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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group H. Alvestrand 3 Internet-Draft Google 4 Intended status: Standards Track June 17, 2014 5 Expires: December 19, 2014 7 Overview: Real Time Protocols for Browser-based Applications 8 draft-ietf-rtcweb-overview-10 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 an Applicability Statement - it does not itself 22 specify any protocol, but specifies which other specifications RTCWEB 23 compliant implementations are supposed to follow. 25 This document is a work item of the RTCWEB working group. 27 Status of This Memo 29 This Internet-Draft is submitted in full conformance with the 30 provisions of BCP 78 and BCP 79. 32 Internet-Drafts are working documents of the Internet Engineering 33 Task Force (IETF). Note that other groups may also distribute 34 working documents as Internet-Drafts. The list of current Internet- 35 Drafts is at http://datatracker.ietf.org/drafts/current/. 37 Internet-Drafts are draft documents valid for a maximum of six months 38 and may be updated, replaced, or obsoleted by other documents at any 39 time. It is inappropriate to use Internet-Drafts as reference 40 material or to cite them other than as "work in progress." 42 This Internet-Draft will expire on December 19, 2014. 44 Copyright Notice 46 Copyright (c) 2014 IETF Trust and the persons identified as the 47 document authors. All rights reserved. 49 This document is subject to BCP 78 and the IETF Trust's Legal 50 Provisions Relating to IETF Documents 51 (http://trustee.ietf.org/license-info) in effect on the date of 52 publication of this document. Please review these documents 53 carefully, as they describe your rights and restrictions with respect 54 to this document. Code Components extracted from this document must 55 include Simplified BSD License text as described in Section 4.e of 56 the Trust Legal Provisions and are provided without warranty as 57 described in the Simplified BSD License. 59 Table of Contents 61 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 62 2. Principles and Terminology . . . . . . . . . . . . . . . . . 4 63 2.1. Goals of this document . . . . . . . . . . . . . . . . . 4 64 2.2. Relationship between API and protocol . . . . . . . . . . 4 65 2.3. On interoperability and innovation . . . . . . . . . . . 5 66 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 67 3. Architecture and Functionality groups . . . . . . . . . . . . 7 68 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . 11 69 5. Data framing and securing . . . . . . . . . . . . . . . . . . 11 70 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . 12 71 7. Connection management . . . . . . . . . . . . . . . . . . . . 12 72 8. Presentation and control . . . . . . . . . . . . . . . . . . 13 73 9. Local system support functions . . . . . . . . . . . . . . . 13 74 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14 75 11. Security Considerations . . . . . . . . . . . . . . . . . . . 14 76 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 15 77 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 15 78 13.1. Normative References . . . . . . . . . . . . . . . . . . 15 79 13.2. Informative References . . . . . . . . . . . . . . . . . 17 80 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 17 81 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 82 to -01 . . . . . . . . . . . . . . . . . . . . . . . . . 17 83 A.2. Changes from draft-alvestrand-dispatch-01 to draft- 84 alvestrand-rtcweb-overview-00 . . . . . . . . . . . . . . 18 85 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . 18 86 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to 87 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 18 88 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 18 89 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 18 90 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 19 91 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 19 92 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 19 93 A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 19 94 A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 19 95 A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 20 96 A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 20 97 A.14. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 20 98 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 20 100 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 This memo uses the term "WebRTC" (note the case used) to refer to the 155 overall effort consisting of both IETF and W3C efforts. 157 2. Principles and Terminology 159 2.1. Goals of this document 161 The goal of the RTCWEB protocol specification is to specify a set of 162 protocols that, if all are implemented, will allow an implementation 163 to communicate with another implementation using audio, video and 164 data sent along the most direct possible path between the 165 participants. 167 This document is intended to serve as the roadmap to the RTCWEB 168 specifications. It defines terms used by other pieces of 169 specification, lists references to other specifications that don't 170 need further elaboration in the RTCWEB context, and gives pointers to 171 other documents that form part of the RTCWEB suite. 173 By reading this document and the documents it refers to, it should be 174 possible to have all information needed to implement an RTCWEB 175 compatible implementation. 177 2.2. Relationship between API and protocol 179 The total WebRTC effort consists of two pieces: 181 o A protocol specification, done in the IETF 183 o A Javascript API specification, done in the W3C 184 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 186 Together, these two specifications aim to provide an environment 187 where Javascript embedded in any page, viewed in any compatible 188 browser, when suitably authorized by its user, is able to set up 189 communication using audio, video and auxiliary data, where the 190 browser environment does not constrain the types of application in 191 which this functionality can be used. 193 The protocol specification does not assume that all implementations 194 implement this API; it is not intended to be necessary for 195 interoperation to know whether the entity one is communicating with 196 is a browser or another device implementing this specification. 198 The goal of cooperation between the protocol specification and the 199 API specification is that for all options and features of the 200 protocol specification, it should be clear which API calls to make to 201 exercise that option or feature; similarly, for any sequence of API 202 calls, it should be clear which protocol options and features will be 203 invoked. Both subject to constraints of the implementation, of 204 course. 206 For the purpose of this document, two classes of things that can 207 claim conformance are defined: 209 o A WebRTC browser is something that conforms to both the protocol 210 specification and the Javascript API defined above. 212 o A WebRTC device is something that conforms to the protocol 213 specification, but does not claim to implement the Javascript API. 215 All WebRTC browsers are WebRTC devices, so any requirement on a 216 WebRTC device also applies to a WebRTC browser. 218 2.3. On interoperability and innovation 220 The "Mission statement of the IETF" [RFC3935] states that "The 221 benefit of a standard to the Internet is in interoperability - that 222 multiple products implementing a standard are able to work together 223 in order to deliver valuable functions to the Internet's users." 225 Communication on the Internet frequently occurs in two phases: 227 o Two parties communicate, through some mechanism, what 228 functionality they both are able to support 230 o They use that shared communicative functionality to communicate, 231 or, failing to find anything in common, give up on communication. 233 There are often many choices that can be made for communicative 234 functionality; the history of the Internet is rife with the proposal, 235 standardization, implementation, and success or failure of many types 236 of options, in all sorts of protocols. 238 The goal of having a mandatory to implement function set is to 239 prevent negotiation failure, not to preempt or prevent negotiation. 241 The presence of a mandatory to implement function set serves as a 242 strong changer of the marketplace of deployment - in that it gives a 243 guarantee that, as long as you conform to a specification, and the 244 other party is willing to accept communication at the base level of 245 that specification, you can communicate successfully. 247 The alternative - that of having no mandatory to implement - does not 248 mean that you cannot communicate, it merely means that in order to be 249 part of the communications partnership, you have to implement the 250 standard "and then some" - that "and then some" usually being called 251 a profile of some sort; in the version most antithetical to the 252 Internet ethos, that "and then some" consists of having to use a 253 specific vendor's product only. 255 2.4. Terminology 257 The following terms are used in this document, and as far as possible 258 across the documents specifying the RTCWEB suite, in the specific 259 meanings given here. Not all terms are used in this document. Other 260 terms are used in their commonly used meaning. 262 The list is in alphabetical order. 264 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 266 API: Application Programming Interface - a specification of a set of 267 calls and events, usually tied to a programming language or an 268 abstract formal specification such as WebIDL, with its defined 269 semantics. 271 Browser: Used synonymously with "Interactive User Agent" as defined 272 in the HTML specification [W3C.WD-html5-20110525]. 274 ICE Agent: An implementation of the Interactive Connectivty 275 Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be 276 an SDP Agent, but there exist ICE Agents that do not use SDP (for 277 instance those that use Jingle). 279 Interactive: Communication between multiple parties, where the 280 expectation is that an action from one party can cause a reaction 281 by another party, and the reaction can be observed by the first 282 party, with the total time required for the action/reaction/ 283 observation is on the order of no more than hundreds of 284 milliseconds. 286 Media: Audio and video content. Not to be confused with 287 "transmission media" such as wires. 289 Media path: The path that media data follows from one browser to 290 another. 292 Protocol: A specification of a set of data units, their 293 representation, and rules for their transmission, with their 294 defined semantics. A protocol is usually thought of as going 295 between systems. 297 Real-time media: Media where generation of content and display of 298 content are intended to occur closely together in time (on the 299 order of no more than hundreds of milliseconds). Real-time media 300 can be used to support interactive communication. 302 SDP Agent: The protocol implementation involved in the SDP offer/ 303 answer exchange, as defined in [RFC3264] section 3. 305 Signaling: Communication that happens in order to establish, manage 306 and control media paths. 308 Signaling Path: The communication channels used between entities 309 participating in signaling to transfer signaling. There may be 310 more entities in the signaling path than in the media path. 312 NOTE: Where common definitions exist for these terms, those 313 definitions should be used to the greatest extent possible. 315 3. Architecture and Functionality groups 317 The model of real-time support for browser-based applications does 318 not assume that the browser will contain all the functions that need 319 to be performed in order to have a function such as a telephone or a 320 video conferencing unit; the vision is that the browser will have the 321 functions that are needed for a Web application, working in 322 conjunction with its backend servers, to implement these functions. 324 This means that two vital interfaces need specification: The 325 protocols that browsers talk to each other, without any intervening 326 servers, and the APIs that are offered for a Javascript application 327 to take advantage of the browser's functionality. 329 +------------------------+ On-the-wire 330 | | Protocols 331 | Servers |---------> 332 | | 333 | | 334 +------------------------+ 335 ^ 336 | 337 | 338 | HTTP/ 339 | Websockets 340 | 341 | 342 +----------------------------+ 343 | Javascript/HTML/CSS | 344 +----------------------------+ 345 Other ^ ^RTC 346 APIs | |APIs 347 +---|-----------------|------+ 348 | | | | 349 | +---------+| 350 | | Browser || On-the-wire 351 | Browser | RTC || Protocols 352 | | Function|-----------> 353 | | || 354 | | || 355 | +---------+| 356 +---------------------|------+ 357 | 358 V 359 Native OS Services 361 Figure 1: Browser Model 363 Note that HTTP and Websockets are also offered to the Javascript 364 application through browser APIs. 366 As for all protocol and API specifications, there is no restriction 367 that the protocols can only be used to talk to another browser; since 368 they are fully specified, any device that implements the protocols 369 faithfully should be able to interoperate with the application 370 running in the browser. 372 A commonly imagined model of deployment is the one depicted below. 374 +-----------+ +-----------+ 375 | Web | | Web | 376 | | Signaling | | 377 | |-------------| | 378 | Server | path | Server | 379 | | | | 380 +-----------+ +-----------+ 381 / \ 382 / \ Application-defined 383 / \ over 384 / \ HTTP/Websockets 385 / Application-defined over \ 386 / HTTP/Websockets \ 387 / \ 388 +-----------+ +-----------+ 389 |JS/HTML/CSS| |JS/HTML/CSS| 390 +-----------+ +-----------+ 391 +-----------+ +-----------+ 392 | | | | 393 | | | | 394 | Browser | ------------------------- | Browser | 395 | | Media path | | 396 | | | | 397 +-----------+ +-----------+ 399 Figure 2: Browser RTC Trapezoid 401 On this drawing, the critical part to note is that the media path 402 ("low path") goes directly between the browsers, so it has to be 403 conformant to the specifications of the RTCWEB protocol suite; the 404 signaling path ("high path") goes via servers that can modify, 405 translate or massage the signals as needed. 407 If the two Web servers are operated by different entities, the inter- 408 server signaling mechanism needs to be agreed upon, either by 409 standardization or by other means of agreement. Existing protocols 410 (for example SIP [RFC3261] or XMPP [RFC6120]) could be used between 411 servers, while either a standards-based or proprietary protocol could 412 be used between the browser and the web server. 414 For example, if both operators' servers implement SIP, SIP could be 415 used for communication between servers, along with either a 416 standardized signaling mechanism (e.g. SIP over Websockets) or a 417 proprietary signaling mechanism used between the application running 418 in the browser and the web server. Similarly, if both operators' 419 servers implement XMPP, XMPP could be used for communication between 420 XMPP servers, with either a standardized signaling mechanism (e.g. 421 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 422 used between the application running in the browser and the web 423 server. 425 The choice of protocols, and definition of the translation between 426 them, is outside the scope of the RTCWEB standards suite described in 427 the document. 429 The functionality groups that are needed in the browser can be 430 specified, more or less from the bottom up, as: 432 o Data transport: TCP, UDP and the means to securely set up 433 connections between entities, as well as the functions for 434 deciding when to send data: Congestion management, bandwidth 435 estimation and so on. 437 o Data framing: RTP and other data formats that serve as containers, 438 and their functions for data confidentiality and integrity. 440 o Data formats: Codec specifications, format specifications and 441 functionality specifications for the data passed between systems. 442 Audio and video codecs, as well as formats for data and document 443 sharing, belong in this category. In order to make use of data 444 formats, a way to describe them, a session description, is needed. 446 o Connection management: Setting up connections, agreeing on data 447 formats, changing data formats during the duration of a call; SIP 448 and Jingle/XMPP belong in this category. 450 o Presentation and control: What needs to happen in order to ensure 451 that interactions behave in a non-surprising manner. This can 452 include floor control, screen layout, voice activated image 453 switching and other such functions - where part of the system 454 require the cooperation between parties. XCON and Cisco/ 455 Tandberg's TIP were some attempts at specifying this kind of 456 functionality; many applications have been built without 457 standardized interfaces to these functions. 459 o Local system support functions: These are things that need not be 460 specified uniformly, because each participant may choose to do 461 these in a way of the participant's choosing, without affecting 462 the bits on the wire in a way that others have to be cognizant of. 463 Examples in this category include echo cancellation (some forms of 464 it), local authentication and authorization mechanisms, OS access 465 control and the ability to do local recording of conversations. 467 Within each functionality group, it is important to preserve both 468 freedom to innovate and the ability for global communication. 469 Freedom to innovate is helped by doing the specification in terms of 470 interfaces, not implementation; any implementation able to 471 communicate according to the interfaces is a valid implementation. 472 Ability to communicate globally is helped both by having core 473 specifications be unencumbered by IPR issues and by having the 474 formats and protocols be fully enough specified to allow for 475 independent implementation. 477 One can think of the three first groups as forming a "media transport 478 infrastructure", and of the three last groups as forming a "media 479 service". In many contexts, it makes sense to use a common 480 specification for the media transport infrastructure, which can be 481 embedded in browsers and accessed using standard interfaces, and "let 482 a thousand flowers bloom" in the "media service" layer; to achieve 483 interoperable services, however, at least the first five of the six 484 groups need to be specified. 486 4. Data transport 488 Data transport refers to the sending and receiving of data over the 489 network interfaces, the choice of network-layer addresses at each end 490 of the communication, and the interaction with any intermediate 491 entities that handle the data, but do not modify it (such as TURN 492 relays). 494 It includes necessary functions for congestion control: When not to 495 send data. 497 WebRTC devices MUST implement the transport protocols described in 498 [I-D.ietf-rtcweb-transports]. 500 5. Data framing and securing 502 The format for media transport is RTP [RFC3550]. Implementation of 503 SRTP [RFC3711] is REQUIRED for all implementations. 505 The detailed considerations for usage of functions from RTP and SRTP 506 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 507 considerations for the RTCWEB use case are in 508 [I-D.ietf-rtcweb-security], and the resulting security functions are 509 described in [I-D.ietf-rtcweb-security-arch]. 511 Considerations for the transfer of data that is not in RTP format is 512 described in [I-D.ietf-rtcweb-data-channel], and a supporting 513 protocol is described in [I-D.ietf-rtcweb-data-protocol]. Webrtc 514 devices MUST implement these two specifications. 516 WebRTC devices MUST implement [I-D.ietf-rtcweb-rtp-usage], 517 [I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the 518 requirements they include. 520 6. Data formats 522 The intent of this specification is to allow each communications 523 event to use the data formats that are best suited for that 524 particular instance, where a format is supported by both sides of the 525 connection. However, a minimum standard is greatly helpful in order 526 to ensure that communication can be achieved. This document 527 specifies a minimum baseline that will be supported by all 528 implementations of this specification, and leaves further codecs to 529 be included at the will of the implementor. 531 WebRTC devices MUST implement the codecs and profiles required in 532 [I-D.ietf-rtcweb-audio] 534 NOTE IN DRAFT: At this time (June 2014) there is no consensus on what 535 to say about video codecs in this section. 537 7. Connection management 539 The methods, mechanisms and requirements for setting up, negotiating 540 and tearing down connections is a large subject, and one where it is 541 desirable to have both interoperability and freedom to innovate. 543 The following principles apply: 545 1. The RTCWEB media negotiations will be capable of representing the 546 same SDP offer/answer semantics that are used in SIP [RFC3264], 547 in such a way that it is possible to build a signaling gateway 548 between SIP and the RTCWEB media negotiation. 550 2. It will be possible to gateway between legacy SIP devices that 551 support ICE and appropriate RTP / SDP mechanisms, codecs and 552 security mechanisms without using a media gateway. A signaling 553 gateway to convert between the signaling on the web side to the 554 SIP signaling may be needed. 556 3. When a new codec is specified, and the SDP for the new codec is 557 specified in the MMUSIC WG, no other standardization should be 558 required for it to be possible to use that in the web browsers. 560 Adding new codecs which might have new SDP parameters should not 561 change the APIs between the browser and Javascript application. 562 As soon as the browsers support the new codecs, old applications 563 written before the codecs were specified should automatically be 564 able to use the new codecs where appropriate with no changes to 565 the JS applications. 567 The particular choices made for RTCWEB, and their implications for 568 the API offered by a browser implementing RTCWEB, are described in 569 [I-D.ietf-rtcweb-jsep]. 571 WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. 573 NOTE IN DRAFT: Is there any part of -jsep that WebRTC devices need to 574 be required to implement, and are not also required via other paths? 576 8. Presentation and control 578 The most important part of control is the user's control over the 579 browser's interaction with input/output devices and communications 580 channels. It is important that the user have some way of figuring 581 out where his audio, video or texting is being sent, for what 582 purported reason, and what guarantees are made by the parties that 583 form part of this control channel. This is largely a local function 584 between the browser, the underlying operating system and the user 585 interface; this is specified in the peer connection API 586 [W3C.WD-webrtc-20120209], and the media capture API 587 [W3C.WD-mediacapture-streams-20120628]. 589 WebRTC browsers MUST implement these two specifications. 591 9. Local system support functions 593 These are characterized by the fact that the quality of these 594 functions strongly influence the user experience, but the exact 595 algorithm does not need coordination. In some cases (for instance 596 echo cancellation, as described below), the overall system definition 597 may need to specify that the overall system needs to have some 598 characteristics for which these facilities are useful, without 599 requiring them to be implemented a certain way. 601 Local functions include echo cancellation, volume control, camera 602 management including focus, zoom, pan/tilt controls (if available), 603 and more. 605 Certain parts of the system SHOULD conform to certain properties, for 606 instance: 608 o Echo cancellation should be good enough to achieve the suppression 609 of acoustical feedback loops below a perceptually noticeable 610 level. 612 o Privacy concerns MUST be satisfied; for instance, if remote 613 control of camera is offered, the APIs should be available to let 614 the local participant figure out who's controlling the camera, and 615 possibly decide to revoke the permission for camera usage. 617 o Automatic gain control, if present, should normalize a speaking 618 voice into a reasonable dB range. 620 The requirements on RTCWEB systems with regard to audio processing 621 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 622 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 624 WebRTC browsers MUST implement the processing functions in 625 [I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6, 626 this means that browsers MUST implement the whole document.) 628 10. IANA Considerations 630 This document makes no request of IANA. 632 Note to RFC Editor: this section may be removed on publication as an 633 RFC. 635 11. Security Considerations 637 Security of the web-enabled real time communications comes in several 638 pieces: 640 o Security of the components: The browsers, and other servers 641 involved. The most target-rich environment here is probably the 642 browser; the aim here should be that the introduction of these 643 components introduces no additional vulnerability. 645 o Security of the communication channels: It should be easy for a 646 participant to reassure himself of the security of his 647 communication - by verifying the crypto parameters of the links he 648 himself participates in, and to get reassurances from the other 649 parties to the communication that they promise that appropriate 650 measures are taken. 652 o Security of the partners' identity: verifying that the 653 participants are who they say they are (when positive 654 identification is appropriate), or that their identity cannot be 655 uncovered (when anonymity is a goal of the application). 657 The security analysis, and the requirements derived from that 658 analysis, is contained in [I-D.ietf-rtcweb-security]. 660 It is also important to read the security sections of 661 [W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209]. 663 12. Acknowledgements 665 The number of people who have taken part in the discussions 666 surrounding this draft are too numerous to list, or even to identify. 667 The ones below have made special, identifiable contributions; this 668 does not mean that others' contributions are less important. 670 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 671 Westerlund and Joerg Ott, who offered technical contributions on 672 various versions of the draft. 674 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 675 the ASCII drawings in section 1. 677 Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric 678 Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage 679 and Simon Leinen for document review. 681 13. References 683 13.1. Normative References 685 [I-D.ietf-rtcweb-audio] 686 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 687 Requirements", draft-ietf-rtcweb-audio-05 (work in 688 progress), February 2014. 690 [I-D.ietf-rtcweb-data-channel] 691 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 692 Channels", draft-ietf-rtcweb-data-channel-10 (work in 693 progress), June 2014. 695 [I-D.ietf-rtcweb-data-protocol] 696 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 697 Establishment Protocol", draft-ietf-rtcweb-data- 698 protocol-06 (work in progress), June 2014. 700 [I-D.ietf-rtcweb-jsep] 701 Uberti, J. and C. Jennings, "Javascript Session 702 Establishment Protocol", draft-ietf-rtcweb-jsep-06 (work 703 in progress), February 2014. 705 [I-D.ietf-rtcweb-rtp-usage] 706 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 707 Communication (WebRTC): Media Transport and Use of RTP", 708 draft-ietf-rtcweb-rtp-usage-15 (work in progress), May 709 2014. 711 [I-D.ietf-rtcweb-security] 712 Rescorla, E., "Security Considerations for WebRTC", draft- 713 ietf-rtcweb-security-06 (work in progress), January 2014. 715 [I-D.ietf-rtcweb-security-arch] 716 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 717 rtcweb-security-arch-09 (work in progress), February 2014. 719 [I-D.ietf-rtcweb-transports] 720 Alvestrand, H., "Transports for RTCWEB", draft-ietf- 721 rtcweb-transports-05 (work in progress), June 2014. 723 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 724 with Session Description Protocol (SDP)", RFC 3264, June 725 2002. 727 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 728 Jacobson, "RTP: A Transport Protocol for Real-Time 729 Applications", STD 64, RFC 3550, July 2003. 731 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 732 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 733 RFC 3711, March 2004. 735 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 736 (ICE): A Protocol for Network Address Translator (NAT) 737 Traversal for Offer/Answer Protocols", RFC 5245, April 738 2010. 740 [W3C.WD-mediacapture-streams-20120628] 741 Burnett, D. and A. Narayanan, "Media Capture and Streams", 742 World Wide Web Consortium WD WD-mediacapture-streams- 743 20120628, June 2012, . 746 [W3C.WD-webrtc-20120209] 747 Bergkvist, A., Burnett, D., Jennings, C., and A. 748 Narayanan, "WebRTC 1.0: Real-time Communication Between 749 Browsers", World Wide Web Consortium WD WD-webrtc- 750 20120209, February 2012, 751 . 753 13.2. Informative References 755 [I-D.ietf-rtcweb-use-cases-and-requirements] 756 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 757 Time Communication Use-cases and Requirements", draft- 758 ietf-rtcweb-use-cases-and-requirements-14 (work in 759 progress), February 2014. 761 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 762 A., Peterson, J., Sparks, R., Handley, M., and E. 763 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 764 June 2002. 766 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", BCP 767 95, RFC 3935, October 2004. 769 [RFC6120] Saint-Andre, P., "Extensible Messaging and Presence 770 Protocol (XMPP): Core", RFC 6120, March 2011. 772 [W3C.WD-html5-20110525] 773 Hickson, I., "HTML5", World Wide Web Consortium LastCall 774 WD-html5-20110525, May 2011, 775 . 777 Appendix A. Change log 779 This section may be deleted by the RFC Editor when preparing for 780 publication. 782 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 784 Added section "On interoperability and innovation" 786 Added data confidentiality and integrity to the "data framing" layer 788 Added congestion management requirements in the "data transport" 789 layer section 791 Changed need for non-media data from "question: do we need this?" to 792 "Open issue: How do we do this?" 794 Strengthened disclaimer that listed codecs are placeholders, not 795 decisions. 797 More details on why the "local system support functions" section is 798 there. 800 A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand- 801 rtcweb-overview-00 803 Added section on "Relationship between API and protocol" 805 Added terminology section 807 Mentioned congestion management as part of the "data transport" layer 808 in the layer list 810 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 812 Removed most technical content, and replaced with pointers to drafts 813 as requested and identified by the RTCWEB WG chairs. 815 Added content to acknowledgments section. 817 Added change log. 819 Spell-checked document. 821 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf- 822 rtcweb-overview-00 824 Changed draft name and document date. 826 Removed unused references 828 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 830 Added architecture figures to section 2. 832 Changed the description of "echo cancellation" under "local system 833 support functions". 835 Added a few more definitions. 837 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 839 Added pointers to use cases, security and rtp-usage drafts (now WG 840 drafts). 842 Changed description of SRTP from mandatory-to-use to mandatory-to- 843 implement. 845 Added the "3 principles of negotiation" to the connection management 846 section. 848 Added an explicit statement that ICE is required for both NAT and 849 consent-to-receive. 851 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 853 Added references to a number of new drafts. 855 Expanded the description text under the "trapezoid" drawing with some 856 more text discussed on the list. 858 Changed the "Connection management" sentence from "will be done using 859 SDP offer/answer" to "will be capable of representing SDP offer/ 860 answer" - this seems more consistent with JSEP. 862 Added "security mechanisms" to the things a non-gatewayed SIP devices 863 must support in order to not need a media gateway. 865 Added a definition for "browser". 867 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 869 Made introduction more normative. 871 Several wording changes in response to review comments from EKR 873 Added an appendix to hold references and notes that are not yet in a 874 separate document. 876 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 878 Minor grammatical fixes. This is mainly a "keepalive" refresh. 880 A.10. Changes from -05 to -06 882 Clarifications in response to Last Call review comments. Inserted 883 reference to draft-ietf-rtcweb-audio. 885 A.11. Changes from -06 to -07 887 Added a reference to the "unified plan" draft, and updated some 888 references. 890 Otherwise, it's a "keepalive" draft. 892 A.12. Changes from -07 to -08 894 Removed the appendix that detailed transports, and replaced it with a 895 reference to draft-ietf-rtcweb-transports. Removed now-unused 896 references. 898 A.13. Changes from -08 to -09 900 Added text to the Abstract indicating that the intended status is an 901 Applicability Statement. 903 A.14. Changes from -09 to -10 905 Defined "WebRTC Browser" and "WebRTC device" as things that do, or 906 don't, conform to the API. 908 Updated reference to data-protocol draft 910 Updated data formats to reference -rtcweb-audio- and not the expired 911 -cbran draft. 913 Deleted references to -unified-plan 915 Deleted reference to -generic-idp (draft expired) 917 Added notes on which referenced documents WebRTC browsers or devices 918 MUST conform to. 920 Added pointer to the security section of the API drafts. 922 Author's Address 924 Harald T. Alvestrand 925 Google 926 Kungsbron 2 927 Stockholm 11122 928 Sweden 930 Email: harald@alvestrand.no