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Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Outdated reference: A later version (-26) exists of draft-ietf-rtcweb-rtp-usage-00 == Outdated reference: A later version (-12) exists of draft-ietf-rtcweb-security-00 == Outdated reference: A later version (-16) exists of draft-ietf-rtcweb-use-cases-and-requirements-05 -- Obsolete informational reference (is this intentional?): RFC 5245 (Obsoleted by RFC 8445, RFC 8839) Summary: 0 errors (**), 0 flaws (~~), 4 warnings (==), 2 comments (--). 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 28, 2011 5 Expires: March 31, 2012 7 Overview: Real Time Protocols for Brower-based Applications 8 draft-ietf-rtcweb-overview-02 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 work is an attempt to synthesize the input of many people, but 22 makes no claims to fully represent the views of any of them. All 23 parts of the document should be regarded as open for discussion, 24 unless the RTCWEB chairs have declared consensus on an item. 26 This document is a work item of the RTCWEB working group. 28 Requirements Language 30 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 31 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 32 document are to be interpreted as described in RFC 2119 [RFC2119]. 34 Status of this Memo 36 This Internet-Draft is submitted in full conformance with the 37 provisions of BCP 78 and BCP 79. 39 Internet-Drafts are working documents of the Internet Engineering 40 Task Force (IETF). Note that other groups may also distribute 41 working documents as Internet-Drafts. The list of current Internet- 42 Drafts is at http://datatracker.ietf.org/drafts/current/. 44 Internet-Drafts are draft documents valid for a maximum of six months 45 and may be updated, replaced, or obsoleted by other documents at any 46 time. It is inappropriate to use Internet-Drafts as reference 47 material or to cite them other than as "work in progress." 48 This Internet-Draft will expire on March 31, 2012. 50 Copyright Notice 52 Copyright (c) 2011 IETF Trust and the persons identified as the 53 document authors. All rights reserved. 55 This document is subject to BCP 78 and the IETF Trust's Legal 56 Provisions Relating to IETF Documents 57 (http://trustee.ietf.org/license-info) in effect on the date of 58 publication of this document. Please review these documents 59 carefully, as they describe your rights and restrictions with respect 60 to this document. Code Components extracted from this document must 61 include Simplified BSD License text as described in Section 4.e of 62 the Trust Legal Provisions and are provided without warranty as 63 described in the Simplified BSD License. 65 Table of Contents 67 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 68 2. Principles and Terminology . . . . . . . . . . . . . . . . . . 5 69 2.1. Goals of this document . . . . . . . . . . . . . . . . . . 5 70 2.2. Relationship between API and protocol . . . . . . . . . . 5 71 2.3. On interoperability and innovation . . . . . . . . . . . . 6 72 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 7 73 3. Architecture and Functionality groups . . . . . . . . . . . . 8 74 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . . 12 75 5. Data framing and securing . . . . . . . . . . . . . . . . . . 12 76 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . . 12 77 7. Connection management . . . . . . . . . . . . . . . . . . . . 13 78 8. Presentation and control . . . . . . . . . . . . . . . . . . . 13 79 9. Local system support functions . . . . . . . . . . . . . . . . 14 80 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14 81 11. Security Considerations . . . . . . . . . . . . . . . . . . . 14 82 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 15 83 13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15 84 13.1. Normative References . . . . . . . . . . . . . . . . . . . 15 85 13.2. Informative References . . . . . . . . . . . . . . . . . . 16 86 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 16 87 A.1. Changes from 88 draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 17 89 A.2. Changes from draft-alvestrand-dispatch-01 to 90 draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 17 91 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 17 92 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to 93 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 17 94 A.5. Changes from draft-ietf-rtcweb-overview -00 to -01 . . . . 18 95 A.6. Changes from draft-ietf-rtcweb-overview -01 to -02 . . . . 18 96 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 18 98 1. Introduction 100 The Internet was, from very early in its lifetime, considered a 101 possible vehicle for the deployment of real-time, interactive 102 applications - with the most easily imaginable being audio 103 conversations (aka "Internet telephony") and videoconferencing. 105 The first attempts to build this were dependent on special networks, 106 special hardware and custom-built software, often at very high prices 107 or at low quality, placing great demands on the infrastructure. 109 As the available bandwidth has increased, and as processors and other 110 hardware has become ever faster, the barriers to participation have 111 decreased, and it is possible to deliver a satisfactory experience on 112 commonly available computing hardware. 114 Still, there are a number of barriers to the ability to communicate 115 universally - one of these is that there is, as of yet, no single set 116 of communication protocols that all agree should be made available 117 for communication; another is the sheer lack of universal 118 identification systems (such as is served by telephone numbers or 119 email addresses in other communications systems). 121 Development of The Universal Solution has proved hard, however, for 122 all the usual reasons. This memo aims to take a more building-block- 123 oriented approach, and try to find consensus on a set of substrate 124 components that we think will be useful in any real-time 125 communications systems. 127 The last few years have also seen a new platform rise for deployment 128 of services: The browser-embedded application, or "Web application". 129 It turns out that as long as the browser platform has the necessary 130 interfaces, it is possible to deliver almost any kind of service on 131 it. 133 Traditionally, these interfaces have been delivered by plugins, which 134 had to be downloaded and installed separately from the browser; in 135 the development of HTML5, much promise is seen by the possibility of 136 making those interfaces available in a standardized way within the 137 browser. 139 This memo specifies a set of building blocks that can be made 140 accessible and controllable through a Javascript API interface in a 141 browser, and which together form a necessary and sufficient set of 142 functions to allow the use of interactive audio and video in 143 applications that communicate directly between browsers across the 144 Internet. The resulting protocol suite is intended to enable all the 145 applications that are described as required scenarios in the RTCWEB 146 use cases document [I-D.ietf-rtcweb-use-cases-and-requirements]. 148 Other efforts, for instance the W3C WebRTC, Web Applications and 149 Device API working groups, focus on making standardized APIs and 150 interfaces available, within or alongside the HTML5 effort, for those 151 functions; this memo concentrates on specifying the protocols and 152 subprotocols that are needed to specify the interactions that happen 153 across the network. 155 2. Principles and Terminology 157 2.1. Goals of this document 159 The goal of the RTCWEB protocol specification is to specify a set of 160 protocols that, if all are implemented, will allow the implementation 161 to communicate with another implementation using audio, video and 162 auxiliary data sent along the most direct possible path between the 163 participants. 165 This document is intended to serve as the roadmap to the RTCWEB 166 specifications. It defines terms used by other pieces of 167 specification, lists references to other specifications that don't 168 need further elaboration in the RTCWEB context, and gives pointers to 169 other documents that form part of the RTCWEB suite. 171 By reading this document and the documents it refers to, it should be 172 possible to have all information needed to implement an RTCWEB 173 compatible implementation. 175 2.2. Relationship between API and protocol 177 The total RTCWEB/WEBRTC effort consists of two pieces: 179 o A protocol specification, done in the IETF 181 o A Javascript API specification, done in the W3C [webrtc-api] 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 possible by observing 192 the bits on the wire whether they come from a browser or from another 193 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 ICE Agent: An implementation of the ICE [RFC5245] protocol. An ICE 257 Agent may also be an SDP Agent, but there exist ICE Agents that do 258 not use SDP (for instance those that use Jingle). 260 Interactive: Communication between multiple parties, where the 261 expectation is that an action from one party can cause a reaction 262 by another party, and the reaction can be observed by the first 263 party, with the total time required for the action/reaction/ 264 observation is on the order of no more than hundreds of 265 milliseconds. 267 Media: Audio and video content. Not to be confused with 268 "transmission media" such as wires. 270 Media path: The path that media data follows from one browser to 271 another. 273 Protocol: A specification of a set of data units, their 274 representation, and rules for their transmission, with their 275 defined semantics. A protocol is usually thought of as going 276 between systems. 278 Real-time media: Media where generation of content and display of 279 content are intended to occur closely together in time (on the 280 order of no more than hundreds of milliseconds). 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 signalling 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 videoconferencing unit; the vision is that the browser will have 303 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 As for all protocol and API specifications, there is no restriction 346 that the protocols can only be used to talk to another browser; since 347 they are fully specified, any device that implements the protocols 348 faithfully should be able to interoperate with the application 349 running in the browser. 351 A commonly imagined model of deployment is the one depicted below. 353 +-----------+ +-----------+ 354 | Web | | Web | 355 | | Signalling | | 356 | |-------------| | 357 | Server | path | Server | 358 | | | | 359 +-----------+ +-----------+ 360 / \ 361 / \ Proprietary over 362 / \ HTTP/Websockets 363 / \ 364 / Proprietary over \ 365 / HTTP/Websockets \ 366 / \ 367 +-----------+ +-----------+ 368 |JS/HTML/CSS| |JS/HTML/CSS| 369 +-----------+ +-----------+ 370 +-----------+ +-----------+ 371 | | | | 372 | | | | 373 | Browser | ------------------------- | Browser | 374 | | Media path | | 375 | | | | 376 +-----------+ +-----------+ 378 Figure 2: Browser RTC Trapezoid 380 If the two Web servers are operated by different entities, the 381 signalling path needs to be agreed upon, either by standardization or 382 by other means of agreement; for example, both servers might 383 implement SIP, and the servers would talk SIP to each other, and each 384 would translate between the SIP protocol and their proprietary 385 representation for sending to their application running in the 386 browser. This part is outside the scope of the RTCWEB standars 387 suite. 389 On this drawing, the critical part to note is that the media path 390 ("low path") goes directly between the browsers, so it has to be 391 conformant to the specifications of the RTCWEB protocol suite; the 392 signalling path ("high path") goes via servers that can modify, 393 translate or massage the signals as needed. 395 The functionality groups that are needed in the browser can be 396 specified, more or less from the bottom up, as: 398 o Data transport: TCP, UDP and the means to securely set up 399 connections between entities, as well as the functions for 400 deciding when to send data: Congestion management, bandwidth 401 estimation and so on. 403 o Data framing: RTP and other data formats that serve as containers, 404 and their functions for data confidentiality and integrity. 406 o Data formats: Codec specifications, format specifications and 407 functionality specifications for the data passed between systems. 408 Audio and video codecs, as well as formats for data and document 409 sharing, belong in this category. In order to make use of data 410 formats, a way to describe them, a session description, is needed. 412 o Connection management: Setting up connections, agreeing on data 413 formats, changing data formats during the duration of a call; SIP 414 and Jingle/XMPP belong in this category. 416 o Presentation and control: What needs to happen in order to ensure 417 that interactions behave in a non-surprising manner. This can 418 include floor control, screen layout, voice activated image 419 switching and other such functions - where part of the system 420 require the cooperation between parties. Cisco/Tandberg's TIP was 421 one attempt at specifying this functionality. 423 o Local system support functions: These are things that need not be 424 specified uniformly, because each participant may choose to do 425 these in a way of the participant's choosing, without affecting 426 the bits on the wire in a way that others have to be cognizant of. 427 Examples in this category include echo cancellation (some forms of 428 it), local authentication and authorization mechanisms, OS access 429 control and the ability to do local recording of conversations. 431 Within each functionality group, it is important to preserve both 432 freedom to innovate and the ability for global communication. 433 Freedom to innovate is helped by doing the specification in terms of 434 interfaces, not implementation; any implementation able to 435 communicate according to the interfaces is a valid implementation. 436 Ability to communicate globally is helped both by having core 437 specifications be unencumbered by IPR issues and by having the 438 formats and protocols be fully enough specified to allow for 439 independent implementation. 441 One can think of the three first groups as forming a "media transport 442 infrastructure", and of the three last groups as forming a "media 443 service". In many contexts, it makes sense to use a common 444 specification for the media transport infrastructure, which can be 445 embedded in browsers and accessed using standard interfaces, and "let 446 a thousand flowers bloom" in the "media service" layer; to achieve 447 interoperable services, however, at least the first five of the six 448 groups need to be specified. 450 4. Data transport 452 Data transport refers to the sending and receiving of data over the 453 network interfaces, the choice of network-layer addresses at each end 454 of the communication, and the interaction with any intermediate 455 entities that handle the data, but do not modify it (such as TURN 456 relays). 458 It includes necessary functions for congestion control: When not to 459 send data. 461 The data transport protocols used by RTCWEB are described in . 464 ICE is required for all media paths that use UDP; in addition to the 465 ability to pass NAT boxes, ICE fulfils the need for guaranteeing that 466 the media path is going to an UDP port that is willing to receive the 467 data. 469 The details of interactions with intermediate boxes, such as 470 firewalls, relays and NAT boxes, is described in . 473 5. Data framing and securing 475 The format for media transport is RTP [RFC3550]. Implementation of 476 SRTP [RFC3711] is required for all implementations. 478 The detailed considerations for usage of functions from RTP and SRTP 479 are given in [I-D.ietf-rtcweb-rtp-usage]. Key negotiation for SRTP 480 is described in . Transfer of data that is not in RTP 481 format is described in . 483 6. Data formats 485 The intent of this specification is to allow each communications 486 event to use the data formats that are best suited for that 487 particular instance, where a format is supported by both sides of the 488 connection. However, a minimum standard is greatly helpful in order 489 to ensure that communication can be achieved. This document 490 specifies a minimum baseline that will be supported by all 491 implementations of this specification, and leaves further codecs to 492 be included at the will of the implementor. 494 The mandatory to implement codecs, as well as any profiling 495 requirements for both mandatory and optional codecs, is described in 496 . 498 7. Connection management 500 The methods, mechanisms and requirements for setting up, negotiating 501 and tearing down connections is a large subject, and one where it is 502 desirable to have both interoperability and freedom to innovate. 504 The following principles apply: 506 1. The media negotiations will be done using the same SDP offer/ 507 answer semantics that are used in SIP [RFC3264], in such a way 508 that it is possible to build a signalling gateway between SIP and 509 the RTCWEB media negotiation. 511 2. It will be possible to gateway between legacy SIP devices that 512 support ICE and appropriate RTP / SDP mechanisms and codecs 513 without using a media gateway. A signaling gateway to convert 514 between the signaling on the web side to the SIP signaling may be 515 needed. 517 3. When a new codec is specified, and the SDP for the new codec is 518 specified in the MMUSIC WG, no other standardization would should 519 be required for it to be possible to use that in the web 520 browsers. Adding new codecs which might have new SDP parameters 521 should not change the APIs between the browser and javascript 522 application. As soon as the browsers support the new codecs, old 523 applications written before the codecs were specified should 524 automatically be able to use the new codecs where appropriate 525 with no changes to the JS applications. 527 The particular choices made for RTCWEB are described in . 530 8. Presentation and control 532 The most important part of control is the user's control over the 533 browser's interaction with input/output devices and communications 534 channels. It is important that the user have some way of figuring 535 out where his audio, video or texting is being sent, for what 536 purported reason, and what guarantees are made by the parties that 537 form part of this control channel. This is largely a local function 538 between the browser, the underlying operating system and the user 539 interface; this is being worked on as part of the W3C API effort, and 540 will be part of [webrtc-api] 542 9. Local system support functions 544 These are characterized by the fact that the quality of these 545 functions strongly influences the user experience, but the exact 546 algorithm does not need coordination. In some cases (for instance 547 echo cancellation, as described below), the overall system definition 548 may need to specify that the overall system needs to have some 549 characteristics for which these facilities are useful, without 550 requiring them to be implemented a certain way. 552 Local functions include echo cancellation, volume control, camera 553 management including focus, zoom, pan/tilt controls (if available), 554 and more. 556 Certain parts of the system SHOULD conform to certain properties, for 557 instance: 559 o Echo cancellation should be good enough to achieve the suppression 560 of acoustical feedback loops below a perceptually noticeable 561 level. 563 o Privacy concerns must be satisfied; for instance, if remote 564 control of camera is offered, the APIs should be available to let 565 the local participant to figure out who's controlling the camera, 566 and possibly decide to revoke the permission for camera usage. 568 o Automatic gain control, if present, should normalize a speaking 569 voice into 572 The requirements on RTCWEB systems in this category are found in 573 . 575 10. IANA Considerations 577 This document makes no request of IANA. 579 Note to RFC Editor: this section may be removed on publication as an 580 RFC. 582 11. Security Considerations 584 Security of the web-enabled real time communications comes in several 585 pieces: 587 o Security of the components: The browsers, and other servers 588 involved. The most target-rich environment here is probably the 589 browser; the aim here should be that the introduction of these 590 components introduces no additional vulnerability. 592 o Security of the communication channels: It should be easy for a 593 participant to reassure himself of the security of his 594 communication - by verifying the crypto parameters of the links he 595 himself participates in, and to get reassurances from the other 596 parties to the communication that they promise that appropriate 597 measures are taken. 599 o Security of the partners' identity: verifying that the 600 participants are who they say they are (when positive 601 identification is appropriate), or that their identity cannot be 602 uncovered (when anonymity is a goal of the application). 604 The security analysis, and the requirements derived from that 605 analysis, is contained in [I-D.ietf-rtcweb-security]. 607 12. Acknowledgements 609 The number of people who have taken part in the discussions 610 surrounding this draft are too numerous to list, or even to identify. 611 The ones below have made special, identifiable contributions; this 612 does not mean that others' contributions are less important. 614 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 615 Westerlund and Joerg Ott, who offered technical contributions on 616 various versions of the draft. 618 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 619 the ASCII drawings in section 1. 621 Thanks to Justin Uberti, Henry Sinnreich, Colin Perkins and Simon 622 Leinen for document review. 624 13. References 626 13.1. Normative References 628 [I-D.ietf-rtcweb-rtp-usage] 629 Perkins, C., Westerlund, M., and J. Ott, "RTP Requirements 630 for RTC-Web", draft-ietf-rtcweb-rtp-usage-00 (work in 631 progress), September 2011. 633 [I-D.ietf-rtcweb-security] 634 Rescorla, E., "Security Considerations for RTC-Web", 635 draft-ietf-rtcweb-security-00 (work in progress), 636 September 2011. 638 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 639 Requirement Levels", BCP 14, RFC 2119, March 1997. 641 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 642 with Session Description Protocol (SDP)", RFC 3264, 643 June 2002. 645 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 646 Jacobson, "RTP: A Transport Protocol for Real-Time 647 Applications", STD 64, RFC 3550, July 2003. 649 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 650 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 651 RFC 3711, March 2004. 653 13.2. Informative References 655 [I-D.ietf-rtcweb-use-cases-and-requirements] 656 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 657 Time Communication Use-cases and Requirements", 658 draft-ietf-rtcweb-use-cases-and-requirements-05 (work in 659 progress), September 2011. 661 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", 662 BCP 95, RFC 3935, October 2004. 664 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 665 (ICE): A Protocol for Network Address Translator (NAT) 666 Traversal for Offer/Answer Protocols", RFC 5245, 667 April 2010. 669 [webrtc-api] 670 Bergkvist, Burnett, Jennings, Narayanan, "WebRTC 1.0: 671 Real-time Communication Between Browsers", August 2011. 673 Available at 674 http://dev.w3.org/2011/webrtc/editor/webrtc.html 676 Appendix A. Change log 678 This section may be deleted by the RFC Editor when preparing for 679 publication. 681 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 683 Added section "On interoperability and innovation" 685 Added data confidentiality and integrity to the "data framing" layer 687 Added congestion management requirements in the "data transport" 688 layer section 690 Changed need for non-media data from "question: do we need this?" to 691 "Open issue: How do we do this?" 693 Strengthened disclaimer that listed codecs are placeholders, not 694 decisions. 696 More details on why the "local system support functions" section is 697 there. 699 A.2. Changes from draft-alvestrand-dispatch-01 to 700 draft-alvestrand-rtcweb-overview-00 702 Added section on "Relationship between API and protocol" 704 Added terminology section 706 Mentioned congestion management as part of the "data transport" layer 707 in the layer list 709 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 711 Removed most technical content, and replaced with pointers to drafts 712 as requested and identified by the RTCWEB WG chairs. 714 Added content to acknowledgements section. 716 Added change log. 718 Spell-checked document. 720 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to 721 draft-ietf-rtcweb-overview-00 723 Changed draft name and document date. 725 Removed unused references 727 A.5. Changes from draft-ietf-rtcweb-overview -00 to -01 729 Added architecture figures to section 2. 731 Changed the description of "echo cancellation" under "local system 732 support functions". 734 Added a few more definitions. 736 A.6. Changes from draft-ietf-rtcweb-overview -01 to -02 738 Added pointers to use cases, security and rtp-usage drafts (now WG 739 drafts). 741 Changed description of SRTP from mandatory-to-use to mandatory-to- 742 implement. 744 Added the "3 principles of negotiation" to the connection management 745 section. 747 Added an explicit statement that ICE is required for both NAT and 748 consent-to-receive. 750 Author's Address 752 Harald T. Alvestrand 753 Google 754 Kungsbron 2 755 Stockholm, 11122 756 Sweden 758 Email: harald@alvestrand.no