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'I-D.roach-mmusic-unified-plan') ** Obsolete normative reference: RFC 5245 (Obsoleted by RFC 8445, RFC 8839) == Outdated reference: A later version (-16) exists of draft-ietf-rtcweb-use-cases-and-requirements-11 Summary: 3 errors (**), 0 flaws (~~), 9 warnings (==), 1 comment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 September 3, 2013 5 Expires: March 7, 2014 7 Overview: Real Time Protocols for Brower-based Applications 8 draft-ietf-rtcweb-overview-08 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 March 7, 2014. 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 . . . . . . . . . . . . . . . . . . 11 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 . . . . . . . . . . . . . . . . . . 16 76 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 17 77 A.1. Changes from 78 draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 17 79 A.2. Changes from draft-alvestrand-dispatch-01 to 80 draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 18 81 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 18 82 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to 83 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 18 84 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 18 85 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 18 86 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 19 87 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 19 88 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 19 89 A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 19 90 A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 19 91 A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 20 92 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 20 94 1. Introduction 96 The Internet was, from very early in its lifetime, considered a 97 possible vehicle for the deployment of real-time, interactive 98 applications - with the most easily imaginable being audio 99 conversations (aka "Internet telephony") and video conferencing. 101 The first attempts to build this were dependent on special networks, 102 special hardware and custom-built software, often at very high prices 103 or at low quality, placing great demands on the infrastructure. 105 As the available bandwidth has increased, and as processors and other 106 hardware has become ever faster, the barriers to participation have 107 decreased, and it has become possible to deliver a satisfactory 108 experience on commonly available computing hardware. 110 Still, there are a number of barriers to the ability to communicate 111 universally - one of these is that there is, as of yet, no single set 112 of communication protocols that all agree should be made available 113 for communication; another is the sheer lack of universal 114 identification systems (such as is served by telephone numbers or 115 email addresses in other communications systems). 117 Development of The Universal Solution has proved hard, however, for 118 all the usual reasons. 120 The last few years have also seen a new platform rise for deployment 121 of services: The browser-embedded application, or "Web application". 122 It turns out that as long as the browser platform has the necessary 123 interfaces, it is possible to deliver almost any kind of service on 124 it. 126 Traditionally, these interfaces have been delivered by plugins, which 127 had to be downloaded and installed separately from the browser; in 128 the development of HTML5, application developers see much promise in 129 the possibility of making those interfaces available in a 130 standardized way within the browser. 132 This memo describes a set of building blocks that can be made 133 accessible and controllable through a Javascript API in a browser, 134 and which together form a sufficient set of functions to allow the 135 use of interactive audio and video in applications that communicate 136 directly between browsers across the Internet. The resulting 137 protocol suite is intended to enable all the applications that are 138 described as required scenarios in the RTCWEB use cases document 139 [I-D.ietf-rtcweb-use-cases-and-requirements]. 141 Other efforts, for instance the W3C WebRTC, Web Applications and 142 Device API working groups, focus on making standardized APIs and 143 interfaces available, within or alongside the HTML5 effort, for those 144 functions; this memo concentrates on specifying the protocols and 145 subprotocols that are needed to specify the interactions that happen 146 across the network. 148 2. Principles and Terminology 150 2.1. Goals of this document 152 The goal of the RTCWEB protocol specification is to specify a set of 153 protocols that, if all are implemented, will allow an implementation 154 to communicate with another implementation using audio, video and 155 data sent along the most direct possible path between the 156 participants. 158 This document is intended to serve as the roadmap to the RTCWEB 159 specifications. It defines terms used by other pieces of 160 specification, lists references to other specifications that don't 161 need further elaboration in the RTCWEB context, and gives pointers to 162 other documents that form part of the RTCWEB suite. 164 By reading this document and the documents it refers to, it should be 165 possible to have all information needed to implement an RTCWEB 166 compatible implementation. 168 2.2. Relationship between API and protocol 170 The total RTCWEB/WEBRTC effort consists of two pieces: 172 o A protocol specification, done in the IETF 174 o A Javascript API specification, done in the W3C 175 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 177 Together, these two specifications aim to provide an environment 178 where Javascript embedded in any page, viewed in any compatible 179 browser, when suitably authorized by its user, is able to set up 180 communication using audio, video and auxiliary data, where the 181 browser environment does not constrain the types of application in 182 which this functionality can be used. 184 The protocol specification does not assume that all implementations 185 implement this API; it is not intended to be necessary for 186 interoperation to know whether the entity one is communicating with 187 is a browser or another device implementing this specification. 189 The goal of cooperation between the protocol specification and the 190 API specification is that for all options and features of the 191 protocol specification, it should be clear which API calls to make to 192 exercise that option or feature; similarly, for any sequence of API 193 calls, it should be clear which protocol options and features will be 194 invoked. Both subject to constraints of the implementation, of 195 course. 197 2.3. On interoperability and innovation 199 The "Mission statement of the IETF" [RFC3935] states that "The 200 benefit of a standard to the Internet is in interoperability - that 201 multiple products implementing a standard are able to work together 202 in order to deliver valuable functions to the Internet's users." 204 Communication on the Internet frequently occurs in two phases: 206 o Two parties communicate, through some mechanism, what 207 functionality they both are able to support 209 o They use that shared communicative functionality to communicate, 210 or, failing to find anything in common, give up on communication. 212 There are often many choices that can be made for communicative 213 functionality; the history of the Internet is rife with the proposal, 214 standardization, implementation, and success or failure of many types 215 of options, in all sorts of protocols. 217 The goal of having a mandatory to implement function set is to 218 prevent negotiation failure, not to preempt or prevent negotiation. 220 The presence of a mandatory to implement function set serves as a 221 strong changer of the marketplace of deployment - in that it gives a 222 guarantee that, as long as you conform to a specification, and the 223 other party is willing to accept communication at the base level of 224 that specification, you can communicate successfully. 226 The alternative - that of having no mandatory to implement - does not 227 mean that you cannot communicate, it merely means that in order to be 228 part of the communications partnership, you have to implement the 229 standard "and then some" - that "and then some" usually being called 230 a profile of some sort; in the version most antithetical to the 231 Internet ethos, that "and then some" consists of having to use a 232 specific vendor's product only. 234 2.4. Terminology 236 The following terms are used in this document, and as far as possible 237 across the documents specifying the RTCWEB suite, in the specific 238 meanings given here. Not all terms are used in this document. Other 239 terms are used in their commonly used meaning. 241 The list is in alphabetical order. 243 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 245 API: Application Programming Interface - a specification of a set of 246 calls and events, usually tied to a programming language or an 247 abstract formal specification such as WebIDL, with its defined 248 semantics. 250 Browser: Used synonymously with "Interactive User Agent" as defined 251 in the HTML specification [W3C.WD-html5-20110525]. 253 ICE Agent: An implementation of the Interactive Connectivty 254 Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be 255 an SDP Agent, but there exist ICE Agents that do not use SDP (for 256 instance those that use Jingle). 258 Interactive: Communication between multiple parties, where the 259 expectation is that an action from one party can cause a reaction 260 by another party, and the reaction can be observed by the first 261 party, with the total time required for the action/reaction/ 262 observation is on the order of no more than hundreds of 263 milliseconds. 265 Media: Audio and video content. Not to be confused with 266 "transmission media" such as wires. 268 Media path: The path that media data follows from one browser to 269 another. 271 Protocol: A specification of a set of data units, their 272 representation, and rules for their transmission, with their 273 defined semantics. A protocol is usually thought of as going 274 between systems. 276 Real-time media: Media where generation of content and display of 277 content are intended to occur closely together in time (on the 278 order of no more than hundreds of milliseconds). Real-time media 279 can be used to support interactive communication. 281 SDP Agent: The protocol implementation involved in the SDP offer/ 282 answer exchange, as defined in [RFC3264] section 3. 284 Signaling: Communication that happens in order to establish, manage 285 and control media paths. 287 Signaling Path: The communication channels used between entities 288 participating in signaling to transfer signaling. There may be 289 more entities in the signaling path than in the media path. 291 NOTE: Where common definitions exist for these terms, those 292 definitions should be used to the greatest extent possible. 294 TODO: Extend this list with other terms that might prove slippery. 296 3. Architecture and Functionality groups 298 The model of real-time support for browser-based applications does 299 not envisage that the browser will contain all the functions that 300 need to be performed in order to have a function such as a telephone 301 or a video conferencing unit; the vision is that the browser will 302 have the functions that are needed for a Web application, working in 303 conjunction with its backend servers, to implement these functions. 305 This means that two vital interfaces need specification: The 306 protocols that browsers talk to each other, without any intervening 307 servers, and the APIs that are offered for a Javascript application 308 to take advantage of the browser's functionality. 310 +------------------------+ On-the-wire 311 | | Protocols 312 | Servers |---------> 313 | | 314 | | 315 +------------------------+ 316 ^ 317 | 318 | 319 | HTTP/ 320 | Websockets 321 | 322 | 323 +----------------------------+ 324 | Javascript/HTML/CSS | 325 +----------------------------+ 326 Other ^ ^RTC 327 APIs | |APIs 328 +---|-----------------|------+ 329 | | | | 330 | +---------+| 331 | | Browser || On-the-wire 332 | Browser | RTC || Protocols 333 | | Function|-----------> 334 | | || 335 | | || 336 | +---------+| 337 +---------------------|------+ 338 | 339 V 340 Native OS Services 342 Figure 1: Browser Model 344 Note that HTTP and Websockets are also offered to the Javascript 345 application through browser APIs. 347 As for all protocol and API specifications, there is no restriction 348 that the protocols can only be used to talk to another browser; since 349 they are fully specified, any device that implements the protocols 350 faithfully should be able to interoperate with the application 351 running in the browser. 353 A commonly imagined model of deployment is the one depicted below. 355 +-----------+ +-----------+ 356 | Web | | Web | 357 | | Signaling | | 358 | |-------------| | 359 | Server | path | Server | 360 | | | | 361 +-----------+ +-----------+ 362 / \ 363 / \ Application-defined 364 / \ over 365 / \ HTTP/Websockets 366 / Application-defined over \ 367 / HTTP/Websockets \ 368 / \ 369 +-----------+ +-----------+ 370 |JS/HTML/CSS| |JS/HTML/CSS| 371 +-----------+ +-----------+ 372 +-----------+ +-----------+ 373 | | | | 374 | | | | 375 | Browser | ------------------------- | Browser | 376 | | Media path | | 377 | | | | 378 +-----------+ +-----------+ 380 Figure 2: Browser RTC Trapezoid 382 On this drawing, the critical part to note is that the media path 383 ("low path") goes directly between the browsers, so it has to be 384 conformant to the specifications of the RTCWEB protocol suite; the 385 signaling path ("high path") goes via servers that can modify, 386 translate or massage the signals as needed. 388 If the two Web servers are operated by different entities, the inter- 389 server signaling mechanism needs to be agreed upon, either by 390 standardization or by other means of agreement. Existing protocols 391 (for example SIP or XMPP) could be used between servers, while either 392 a standards-based or proprietary protocol could be used between the 393 browser and the web server. 395 For example, if both operators' servers implement SIP, SIP could be 396 used for communication between servers, along with either a 397 standardized signaling mechanism (e.g. SIP over Websockets) or a 398 proprietary signaling mechanism used between the application running 399 in the browser and the web server. Similarly, if both operators' 400 servers implement XMPP, XMPP could be used for communication between 401 XMPP servers, with either a standardized signaling mechanism (e.g. 402 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 403 used between the application running in the browser and the web 404 server. 406 The choice of protocols, and definition of the translation between 407 them, is outside the scope of the RTCWEB standards suite described in 408 the document. 410 The functionality groups that are needed in the browser can be 411 specified, more or less from the bottom up, as: 413 o Data transport: TCP, UDP and the means to securely set up 414 connections between entities, as well as the functions for 415 deciding when to send data: Congestion management, bandwidth 416 estimation and so on. 418 o Data framing: RTP and other data formats that serve as containers, 419 and their functions for data confidentiality and integrity. 421 o Data formats: Codec specifications, format specifications and 422 functionality specifications for the data passed between systems. 423 Audio and video codecs, as well as formats for data and document 424 sharing, belong in this category. In order to make use of data 425 formats, a way to describe them, a session description, is needed. 427 o Connection management: Setting up connections, agreeing on data 428 formats, changing data formats during the duration of a call; SIP 429 and Jingle/XMPP belong in this category. 431 o Presentation and control: What needs to happen in order to ensure 432 that interactions behave in a non-surprising manner. This can 433 include floor control, screen layout, voice activated image 434 switching and other such functions - where part of the system 435 require the cooperation between parties. XCON and Cisco/ 436 Tandberg's TIP were some attempts at specifying this kind of 437 functionality; many applications have been built without 438 standardized interfaces to these functions. 440 o Local system support functions: These are things that need not be 441 specified uniformly, because each participant may choose to do 442 these in a way of the participant's choosing, without affecting 443 the bits on the wire in a way that others have to be cognizant of. 444 Examples in this category include echo cancellation (some forms of 445 it), local authentication and authorization mechanisms, OS access 446 control and the ability to do local recording of conversations. 448 Within each functionality group, it is important to preserve both 449 freedom to innovate and the ability for global communication. 450 Freedom to innovate is helped by doing the specification in terms of 451 interfaces, not implementation; any implementation able to 452 communicate according to the interfaces is a valid implementation. 453 Ability to communicate globally is helped both by having core 454 specifications be unencumbered by IPR issues and by having the 455 formats and protocols be fully enough specified to allow for 456 independent implementation. 458 One can think of the three first groups as forming a "media transport 459 infrastructure", and of the three last groups as forming a "media 460 service". In many contexts, it makes sense to use a common 461 specification for the media transport infrastructure, which can be 462 embedded in browsers and accessed using standard interfaces, and "let 463 a thousand flowers bloom" in the "media service" layer; to achieve 464 interoperable services, however, at least the first five of the six 465 groups need to be specified. 467 4. Data transport 469 Data transport refers to the sending and receiving of data over the 470 network interfaces, the choice of network-layer addresses at each end 471 of the communication, and the interaction with any intermediate 472 entities that handle the data, but do not modify it (such as TURN 473 relays). 475 It includes necessary functions for congestion control: When not to 476 send data. 478 The data transport protocols used by RTCWEB are described in 479 [I-D.ietf-rtcweb-transports]. 481 5. Data framing and securing 483 The format for media transport is RTP [RFC3550]. Implementation of 484 SRTP [RFC3711] is required for all implementations. 486 The detailed considerations for usage of functions from RTP and SRTP 487 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 488 considerations for the RTCWEB use case are in 489 [I-D.ietf-rtcweb-security], and the resulting security functions are 490 described in [I-D.ietf-rtcweb-security-arch]. 492 Considerations for the transfer of data that is not in RTP format is 493 described in [I-D.ietf-rtcweb-data-channel], and the resulting 494 protocol is described in [I-D.jesup-rtcweb-data-protocol] (not yet a 495 WG document) 497 6. Data formats 499 The intent of this specification is to allow each communications 500 event to use the data formats that are best suited for that 501 particular instance, where a format is supported by both sides of the 502 connection. However, a minimum standard is greatly helpful in order 503 to ensure that communication can be achieved. This document 504 specifies a minimum baseline that will be supported by all 505 implementations of this specification, and leaves further codecs to 506 be included at the will of the implementor. 508 The mandatory to implement codecs, as well as any profiling 509 requirements for both mandatory and optional codecs, is described in 510 (candidate draft: 511 [I-D.cbran-rtcweb-codec]. 513 7. Connection management 515 The methods, mechanisms and requirements for setting up, negotiating 516 and tearing down connections is a large subject, and one where it is 517 desirable to have both interoperability and freedom to innovate. 519 The following principles apply: 521 1. The RTCWEB media negotiations will be capable of representing the 522 same SDP offer/answer semantics that are used in SIP [RFC3264], 523 in such a way that it is possible to build a signaling gateway 524 between SIP and the RTCWEB media negotiation. 526 2. It will be possible to gateway between legacy SIP devices that 527 support ICE and appropriate RTP / SDP mechanisms, codecs and 528 security mechanisms without using a media gateway. A signaling 529 gateway to convert between the signaling on the web side to the 530 SIP signaling may be needed. 532 3. When a new codec is specified, and the SDP for the new codec is 533 specified in the MMUSIC WG, no other standardization should be 534 required for it to be possible to use that in the web browsers. 535 Adding new codecs which might have new SDP parameters should not 536 change the APIs between the browser and Javascript application. 537 As soon as the browsers support the new codecs, old applications 538 written before the codecs were specified should automatically be 539 able to use the new codecs where appropriate with no changes to 540 the JS applications. 542 The particular choices made for RTCWEB, and their implications for 543 the API offered by a browser implementing RTCWEB, are described in 544 [I-D.ietf-rtcweb-jsep]. This document in turn implements the 545 solutions described in [I-D.roach-mmusic-unified-plan]. 547 8. Presentation and control 549 The most important part of control is the user's control over the 550 browser's interaction with input/output devices and communications 551 channels. It is important that the user have some way of figuring 552 out where his audio, video or texting is being sent, for what 553 purported reason, and what guarantees are made by the parties that 554 form part of this control channel. This is largely a local function 555 between the browser, the underlying operating system and the user 556 interface; this is being worked on as part of the W3C API effort, and 557 will be part of the peer connection API [W3C.WD-webrtc-20120209], and 558 the media capture API [W3C.WD-mediacapture-streams-20120628]. 559 Considerations for the implications of wanting to identify 560 correspondents are described in [I-D.rescorla-rtcweb-generic-idp] 561 (not a WG item). 563 9. Local system support functions 565 These are characterized by the fact that the quality of these 566 functions strongly influence the user experience, but the exact 567 algorithm does not need coordination. In some cases (for instance 568 echo cancellation, as described below), the overall system definition 569 may need to specify that the overall system needs to have some 570 characteristics for which these facilities are useful, without 571 requiring them to be implemented a certain way. 573 Local functions include echo cancellation, volume control, camera 574 management including focus, zoom, pan/tilt controls (if available), 575 and more. 577 Certain parts of the system SHOULD conform to certain properties, for 578 instance: 580 o Echo cancellation should be good enough to achieve the suppression 581 of acoustical feedback loops below a perceptually noticeable 582 level. 584 o Privacy concerns must be satisfied; for instance, if remote 585 control of camera is offered, the APIs should be available to let 586 the local participant figure out who's controlling the camera, and 587 possibly decide to revoke the permission for camera usage. 589 o Automatic gain control, if present, should normalize a speaking 590 voice into a reasonable dB range. 592 The requirements on RTCWEB systems with regard to audio processing 593 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 594 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 596 10. IANA Considerations 598 This document makes no request of IANA. 600 Note to RFC Editor: this section may be removed on publication as an 601 RFC. 603 11. Security Considerations 605 Security of the web-enabled real time communications comes in several 606 pieces: 608 o Security of the components: The browsers, and other servers 609 involved. The most target-rich environment here is probably the 610 browser; the aim here should be that the introduction of these 611 components introduces no additional vulnerability. 613 o Security of the communication channels: It should be easy for a 614 participant to reassure himself of the security of his 615 communication - by verifying the crypto parameters of the links he 616 himself participates in, and to get reassurances from the other 617 parties to the communication that they promise that appropriate 618 measures are taken. 620 o Security of the partners' identity: verifying that the 621 participants are who they say they are (when positive 622 identification is appropriate), or that their identity cannot be 623 uncovered (when anonymity is a goal of the application). 625 The security analysis, and the requirements derived from that 626 analysis, is contained in [I-D.ietf-rtcweb-security]. 628 12. Acknowledgements 630 The number of people who have taken part in the discussions 631 surrounding this draft are too numerous to list, or even to identify. 632 The ones below have made special, identifiable contributions; this 633 does not mean that others' contributions are less important. 635 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 636 Westerlund and Joerg Ott, who offered technical contributions on 637 various versions of the draft. 639 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 640 the ASCII drawings in section 1. 642 Thanks to Eric Rescorla, Justin Uberti, Henry Sinnreich, Colin 643 Perkins, Bjoern Hoehrmann and Simon Leinen for document review, and 644 to Heath Matlock for grammatical review. 646 13. References 648 13.1. Normative References 650 [I-D.ietf-rtcweb-audio] 651 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 652 Requirements", draft-ietf-rtcweb-audio-02 (work in 653 progress), August 2013. 655 [I-D.ietf-rtcweb-data-channel] 656 Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Data 657 Channels", draft-ietf-rtcweb-data-channel-05 (work in 658 progress), July 2013. 660 [I-D.ietf-rtcweb-jsep] 661 Uberti, J. and C. Jennings, "Javascript Session 662 Establishment Protocol", draft-ietf-rtcweb-jsep-03 (work 663 in progress), February 2013. 665 [I-D.ietf-rtcweb-rtp-usage] 666 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 667 Communication (WebRTC): Media Transport and Use of RTP", 668 draft-ietf-rtcweb-rtp-usage-07 (work in progress), 669 July 2013. 671 [I-D.ietf-rtcweb-security] 672 Rescorla, E., "Security Considerations for WebRTC", 673 draft-ietf-rtcweb-security-05 (work in progress), 674 July 2013. 676 [I-D.ietf-rtcweb-security-arch] 677 Rescorla, E., "WebRTC Security Architecture", 678 draft-ietf-rtcweb-security-arch-07 (work in progress), 679 July 2013. 681 [I-D.ietf-rtcweb-transports] 682 Alvestrand, H., "Transports for RTCWEB", 683 draft-ietf-rtcweb-transports-00 (work in progress), 684 August 2013. 686 [I-D.roach-mmusic-unified-plan] 687 Roach, A., Uberti, J., and M. Thomson, "A Unified Plan for 688 Using SDP with Large Numbers of Media Flows", 689 draft-roach-mmusic-unified-plan-00 (work in progress), 690 July 2013. 692 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 693 with Session Description Protocol (SDP)", RFC 3264, 694 June 2002. 696 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 697 Jacobson, "RTP: A Transport Protocol for Real-Time 698 Applications", STD 64, RFC 3550, July 2003. 700 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 701 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 702 RFC 3711, March 2004. 704 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 705 (ICE): A Protocol for Network Address Translator (NAT) 706 Traversal for Offer/Answer Protocols", RFC 5245, 707 April 2010. 709 13.2. Informative References 711 [I-D.cbran-rtcweb-codec] 712 Bran, C., Jennings, C., and J. Valin, "WebRTC Codec and 713 Media Processing Requirements", 714 draft-cbran-rtcweb-codec-02 (work in progress), 715 March 2012. 717 [I-D.ietf-rtcweb-use-cases-and-requirements] 718 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 719 Time Communication Use-cases and Requirements", 720 draft-ietf-rtcweb-use-cases-and-requirements-11 (work in 721 progress), June 2013. 723 [I-D.jesup-rtcweb-data-protocol] 724 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 725 Protocol", draft-jesup-rtcweb-data-protocol-04 (work in 726 progress), February 2013. 728 [I-D.rescorla-rtcweb-generic-idp] 729 Rescorla, E., "RTCWEB Generic Identity Provider 730 Interface", draft-rescorla-rtcweb-generic-idp-01 (work in 731 progress), March 2012. 733 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", 734 BCP 95, RFC 3935, October 2004. 736 [W3C.WD-html5-20110525] 737 Hickson, I., "HTML5", World Wide Web Consortium 738 LastCall WD-html5-20110525, May 2011, 739 . 741 [W3C.WD-mediacapture-streams-20120628] 742 Burnett, D. and A. Narayanan, "Media Capture and Streams", 743 World Wide Web Consortium WD WD-mediacapture-streams- 744 20120628, June 2012, . 747 [W3C.WD-webrtc-20120209] 748 Bergkvist, A., Burnett, D., Jennings, C., and A. 749 Narayanan, "WebRTC 1.0: Real-time Communication Between 750 Browsers", World Wide Web Consortium WD WD-webrtc- 751 20120209, February 2012, 752 . 754 Appendix A. Change log 756 This section may be deleted by the RFC Editor when preparing for 757 publication. 759 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 761 Added section "On interoperability and innovation" 763 Added data confidentiality and integrity to the "data framing" layer 765 Added congestion management requirements in the "data transport" 766 layer section 768 Changed need for non-media data from "question: do we need this?" to 769 "Open issue: How do we do this?" 770 Strengthened disclaimer that listed codecs are placeholders, not 771 decisions. 773 More details on why the "local system support functions" section is 774 there. 776 A.2. Changes from draft-alvestrand-dispatch-01 to 777 draft-alvestrand-rtcweb-overview-00 779 Added section on "Relationship between API and protocol" 781 Added terminology section 783 Mentioned congestion management as part of the "data transport" layer 784 in the layer list 786 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 788 Removed most technical content, and replaced with pointers to drafts 789 as requested and identified by the RTCWEB WG chairs. 791 Added content to acknowledgments section. 793 Added change log. 795 Spell-checked document. 797 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to 798 draft-ietf-rtcweb-overview-00 800 Changed draft name and document date. 802 Removed unused references 804 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 806 Added architecture figures to section 2. 808 Changed the description of "echo cancellation" under "local system 809 support functions". 811 Added a few more definitions. 813 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 815 Added pointers to use cases, security and rtp-usage drafts (now WG 816 drafts). 818 Changed description of SRTP from mandatory-to-use to mandatory-to- 819 implement. 821 Added the "3 principles of negotiation" to the connection management 822 section. 824 Added an explicit statement that ICE is required for both NAT and 825 consent-to-receive. 827 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 829 Added references to a number of new drafts. 831 Expanded the description text under the "trapezoid" drawing with some 832 more text discussed on the list. 834 Changed the "Connection management" sentence from "will be done using 835 SDP offer/answer" to "will be capable of representing SDP offer/ 836 answer" - this seems more consistent with JSEP. 838 Added "security mechanisms" to the things a non-gatewayed SIP devices 839 must support in order to not need a media gateway. 841 Added a definition for "browser". 843 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 845 Made introduction more normative. 847 Several wording changes in response to review comments from EKR 849 Added an appendix to hold references and notes that are not yet in a 850 separate document. 852 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 854 Minor grammatical fixes. This is mainly a "keepalive" refresh. 856 A.10. Changes from -05 to -06 858 Clarifications in response to Last Call review comments. Inserted 859 reference to draft-ietf-rtcweb-audio. 861 A.11. Changes from -06 to -07 863 Added a refereence to the "unified plan" draft, and updated some 864 references. 866 Otherwise, it's a "keepalive" draft. 868 A.12. Changes from -07 to -08 870 Removed the appendix that detailed transports, and replaced it with a 871 reference to draft-ietf-rtcweb-transports. Removed now-unused 872 references. 874 Author's Address 876 Harald T. Alvestrand 877 Google 878 Kungsbron 2 879 Stockholm, 11122 880 Sweden 882 Email: harald@alvestrand.no