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'I-D.roach-mmusic-unified-plan') ** Obsolete normative reference: RFC 5245 (Obsoleted by RFC 8445, RFC 8839) == Outdated reference: A later version (-16) exists of draft-ietf-rtcweb-use-cases-and-requirements-11 Summary: 3 errors (**), 0 flaws (~~), 9 warnings (==), 1 comment (--). 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 14, 2014 5 Expires: August 18, 2014 7 Overview: Real Time Protocols for Brower-based Applications 8 draft-ietf-rtcweb-overview-09 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 The document will be publishd as an Applicability Statement - it does 22 not itself specify any protocol, but specifies which other 23 specifications RTCWEB compliant implementations are supposed to 24 follow. 26 This document is a work item of the RTCWEB working group. 28 Status of this Memo 30 This Internet-Draft is submitted in full conformance with the 31 provisions of BCP 78 and BCP 79. 33 Internet-Drafts are working documents of the Internet Engineering 34 Task Force (IETF). Note that other groups may also distribute 35 working documents as Internet-Drafts. The list of current Internet- 36 Drafts is at http://datatracker.ietf.org/drafts/current/. 38 Internet-Drafts are draft documents valid for a maximum of six months 39 and may be updated, replaced, or obsoleted by other documents at any 40 time. It is inappropriate to use Internet-Drafts as reference 41 material or to cite them other than as "work in progress." 43 This Internet-Draft will expire on August 18, 2014. 45 Copyright Notice 47 Copyright (c) 2014 IETF Trust and the persons identified as the 48 document authors. All rights reserved. 50 This document is subject to BCP 78 and the IETF Trust's Legal 51 Provisions Relating to IETF Documents 52 (http://trustee.ietf.org/license-info) in effect on the date of 53 publication of this document. Please review these documents 54 carefully, as they describe your rights and restrictions with respect 55 to this document. Code Components extracted from this document must 56 include Simplified BSD License text as described in Section 4.e of 57 the Trust Legal Provisions and are provided without warranty as 58 described in the Simplified BSD License. 60 Table of Contents 62 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 63 2. Principles and Terminology . . . . . . . . . . . . . . . . . . 5 64 2.1. Goals of this document . . . . . . . . . . . . . . . . . . 5 65 2.2. Relationship between API and protocol . . . . . . . . . . 5 66 2.3. On interoperability and innovation . . . . . . . . . . . . 6 67 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 7 68 3. Architecture and Functionality groups . . . . . . . . . . . . 8 69 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . . 12 70 5. Data framing and securing . . . . . . . . . . . . . . . . . . 12 71 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . . 13 72 7. Connection management . . . . . . . . . . . . . . . . . . . . 13 73 8. Presentation and control . . . . . . . . . . . . . . . . . . . 14 74 9. Local system support functions . . . . . . . . . . . . . . . . 14 75 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 76 11. Security Considerations . . . . . . . . . . . . . . . . . . . 15 77 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 16 78 13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 16 79 13.1. Normative References . . . . . . . . . . . . . . . . . . . 16 80 13.2. Informative References . . . . . . . . . . . . . . . . . . 17 81 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 18 82 A.1. Changes from 83 draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 18 84 A.2. Changes from draft-alvestrand-dispatch-01 to 85 draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 19 86 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 19 87 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to 88 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 19 89 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 19 90 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 19 91 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 20 92 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 20 93 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 20 94 A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 20 95 A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 20 96 A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 21 97 A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 21 98 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21 100 1. Introduction 102 The Internet was, from very early in its lifetime, considered a 103 possible vehicle for the deployment of real-time, interactive 104 applications - with the most easily imaginable being audio 105 conversations (aka "Internet telephony") and video conferencing. 107 The first attempts to build this were dependent on special networks, 108 special hardware and custom-built software, often at very high prices 109 or at low quality, placing great demands on the infrastructure. 111 As the available bandwidth has increased, and as processors and other 112 hardware has become ever faster, the barriers to participation have 113 decreased, and it has become possible to deliver a satisfactory 114 experience on commonly available computing hardware. 116 Still, there are a number of barriers to the ability to communicate 117 universally - one of these is that there is, as of yet, no single set 118 of communication protocols that all agree should be made available 119 for communication; another is the sheer lack of universal 120 identification systems (such as is served by telephone numbers or 121 email addresses in other communications systems). 123 Development of The Universal Solution has proved hard, however, for 124 all the usual reasons. 126 The last few years have also seen a new platform rise for deployment 127 of services: The browser-embedded application, or "Web application". 128 It turns out that as long as the browser platform has the necessary 129 interfaces, it is possible to deliver almost any kind of service on 130 it. 132 Traditionally, these interfaces have been delivered by plugins, which 133 had to be downloaded and installed separately from the browser; in 134 the development of HTML5, application developers see much promise in 135 the possibility of making those interfaces available in a 136 standardized way within the browser. 138 This memo describes a set of building blocks that can be made 139 accessible and controllable through a Javascript API in a browser, 140 and which together form a sufficient set of functions to allow the 141 use of interactive audio and video in applications that communicate 142 directly between browsers across the Internet. The resulting 143 protocol suite is intended to enable all the applications that are 144 described as required scenarios in the RTCWEB use cases document 145 [I-D.ietf-rtcweb-use-cases-and-requirements]. 147 Other efforts, for instance the W3C WebRTC, Web Applications and 148 Device API working groups, focus on making standardized APIs and 149 interfaces available, within or alongside the HTML5 effort, for those 150 functions; this memo concentrates on specifying the protocols and 151 subprotocols that are needed to specify the interactions that happen 152 across the network. 154 2. Principles and Terminology 156 2.1. Goals of this document 158 The goal of the RTCWEB protocol specification is to specify a set of 159 protocols that, if all are implemented, will allow an implementation 160 to communicate with another implementation using audio, video and 161 data sent along the most direct possible path between the 162 participants. 164 This document is intended to serve as the roadmap to the RTCWEB 165 specifications. It defines terms used by other pieces of 166 specification, lists references to other specifications that don't 167 need further elaboration in the RTCWEB context, and gives pointers to 168 other documents that form part of the RTCWEB suite. 170 By reading this document and the documents it refers to, it should be 171 possible to have all information needed to implement an RTCWEB 172 compatible implementation. 174 2.2. Relationship between API and protocol 176 The total RTCWEB/WEBRTC effort consists of two pieces: 178 o A protocol specification, done in the IETF 180 o A Javascript API specification, done in the W3C 181 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 183 Together, these two specifications aim to provide an environment 184 where Javascript embedded in any page, viewed in any compatible 185 browser, when suitably authorized by its user, is able to set up 186 communication using audio, video and auxiliary data, where the 187 browser environment does not constrain the types of application in 188 which this functionality can be used. 190 The protocol specification does not assume that all implementations 191 implement this API; it is not intended to be necessary for 192 interoperation to know whether the entity one is communicating with 193 is a browser or another device implementing this specification. 195 The goal of cooperation between the protocol specification and the 196 API specification is that for all options and features of the 197 protocol specification, it should be clear which API calls to make to 198 exercise that option or feature; similarly, for any sequence of API 199 calls, it should be clear which protocol options and features will be 200 invoked. Both subject to constraints of the implementation, of 201 course. 203 2.3. On interoperability and innovation 205 The "Mission statement of the IETF" [RFC3935] states that "The 206 benefit of a standard to the Internet is in interoperability - that 207 multiple products implementing a standard are able to work together 208 in order to deliver valuable functions to the Internet's users." 210 Communication on the Internet frequently occurs in two phases: 212 o Two parties communicate, through some mechanism, what 213 functionality they both are able to support 215 o They use that shared communicative functionality to communicate, 216 or, failing to find anything in common, give up on communication. 218 There are often many choices that can be made for communicative 219 functionality; the history of the Internet is rife with the proposal, 220 standardization, implementation, and success or failure of many types 221 of options, in all sorts of protocols. 223 The goal of having a mandatory to implement function set is to 224 prevent negotiation failure, not to preempt or prevent negotiation. 226 The presence of a mandatory to implement function set serves as a 227 strong changer of the marketplace of deployment - in that it gives a 228 guarantee that, as long as you conform to a specification, and the 229 other party is willing to accept communication at the base level of 230 that specification, you can communicate successfully. 232 The alternative - that of having no mandatory to implement - does not 233 mean that you cannot communicate, it merely means that in order to be 234 part of the communications partnership, you have to implement the 235 standard "and then some" - that "and then some" usually being called 236 a profile of some sort; in the version most antithetical to the 237 Internet ethos, that "and then some" consists of having to use a 238 specific vendor's product only. 240 2.4. Terminology 242 The following terms are used in this document, and as far as possible 243 across the documents specifying the RTCWEB suite, in the specific 244 meanings given here. Not all terms are used in this document. Other 245 terms are used in their commonly used meaning. 247 The list is in alphabetical order. 249 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 251 API: Application Programming Interface - a specification of a set of 252 calls and events, usually tied to a programming language or an 253 abstract formal specification such as WebIDL, with its defined 254 semantics. 256 Browser: Used synonymously with "Interactive User Agent" as defined 257 in the HTML specification [W3C.WD-html5-20110525]. 259 ICE Agent: An implementation of the Interactive Connectivty 260 Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be 261 an SDP Agent, but there exist ICE Agents that do not use SDP (for 262 instance those that use Jingle). 264 Interactive: Communication between multiple parties, where the 265 expectation is that an action from one party can cause a reaction 266 by another party, and the reaction can be observed by the first 267 party, with the total time required for the action/reaction/ 268 observation is on the order of no more than hundreds of 269 milliseconds. 271 Media: Audio and video content. Not to be confused with 272 "transmission media" such as wires. 274 Media path: The path that media data follows from one browser to 275 another. 277 Protocol: A specification of a set of data units, their 278 representation, and rules for their transmission, with their 279 defined semantics. A protocol is usually thought of as going 280 between systems. 282 Real-time media: Media where generation of content and display of 283 content are intended to occur closely together in time (on the 284 order of no more than hundreds of milliseconds). Real-time media 285 can be used to support interactive communication. 287 SDP Agent: The protocol implementation involved in the SDP offer/ 288 answer exchange, as defined in [RFC3264] section 3. 290 Signaling: Communication that happens in order to establish, manage 291 and control media paths. 293 Signaling Path: The communication channels used between entities 294 participating in signaling to transfer signaling. There may be 295 more entities in the signaling path than in the media path. 297 NOTE: Where common definitions exist for these terms, those 298 definitions should be used to the greatest extent possible. 300 TODO: Extend this list with other terms that might prove slippery. 302 3. Architecture and Functionality groups 304 The model of real-time support for browser-based applications does 305 not envisage that the browser will contain all the functions that 306 need to be performed in order to have a function such as a telephone 307 or a video conferencing unit; the vision is that the browser will 308 have the functions that are needed for a Web application, working in 309 conjunction with its backend servers, to implement these functions. 311 This means that two vital interfaces need specification: The 312 protocols that browsers talk to each other, without any intervening 313 servers, and the APIs that are offered for a Javascript application 314 to take advantage of the browser's functionality. 316 +------------------------+ On-the-wire 317 | | Protocols 318 | Servers |---------> 319 | | 320 | | 321 +------------------------+ 322 ^ 323 | 324 | 325 | HTTP/ 326 | Websockets 327 | 328 | 329 +----------------------------+ 330 | Javascript/HTML/CSS | 331 +----------------------------+ 332 Other ^ ^RTC 333 APIs | |APIs 334 +---|-----------------|------+ 335 | | | | 336 | +---------+| 337 | | Browser || On-the-wire 338 | Browser | RTC || Protocols 339 | | Function|-----------> 340 | | || 341 | | || 342 | +---------+| 343 +---------------------|------+ 344 | 345 V 346 Native OS Services 348 Figure 1: Browser Model 350 Note that HTTP and Websockets are also offered to the Javascript 351 application through browser APIs. 353 As for all protocol and API specifications, there is no restriction 354 that the protocols can only be used to talk to another browser; since 355 they are fully specified, any device that implements the protocols 356 faithfully should be able to interoperate with the application 357 running in the browser. 359 A commonly imagined model of deployment is the one depicted below. 361 +-----------+ +-----------+ 362 | Web | | Web | 363 | | Signaling | | 364 | |-------------| | 365 | Server | path | Server | 366 | | | | 367 +-----------+ +-----------+ 368 / \ 369 / \ Application-defined 370 / \ over 371 / \ HTTP/Websockets 372 / Application-defined over \ 373 / HTTP/Websockets \ 374 / \ 375 +-----------+ +-----------+ 376 |JS/HTML/CSS| |JS/HTML/CSS| 377 +-----------+ +-----------+ 378 +-----------+ +-----------+ 379 | | | | 380 | | | | 381 | Browser | ------------------------- | Browser | 382 | | Media path | | 383 | | | | 384 +-----------+ +-----------+ 386 Figure 2: Browser RTC Trapezoid 388 On this drawing, the critical part to note is that the media path 389 ("low path") goes directly between the browsers, so it has to be 390 conformant to the specifications of the RTCWEB protocol suite; the 391 signaling path ("high path") goes via servers that can modify, 392 translate or massage the signals as needed. 394 If the two Web servers are operated by different entities, the inter- 395 server signaling mechanism needs to be agreed upon, either by 396 standardization or by other means of agreement. Existing protocols 397 (for example SIP or XMPP) could be used between servers, while either 398 a standards-based or proprietary protocol could be used between the 399 browser and the web server. 401 For example, if both operators' servers implement SIP, SIP could be 402 used for communication between servers, along with either a 403 standardized signaling mechanism (e.g. SIP over Websockets) or a 404 proprietary signaling mechanism used between the application running 405 in the browser and the web server. Similarly, if both operators' 406 servers implement XMPP, XMPP could be used for communication between 407 XMPP servers, with either a standardized signaling mechanism (e.g. 408 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 409 used between the application running in the browser and the web 410 server. 412 The choice of protocols, and definition of the translation between 413 them, is outside the scope of the RTCWEB standards suite described in 414 the document. 416 The functionality groups that are needed in the browser can be 417 specified, more or less from the bottom up, as: 419 o Data transport: TCP, UDP and the means to securely set up 420 connections between entities, as well as the functions for 421 deciding when to send data: Congestion management, bandwidth 422 estimation and so on. 424 o Data framing: RTP and other data formats that serve as containers, 425 and their functions for data confidentiality and integrity. 427 o Data formats: Codec specifications, format specifications and 428 functionality specifications for the data passed between systems. 429 Audio and video codecs, as well as formats for data and document 430 sharing, belong in this category. In order to make use of data 431 formats, a way to describe them, a session description, is needed. 433 o Connection management: Setting up connections, agreeing on data 434 formats, changing data formats during the duration of a call; SIP 435 and Jingle/XMPP belong in this category. 437 o Presentation and control: What needs to happen in order to ensure 438 that interactions behave in a non-surprising manner. This can 439 include floor control, screen layout, voice activated image 440 switching and other such functions - where part of the system 441 require the cooperation between parties. XCON and Cisco/ 442 Tandberg's TIP were some attempts at specifying this kind of 443 functionality; many applications have been built without 444 standardized interfaces to these functions. 446 o Local system support functions: These are things that need not be 447 specified uniformly, because each participant may choose to do 448 these in a way of the participant's choosing, without affecting 449 the bits on the wire in a way that others have to be cognizant of. 450 Examples in this category include echo cancellation (some forms of 451 it), local authentication and authorization mechanisms, OS access 452 control and the ability to do local recording of conversations. 454 Within each functionality group, it is important to preserve both 455 freedom to innovate and the ability for global communication. 456 Freedom to innovate is helped by doing the specification in terms of 457 interfaces, not implementation; any implementation able to 458 communicate according to the interfaces is a valid implementation. 459 Ability to communicate globally is helped both by having core 460 specifications be unencumbered by IPR issues and by having the 461 formats and protocols be fully enough specified to allow for 462 independent implementation. 464 One can think of the three first groups as forming a "media transport 465 infrastructure", and of the three last groups as forming a "media 466 service". In many contexts, it makes sense to use a common 467 specification for the media transport infrastructure, which can be 468 embedded in browsers and accessed using standard interfaces, and "let 469 a thousand flowers bloom" in the "media service" layer; to achieve 470 interoperable services, however, at least the first five of the six 471 groups need to be specified. 473 4. Data transport 475 Data transport refers to the sending and receiving of data over the 476 network interfaces, the choice of network-layer addresses at each end 477 of the communication, and the interaction with any intermediate 478 entities that handle the data, but do not modify it (such as TURN 479 relays). 481 It includes necessary functions for congestion control: When not to 482 send data. 484 The data transport protocols used by RTCWEB are described in 485 [I-D.ietf-rtcweb-transports]. 487 5. Data framing and securing 489 The format for media transport is RTP [RFC3550]. Implementation of 490 SRTP [RFC3711] is required for all implementations. 492 The detailed considerations for usage of functions from RTP and SRTP 493 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 494 considerations for the RTCWEB use case are in 495 [I-D.ietf-rtcweb-security], and the resulting security functions are 496 described in [I-D.ietf-rtcweb-security-arch]. 498 Considerations for the transfer of data that is not in RTP format is 499 described in [I-D.ietf-rtcweb-data-channel], and the resulting 500 protocol is described in [I-D.jesup-rtcweb-data-protocol] (not yet a 501 WG document) 503 6. Data formats 505 The intent of this specification is to allow each communications 506 event to use the data formats that are best suited for that 507 particular instance, where a format is supported by both sides of the 508 connection. However, a minimum standard is greatly helpful in order 509 to ensure that communication can be achieved. This document 510 specifies a minimum baseline that will be supported by all 511 implementations of this specification, and leaves further codecs to 512 be included at the will of the implementor. 514 The mandatory to implement codecs, as well as any profiling 515 requirements for both mandatory and optional codecs, is described in 516 (candidate draft: 517 [I-D.cbran-rtcweb-codec]. 519 7. Connection management 521 The methods, mechanisms and requirements for setting up, negotiating 522 and tearing down connections is a large subject, and one where it is 523 desirable to have both interoperability and freedom to innovate. 525 The following principles apply: 527 1. The RTCWEB media negotiations will be capable of representing the 528 same SDP offer/answer semantics that are used in SIP [RFC3264], 529 in such a way that it is possible to build a signaling gateway 530 between SIP and the RTCWEB media negotiation. 532 2. It will be possible to gateway between legacy SIP devices that 533 support ICE and appropriate RTP / SDP mechanisms, codecs and 534 security mechanisms without using a media gateway. A signaling 535 gateway to convert between the signaling on the web side to the 536 SIP signaling may be needed. 538 3. When a new codec is specified, and the SDP for the new codec is 539 specified in the MMUSIC WG, no other standardization should be 540 required for it to be possible to use that in the web browsers. 541 Adding new codecs which might have new SDP parameters should not 542 change the APIs between the browser and Javascript application. 543 As soon as the browsers support the new codecs, old applications 544 written before the codecs were specified should automatically be 545 able to use the new codecs where appropriate with no changes to 546 the JS applications. 548 The particular choices made for RTCWEB, and their implications for 549 the API offered by a browser implementing RTCWEB, are described in 550 [I-D.ietf-rtcweb-jsep]. This document in turn implements the 551 solutions described in [I-D.roach-mmusic-unified-plan]. 553 8. Presentation and control 555 The most important part of control is the user's control over the 556 browser's interaction with input/output devices and communications 557 channels. It is important that the user have some way of figuring 558 out where his audio, video or texting is being sent, for what 559 purported reason, and what guarantees are made by the parties that 560 form part of this control channel. This is largely a local function 561 between the browser, the underlying operating system and the user 562 interface; this is being worked on as part of the W3C API effort, and 563 will be part of the peer connection API [W3C.WD-webrtc-20120209], and 564 the media capture API [W3C.WD-mediacapture-streams-20120628]. 565 Considerations for the implications of wanting to identify 566 correspondents are described in [I-D.rescorla-rtcweb-generic-idp] 567 (not a WG item). 569 9. Local system support functions 571 These are characterized by the fact that the quality of these 572 functions strongly influence the user experience, but the exact 573 algorithm does not need coordination. In some cases (for instance 574 echo cancellation, as described below), the overall system definition 575 may need to specify that the overall system needs to have some 576 characteristics for which these facilities are useful, without 577 requiring them to be implemented a certain way. 579 Local functions include echo cancellation, volume control, camera 580 management including focus, zoom, pan/tilt controls (if available), 581 and more. 583 Certain parts of the system SHOULD conform to certain properties, for 584 instance: 586 o Echo cancellation should be good enough to achieve the suppression 587 of acoustical feedback loops below a perceptually noticeable 588 level. 590 o Privacy concerns must be satisfied; for instance, if remote 591 control of camera is offered, the APIs should be available to let 592 the local participant figure out who's controlling the camera, and 593 possibly decide to revoke the permission for camera usage. 595 o Automatic gain control, if present, should normalize a speaking 596 voice into a reasonable dB range. 598 The requirements on RTCWEB systems with regard to audio processing 599 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 600 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 602 10. IANA Considerations 604 This document makes no request of IANA. 606 Note to RFC Editor: this section may be removed on publication as an 607 RFC. 609 11. Security Considerations 611 Security of the web-enabled real time communications comes in several 612 pieces: 614 o Security of the components: The browsers, and other servers 615 involved. The most target-rich environment here is probably the 616 browser; the aim here should be that the introduction of these 617 components introduces no additional vulnerability. 619 o Security of the communication channels: It should be easy for a 620 participant to reassure himself of the security of his 621 communication - by verifying the crypto parameters of the links he 622 himself participates in, and to get reassurances from the other 623 parties to the communication that they promise that appropriate 624 measures are taken. 626 o Security of the partners' identity: verifying that the 627 participants are who they say they are (when positive 628 identification is appropriate), or that their identity cannot be 629 uncovered (when anonymity is a goal of the application). 631 The security analysis, and the requirements derived from that 632 analysis, is contained in [I-D.ietf-rtcweb-security]. 634 12. Acknowledgements 636 The number of people who have taken part in the discussions 637 surrounding this draft are too numerous to list, or even to identify. 638 The ones below have made special, identifiable contributions; this 639 does not mean that others' contributions are less important. 641 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 642 Westerlund and Joerg Ott, who offered technical contributions on 643 various versions of the draft. 645 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 646 the ASCII drawings in section 1. 648 Thanks to Eric Rescorla, Justin Uberti, Henry Sinnreich, Colin 649 Perkins, Bjoern Hoehrmann and Simon Leinen for document review, and 650 to Heath Matlock for grammatical review. 652 13. References 654 13.1. Normative References 656 [I-D.ietf-rtcweb-audio] 657 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 658 Requirements", draft-ietf-rtcweb-audio-02 (work in 659 progress), August 2013. 661 [I-D.ietf-rtcweb-data-channel] 662 Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Data 663 Channels", draft-ietf-rtcweb-data-channel-05 (work in 664 progress), July 2013. 666 [I-D.ietf-rtcweb-jsep] 667 Uberti, J. and C. Jennings, "Javascript Session 668 Establishment Protocol", draft-ietf-rtcweb-jsep-04 (work 669 in progress), September 2013. 671 [I-D.ietf-rtcweb-rtp-usage] 672 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 673 Communication (WebRTC): Media Transport and Use of RTP", 674 draft-ietf-rtcweb-rtp-usage-09 (work in progress), 675 September 2013. 677 [I-D.ietf-rtcweb-security] 678 Rescorla, E., "Security Considerations for WebRTC", 679 draft-ietf-rtcweb-security-05 (work in progress), 680 July 2013. 682 [I-D.ietf-rtcweb-security-arch] 683 Rescorla, E., "WebRTC Security Architecture", 684 draft-ietf-rtcweb-security-arch-07 (work in progress), 685 July 2013. 687 [I-D.ietf-rtcweb-transports] 688 Alvestrand, H., "Transports for RTCWEB", 689 draft-ietf-rtcweb-transports-01 (work in progress), 690 September 2013. 692 [I-D.roach-mmusic-unified-plan] 693 Roach, A., Uberti, J., and M. Thomson, "A Unified Plan for 694 Using SDP with Large Numbers of Media Flows", 695 draft-roach-mmusic-unified-plan-00 (work in progress), 696 July 2013. 698 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 699 with Session Description Protocol (SDP)", RFC 3264, 700 June 2002. 702 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 703 Jacobson, "RTP: A Transport Protocol for Real-Time 704 Applications", STD 64, RFC 3550, July 2003. 706 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 707 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 708 RFC 3711, March 2004. 710 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 711 (ICE): A Protocol for Network Address Translator (NAT) 712 Traversal for Offer/Answer Protocols", RFC 5245, 713 April 2010. 715 13.2. Informative References 717 [I-D.cbran-rtcweb-codec] 718 Bran, C., Jennings, C., and J. Valin, "WebRTC Codec and 719 Media Processing Requirements", 720 draft-cbran-rtcweb-codec-02 (work in progress), 721 March 2012. 723 [I-D.ietf-rtcweb-use-cases-and-requirements] 724 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 725 Time Communication Use-cases and Requirements", 726 draft-ietf-rtcweb-use-cases-and-requirements-11 (work in 727 progress), June 2013. 729 [I-D.jesup-rtcweb-data-protocol] 730 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 731 Protocol", draft-jesup-rtcweb-data-protocol-04 (work in 732 progress), February 2013. 734 [I-D.rescorla-rtcweb-generic-idp] 735 Rescorla, E., "RTCWEB Generic Identity Provider 736 Interface", draft-rescorla-rtcweb-generic-idp-01 (work in 737 progress), March 2012. 739 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", 740 BCP 95, RFC 3935, October 2004. 742 [W3C.WD-html5-20110525] 743 Hickson, I., "HTML5", World Wide Web Consortium 744 LastCall WD-html5-20110525, May 2011, 745 . 747 [W3C.WD-mediacapture-streams-20120628] 748 Burnett, D. and A. Narayanan, "Media Capture and Streams", 749 World Wide Web Consortium WD WD-mediacapture-streams- 750 20120628, June 2012, . 753 [W3C.WD-webrtc-20120209] 754 Bergkvist, A., Burnett, D., Jennings, C., and A. 755 Narayanan, "WebRTC 1.0: Real-time Communication Between 756 Browsers", World Wide Web Consortium WD WD-webrtc- 757 20120209, February 2012, 758 . 760 Appendix A. Change log 762 This section may be deleted by the RFC Editor when preparing for 763 publication. 765 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 767 Added section "On interoperability and innovation" 769 Added data confidentiality and integrity to the "data framing" layer 771 Added congestion management requirements in the "data transport" 772 layer section 774 Changed need for non-media data from "question: do we need this?" to 775 "Open issue: How do we do this?" 776 Strengthened disclaimer that listed codecs are placeholders, not 777 decisions. 779 More details on why the "local system support functions" section is 780 there. 782 A.2. Changes from draft-alvestrand-dispatch-01 to 783 draft-alvestrand-rtcweb-overview-00 785 Added section on "Relationship between API and protocol" 787 Added terminology section 789 Mentioned congestion management as part of the "data transport" layer 790 in the layer list 792 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 794 Removed most technical content, and replaced with pointers to drafts 795 as requested and identified by the RTCWEB WG chairs. 797 Added content to acknowledgments section. 799 Added change log. 801 Spell-checked document. 803 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to 804 draft-ietf-rtcweb-overview-00 806 Changed draft name and document date. 808 Removed unused references 810 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 812 Added architecture figures to section 2. 814 Changed the description of "echo cancellation" under "local system 815 support functions". 817 Added a few more definitions. 819 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 821 Added pointers to use cases, security and rtp-usage drafts (now WG 822 drafts). 824 Changed description of SRTP from mandatory-to-use to mandatory-to- 825 implement. 827 Added the "3 principles of negotiation" to the connection management 828 section. 830 Added an explicit statement that ICE is required for both NAT and 831 consent-to-receive. 833 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 835 Added references to a number of new drafts. 837 Expanded the description text under the "trapezoid" drawing with some 838 more text discussed on the list. 840 Changed the "Connection management" sentence from "will be done using 841 SDP offer/answer" to "will be capable of representing SDP offer/ 842 answer" - this seems more consistent with JSEP. 844 Added "security mechanisms" to the things a non-gatewayed SIP devices 845 must support in order to not need a media gateway. 847 Added a definition for "browser". 849 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 851 Made introduction more normative. 853 Several wording changes in response to review comments from EKR 855 Added an appendix to hold references and notes that are not yet in a 856 separate document. 858 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 860 Minor grammatical fixes. This is mainly a "keepalive" refresh. 862 A.10. Changes from -05 to -06 864 Clarifications in response to Last Call review comments. Inserted 865 reference to draft-ietf-rtcweb-audio. 867 A.11. Changes from -06 to -07 869 Added a refereence to the "unified plan" draft, and updated some 870 references. 872 Otherwise, it's a "keepalive" draft. 874 A.12. Changes from -07 to -08 876 Removed the appendix that detailed transports, and replaced it with a 877 reference to draft-ietf-rtcweb-transports. Removed now-unused 878 references. 880 A.13. Changes from -08 to -09 882 Added text to the Abstract indicating that the intended status is an 883 Applicability Statement. 885 Author's Address 887 Harald T. Alvestrand 888 Google 889 Kungsbron 2 890 Stockholm, 11122 891 Sweden 893 Email: harald@alvestrand.no