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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 February 20, 2013 5 Expires: August 24, 2013 7 Overview: Real Time Protocols for Brower-based Applications 8 draft-ietf-rtcweb-overview-06 10 Abstract 12 This document gives an overview and context of a protocol suite 13 intended for use with real-time applications that can be deployed in 14 browsers - "real time communication on the Web". 16 It intends to serve as a starting and coordination point to make sure 17 all the parts that are needed to achieve this goal are findable, and 18 that the parts that belong in the Internet protocol suite are fully 19 specified and on the right publication track. 21 This document is a work item of the RTCWEB working group. 23 Status of this Memo 25 This Internet-Draft is submitted in full conformance with the 26 provisions of BCP 78 and BCP 79. 28 Internet-Drafts are working documents of the Internet Engineering 29 Task Force (IETF). Note that other groups may also distribute 30 working documents as Internet-Drafts. The list of current Internet- 31 Drafts is at http://datatracker.ietf.org/drafts/current/. 33 Internet-Drafts are draft documents valid for a maximum of six months 34 and may be updated, replaced, or obsoleted by other documents at any 35 time. It is inappropriate to use Internet-Drafts as reference 36 material or to cite them other than as "work in progress." 38 This Internet-Draft will expire on August 24, 2013. 40 Copyright Notice 42 Copyright (c) 2013 IETF Trust and the persons identified as the 43 document authors. All rights reserved. 45 This document is subject to BCP 78 and the IETF Trust's Legal 46 Provisions Relating to IETF Documents 47 (http://trustee.ietf.org/license-info) in effect on the date of 48 publication of this document. Please review these documents 49 carefully, as they describe your rights and restrictions with respect 50 to this document. Code Components extracted from this document must 51 include Simplified BSD License text as described in Section 4.e of 52 the Trust Legal Provisions and are provided without warranty as 53 described in the Simplified BSD License. 55 Table of Contents 57 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 58 2. Principles and Terminology . . . . . . . . . . . . . . . . . . 4 59 2.1. Goals of this document . . . . . . . . . . . . . . . . . . 4 60 2.2. Relationship between API and protocol . . . . . . . . . . 4 61 2.3. On interoperability and innovation . . . . . . . . . . . . 5 62 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 63 3. Architecture and Functionality groups . . . . . . . . . . . . 7 64 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . . 11 65 5. Data framing and securing . . . . . . . . . . . . . . . . . . 12 66 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . . 12 67 7. Connection management . . . . . . . . . . . . . . . . . . . . 12 68 8. Presentation and control . . . . . . . . . . . . . . . . . . . 13 69 9. Local system support functions . . . . . . . . . . . . . . . . 13 70 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14 71 11. Security Considerations . . . . . . . . . . . . . . . . . . . 14 72 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 15 73 13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15 74 13.1. Normative References . . . . . . . . . . . . . . . . . . . 15 75 13.2. Informative References . . . . . . . . . . . . . . . . . . 17 76 Appendix A. Transport and Middlebox specification . . . . . . . . 18 77 A.1. System-provided interfaces . . . . . . . . . . . . . . . . 18 78 A.2. Middle box related functions . . . . . . . . . . . . . . . 18 79 A.3. Transport protocols implemented . . . . . . . . . . . . . 19 80 Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 19 81 B.1. Changes from 82 draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 19 83 B.2. Changes from draft-alvestrand-dispatch-01 to 84 draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 19 85 B.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 19 86 B.4. Changes from draft-alvestrand-rtcweb-overview-01 to 87 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 20 88 B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 20 89 B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 20 90 B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 20 91 B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 21 92 B.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 21 93 B.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 21 94 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21 96 1. Introduction 98 The Internet was, from very early in its lifetime, considered a 99 possible vehicle for the deployment of real-time, interactive 100 applications - with the most easily imaginable being audio 101 conversations (aka "Internet telephony") and video conferencing. 103 The first attempts to build this were dependent on special networks, 104 special hardware and custom-built software, often at very high prices 105 or at low quality, placing great demands on the infrastructure. 107 As the available bandwidth has increased, and as processors and other 108 hardware has become ever faster, the barriers to participation have 109 decreased, and it has become possible to deliver a satisfactory 110 experience on commonly available computing hardware. 112 Still, there are a number of barriers to the ability to communicate 113 universally - one of these is that there is, as of yet, no single set 114 of communication protocols that all agree should be made available 115 for communication; another is the sheer lack of universal 116 identification systems (such as is served by telephone numbers or 117 email addresses in other communications systems). 119 Development of The Universal Solution has proved hard, however, for 120 all the usual reasons. 122 The last few years have also seen a new platform rise for deployment 123 of services: The browser-embedded application, or "Web application". 124 It turns out that as long as the browser platform has the necessary 125 interfaces, it is possible to deliver almost any kind of service on 126 it. 128 Traditionally, these interfaces have been delivered by plugins, which 129 had to be downloaded and installed separately from the browser; in 130 the development of HTML5, application developers see much promise in 131 the possibility of making those interfaces available in a 132 standardized way within the browser. 134 This memo describes a set of building blocks that can be made 135 accessible and controllable through a Javascript API in a browser, 136 and which together form a sufficient set of functions to allow the 137 use of interactive audio and video in applications that communicate 138 directly between browsers across the Internet. The resulting 139 protocol suite is intended to enable all the applications that are 140 described as required scenarios in the RTCWEB use cases document 141 [I-D.ietf-rtcweb-use-cases-and-requirements]. 143 Other efforts, for instance the W3C WebRTC, Web Applications and 144 Device API working groups, focus on making standardized APIs and 145 interfaces available, within or alongside the HTML5 effort, for those 146 functions; this memo concentrates on specifying the protocols and 147 subprotocols that are needed to specify the interactions that happen 148 across the network. 150 2. Principles and Terminology 152 2.1. Goals of this document 154 The goal of the RTCWEB protocol specification is to specify a set of 155 protocols that, if all are implemented, will allow an implementation 156 to communicate with another implementation using audio, video and 157 data sent along the most direct possible path between the 158 participants. 160 This document is intended to serve as the roadmap to the RTCWEB 161 specifications. It defines terms used by other pieces of 162 specification, lists references to other specifications that don't 163 need further elaboration in the RTCWEB context, and gives pointers to 164 other documents that form part of the RTCWEB suite. 166 By reading this document and the documents it refers to, it should be 167 possible to have all information needed to implement an RTCWEB 168 compatible implementation. 170 2.2. Relationship between API and protocol 172 The total RTCWEB/WEBRTC effort consists of two pieces: 174 o A protocol specification, done in the IETF 176 o A Javascript API specification, done in the W3C 177 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 179 Together, these two specifications aim to provide an environment 180 where Javascript embedded in any page, viewed in any compatible 181 browser, when suitably authorized by its user, is able to set up 182 communication using audio, video and auxiliary data, where the 183 browser environment does not constrain the types of application in 184 which this functionality can be used. 186 The protocol specification does not assume that all implementations 187 implement this API; it is not intended to be necessary for 188 interoperation to know whether the entity one is communicating with 189 is a browser or another device implementing this specification. 191 The goal of cooperation between the protocol specification and the 192 API specification is that for all options and features of the 193 protocol specification, it should be clear which API calls to make to 194 exercise that option or feature; similarly, for any sequence of API 195 calls, it should be clear which protocol options and features will be 196 invoked. Both subject to constraints of the implementation, of 197 course. 199 2.3. On interoperability and innovation 201 The "Mission statement of the IETF" [RFC3935] states that "The 202 benefit of a standard to the Internet is in interoperability - that 203 multiple products implementing a standard are able to work together 204 in order to deliver valuable functions to the Internet's users." 206 Communication on the Internet frequently occurs in two phases: 208 o Two parties communicate, through some mechanism, what 209 functionality they both are able to support 211 o They use that shared communicative functionality to communicate, 212 or, failing to find anything in common, give up on communication. 214 There are often many choices that can be made for communicative 215 functionality; the history of the Internet is rife with the proposal, 216 standardization, implementation, and success or failure of many types 217 of options, in all sorts of protocols. 219 The goal of having a mandatory to implement function set is to 220 prevent negotiation failure, not to preempt or prevent negotiation. 222 The presence of a mandatory to implement function set serves as a 223 strong changer of the marketplace of deployment - in that it gives a 224 guarantee that, as long as you conform to a specification, and the 225 other party is willing to accept communication at the base level of 226 that specification, you can communicate successfully. 228 The alternative - that of having no mandatory to implement - does not 229 mean that you cannot communicate, it merely means that in order to be 230 part of the communications partnership, you have to implement the 231 standard "and then some" - that "and then some" usually being called 232 a profile of some sort; in the version most antithetical to the 233 Internet ethos, that "and then some" consists of having to use a 234 specific vendor's product only. 236 2.4. Terminology 238 The following terms are used in this document, and as far as possible 239 across the documents specifying the RTCWEB suite, in the specific 240 meanings given here. Not all terms are used in this document. Other 241 terms are used in their commonly used meaning. 243 The list is in alphabetical order. 245 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 247 API: Application Programming Interface - a specification of a set of 248 calls and events, usually tied to a programming language or an 249 abstract formal specification such as WebIDL, with its defined 250 semantics. 252 Browser: Used synonymously with "Interactive User Agent" as defined 253 in the HTML specification [W3C.WD-html5-20110525]. 255 ICE Agent: An implementation of the ICE [RFC5245] protocol. An ICE 256 Agent may also be an SDP Agent, but there exist ICE Agents that do 257 not use SDP (for instance those that use Jingle). 259 Interactive: Communication between multiple parties, where the 260 expectation is that an action from one party can cause a reaction 261 by another party, and the reaction can be observed by the first 262 party, with the total time required for the action/reaction/ 263 observation is on the order of no more than hundreds of 264 milliseconds. 266 Media: Audio and video content. Not to be confused with 267 "transmission media" such as wires. 269 Media path: The path that media data follows from one browser to 270 another. 272 Protocol: A specification of a set of data units, their 273 representation, and rules for their transmission, with their 274 defined semantics. A protocol is usually thought of as going 275 between systems. 277 Real-time media: Media where generation of content and display of 278 content are intended to occur closely together in time (on the 279 order of no more than hundreds of milliseconds). Real-time media 280 can be used to support interactive communication. 282 SDP Agent: The protocol implementation involved in the SDP offer/ 283 answer exchange, as defined in [RFC3264] section 3. 285 Signaling: Communication that happens in order to establish, manage 286 and control media paths. 288 Signaling Path: The communication channels used between entities 289 participating in signaling to transfer signaling. There may be 290 more entities in the signaling path than in the media path. 292 NOTE: Where common definitions exist for these terms, those 293 definitions should be used to the greatest extent possible. 295 TODO: Extend this list with other terms that might prove slippery. 297 3. Architecture and Functionality groups 299 The model of real-time support for browser-based applications does 300 not envisage that the browser will contain all the functions that 301 need to be performed in order to have a function such as a telephone 302 or a video conferencing unit; the vision is that the browser will 303 have the functions that are needed for a Web application, working in 304 conjunction with its backend servers, to implement these functions. 306 This means that two vital interfaces need specification: The 307 protocols that browsers talk to each other, without any intervening 308 servers, and the APIs that are offered for a Javascript application 309 to take advantage of the browser's functionality. 311 +------------------------+ On-the-wire 312 | | Protocols 313 | Servers |---------> 314 | | 315 | | 316 +------------------------+ 317 ^ 318 | 319 | 320 | HTTP/ 321 | Websockets 322 | 323 | 324 +----------------------------+ 325 | Javascript/HTML/CSS | 326 +----------------------------+ 327 Other ^ ^RTC 328 APIs | |APIs 329 +---|-----------------|------+ 330 | | | | 331 | +---------+| 332 | | Browser || On-the-wire 333 | Browser | RTC || Protocols 334 | | Function|-----------> 335 | | || 336 | | || 337 | +---------+| 338 +---------------------|------+ 339 | 340 V 341 Native OS Services 343 Figure 1: Browser Model 345 Note that HTTP and Websockets are also offered to the Javascript 346 application through browser APIs. 348 As for all protocol and API specifications, there is no restriction 349 that the protocols can only be used to talk to another browser; since 350 they are fully specified, any device that implements the protocols 351 faithfully should be able to interoperate with the application 352 running in the browser. 354 A commonly imagined model of deployment is the one depicted below. 356 +-----------+ +-----------+ 357 | Web | | Web | 358 | | Signaling | | 359 | |-------------| | 360 | Server | path | Server | 361 | | | | 362 +-----------+ +-----------+ 363 / \ 364 / \ Application-defined 365 / \ over 366 / \ HTTP/Websockets 367 / Application-defined over \ 368 / HTTP/Websockets \ 369 / \ 370 +-----------+ +-----------+ 371 |JS/HTML/CSS| |JS/HTML/CSS| 372 +-----------+ +-----------+ 373 +-----------+ +-----------+ 374 | | | | 375 | | | | 376 | Browser | ------------------------- | Browser | 377 | | Media path | | 378 | | | | 379 +-----------+ +-----------+ 381 Figure 2: Browser RTC Trapezoid 383 On this drawing, the critical part to note is that the media path 384 ("low path") goes directly between the browsers, so it has to be 385 conformant to the specifications of the RTCWEB protocol suite; the 386 signaling path ("high path") goes via servers that can modify, 387 translate or massage the signals as needed. 389 If the two Web servers are operated by different entities, the inter- 390 server signaling mechanism needs to be agreed upon, either by 391 standardization or by other means of agreement. Existing protocols 392 (for example SIP or XMPP) could be used between servers, while either 393 a standards-based or proprietary protocol could be used between the 394 browser and the web server. 396 For example, if both operators' servers implement SIP, SIP could be 397 used for communication between servers, along with either a 398 standardized signaling mechanism (e.g. SIP over Websockets) or a 399 proprietary signaling mechanism used between the application running 400 in the browser and the web server. Similarly, if both operators' 401 servers implement XMPP, XMPP could be used for communication between 402 XMPP servers, with either a standardized signaling mechanism (e.g. 403 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 404 used between the application running in the browser and the web 405 server. 407 The choice of protocols, and definition of the translation between 408 them, is outside the scope of the RTCWEB standards suite described in 409 the document. 411 The functionality groups that are needed in the browser can be 412 specified, more or less from the bottom up, as: 414 o Data transport: TCP, UDP and the means to securely set up 415 connections between entities, as well as the functions for 416 deciding when to send data: Congestion management, bandwidth 417 estimation and so on. 419 o Data framing: RTP and other data formats that serve as containers, 420 and their functions for data confidentiality and integrity. 422 o Data formats: Codec specifications, format specifications and 423 functionality specifications for the data passed between systems. 424 Audio and video codecs, as well as formats for data and document 425 sharing, belong in this category. In order to make use of data 426 formats, a way to describe them, a session description, is needed. 428 o Connection management: Setting up connections, agreeing on data 429 formats, changing data formats during the duration of a call; SIP 430 and Jingle/XMPP belong in this category. 432 o Presentation and control: What needs to happen in order to ensure 433 that interactions behave in a non-surprising manner. This can 434 include floor control, screen layout, voice activated image 435 switching and other such functions - where part of the system 436 require the cooperation between parties. XCON and Cisco/ 437 Tandberg's TIP were some attempts at specifying this kind of 438 functionality; many applications have been built without 439 standardized interfaces to these functions. 441 o Local system support functions: These are things that need not be 442 specified uniformly, because each participant may choose to do 443 these in a way of the participant's choosing, without affecting 444 the bits on the wire in a way that others have to be cognizant of. 445 Examples in this category include echo cancellation (some forms of 446 it), local authentication and authorization mechanisms, OS access 447 control and the ability to do local recording of conversations. 449 Within each functionality group, it is important to preserve both 450 freedom to innovate and the ability for global communication. 451 Freedom to innovate is helped by doing the specification in terms of 452 interfaces, not implementation; any implementation able to 453 communicate according to the interfaces is a valid implementation. 454 Ability to communicate globally is helped both by having core 455 specifications be unencumbered by IPR issues and by having the 456 formats and protocols be fully enough specified to allow for 457 independent implementation. 459 One can think of the three first groups as forming a "media transport 460 infrastructure", and of the three last groups as forming a "media 461 service". In many contexts, it makes sense to use a common 462 specification for the media transport infrastructure, which can be 463 embedded in browsers and accessed using standard interfaces, and "let 464 a thousand flowers bloom" in the "media service" layer; to achieve 465 interoperable services, however, at least the first five of the six 466 groups need to be specified. 468 4. Data transport 470 Data transport refers to the sending and receiving of data over the 471 network interfaces, the choice of network-layer addresses at each end 472 of the communication, and the interaction with any intermediate 473 entities that handle the data, but do not modify it (such as TURN 474 relays). 476 It includes necessary functions for congestion control: When not to 477 send data. 479 T are described in . 481 ICE is required for all media paths that use UDP; in addition to the 482 ability to pass NAT boxes, ICE fulfills the need for guaranteeing 483 that the media path is going to a UDP port that is willing to receive 484 the data. 486 The data transport protocols used by RTCWEB, as well as the details 487 of interactions with intermediate boxes, such as firewalls, relays 488 and NAT boxes, are intended to be described in a separate document; 489 for now, notes are gathered in Appendix A. 491 5. Data framing and securing 493 The format for media transport is RTP [RFC3550]. Implementation of 494 SRTP [RFC3711] is required for all implementations. 496 The detailed considerations for usage of functions from RTP and SRTP 497 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 498 considerations for the RTCWEB use case are in 499 [I-D.ietf-rtcweb-security], and the resulting security functions are 500 described in [I-D.ietf-rtcweb-security-arch]. 502 Considerations for the transfer of data that is not in RTP format is 503 described in [I-D.ietf-rtcweb-data-channel], and the resulting 504 protocol is described in [I-D.jesup-rtcweb-data-protocol] (not yet a 505 WG document) 507 6. Data formats 509 The intent of this specification is to allow each communications 510 event to use the data formats that are best suited for that 511 particular instance, where a format is supported by both sides of the 512 connection. However, a minimum standard is greatly helpful in order 513 to ensure that communication can be achieved. This document 514 specifies a minimum baseline that will be supported by all 515 implementations of this specification, and leaves further codecs to 516 be included at the will of the implementor. 518 The mandatory to implement codecs, as well as any profiling 519 requirements for both mandatory and optional codecs, is described in 520 (candidate draft: 521 [I-D.cbran-rtcweb-codec]. 523 7. Connection management 525 The methods, mechanisms and requirements for setting up, negotiating 526 and tearing down connections is a large subject, and one where it is 527 desirable to have both interoperability and freedom to innovate. 529 The following principles apply: 531 1. The RTCWEB media negotiations will be capable of representing the 532 same SDP offer/answer semantics that are used in SIP [RFC3264], 533 in such a way that it is possible to build a signaling gateway 534 between SIP and the RTCWEB media negotiation. 536 2. It will be possible to gateway between legacy SIP devices that 537 support ICE and appropriate RTP / SDP mechanisms, codecs and 538 security mechanisms without using a media gateway. A signaling 539 gateway to convert between the signaling on the web side to the 540 SIP signaling may be needed. 542 3. When a new codec is specified, and the SDP for the new codec is 543 specified in the MMUSIC WG, no other standardization should be 544 required for it to be possible to use that in the web browsers. 545 Adding new codecs which might have new SDP parameters should not 546 change the APIs between the browser and Javascript application. 547 As soon as the browsers support the new codecs, old applications 548 written before the codecs were specified should automatically be 549 able to use the new codecs where appropriate with no changes to 550 the JS applications. 552 The particular choices made for RTCWEB, and their implications for 553 the API offered by a browser implementing RTCWEB, are described in 554 [I-D.ietf-rtcweb-jsep] 556 8. Presentation and control 558 The most important part of control is the user's control over the 559 browser's interaction with input/output devices and communications 560 channels. It is important that the user have some way of figuring 561 out where his audio, video or texting is being sent, for what 562 purported reason, and what guarantees are made by the parties that 563 form part of this control channel. This is largely a local function 564 between the browser, the underlying operating system and the user 565 interface; this is being worked on as part of the W3C API effort, and 566 will be part of the peer connection API [W3C.WD-webrtc-20120209], and 567 the media capture API [W3C.WD-mediacapture-streams-20120628]. 568 Considerations for the implications of wanting to identify 569 correspondents are described in [I-D.rescorla-rtcweb-generic-idp] 570 (not a WG item). 572 9. Local system support functions 574 These are characterized by the fact that the quality of these 575 functions strongly influence the user experience, but the exact 576 algorithm does not need coordination. In some cases (for instance 577 echo cancellation, as described below), the overall system definition 578 may need to specify that the overall system needs to have some 579 characteristics for which these facilities are useful, without 580 requiring them to be implemented a certain way. 582 Local functions include echo cancellation, volume control, camera 583 management including focus, zoom, pan/tilt controls (if available), 584 and more. 586 Certain parts of the system SHOULD conform to certain properties, for 587 instance: 589 o Echo cancellation should be good enough to achieve the suppression 590 of acoustical feedback loops below a perceptually noticeable 591 level. 593 o Privacy concerns must be satisfied; for instance, if remote 594 control of camera is offered, the APIs should be available to let 595 the local participant figure out who's controlling the camera, and 596 possibly decide to revoke the permission for camera usage. 598 o Automatic gain control, if present, should normalize a speaking 599 voice into a reasonable dB range. 601 The requirements on RTCWEB systems with regard to audio processing 602 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 603 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 605 10. IANA Considerations 607 This document makes no request of IANA. 609 Note to RFC Editor: this section may be removed on publication as an 610 RFC. 612 11. Security Considerations 614 Security of the web-enabled real time communications comes in several 615 pieces: 617 o Security of the components: The browsers, and other servers 618 involved. The most target-rich environment here is probably the 619 browser; the aim here should be that the introduction of these 620 components introduces no additional vulnerability. 622 o Security of the communication channels: It should be easy for a 623 participant to reassure himself of the security of his 624 communication - by verifying the crypto parameters of the links he 625 himself participates in, and to get reassurances from the other 626 parties to the communication that they promise that appropriate 627 measures are taken. 629 o Security of the partners' identity: verifying that the 630 participants are who they say they are (when positive 631 identification is appropriate), or that their identity cannot be 632 uncovered (when anonymity is a goal of the application). 634 The security analysis, and the requirements derived from that 635 analysis, is contained in [I-D.ietf-rtcweb-security]. 637 12. Acknowledgements 639 The number of people who have taken part in the discussions 640 surrounding this draft are too numerous to list, or even to identify. 641 The ones below have made special, identifiable contributions; this 642 does not mean that others' contributions are less important. 644 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 645 Westerlund and Joerg Ott, who offered technical contributions on 646 various versions of the draft. 648 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 649 the ASCII drawings in section 1. 651 Thanks to Eric Rescorla, Justin Uberti, Henry Sinnreich, Colin 652 Perkins, Bjoern Hoehrmann and Simon Leinen for document review, and 653 to Heath Matlock for grammatical review. 655 13. References 657 13.1. Normative References 659 [I-D.ietf-mmusic-sctp-sdp] 660 Loreto, S. and G. Camarillo, "Stream Control Transmission 661 Protocol (SCTP)-Based Media Transport in the Session 662 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-03 663 (work in progress), January 2013. 665 [I-D.ietf-rtcweb-audio] 666 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 667 Requirements", draft-ietf-rtcweb-audio-01 (work in 668 progress), November 2012. 670 [I-D.ietf-rtcweb-data-channel] 671 Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Datagram 672 Connection", draft-ietf-rtcweb-data-channel-02 (work in 673 progress), October 2012. 675 [I-D.ietf-rtcweb-jsep] 676 Uberti, J. and C. Jennings, "Javascript Session 677 Establishment Protocol", draft-ietf-rtcweb-jsep-02 (work 678 in progress), October 2012. 680 [I-D.ietf-rtcweb-rtp-usage] 681 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 682 Communication (WebRTC): Media Transport and Use of RTP", 683 draft-ietf-rtcweb-rtp-usage-05 (work in progress), 684 October 2012. 686 [I-D.ietf-rtcweb-security] 687 Rescorla, E., "Security Considerations for RTC-Web", 688 draft-ietf-rtcweb-security-04 (work in progress), 689 January 2013. 691 [I-D.ietf-rtcweb-security-arch] 692 Rescorla, E., "RTCWEB Security Architecture", 693 draft-ietf-rtcweb-security-arch-06 (work in progress), 694 January 2013. 696 [I-D.nandakumar-rtcweb-stun-uri] 697 Nandakumar, S., Salgueiro, G., Jones, P., and M. Petit- 698 Huguenin, "URI Scheme for Session Traversal Utilities for 699 NAT (STUN) Protocol", draft-nandakumar-rtcweb-stun-uri-03 700 (work in progress), January 2013. 702 [I-D.tuexen-tsvwg-sctp-dtls-encaps] 703 Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS 704 Encapsulation of SCTP Packets for RTCWEB", 705 draft-tuexen-tsvwg-sctp-dtls-encaps-01 (work in progress), 706 July 2012. 708 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 709 with Session Description Protocol (SDP)", RFC 3264, 710 June 2002. 712 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 713 Jacobson, "RTP: A Transport Protocol for Real-Time 714 Applications", RFC 3550, July 2003. 716 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 717 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 718 RFC 3711, March 2004. 720 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 721 (ICE): A Protocol for Network Address Translator (NAT) 722 Traversal for Offer/Answer Protocols", RFC 5245, 723 April 2010. 725 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 726 Relays around NAT (TURN): Relay Extensions to Session 727 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 729 13.2. Informative References 731 [I-D.cbran-rtcweb-codec] 732 Bran, C., Jennings, C., and J. Valin, "WebRTC Codec and 733 Media Processing Requirements", 734 draft-cbran-rtcweb-codec-02 (work in progress), 735 March 2012. 737 [I-D.ietf-rtcweb-use-cases-and-requirements] 738 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 739 Time Communication Use-cases and Requirements", 740 draft-ietf-rtcweb-use-cases-and-requirements-10 (work in 741 progress), December 2012. 743 [I-D.jesup-rtcweb-data-protocol] 744 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 745 Protocol", draft-jesup-rtcweb-data-protocol-03 (work in 746 progress), September 2012. 748 [I-D.rescorla-rtcweb-generic-idp] 749 Rescorla, E., "RTCWEB Generic Identity Provider 750 Interface", draft-rescorla-rtcweb-generic-idp-01 (work in 751 progress), March 2012. 753 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", 754 BCP 95, RFC 3935, October 2004. 756 [W3C.WD-html5-20110525] 757 Hickson, I., "HTML5", World Wide Web Consortium 758 LastCall WD-html5-20110525, May 2011, 759 . 761 [W3C.WD-mediacapture-streams-20120628] 762 Burnett, D. and A. Narayanan, "Media Capture and Streams", 763 World Wide Web Consortium WD WD-mediacapture-streams- 764 20120628, June 2012, . 767 [W3C.WD-webrtc-20120209] 768 Bergkvist, A., Burnett, D., Narayanan, A., and C. 769 Jennings, "WebRTC 1.0: Real-time Communication Between 770 Browsers", World Wide Web Consortium WD WD-webrtc- 771 20120209, February 2012, 772 . 774 Appendix A. Transport and Middlebox specification 776 The draft referred to as "transport and middle boxes" in Section 4 777 has not been written yet. This appendix contains some keywords to 778 what it should say; this also serves the purpose of linking to the 779 drafts-in-progress that are relevant to this specification. 781 A.1. System-provided interfaces 783 The protocol specifications used here assume that the following 784 protocols are available as system-level interfaces: 786 o UDP. This is the protocol assumed by most protocol elements 787 described. 789 o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL 790 and ICE-TCP. 792 For both protocols, this specification assumes the ability to set the 793 DSCP code point of the sockets opened. It does not assume that the 794 DSCP codepoints will be honored, and does assume that they may be 795 zeroed or changed, since this is a local configuration issue. 797 This specification does not assume that the implementation will have 798 access to ICMP or raw IP. 800 A.2. Middle box related functions 802 The primary mechanism to deal with middle boxes is ICE, which is an 803 appropriate way to deal with NAT boxes and firewalls that accept 804 traffic from the inside, but only from the outside if it's in 805 response to inside traffic (simple stateful firewalls). 807 In order to deal with symmetric NATs, TURN MUST be supported. 809 In order to deal with firewalls that block all UDP traffic, TURN over 810 TCP MUST be supported. (QUESTION: What about ICE-TCP?) 812 The following specifications MUST be supported: 814 o ICE [RFC5245] 816 o TURN, including TURN over TCP [[QUESTION: and TURN over TLS]], 817 [RFC5766]. 819 For referring to STUN and TURN servers, this specification depends on 820 the STUN URI, [I-D.nandakumar-rtcweb-stun-uri]. 822 A.3. Transport protocols implemented 824 For data transport, RTCWEB implementations support SCTP over DTLS 825 over ICE. This is specified in [I-D.tuexen-tsvwg-sctp-dtls-encaps]. 826 Negotiation of this transport in SCTP is defined in 827 [I-D.ietf-mmusic-sctp-sdp]. 829 Appendix B. Change log 831 This section may be deleted by the RFC Editor when preparing for 832 publication. 834 B.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 836 Added section "On interoperability and innovation" 838 Added data confidentiality and integrity to the "data framing" layer 840 Added congestion management requirements in the "data transport" 841 layer section 843 Changed need for non-media data from "question: do we need this?" to 844 "Open issue: How do we do this?" 846 Strengthened disclaimer that listed codecs are placeholders, not 847 decisions. 849 More details on why the "local system support functions" section is 850 there. 852 B.2. Changes from draft-alvestrand-dispatch-01 to 853 draft-alvestrand-rtcweb-overview-00 855 Added section on "Relationship between API and protocol" 857 Added terminology section 859 Mentioned congestion management as part of the "data transport" layer 860 in the layer list 862 B.3. Changes from draft-alvestrand-rtcweb-00 to -01 864 Removed most technical content, and replaced with pointers to drafts 865 as requested and identified by the RTCWEB WG chairs. 867 Added content to acknowledgments section. 869 Added change log. 871 Spell-checked document. 873 B.4. Changes from draft-alvestrand-rtcweb-overview-01 to 874 draft-ietf-rtcweb-overview-00 876 Changed draft name and document date. 878 Removed unused references 880 B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 882 Added architecture figures to section 2. 884 Changed the description of "echo cancellation" under "local system 885 support functions". 887 Added a few more definitions. 889 B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 891 Added pointers to use cases, security and rtp-usage drafts (now WG 892 drafts). 894 Changed description of SRTP from mandatory-to-use to mandatory-to- 895 implement. 897 Added the "3 principles of negotiation" to the connection management 898 section. 900 Added an explicit statement that ICE is required for both NAT and 901 consent-to-receive. 903 B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 905 Added references to a number of new drafts. 907 Expanded the description text under the "trapezoid" drawing with some 908 more text discussed on the list. 910 Changed the "Connection management" sentence from "will be done using 911 SDP offer/answer" to "will be capable of representing SDP offer/ 912 answer" - this seems more consistent with JSEP. 914 Added "security mechanisms" to the things a non-gatewayed SIP devices 915 must support in order to not need a media gateway. 917 Added a definition for "browser". 919 B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 921 Made introduction more normative. 923 Several wording changes in response to review comments from EKR 925 Added Appendix A to hold references and notes that are not yet in a 926 separate document. 928 B.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 930 Minor grammatical fixes. This is mainly a "keepalive" refresh. 932 B.10. Changes from -05 to -06 934 Clarifications in response to Last Call review comments. Inserted 935 reference to draft-ietf-rtcweb-audio. 937 Author's Address 939 Harald T. Alvestrand 940 Google 941 Kungsbron 2 942 Stockholm, 11122 943 Sweden 945 Email: harald@alvestrand.no