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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 April 25, 2014 5 Expires: October 27, 2014 7 Transports for RTCWEB 8 draft-ietf-rtcweb-transports-04 10 Abstract 12 This document describes the data transport protocols used by RTCWEB, 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 October 27, 2014. 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 . . . . . . . . . . . . . . . . . . . . . . . . . 3 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 . . . . . . . . . . . . . . . . . . . 7 61 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 62 6. Security Considerations . . . . . . . . . . . . . . . . . . . 8 63 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 8 64 8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 9 65 8.1. Normative References . . . . . . . . . . . . . . . . . . . 9 66 8.2. Informative References . . . . . . . . . . . . . . . . . . 11 67 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 11 68 A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 11 69 A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 12 70 A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 12 71 A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 13 72 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 13 74 1. Introduction 76 The IETF RTCWEB effort, part of the WebRTC effort carried out in 77 cooperation between the IETF and the W3C, is aimed at specifying a 78 protocol suite that is useful for real time multimedia exchange 79 between browsers. 81 The overall effort is described in the RTCWEB overview document, 82 [I-D.ietf-rtcweb-overview]. This document focuses on the data 83 transport protocols that are used by conforming implementations. 85 This protocol suite is designed for WebRTC, and intends to satisfy 86 the security considerations described in the WebRTC security 87 documents, [I-D.ietf-rtcweb-security] and 88 [I-D.ietf-rtcweb-security-arch]. 90 2. Requirements language 92 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 93 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 94 document are to be interpreted as described in RFC 2119 [RFC2119]. 96 3. Transport and Middlebox specification 98 3.1. System-provided interfaces 100 The protocol specifications used here assume that the following 101 protocols are available to the implementations of the RTCWEB 102 protocols: 104 o UDP. This is the protocol assumed by most protocol elements 105 described. 107 o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL 108 and ICE-TCP. 110 For both protocols, IPv4 and IPv6 support is assumed. 112 For UDP, this specification assumes the ability to set the DSCP code 113 point of the sockets opened on a per-packet basis, in order to 114 achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos] 115 (see Section 4.1) when multiple media types are multiplexed. It does 116 not assume that the DSCP codepoints will be honored, and does assume 117 that they may be zeroed or changed, since this is a local 118 configuration issue. 120 Platforms that do not give access to these interfaces will not be 121 able to support a conforming RTCWEB implementation. 123 This specification does not assume that the implementation will have 124 access to ICMP or raw IP. 126 3.2. Ability to use IPv4 and IPv6 128 Web applications running on top of the RTCWEB implementation MUST be 129 able to utilize both IPv4 and IPv6 where available - that is, when 130 two peers have only IPv4 connectivty to each other, or they have only 131 IPv6 connectivity to each other, applications running on top of the 132 RTCWEB implementation MUST be able to communicate. 134 When TURN is used, and the TURN server has IPv4 or IPv6 connectivity 135 to the peer or its TURN server, candidates of the appropriate types 136 MUST be supported. The "Happy Eyeballs" specification for ICE 137 [I-D.reddy-mmusic-ice-happy-eyeballs] SHOULD be supported. 139 3.3. Usage of temporary IPv6 addresses 141 The IPv6 default address selection specification [RFC6724] specifies 142 that temporary addresses [RFC4941] are to be preferred over permanent 143 addresses. This is a change from the rules specified by [RFC3484]. 144 For applications that select a single address, this is usually done 145 by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014]. 146 However, this rule is not completely obvious in the ICE scope. This 147 is therefore clarified as follows: 149 When a client gathers all IPv6 addresses on a host, and both 150 temporary addresses and permanent addresses of the same scope are 151 present, the client SHOULD discard the permanent addresses before 152 forming pairs. This is consistent with the default policy described 153 in [RFC6724]. 155 3.4. Middle box related functions 157 The primary mechanism to deal with middle boxes is ICE, which is an 158 appropriate way to deal with NAT boxes and firewalls that accept 159 traffic from the inside, but only from the outside if it's in 160 response to inside traffic (simple stateful firewalls). 162 ICE [RFC5245] MUST be supported. The implementation MUST be a full 163 ICE implementation, not ICE-Lite; this allows interworking with both 164 ICE and ICE-Lite implementations when they are deployed 165 appropriately. 167 In order to deal with situations where both parties are behind NATs 168 which perform endpoint-dependent mapping (as defined in [RFC5128] 169 section 2.4), TURN [RFC5766] MUST be supported. 171 Configuration of STUN and TURN servers, both from browser 172 configuration and from an applicaiton, MUST be supported. 174 In order to deal with firewalls that block all UDP traffic, TURN 175 using TCP between the client and the server MUST be supported, and 176 TURN using TLS over TCP between the client and the server MUST be 177 supported. See [RFC5766] section 2.1 for details. 179 In order to deal with situations where one party is on an IPv4 180 network and the other party is on an IPv6 network, TURN extensions 181 for IPv6 [RFC6156] MUST be supported. 183 TURN TCP candidates [RFC6062] MAY be supported. 185 However, such candidates are not seen as providing any significant 186 benefit. First, use of TURN TCP would only be relevant in cases 187 which both peers are required to use TCP to establish a 188 PeerConnection. Secondly, that use case is anyway supported by both 189 sides establishing UDP relay candidates using TURN over TCP to 190 connect to the relay server. Thirdly, using TCP only between the 191 endpoint and its relay may result in less issues with TCP in regards 192 to real-time constraints, e.g. due to head of line blocking. 194 ICE-TCP candidates [RFC6544] MUST be supported; this may allow 195 applications to communicate to peers with public IP addresses across 196 UDP-blocking firewalls without using a TURN server. 198 If TCP connections are used, RTP framing according to [RFC4571] MUST 199 be used, both for the RTP packets and for the DTLS packets used to 200 carry data channels. 202 The ALTERNATE-SERVER mechanism specified in [RFC5389] (STUN) section 203 11 (300 Try Alternate) MUST be supported. 205 Further discussion of the interaction of RTCWEB with firewalls is 206 contained in [I-D.hutton-rtcweb-nat-firewall-considerations]. This 207 document makes no requirements on interacting with HTTP proxies or 208 HTTP proxy configuration methods. 210 NOTE IN DRAFT: This may be added. 212 3.5. Transport protocols implemented 214 For transport of media, secure RTP is used. The details of the 215 profile of RTP used are described in "RTP Usage" 217 [I-D.ietf-rtcweb-rtp-usage]. 219 For data transport over the RTCWEB data channel 220 [I-D.ietf-rtcweb-data-channel], RTCWEB implementations MUST support 221 SCTP over DTLS over ICE. This encapsulation is specified in 222 [I-D.ietf-tsvwg-sctp-dtls-encaps]. Negotiation of this transport in 223 SDP is defined in [I-D.ietf-mmusic-sctp-sdp]. The SCTP extension for 224 NDATA, [I-D.ietf-tsvwg-sctp-ndata], MUST be supported. 226 The setup protocol for RTCWEB data channels is described in 227 [I-D.jesup-rtcweb-data-protocol]. 229 RTCWEB implementations MUST support multiplexing of DTLS and RTP over 230 the same port pair, as described in the DTLS_SRTP specification 231 [RFC5764], section 5.1.2. All application layer protocol payloads 232 over this DTLS connection are SCTP packets. 234 4. Media Prioritization 236 The RTCWEB prioritization model is that the application tells the 237 RTCWEB implementation about the priority of media and data flows 238 through an API. 240 The priority associated with a media or data flow is classified as 241 "normal", "below normal", "high" or "very high". There are only four 242 priority levels at the API. 244 The priority settings affect two pieces of behavior: Packet markings 245 and packet send sequence decisions. Each is described in its own 246 section below. 248 4.1. Usage of Quality of Service - DSCP and Multiplexing 250 WebRTC implementations SHOULD attempt to set QoS on the packets sent, 251 according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos]. It is 252 appropriate to depart from this recommendation when running on 253 platforms where QoS marking is not implemented. 255 The implementation MAY turn off use of DSCP markings if it detects 256 symptoms of unexpected behaviour like priority inversion or blocking 257 of packets with certain DSCP markings. The detection of these 258 conditions is implementation dependent. (Question: Does there need 259 to be an API knob to turn off DSCP markings?) 261 There exist a number of schemes for achieving quality of service that 262 do not depend solely on DSCP code points. Some of these schemes 263 depend on classifying the traffic into flows based on 5-tuple (source 264 address, source port, protocol, destination address, destination 265 port) or 6-tuple (same as above + DSCP code point). Under differing 266 conditions, it may therefore make sense for a sending application to 267 choose any of the configurations: 269 o Each media stream carried on its own 5-tuple 271 o Media streams grouped by media type into 5-tuples (such as 272 carrying all audio on one 5-tuple) 274 o All media sent over a single 5-tuple, with or without 275 differentiation into 6-tuples based on DSCP code points 277 In each of the configurations mentioned, data channels may be carried 278 in its own 5-tuple, or multiplexed together with one of the media 279 flows. 281 More complex configurations, such as sending a high priority video 282 stream on one 5-tuple and sending all other video streams multiplexed 283 together over another 5-tuple, can also be envisioned. More 284 information on mapping media flows to 5-tuples can be found in 285 [I-D.ietf-rtcweb-rtp-usage]. 287 A sending implementation MUST be able to multiplex all media and data 288 on a single 5-tuple (fully bundled), MUST be able to send each media 289 stream on its own 5-tuple and data on its own 5-tuple (fully 290 unbundled), and MAY choose to support other configurations. 292 Sending data over multiple 5-tuples is not supported. 294 NOTE IN DRAFT: is there a need to place the "group by media type, 295 with data multiplexed on the video" as a MUST or SHOULD 296 configuration? Are there other MUST configurations? 298 NOTE IN DRAFT: It's been suggested that at least one "MUST" 299 configuration should be with data channels on its own 5-tuple, 300 separate from the media. Opinions sought. 302 A receiving implementation MUST be able to receive media and data in 303 all these configurations. 305 4.2. Local prioritization 307 When an RTCWEB implementation has packets to send on multiple streams 308 that are congestion-controlled under the same congestion controller, 309 the RTCWEB implementation SHOULD serve the streams in a weighted 310 round-robin fashion, with each stream at each level of priority being 311 given approximately twice the transmission capacity (measured in 312 payload bytes) of the level below. 314 Thus, when congestion occurs, a "very high" priority flow will have 315 the ability to send 8 times as much data as a "below normal" flow if 316 both have data to send. This prioritization is independent of the 317 media type, but will lead to packet loss due to full send buffers 318 occuring first on the highest volume flows at any given priority 319 level. The details of which packet to send first are implementation 320 defined. 322 For example: If there is a very high priority audio flow sending 100 323 byte packets, and a normal priority video flow sending 1000 byte 324 packets, and outgoing capacity exists for sending >5000 payload 325 bytes, it would be appropriate to send 4000 bytes (40 packets) of 326 audio and 1000 bytes (one packet) of video as the result of a single 327 pass of sending decisions. 329 Conversely, if the audio flow is marked normal priority and the video 330 flow is marked very high priority, the scheduler may decide to send 2 331 video packets (2000 bytes) and 5 audio packets (500 bytes) when 332 outgoing capacity exists for sending > 2500 payload bytes. 334 If there are two very high priority audio flows, each will be able to 335 send 4000 bytes in the same period where a normal priority video flow 336 is able to send 1000 bytes. 338 NOTE: The appropriate algorithm for deciding when to send SCTP data 339 vs media data is not described yet. 341 5. IANA Considerations 343 This document makes no request of IANA. 345 Note to RFC Editor: this section may be removed on publication as an 346 RFC. 348 6. Security Considerations 350 Security considerations are enumerated in [I-D.ietf-rtcweb-security]. 352 7. Acknowledgements 354 This document is based on earlier versions embedded in 355 [I-D.ietf-rtcweb-overview], which were the results of contributions 356 from many RTCWEB WG members. 358 Special thanks for reviews of earlier versions of this draft go to 359 Magnus Westerlund, Markus Isomaki and Dan Wing; the contributions 360 from Andrew Hutton also deserve special mention. 362 8. References 364 8.1. Normative References 366 [I-D.ietf-mmusic-sctp-sdp] 367 Loreto, S. and G. Camarillo, "Stream Control Transmission 368 Protocol (SCTP)-Based Media Transport in the Session 369 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-06 370 (work in progress), February 2014. 372 [I-D.ietf-rtcweb-data-channel] 373 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 374 Channels", draft-ietf-rtcweb-data-channel-08 (work in 375 progress), April 2014. 377 [I-D.ietf-rtcweb-rtp-usage] 378 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 379 Communication (WebRTC): Media Transport and Use of RTP", 380 draft-ietf-rtcweb-rtp-usage-13 (work in progress), 381 April 2014. 383 [I-D.ietf-rtcweb-security] 384 Rescorla, E., "Security Considerations for WebRTC", 385 draft-ietf-rtcweb-security-06 (work in progress), 386 January 2014. 388 [I-D.ietf-rtcweb-security-arch] 389 Rescorla, E., "WebRTC Security Architecture", 390 draft-ietf-rtcweb-security-arch-09 (work in progress), 391 February 2014. 393 [I-D.ietf-tsvwg-rtcweb-qos] 394 Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and 395 other packet markings for RTCWeb QoS", 396 draft-ietf-tsvwg-rtcweb-qos-00 (work in progress), 397 April 2014. 399 [I-D.ietf-tsvwg-sctp-dtls-encaps] 400 Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS 401 Encapsulation of SCTP Packets", 402 draft-ietf-tsvwg-sctp-dtls-encaps-03 (work in progress), 403 February 2014. 405 [I-D.ietf-tsvwg-sctp-ndata] 406 Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "A 407 New Data Chunk for Stream Control Transmission Protocol", 408 draft-ietf-tsvwg-sctp-ndata-00 (work in progress), 409 February 2014. 411 [I-D.reddy-mmusic-ice-happy-eyeballs] 412 Reddy, T., Patil, P., and P. Martinsen, "Happy Eyeballs 413 Extension for ICE", 414 draft-reddy-mmusic-ice-happy-eyeballs-06 (work in 415 progress), February 2014. 417 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 418 Requirement Levels", BCP 14, RFC 2119, March 1997. 420 [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) 421 and RTP Control Protocol (RTCP) Packets over Connection- 422 Oriented Transport", RFC 4571, July 2006. 424 [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy 425 Extensions for Stateless Address Autoconfiguration in 426 IPv6", RFC 4941, September 2007. 428 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 429 (ICE): A Protocol for Network Address Translator (NAT) 430 Traversal for Offer/Answer Protocols", RFC 5245, 431 April 2010. 433 [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, 434 "Session Traversal Utilities for NAT (STUN)", RFC 5389, 435 October 2008. 437 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 438 Security (DTLS) Extension to Establish Keys for the Secure 439 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 441 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 442 Relays around NAT (TURN): Relay Extensions to Session 443 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 445 [RFC6062] Perreault, S. and J. Rosenberg, "Traversal Using Relays 446 around NAT (TURN) Extensions for TCP Allocations", 447 RFC 6062, November 2010. 449 [RFC6156] Camarillo, G., Novo, O., and S. Perreault, "Traversal 450 Using Relays around NAT (TURN) Extension for IPv6", 451 RFC 6156, April 2011. 453 [RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach, 454 "TCP Candidates with Interactive Connectivity 455 Establishment (ICE)", RFC 6544, March 2012. 457 [RFC6724] Thaler, D., Draves, R., Matsumoto, A., and T. Chown, 458 "Default Address Selection for Internet Protocol Version 6 459 (IPv6)", RFC 6724, September 2012. 461 8.2. Informative References 463 [I-D.hutton-rtcweb-nat-firewall-considerations] 464 Stach, T., Hutton, A., and J. Uberti, "RTCWEB 465 Considerations for NATs, Firewalls and HTTP proxies", 466 draft-hutton-rtcweb-nat-firewall-considerations-03 (work 467 in progress), January 2014. 469 [I-D.ietf-rtcweb-overview] 470 Alvestrand, H., "Overview: Real Time Protocols for Brower- 471 based Applications", draft-ietf-rtcweb-overview-09 (work 472 in progress), February 2014. 474 [I-D.jesup-rtcweb-data-protocol] 475 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 476 Protocol", draft-jesup-rtcweb-data-protocol-04 (work in 477 progress), February 2013. 479 [RFC3484] Draves, R., "Default Address Selection for Internet 480 Protocol version 6 (IPv6)", RFC 3484, February 2003. 482 [RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6 483 Socket API for Source Address Selection", RFC 5014, 484 September 2007. 486 [RFC5128] Srisuresh, P., Ford, B., and D. Kegel, "State of Peer-to- 487 Peer (P2P) Communication across Network Address 488 Translators (NATs)", RFC 5128, March 2008. 490 Appendix A. Change log 492 A.1. Changes from -00 to -01 494 o Clarified DSCP requirements, with reference to -qos- 496 o Clarified "symmetric NAT" -> "NATs which perform endpoint- 497 dependent mapping" 499 o Made support of TURN over TCP mandatory 501 o Made support of TURN over TLS a MAY, and added open question 503 o Added an informative reference to -firewalls- 505 o Called out that we don't make requirements on HTTP proxy 506 interaction (yet 508 A.2. Changes from -01 to -02 510 o Required support for 300 Alternate Server from STUN. 512 o Separated the ICE-TCP candidate requirement from the TURN-TCP 513 requirement. 515 o Added new sections on using QoS functions, and on multiplexing 516 considerations. 518 o Removed all mention of RTP profiles. Those are the business of 519 the RTP usage draft, not this one. 521 o Required support for TURN IPv6 extensions. 523 o Removed reference to the TURN URI scheme, as it was unnecessary. 525 o Made an explicit statement that multiplexing (or not) is an 526 application matter. 528 . 530 A.3. Changes from -02 to -03 532 o Added required support for draft-ietf-tsvwg-sctp-ndata 534 o Removed discussion of multiplexing, since this is present in rtp- 535 usage. 537 o Added RFC 4571 reference for framing RTP packets over TCP. 539 o Downgraded TURN TCP candidates from SHOULD to MAY, and added more 540 language discussing TCP usage. 542 o Added language on IPv6 temporary addresses. 544 o Added language describing multiplexing choices. 546 o Added a separate section detailing what it means when we say that 547 an RTCWEB implementation MUST support both IPv4 and IPv6. 549 A.4. Changes from -03 to -04 551 o Added a section on prioritization, moved the DSCP section into it, 552 and added a section on local prioritization, giving a specific 553 algorithm for interpreting "priority" in local prioritization. 555 o ICE-TCP candidates was changed from MAY to MUST, in recognition of 556 the sense of the room at the London IETF. 558 Author's Address 560 Harald Alvestrand 561 Google 563 Email: harald@alvestrand.no