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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 19, 2015 5 Expires: April 21, 2016 7 Transports for WebRTC 8 draft-ietf-rtcweb-transports-10 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 21, 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 . . . . . . . . . . . . . 6 58 4. Media Prioritization . . . . . . . . . . . . . . . . . . . . 6 59 4.1. Local prioritization . . . . . . . . . . . . . . . . . . 7 60 4.2. Usage of Quality of Service - DSCP and Multiplexing . . . 8 61 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 62 6. Security Considerations . . . . . . . . . . . . . . . . . . . 9 63 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 10 64 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 10 65 8.1. Normative References . . . . . . . . . . . . . . . . . . 10 66 8.2. Informative References . . . . . . . . . . . . . . . . . 12 67 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 13 68 A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 13 69 A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 13 70 A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 14 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 . . . . . . . . . . . . . . . . . 15 74 A.7. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 15 75 A.8. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 15 76 A.9. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 15 77 A.10. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 15 78 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 16 80 1. Introduction 82 WebRTC is a protocol suite aimed at real time multimedia exchange 83 between browsers, and between browsers and other entities. 85 WebRTC is described in the WebRTC overview document, 86 [I-D.ietf-rtcweb-overview], which also defines terminology used in 87 this document, including the terms "WebRTC device" and "WebRTC 88 browser". 90 This document focuses on the data transport protocols that are used 91 by conforming implementations, including the protocols used for 92 interaction with intermediate boxes such as firewalls, relays and NAT 93 boxes. 95 This protocol suite intends to satisfy the security considerations 96 described in the WebRTC security documents, 97 [I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch]. 99 This document describes requirements that apply to all WebRTC 100 devices. When there are requirements that apply only to WebRTC 101 browsers, this is called out explicitly. 103 2. Requirements language 105 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 106 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 107 document are to be interpreted as described in RFC 2119 [RFC2119]. 109 3. Transport and Middlebox specification 111 3.1. System-provided interfaces 113 The protocol specifications used here assume that the following 114 protocols are available to the implementations of the WebRTC 115 protocols: 117 o UDP [RFC0768]. This is the protocol assumed by most protocol 118 elements described. 120 o TCP [RFC0793]. This is used for HTTP/WebSockets, as well as for 121 TURN/SSL and ICE-TCP. 123 For both protocols, IPv4 and IPv6 support is assumed. 125 For UDP, this specification assumes the ability to set the DSCP code 126 point of the sockets opened on a per-packet basis, in order to 127 achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos] 128 (see Section 4.2) when multiple media types are multiplexed. It does 129 not assume that the DSCP codepoints will be honored, and does assume 130 that they may be zeroed or changed, since this is a local 131 configuration issue. 133 Platforms that do not give access to these interfaces will not be 134 able to support a conforming WebRTC implementation. 136 This specification does not assume that the implementation will have 137 access to ICMP or raw IP. 139 3.2. Ability to use IPv4 and IPv6 141 Web applications running in a WebRTC browser MUST be able to utilize 142 both IPv4 and IPv6 where available - that is, when two peers have 143 only IPv4 connectivity to each other, or they have only IPv6 144 connectivity to each other, applications running in the WebRTC 145 browser MUST be able to communicate. 147 When TURN is used, and the TURN server has IPv4 or IPv6 connectivity 148 to the peer or its TURN server, candidates of the appropriate types 149 MUST be supported. The "Happy Eyeballs" specification for ICE 150 [I-D.martinsen-mmusic-ice-dualstack-fairness] SHOULD be supported. 152 3.3. Usage of temporary IPv6 addresses 154 The IPv6 default address selection specification [RFC6724] specifies 155 that temporary addresses [RFC4941] are to be preferred over permanent 156 addresses. This is a change from the rules specified by [RFC3484]. 157 For applications that select a single address, this is usually done 158 by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014]. 159 However, this rule is not completely obvious in the ICE scope. This 160 is therefore clarified as follows: 162 When a client gathers all IPv6 addresses on a host, and both 163 temporary addresses and permanent addresses of the same scope are 164 present, the client SHOULD discard the permanent addresses before 165 exposing addresses to the application or using them in ICE. This is 166 consistent with the default policy described in [RFC6724]. 168 3.4. Middle box related functions 170 The primary mechanism to deal with middle boxes is ICE, which is an 171 appropriate way to deal with NAT boxes and firewalls that accept 172 traffic from the inside, but only from the outside if it is in 173 response to inside traffic (simple stateful firewalls). 175 ICE [RFC5245] MUST be supported. The implementation MUST be a full 176 ICE implementation, not ICE-Lite. A full ICE implementation allows 177 interworking with both ICE and ICE-Lite implementations when they are 178 deployed appropriately. 180 In order to deal with situations where both parties are behind NATs 181 of the type that perform endpoint-dependent mapping (as defined in 182 [RFC5128] section 2.4), TURN [RFC5766] MUST be supported. 184 WebRTC browsers MUST support configuration of STUN and TURN servers, 185 both from browser configuration and from an application. 187 In order to deal with firewalls that block all UDP traffic, the mode 188 of TURN that uses TCP between the client and the server MUST be 189 supported, and the mode of TURN that uses TLS over TCP between the 190 client and the server MUST be supported. See [RFC5766] section 2.1 191 for details. 193 In order to deal with situations where one party is on an IPv4 194 network and the other party is on an IPv6 network, TURN extensions 195 for IPv6 [RFC6156] MUST be supported. 197 TURN TCP candidates, where the connection from the client's TURN 198 server to the peer is a TCP connection, [RFC6062] MAY be supported. 200 However, such candidates are not seen as providing any significant 201 benefit, for the following reasons. 203 First, use of TURN TCP candidates would only be relevant in cases 204 which both peers are required to use TCP to establish a 205 PeerConnection. 207 Second, that use case is supported in a different way by both sides 208 establishing UDP relay candidates using TURN over TCP to connect to 209 their respective relay servers. 211 Third, using TCP only between the endpoint and its relay may result 212 in less issues with TCP in regards to real-time constraints, e.g. due 213 to head of line blocking. 215 ICE-TCP candidates [RFC6544] MUST be supported; this may allow 216 applications to communicate to peers with public IP addresses across 217 UDP-blocking firewalls without using a TURN server. 219 If TCP connections are used, RTP framing according to [RFC4571] MUST 220 be used, both for the RTP packets and for the DTLS packets used to 221 carry data channels. 223 The ALTERNATE-SERVER mechanism specified in [RFC5389] (STUN) section 224 11 (300 Try Alternate) MUST be supported. 226 The WebRTC implementation MAY support accessing the Internet through 227 an HTTP proxy. If it does so, it MUST support the "ALPN" header as 228 specified in [RFC7639], and proxy authentication as described in 229 Section 4.3.6 of [RFC7231] and [RFC7235] MUST also be supported. 231 3.5. Transport protocols implemented 233 For transport of media, secure RTP is used. The details of the 234 profile of RTP used are described in "RTP Usage" 235 [I-D.ietf-rtcweb-rtp-usage]. Key exchange MUST be done using DTLS- 236 SRTP, as described in [I-D.ietf-rtcweb-security-arch]. 238 For data transport over the WebRTC data channel 239 [I-D.ietf-rtcweb-data-channel], WebRTC implementations MUST support 240 SCTP over DTLS over ICE. This encapsulation is specified in 241 [I-D.ietf-tsvwg-sctp-dtls-encaps]. Negotiation of this transport in 242 SDP is defined in [I-D.ietf-mmusic-sctp-sdp]. The SCTP extension for 243 NDATA, [I-D.ietf-tsvwg-sctp-ndata], MUST be supported. 245 The setup protocol for WebRTC data channels described in 246 [I-D.ietf-rtcweb-data-protocol] MUST be supported. 248 Note: DTLS-SRTP as defined in [RFC5764] section 6.7.1 defines the 249 interaction between DTLS and ICE ( [RFC5245]). The effect of this 250 specification is that all ICE candidate pairs associated with a 251 single component are part of the same DTLS association. Thus, there 252 will only be one DTLS handshake even if there are multiple valid 253 candidate pairs. 255 WebRTC implementations MUST support multiplexing of DTLS and RTP over 256 the same port pair, as described in the DTLS-SRTP specification 257 [RFC5764], section 5.1.2. All application layer protocol payloads 258 over this DTLS connection are SCTP packets. 260 Protocol identification MUST be supplied as part of the DTLS 261 handshake, as specified in [I-D.ietf-rtcweb-alpn]. 263 4. Media Prioritization 265 The WebRTC prioritization model is that the application tells the 266 WebRTC implementation about the priority of media and data flows 267 through an API. 269 The priority associated with a media or data flow is classified as 270 "normal", "below normal", "high" or "very high". There are only four 271 priority levels at the API. 273 The priority settings affect two pieces of behavior: Packet send 274 sequence decisions and packet markings. Each is described in its own 275 section below. 277 4.1. Local prioritization 279 Local prioritization is applied at the local node, before the packet 280 is sent. This means that the prioritization has full access to the 281 data about the individual packets, and can choose differing treatment 282 based on the stream a packet belongs to. 284 When an WebRTC implementation has packets to send on multiple streams 285 (with each media stream and each data channel considered as one 286 "stream" for this purpose) that are congestion-controlled under the 287 same congestion controller, the WebRTC implementation SHOULD cause 288 data to be emitted in such a way that each stream at each level of 289 priority is being given approximately twice the transmission capacity 290 (measured in payload bytes) of the level below. 292 Thus, when congestion occurs, a "very high" priority flow will have 293 the ability to send 8 times as much data as a "below normal" flow if 294 both have data to send. This prioritization is independent of the 295 media type. The details of which packet to send first are 296 implementation defined. 298 For example: If there is a very high priority audio flow sending 100 299 byte packets, and a normal priority video flow sending 1000 byte 300 packets, and outgoing capacity exists for sending >5000 payload 301 bytes, it would be appropriate to send 4000 bytes (40 packets) of 302 audio and 1000 bytes (one packet) of video as the result of a single 303 pass of sending decisions. 305 Conversely, if the audio flow is marked normal priority and the video 306 flow is marked very high priority, the scheduler may decide to send 2 307 video packets (2000 bytes) and 5 audio packets (500 bytes) when 308 outgoing capacity exists for sending > 2500 payload bytes. 310 If there are two very high priority audio flows, each will be able to 311 send 4000 bytes in the same period where a normal priority video flow 312 is able to send 1000 bytes. 314 Two example implementation strategies are: 316 o When the available bandwidth is known from the congestion control 317 algorithm, configure each codec and each data channel with a 318 target send rate that is appropriate to its share of the available 319 bandwidth. 321 o When congestion control indicates that a specified number of 322 packets can be sent, send packets that are available to send using 323 a weighted round robin scheme across the connections. 325 Any combination of these, or other schemes that have the same effect, 326 is valid, as long as the distribution of transmission capacity is 327 approximately correct. 329 For media, it is usually inappropriate to use deep queues for 330 sending; it is more useful to, for instance, skip intermediate frames 331 that have no dependencies on them in order to achieve a lower 332 bitrate. For reliable data, queues are useful. 334 4.2. Usage of Quality of Service - DSCP and Multiplexing 336 When the packet is sent, the network will make decisions about 337 queueing and/or discarding the packet that can affect the quality of 338 the communication. The sender can attempt to set the DSCP field of 339 the packet to influence these decisions. 341 Implementations SHOULD attempt to set QoS on the packets sent, 342 according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos]. It is 343 appropriate to depart from this recommendation when running on 344 platforms where QoS marking is not implemented. 346 The implementation MAY turn off use of DSCP markings if it detects 347 symptoms of unexpected behaviour like priority inversion or blocking 348 of packets with certain DSCP markings. The detection of these 349 conditions is implementation dependent. 351 All packets carrying data from the SCTP association supporting the 352 data channels MUST use a single DSCP code point. The code point used 353 SHOULD be that recommended by [I-D.ietf-tsvwg-rtcweb-qos] for the 354 highest priority data channel carried. Note that this means that all 355 data packets, no matter what their relative priority is, will be 356 treated the same by the network. 358 All packets on one TCP connection, no matter what it carries, MUST 359 use a single DSCP code point. 361 More advice on the use of DSCP code points with RTP is given in 362 [I-D.ietf-dart-dscp-rtp]. 364 There exist a number of schemes for achieving quality of service that 365 do not depend solely on DSCP code points. Some of these schemes 366 depend on classifying the traffic into flows based on 5-tuple (source 367 address, source port, protocol, destination address, destination 368 port) or 6-tuple (5-tuple + DSCP code point). Under differing 369 conditions, it may therefore make sense for a sending application to 370 choose any of the configurations: 372 o Each media stream carried on its own 5-tuple 373 o Media streams grouped by media type into 5-tuples (such as 374 carrying all audio on one 5-tuple) 376 o All media sent over a single 5-tuple, with or without 377 differentiation into 6-tuples based on DSCP code points 379 In each of the configurations mentioned, data channels may be carried 380 in its own 5-tuple, or multiplexed together with one of the media 381 flows. 383 More complex configurations, such as sending a high priority video 384 stream on one 5-tuple and sending all other video streams multiplexed 385 together over another 5-tuple, can also be envisioned. More 386 information on mapping media flows to 5-tuples can be found in 387 [I-D.ietf-rtcweb-rtp-usage]. 389 A sending implementation MUST be able to support the following 390 configurations: 392 o multiplex all media and data on a single 5-tuple (fully bundled) 394 o send each media stream on its own 5-tuple and data on its own 395 5-tuple (fully unbundled) 397 It MAY choose to support other configurations, such as bundling each 398 media type (audio, video or data) into its own 5-tuple (bundling by 399 media type). 401 Sending data over multiple 5-tuples is not supported. 403 A receiving implementation MUST be able to receive media and data in 404 all these configurations. 406 5. IANA Considerations 408 This document makes no request of IANA. 410 Note to RFC Editor: this section may be removed on publication as an 411 RFC. 413 6. Security Considerations 415 Security considerations are enumerated in [I-D.ietf-rtcweb-security]. 417 7. Acknowledgements 419 This document is based on earlier versions embedded in 420 [I-D.ietf-rtcweb-overview], which were the results of contributions 421 from many RTCWEB WG members. 423 Special thanks for reviews of earlier versions of this draft go to 424 Eduardo Gueiros, Magnus Westerlund, Markus Isomaki and Dan Wing; the 425 contributions from Andrew Hutton also deserve special mention. 427 8. References 429 8.1. Normative References 431 [I-D.ietf-mmusic-sctp-sdp] 432 Holmberg, C., Loreto, S., and G. Camarillo, "Stream 433 Control Transmission Protocol (SCTP)-Based Media Transport 434 in the Session Description Protocol (SDP)", draft-ietf- 435 mmusic-sctp-sdp-14 (work in progress), March 2015. 437 [I-D.ietf-rtcweb-alpn] 438 Thomson, M., "Application Layer Protocol Negotiation for 439 Web Real-Time Communications (WebRTC)", draft-ietf-rtcweb- 440 alpn-01 (work in progress), February 2015. 442 [I-D.ietf-rtcweb-data-channel] 443 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 444 Channels", draft-ietf-rtcweb-data-channel-13 (work in 445 progress), January 2015. 447 [I-D.ietf-rtcweb-data-protocol] 448 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 449 Establishment Protocol", draft-ietf-rtcweb-data- 450 protocol-09 (work in progress), January 2015. 452 [I-D.ietf-rtcweb-rtp-usage] 453 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 454 Communication (WebRTC): Media Transport and Use of RTP", 455 draft-ietf-rtcweb-rtp-usage-22 (work in progress), 456 February 2015. 458 [I-D.ietf-rtcweb-security] 459 Rescorla, E., "Security Considerations for WebRTC", draft- 460 ietf-rtcweb-security-08 (work in progress), February 2015. 462 [I-D.ietf-rtcweb-security-arch] 463 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 464 rtcweb-security-arch-11 (work in progress), March 2015. 466 [I-D.ietf-tsvwg-rtcweb-qos] 467 Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. 468 Polk, "DSCP and other packet markings for RTCWeb QoS", 469 draft-ietf-tsvwg-rtcweb-qos-03 (work in progress), 470 November 2014. 472 [I-D.ietf-tsvwg-sctp-dtls-encaps] 473 Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS 474 Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- 475 dtls-encaps-09 (work in progress), January 2015. 477 [I-D.ietf-tsvwg-sctp-ndata] 478 Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, 479 "Stream Schedulers and User Message Interleaving for the 480 Stream Control Transmission Protocol", draft-ietf-tsvwg- 481 sctp-ndata-03 (work in progress), March 2015. 483 [I-D.martinsen-mmusic-ice-dualstack-fairness] 484 Martinsen, P., Reddy, T., and P. Patil, "ICE IPv4/IPv6 485 Dual Stack Fairness", draft-martinsen-mmusic-ice- 486 dualstack-fairness-02 (work in progress), February 2015. 488 [RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, DOI 489 10.17487/RFC0768, August 1980, 490 . 492 [RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC 493 793, DOI 10.17487/RFC0793, September 1981, 494 . 496 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 497 Requirement Levels", BCP 14, RFC 2119, March 1997. 499 [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) 500 and RTP Control Protocol (RTCP) Packets over Connection- 501 Oriented Transport", RFC 4571, July 2006. 503 [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy 504 Extensions for Stateless Address Autoconfiguration in 505 IPv6", RFC 4941, September 2007. 507 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 508 (ICE): A Protocol for Network Address Translator (NAT) 509 Traversal for Offer/Answer Protocols", RFC 5245, April 510 2010. 512 [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, 513 "Session Traversal Utilities for NAT (STUN)", RFC 5389, 514 October 2008. 516 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 517 Security (DTLS) Extension to Establish Keys for the Secure 518 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 520 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 521 Relays around NAT (TURN): Relay Extensions to Session 522 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 524 [RFC6062] Perreault, S. and J. Rosenberg, "Traversal Using Relays 525 around NAT (TURN) Extensions for TCP Allocations", RFC 526 6062, November 2010. 528 [RFC6156] Camarillo, G., Novo, O., and S. Perreault, "Traversal 529 Using Relays around NAT (TURN) Extension for IPv6", RFC 530 6156, April 2011. 532 [RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach, 533 "TCP Candidates with Interactive Connectivity 534 Establishment (ICE)", RFC 6544, March 2012. 536 [RFC6724] Thaler, D., Draves, R., Matsumoto, A., and T. Chown, 537 "Default Address Selection for Internet Protocol Version 6 538 (IPv6)", RFC 6724, September 2012. 540 [RFC7231] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol 541 (HTTP/1.1): Semantics and Content", RFC 7231, June 2014. 543 [RFC7235] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol 544 (HTTP/1.1): Authentication", RFC 7235, June 2014. 546 [RFC7639] Hutton, A., Uberti, J., and M. Thomson, "The ALPN HTTP 547 Header Field", RFC 7639, DOI 10.17487/RFC7639, August 548 2015, . 550 8.2. Informative References 552 [I-D.ietf-dart-dscp-rtp] 553 Black, D. and P. Jones, "Differentiated Services 554 (DiffServ) and Real-time Communication", draft-ietf-dart- 555 dscp-rtp-08 (work in progress), October 2014. 557 [I-D.ietf-rtcweb-overview] 558 Alvestrand, H., "Overview: Real Time Protocols for 559 Browser-based Applications", draft-ietf-rtcweb-overview-13 560 (work in progress), November 2014. 562 [RFC3484] Draves, R., "Default Address Selection for Internet 563 Protocol version 6 (IPv6)", RFC 3484, February 2003. 565 [RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6 566 Socket API for Source Address Selection", RFC 5014, 567 September 2007. 569 [RFC5128] Srisuresh, P., Ford, B., and D. Kegel, "State of Peer-to- 570 Peer (P2P) Communication across Network Address 571 Translators (NATs)", RFC 5128, March 2008. 573 Appendix A. Change log 575 This section should be removed before publication as an RFC. 577 A.1. Changes from -00 to -01 579 o Clarified DSCP requirements, with reference to -qos- 581 o Clarified "symmetric NAT" -> "NATs which perform endpoint- 582 dependent mapping" 584 o Made support of TURN over TCP mandatory 586 o Made support of TURN over TLS a MAY, and added open question 588 o Added an informative reference to -firewalls- 590 o Called out that we don't make requirements on HTTP proxy 591 interaction (yet 593 A.2. Changes from -01 to -02 595 o Required support for 300 Alternate Server from STUN. 597 o Separated the ICE-TCP candidate requirement from the TURN-TCP 598 requirement. 600 o Added new sections on using QoS functions, and on multiplexing 601 considerations. 603 o Removed all mention of RTP profiles. Those are the business of 604 the RTP usage draft, not this one. 606 o Required support for TURN IPv6 extensions. 608 o Removed reference to the TURN URI scheme, as it was unnecessary. 610 o Made an explicit statement that multiplexing (or not) is an 611 application matter. 613 . 615 A.3. Changes from -02 to -03 617 o Added required support for draft-ietf-tsvwg-sctp-ndata 619 o Removed discussion of multiplexing, since this is present in rtp- 620 usage. 622 o Added RFC 4571 reference for framing RTP packets over TCP. 624 o Downgraded TURN TCP candidates from SHOULD to MAY, and added more 625 language discussing TCP usage. 627 o Added language on IPv6 temporary addresses. 629 o Added language describing multiplexing choices. 631 o Added a separate section detailing what it means when we say that 632 an WebRTC implementation MUST support both IPv4 and IPv6. 634 A.4. Changes from -03 to -04 636 o Added a section on prioritization, moved the DSCP section into it, 637 and added a section on local prioritization, giving a specific 638 algorithm for interpreting "priority" in local prioritization. 640 o ICE-TCP candidates was changed from MAY to MUST, in recognition of 641 the sense of the room at the London IETF. 643 A.5. Changes from -04 to -05 645 o Reworded introduction 647 o Removed all references to "WebRTC". It now uses only the term 648 RTCWEB. 650 o Addressed a number of clarity / language comments 652 o Rewrote the prioritization to cover data channels and to describe 653 multiple ways of prioritizing flows 655 o Made explicit reference to "MUST do DTLS-SRTP", and referred to 656 security-arch for details 658 A.6. Changes from -05 to -06 660 o Changed all references to "RTCWEB" to "WebRTC", except one 661 reference to the working group 663 o Added reference to the httpbis "connect" protocol (being adopted 664 by HTTPBIS) 666 o Added reference to the ALPN header (being adopted by RTCWEB) 668 o Added reference to the DART RTP document 670 o Said explicitly that SCTP for data channels has a single DSCP 671 codepoint 673 A.7. Changes from -06 to -07 675 o Updated references 677 o Removed reference to draft-hutton-rtcweb-nat-firewall- 678 considerations 680 A.8. Changes from -07 to -08 682 o Updated references 684 o Deleted "bundle each media type (audio, video or data) into its 685 own 5-tuple (bundling by media type)" from MUST support 686 configuration, since JSEP does not have a means to negotiate this 687 configuration 689 A.9. Changes from -08 to -09 691 o Added a clarifying note about DTLS-SRTP and ICE interaction. 693 A.10. Changes from -09 to -10 695 o Re-added references to proxy authentication lost in 07-08 696 transition (Bug #5) 698 o Rearranged and rephrased text in section 4 about prioritization to 699 reflect discussions in TSVWG. 701 o Changed the "Connect" header to "ALPN", and updated reference. 702 (Bug #6) 704 Author's Address 706 Harald Alvestrand 707 Google 709 Email: harald@alvestrand.no