firewalls). 170 ICE [RFC5245] MUST be supported. The implementation MUST be a full 171 ICE implementation, not ICE-Lite. A full ICE implementation allows 172 interworking with both ICE and ICE-Lite implementations when they are 173 deployed appropriately. 175 In order to deal with situations where both parties are behind NATs 176 of the type that perform endpoint-dependent mapping (as defined in 177 [RFC5128] section 2.4), TURN [RFC5766] MUST be supported. 179 WebRTC browsers MUST support configuration of STUN and TURN servers, 180 both from browser configuration and from an application. 182 In order to deal with firewalls that block all UDP traffic, the mode 183 of TURN that uses TCP between the client and the server MUST be 184 supported, and the mode of TURN that uses TLS over TCP between the 185 client and the server MUST be supported. See [RFC5766] section 2.1 186 for details. 188 In order to deal with situations where one party is on an IPv4 189 network and the other party is on an IPv6 network, TURN extensions 190 for IPv6 [RFC6156] MUST be supported. 192 TURN TCP candidates, where the connection from the client's TURN 193 server to the peer is a TCP connection, [RFC6062] MAY be supported. 195 However, such candidates are not seen as providing any significant 196 benefit, for the following reasons. 198 First, use of TURN TCP candidates would only be relevant in cases 199 which both peers are required to use TCP to establish a 200 PeerConnection. 202 Second, that use case is supported in a different way by both sides 203 establishing UDP relay candidates using TURN over TCP to connect to 204 their respective relay servers. 206 Third, using TCP only between the endpoint and its relay may result 207 in less issues with TCP in regards to real-time constraints, e.g. due 208 to head of line blocking. 210 ICE-TCP candidates [RFC6544] MUST be supported; this may allow 211 applications to communicate to peers with public IP addresses across 212 UDP-blocking firewalls without using a TURN server. 214 If TCP connections are used, RTP framing according to [RFC4571] MUST 215 be used, both for the RTP packets and for the DTLS packets used to 216 carry data channels. 218 The ALTERNATE-SERVER mechanism specified in [RFC5389] (STUN) section 219 11 (300 Try Alternate) MUST be supported. 221 Further discussion of the interaction of WebRTC with firewalls is 222 contained in [I-D.hutton-rtcweb-nat-firewall-considerations]. This 223 document makes no requirements on interacting with HTTP proxies or 224 HTTP proxy configuration methods. 226 The WebRTC implementation MAY support accessing the Internet through 227 an HTTP proxy. If it does so, it MUST support the "connect" header 228 as specified in [I-D.hutton-httpbis-connect-protocol]. 230 3.5. Transport protocols implemented 232 For transport of media, secure RTP is used. The details of the 233 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 is described in 245 [I-D.jesup-rtcweb-data-protocol]. 247 WebRTC implementations MUST support multiplexing of DTLS and RTP over 248 the same port pair, as described in the DTLS_SRTP specification 249 [RFC5764], section 5.1.2. All application layer protocol payloads 250 over this DTLS connection are SCTP packets. 252 Protocol identification MUST be supplied as part of the DTLS 253 handshake, as specified in [I-D.thomson-rtcweb-alpn]. 255 4. Media Prioritization 257 The WebRTC prioritization model is that the application tells the 258 WebRTC implementation about the priority of media and data flows 259 through an API. 261 The priority associated with a media or data flow is classified as 262 "normal", "below normal", "high" or "very high". There are only four 263 priority levels at the API. 265 The priority settings affect two pieces of behavior: Packet markings 266 and packet send sequence decisions. Each is described in its own 267 section below. 269 4.1. Usage of Quality of Service - DSCP and Multiplexing 271 Implementations SHOULD attempt to set QoS on the packets sent, 272 according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos]. It is 273 appropriate to depart from this recommendation when running on 274 platforms where QoS marking is not implemented. 276 The implementation MAY turn off use of DSCP markings if it detects 277 symptoms of unexpected behaviour like priority inversion or blocking 278 of packets with certain DSCP markings. The detection of these 279 conditions is implementation dependent. (Question: Does there need 280 to be an API knob to turn off DSCP markings?) 282 All packets arrying data from the SCTP association supporting the 283 data channels MUST use a single DSCP code point. 285 All packets on one TCP connection, no matter what it carries, MUST 286 use a single DSCP code point. 288 More advice on the use of DSCP code points with RTP is given in 289 [I-D.ietf-dart-dscp-rtp]. 291 There exist a number of schemes for achieving quality of service that 292 do not depend solely on DSCP code points. Some of these schemes 293 depend on classifying the traffic into flows based on 5-tuple (source 294 address, source port, protocol, destination address, destination 295 port) or 6-tuple (5-tuple + DSCP code point). Under differing 296 conditions, it may therefore make sense for a sending application to 297 choose any of the configurations: 299 o Each media stream carried on its own 5-tuple 301 o Media streams grouped by media type into 5-tuples (such as 302 carrying all audio on one 5-tuple) 304 o All media sent over a single 5-tuple, with or without 305 differentiation into 6-tuples based on DSCP code points 307 In each of the configurations mentioned, data channels may be carried 308 in its own 5-tuple, or multiplexed together with one of the media 309 flows. 311 More complex configurations, such as sending a high priority video 312 stream on one 5-tuple and sending all other video streams multiplexed 313 together over another 5-tuple, can also be envisioned. More 314 information on mapping media flows to 5-tuples can be found in 315 [I-D.ietf-rtcweb-rtp-usage]. 317 A sending implementation MUST be able to support the following 318 configurations: 320 o multiplex all media and data on a single 5-tuple (fully bundled) 322 o send each media stream on its own 5-tuple and data on its own 323 5-tuple (fully unbundled) 325 o bundle each media type (audio, video or data) into its own 5-tuple 326 (bundling by media type) 328 It MAY choose to support other configurations. 330 Sending data over multiple 5-tuples is not supported. 332 A receiving implementation MUST be able to receive media and data in 333 all these configurations. 335 4.2. Local prioritization 337 When an WebRTC implementation has packets to send on multiple streams 338 (with each media stream and each data channel considered as one 339 "stream" for this purpose) that are congestion-controlled under the 340 same congestion controller, the WebRTC implementation SHOULD cause 341 data to be emitted in such a way that each stream at each level of 342 priority is being given approximately twice the transmission capacity 343 (measured in payload bytes) of the level below. 345 Thus, when congestion occurs, a "very high" priority flow will have 346 the ability to send 8 times as much data as a "below normal" flow if 347 both have data to send. This prioritization is independent of the 348 media type. The details of which packet to send first are 349 implementation defined. 351 For example: If there is a very high priority audio flow sending 100 352 byte packets, and a normal priority video flow sending 1000 byte 353 packets, and outgoing capacity exists for sending >5000 payload 354 bytes, it would be appropriate to send 4000 bytes (40 packets) of 355 audio and 1000 bytes (one packet) of video as the result of a single 356 pass of sending decisions. 358 Conversely, if the audio flow is marked normal priority and the video 359 flow is marked very high priority, the scheduler may decide to send 2 360 video packets (2000 bytes) and 5 audio packets (500 bytes) when 361 outgoing capacity exists for sending > 2500 payload bytes. 363 If there are two very high priority audio flows, each will be able to 364 send 4000 bytes in the same period where a normal priority video flow 365 is able to send 1000 bytes. 367 Two example implementation strategies are: 369 o When the available bandwidth is known from the congestion control 370 algorithm, configure each codec and each data channel with a 371 target send rate that is appropriate to its share of the available 372 bandwidth. 374 o When congestion control indicates that a specified number of 375 packets can be sent, send packets that are available to send using 376 a weighted round robin scheme across the connections. 378 Any combination of these, or other schemes that have the same effect, 379 is valid, as long as the distribution of transmission capacity is 380 approximately correct. 382 For media, it is usually inappropriate to use deep queues for 383 sending; it is more useful to, for instance, skip intermediate frames 384 that have no dependencies on them in order to achieve a lower 385 bitrate. For reliable data, queues are useful. 387 5. IANA Considerations 389 This document makes no request of IANA. 391 Note to RFC Editor: this section may be removed on publication as an 392 RFC. 394 6. Security Considerations 396 Security considerations are enumerated in [I-D.ietf-rtcweb-security]. 398 7. Acknowledgements 400 This document is based on earlier versions embedded in 401 [I-D.ietf-rtcweb-overview], which were the results of contributions 402 from many RTCWEB WG members. 404 Special thanks for reviews of earlier versions of this draft go to 405 Eduardo Gueiros, Magnus Westerlund, Markus Isomaki and Dan Wing; the 406 contributions from Andrew Hutton also deserve special mention. 408 8. References 410 8.1. Normative References 412 [I-D.hutton-httpbis-connect-protocol] 413 Hutton, A., Uberti, J., and M. Thomson, "HTTP Connect - 414 Tunnel Protocol For WebRTC", draft-hutton-httpbis-connect- 415 protocol-00 (work in progress), June 2014. 417 [I-D.ietf-mmusic-sctp-sdp] 418 Loreto, S. and G. Camarillo, "Stream Control Transmission 419 Protocol (SCTP)-Based Media Transport in the Session 420 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-07 421 (work in progress), July 2014. 423 [I-D.ietf-rtcweb-data-channel] 424 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 425 Channels", draft-ietf-rtcweb-data-channel-11 (work in 426 progress), July 2014. 428 [I-D.ietf-rtcweb-rtp-usage] 429 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 430 Communication (WebRTC): Media Transport and Use of RTP", 431 draft-ietf-rtcweb-rtp-usage-16 (work in progress), July 432 2014. 434 [I-D.ietf-rtcweb-security] 435 Rescorla, E., "Security Considerations for WebRTC", draft- 436 ietf-rtcweb-security-07 (work in progress), July 2014. 438 [I-D.ietf-rtcweb-security-arch] 439 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 440 rtcweb-security-arch-10 (work in progress), July 2014. 442 [I-D.ietf-tsvwg-rtcweb-qos] 443 Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. 444 Polk, "DSCP and other packet markings for RTCWeb QoS", 445 draft-ietf-tsvwg-rtcweb-qos-02 (work in progress), June 446 2014. 448 [I-D.ietf-tsvwg-sctp-dtls-encaps] 449 Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS 450 Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- 451 dtls-encaps-05 (work in progress), July 2014. 453 [I-D.ietf-tsvwg-sctp-ndata] 454 Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, 455 "Stream Schedulers and a New Data Chunk for the Stream 456 Control Transmission Protocol", draft-ietf-tsvwg-sctp- 457 ndata-01 (work in progress), July 2014. 459 [I-D.reddy-mmusic-ice-happy-eyeballs] 460 Reddy, T., Patil, P., and P. Martinsen, "Happy Eyeballs 461 Extension for ICE", draft-reddy-mmusic-ice-happy- 462 eyeballs-07 (work in progress), June 2014. 464 [I-D.thomson-rtcweb-alpn] 465 Thomson, M., "Application Layer Protocol Negotiation for 466 Web Real-Time Communications (WebRTC)", draft-thomson- 467 rtcweb-alpn-00 (work in progress), April 2014. 469 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 470 Requirement Levels", BCP 14, RFC 2119, March 1997. 472 [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) 473 and RTP Control Protocol (RTCP) Packets over Connection- 474 Oriented Transport", RFC 4571, July 2006. 476 [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy 477 Extensions for Stateless Address Autoconfiguration in 478 IPv6", RFC 4941, September 2007. 480 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 481 (ICE): A Protocol for Network Address Translator (NAT) 482 Traversal for Offer/Answer Protocols", RFC 5245, April 483 2010. 485 [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, 486 "Session Traversal Utilities for NAT (STUN)", RFC 5389, 487 October 2008. 489 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 490 Security (DTLS) Extension to Establish Keys for the Secure 491 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 493 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 494 Relays around NAT (TURN): Relay Extensions to Session 495 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 497 [RFC6062] Perreault, S. and J. Rosenberg, "Traversal Using Relays 498 around NAT (TURN) Extensions for TCP Allocations", RFC 499 6062, November 2010. 501 [RFC6156] Camarillo, G., Novo, O., and S. Perreault, "Traversal 502 Using Relays around NAT (TURN) Extension for IPv6", RFC 503 6156, April 2011. 505 [RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach, 506 "TCP Candidates with Interactive Connectivity 507 Establishment (ICE)", RFC 6544, March 2012. 509 [RFC6724] Thaler, D., Draves, R., Matsumoto, A., and T. Chown, 510 "Default Address Selection for Internet Protocol Version 6 511 (IPv6)", RFC 6724, September 2012. 513 8.2. Informative References 515 [I-D.hutton-rtcweb-nat-firewall-considerations] 516 Stach, T., Hutton, A., and J. Uberti, "RTCWEB 517 Considerations for NATs, Firewalls and HTTP proxies", 518 draft-hutton-rtcweb-nat-firewall-considerations-03 (work 519 in progress), January 2014. 521 [I-D.ietf-dart-dscp-rtp] 522 Black, D. and P. Jones, "Differentiated Services 523 (DiffServ) and Real-time Communication", draft-ietf-dart- 524 dscp-rtp-02 (work in progress), August 2014. 526 [I-D.ietf-rtcweb-overview] 527 Alvestrand, H., "Overview: Real Time Protocols for 528 Browser-based Applications", draft-ietf-rtcweb-overview-10 529 (work in progress), June 2014. 531 [I-D.jesup-rtcweb-data-protocol] 532 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 533 Protocol", draft-jesup-rtcweb-data-protocol-04 (work in 534 progress), February 2013. 536 [RFC3484] Draves, R., "Default Address Selection for Internet 537 Protocol version 6 (IPv6)", RFC 3484, February 2003. 539 [RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6 540 Socket API for Source Address Selection", RFC 5014, 541 September 2007. 543 [RFC5128] Srisuresh, P., Ford, B., and D. Kegel, "State of Peer-to- 544 Peer (P2P) Communication across Network Address 545 Translators (NATs)", RFC 5128, March 2008. 547 Appendix A. Change log 549 This section should be removed before publication as an RFC. 551 A.1. Changes from -00 to -01 553 o Clarified DSCP requirements, with reference to -qos- 555 o Clarified "symmetric NAT" -> "NATs which perform endpoint- 556 dependent mapping" 558 o Made support of TURN over TCP mandatory 560 o Made support of TURN over TLS a MAY, and added open question 562 o Added an informative reference to -firewalls- 564 o Called out that we don't make requirements on HTTP proxy 565 interaction (yet 567 A.2. Changes from -01 to -02 569 o Required support for 300 Alternate Server from STUN. 571 o Separated the ICE-TCP candidate requirement from the TURN-TCP 572 requirement. 574 o Added new sections on using QoS functions, and on multiplexing 575 considerations. 577 o Removed all mention of RTP profiles. Those are the business of 578 the RTP usage draft, not this one. 580 o Required support for TURN IPv6 extensions. 582 o Removed reference to the TURN URI scheme, as it was unnecessary. 584 o Made an explicit statement that multiplexing (or not) is an 585 application matter. 587 . 589 A.3. Changes from -02 to -03 591 o Added required support for draft-ietf-tsvwg-sctp-ndata 593 o Removed discussion of multiplexing, since this is present in rtp- 594 usage. 596 o Added RFC 4571 reference for framing RTP packets over TCP. 598 o Downgraded TURN TCP candidates from SHOULD to MAY, and added more 599 language discussing TCP usage. 601 o Added language on IPv6 temporary addresses. 603 o Added language describing multiplexing choices. 605 o Added a separate section detailing what it means when we say that 606 an WebRTC implementation MUST support both IPv4 and IPv6. 608 A.4. Changes from -03 to -04 610 o Added a section on prioritization, moved the DSCP section into it, 611 and added a section on local prioritization, giving a specific 612 algorithm for interpreting "priority" in local prioritization. 614 o ICE-TCP candidates was changed from MAY to MUST, in recognition of 615 the sense of the room at the London IETF. 617 A.5. Changes from -04 to -05 619 o Reworded introduction 621 o Removed all references to "WebRTC". It now uses only the term 622 RTCWEB. 624 o Addressed a number of clarity / language comments 626 o Rewrote the prioritization to cover data channels and to describe 627 multiple ways of prioritizing flows 629 o Made explicit reference to "MUST do DTLS-SRTP", and referred to 630 security-arch for details 632 A.6. Changes from -05 to -06 634 o Changed all references to "RTCWEB" to "WebRTC", except one 635 reference to the working group 637 o Added reference to the httpbis "connect" protocol (being adopted 638 by HTTPBIS) 640 o Added reference to the ALPN header (being adopted by RTCWEB) 642 o Added reference to the DART RTP document 644 o Said explicitly that SCTP for data channels has a single DSCP 645 codepoint 647 Author's Address 649 Harald Alvestrand 650 Google 652 Email: harald@alvestrand.no