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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 November 28, 2014 5 Expires: June 1, 2015 7 Overview: Real Time Protocols for Browser-based Applications 8 draft-ietf-rtcweb-overview-13 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 WebRTC 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 June 1, 2015. 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 . . . . . . . . . . . 6 66 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 7 67 3. Architecture and Functionality groups . . . . . . . . . . . . 8 68 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . 12 69 5. Data framing and securing . . . . . . . . . . . . . . . . . . 12 70 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . 13 71 7. Connection management . . . . . . . . . . . . . . . . . . . . 13 72 8. Presentation and control . . . . . . . . . . . . . . . . . . 14 73 9. Local system support functions . . . . . . . . . . . . . . . 14 74 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 75 11. Security Considerations . . . . . . . . . . . . . . . . . . . 15 76 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 16 77 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 16 78 13.1. Normative References . . . . . . . . . . . . . . . . . . 16 79 13.2. Informative References . . . . . . . . . . . . . . . . . 18 80 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 18 81 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 82 to -01 . . . . . . . . . . . . . . . . . . . . . . . . . 18 83 A.2. Changes from draft-alvestrand-dispatch-01 to draft- 84 alvestrand-rtcweb-overview-00 . . . . . . . . . . . . . . 19 85 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . 19 86 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to 87 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 19 88 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 19 89 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 20 90 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 20 91 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 20 92 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 20 93 A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 20 94 A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 21 95 A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 21 96 A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 21 97 A.14. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 21 98 A.15. Changes from -10 to -11 . . . . . . . . . . . . . . . . . 21 99 A.16. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 22 100 A.17. Changes from -12 to -13 . . . . . . . . . . . . . . . . . 22 101 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 22 103 1. Introduction 105 The Internet was, from very early in its lifetime, considered a 106 possible vehicle for the deployment of real-time, interactive 107 applications - with the most easily imaginable being audio 108 conversations (aka "Internet telephony") and video conferencing. 110 The first attempts to build this were dependent on special networks, 111 special hardware and custom-built software, often at very high prices 112 or at low quality, placing great demands on the infrastructure. 114 As the available bandwidth has increased, and as processors an other 115 hardware has become ever faster, the barriers to participation have 116 decreased, and it has become possible to deliver a satisfactory 117 experience on commonly available computing hardware. 119 Still, there are a number of barriers to the ability to communicate 120 universally - one of these is that there is, as of yet, no single set 121 of communication protocols that all agree should be made available 122 for communication; another is the sheer lack of universal 123 identification systems (such as is served by telephone numbers or 124 email addresses in other communications systems). 126 Development of The Universal Solution has proved hard, however, for 127 all the usual reasons. 129 The last few years have also seen a new platform rise for deployment 130 of services: The browser-embedded application, or "Web application". 131 It turns out that as long as the browser platform has the necessary 132 interfaces, it is possible to deliver almost any kind of service on 133 it. 135 Traditionally, these interfaces have been delivered by plugins, which 136 had to be downloaded and installed separately from the browser; in 137 the development of HTML5, application developers see much promise in 138 the possibility of making those interfaces available in a 139 standardized way within the browser. 141 This memo describes a set of building blocks that can be made 142 accessible and controllable through a Javascript API in a browser, 143 and which together form a sufficient set of functions to allow the 144 use of interactive audio and video in applications that communicate 145 directly between browsers across the Internet. The resulting 146 protocol suite is intended to enable all the applications that are 147 described as required scenarios in the use cases document 148 [I-D.ietf-rtcweb-use-cases-and-requirements]. 150 Other efforts, for instance the W3C WEBRTC, Web Applications and 151 Device API working groups, focus on making standardized APIs and 152 interfaces available, within or alongside the HTML5 effort, for those 153 functions; this memo concentrates on specifying the protocols and 154 subprotocols that are needed to specify the interactions that happen 155 across the network. 157 This memo uses the term "WebRTC" (note the case used) to refer to the 158 overall effort consisting of both IETF and W3C efforts. 160 2. Principles and Terminology 162 2.1. Goals of this document 164 The goal of the WebRTC protocol specification is to specify a set of 165 protocols that, if all are implemented, will allow an implementation 166 to communicate with another implementation using audio, video and 167 data sent along the most direct possible path between the 168 participants. 170 This document is intended to serve as the roadmap to the WebRTC 171 specifications. It defines terms used by other pieces of 172 specification, lists references to other specifications that don't 173 need further elaboration in the WebRTC context, and gives pointers to 174 other documents that form part of the WebRTC suite. 176 By reading this document and the documents it refers to, it should be 177 possible to have all information needed to implement an WebRTC 178 compatible implementation. 180 2.2. Relationship between API and protocol 182 The total WebRTC effort consists of two pieces: 184 o A protocol specification, done in the IETF 186 o A Javascript API specification, done in the W3C 187 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 189 Together, these two specifications aim to provide an environment 190 where Javascript embedded in any page, viewed in any compatible 191 browser, when suitably authorized by its user, is able to set up 192 communication using audio, video and auxiliary data, where the 193 browser environment does not constrain the types of application in 194 which this functionality can be used. 196 The protocol specification does not assume that all implementations 197 implement this API; it is not intended to be necessary for 198 interoperation to know whether the entity one is communicating with 199 is a browser or another device implementing this specification. 201 The goal of cooperation between the protocol specification and the 202 API specification is that for all options and features of the 203 protocol specification, it should be clear which API calls to make to 204 exercise that option or feature; similarly, for any sequence of API 205 calls, it should be clear which protocol options and features will be 206 invoked. Both subject to constraints of the implementation, of 207 course. 209 For the purpose of this document, we define the following terminology 210 to talk about WebRTC things: 212 o A WebRTC browser (also called a WebRTC User Agent or WebRTC UA) is 213 something that conforms to both the protocol specification and the 214 Javascript API defined above. 216 o A WebRTC non-browser is something that conforms to the protocol 217 specification, but does not claim to implement the Javascript API. 218 This can also be called a "WebRTC device" or "WebRTC native 219 application". 221 o A WebRTC endpoint is either a WebRTC browser or a WebRTC non- 222 browser. It conforms to the protocol specification. 224 o A WebRTC-compatible endpoint is an endpoint that is able to 225 successfully communicate with a WebRTC endpoint, but may fail to 226 meet some requirements of a WebRTC endpoint. This may limit where 227 in the network such an endpoint can be attached, or may limit the 228 security guarantees that it offers to others. It is not 229 constrained by this specification; when it is mentioned at all, it 230 is to note the implications on WebRTC-compatible endpoints of the 231 requirements placed on WebRTC endpoints. 233 o A WebRTC gateway is a WebRTC-compatible endpoint that mediates 234 media traffic to non-WebRTC entities. 236 All WebRTC browsers are WebRTC endpoints, so any requirement on a 237 WebRTC endpoint also applies to a WebRTC browser. 239 A WebRTC non-browser may be capable of hosting applications in a 240 similar way to the way in which a browser can host Javascript 241 applications, typically by offering APIs in other languages. For 242 instance it may be implemented as a library that offers a C++ API 243 intended to be loaded into applications. In this case, similar 244 security considerations as for Javascript may be needed; however, 245 since such APIs are not defined or referenced here, this document 246 cannot give any specific rules for those interfaces. 248 WebRTC gateways are described in a separate document, 249 [I-D.alvestrand-rtcweb-gateways]. 251 2.3. On interoperability and innovation 253 The "Mission statement of the IETF" [RFC3935] states that "The 254 benefit of a standard to the Internet is in interoperability - that 255 multiple products implementing a standard are able to work together 256 in order to deliver valuable functions to the Internet's users." 258 Communication on the Internet frequently occurs in two phases: 260 o Two parties communicate, through some mechanism, what 261 functionality they both are able to support 263 o They use that shared communicative functionality to communicate, 264 or, failing to find anything in common, give up on communication. 266 There are often many choices that can be made for communicative 267 functionality; the history of the Internet is rife with the proposal, 268 standardization, implementation, and success or failure of many types 269 of options, in all sorts of protocols. 271 The goal of having a mandatory to implement function set is to 272 prevent negotiation failure, not to preempt or prevent negotiation. 274 The presence of a mandatory to implement function set serves as a 275 strong changer of the marketplace of deployment - in that it gives a 276 guarantee that, as long as you conform to a specification, and the 277 other party is willing to accept communication at the base level of 278 that specification, you can communicate successfully. 280 The alternative - that of having no mandatory to implement - does not 281 mean that you cannot communicate, it merely means that in order to be 282 part of the communications partnership, you have to implement the 283 standard "and then some" - that "and then some" usually being called 284 a profile of some sort; in the version most antithetical to the 285 Internet ethos, that "and then some" consists of having to use a 286 specific vendor's product only. 288 2.4. Terminology 290 The following terms are used across the documents specifying the 291 WebRTC suite, in the specific meanings given here. Not all terms are 292 used in this document. Other terms are used in their commonly used 293 meaning. 295 The list is in alphabetical order. 297 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 299 API: Application Programming Interface - a specification of a set of 300 calls and events, usually tied to a programming language or an 301 abstract formal specification such as WebIDL, with its defined 302 semantics. 304 Browser: Used synonymously with "Interactive User Agent" as defined 305 in the HTML specification [W3C.WD-html5-20110525]. See also 306 "WebRTC User Agent". 308 ICE Agent: An implementation of the Interactive Connectivty 309 Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be 310 an SDP Agent, but there exist ICE Agents that do not use SDP (for 311 instance those that use Jingle). 313 Interactive: Communication between multiple parties, where the 314 expectation is that an action from one party can cause a reaction 315 by another party, and the reaction can be observed by the first 316 party, with the total time required for the action/reaction/ 317 observation is on the order of no more than hundreds of 318 milliseconds. 320 Media: Audio and video content. Not to be confused with 321 "transmission media" such as wires. 323 Media path: The path that media data follows from one WebRTC 324 endpoint to another. 326 Protocol: A specification of a set of data units, their 327 representation, and rules for their transmission, with their 328 defined semantics. A protocol is usually thought of as going 329 between systems. 331 Real-time media: Media where generation of content and display of 332 content are intended to occur closely together in time (on the 333 order of no more than hundreds of milliseconds). Real-time media 334 can be used to support interactive communication. 336 SDP Agent: The protocol implementation involved in the SDP offer/ 337 answer exchange, as defined in [RFC3264] section 3. 339 Signaling: Communication that happens in order to establish, manage 340 and control media paths. 342 Signaling Path: The communication channels used between entities 343 participating in signaling to transfer signaling. There may be 344 more entities in the signaling path than in the media path. 346 NOTE: Where common definitions exist for these terms, those 347 definitions should be used to the greatest extent possible. 349 3. Architecture and Functionality groups 351 The model of real-time support for browser-based applications does 352 not assume that the browser will contain all the functions that need 353 to be performed in order to have a function such as a telephone or a 354 video conferencing unit; the vision is that the browser will have the 355 functions that are needed for a Web application, working in 356 conjunction with its backend servers, to implement these functions. 358 This means that two vital interfaces need specification: The 359 protocols that browsers talk to each other, without any intervening 360 servers, and the APIs that are offered for a Javascript application 361 to take advantage of the browser's functionality. 363 +------------------------+ On-the-wire 364 | | Protocols 365 | Servers |---------> 366 | | 367 | | 368 +------------------------+ 369 ^ 370 | 371 | 372 | HTTP/ 373 | Websockets 374 | 375 | 376 +----------------------------+ 377 | Javascript/HTML/CSS | 378 +----------------------------+ 379 Other ^ ^RTC 380 APIs | |APIs 381 +---|-----------------|------+ 382 | | | | 383 | +---------+| 384 | | Browser || On-the-wire 385 | Browser | RTC || Protocols 386 | | Function|-----------> 387 | | || 388 | | || 389 | +---------+| 390 +---------------------|------+ 391 | 392 V 393 Native OS Services 395 Figure 1: Browser Model 397 Note that HTTP and Websockets are also offered to the Javascript 398 application through browser APIs. 400 As for all protocol and API specifications, there is no restriction 401 that the protocols can only be used to talk to another browser; since 402 they are fully specified, any endpoint that implements the protocols 403 faithfully should be able to interoperate with the application 404 running in the browser. 406 A commonly imagined model of deployment is the one depicted below. 408 +-----------+ +-----------+ 409 | Web | | Web | 410 | | Signaling | | 411 | |-------------| | 412 | Server | path | Server | 413 | | | | 414 +-----------+ +-----------+ 415 / \ 416 / \ Application-defined 417 / \ over 418 / \ HTTP/Websockets 419 / Application-defined over \ 420 / HTTP/Websockets \ 421 / \ 422 +-----------+ +-----------+ 423 |JS/HTML/CSS| |JS/HTML/CSS| 424 +-----------+ +-----------+ 425 +-----------+ +-----------+ 426 | | | | 427 | | | | 428 | Browser | ------------------------- | Browser | 429 | | Media path | | 430 | | | | 431 +-----------+ +-----------+ 433 Figure 2: Browser RTC Trapezoid 435 On this drawing, the critical part to note is that the media path 436 ("low path") goes directly between the browsers, so it has to be 437 conformant to the specifications of the WebRTC protocol suite; the 438 signaling path ("high path") goes via servers that can modify, 439 translate or massage the signals as needed. 441 If the two Web servers are operated by different entities, the inter- 442 server signaling mechanism needs to be agreed upon, either by 443 standardization or by other means of agreement. Existing protocols 444 (for example SIP [RFC3261] or XMPP [RFC6120]) could be used between 445 servers, while either a standards-based or proprietary protocol could 446 be used between the browser and the web server. 448 For example, if both operators' servers implement SIP, SIP could be 449 used for communication between servers, along with either a 450 standardized signaling mechanism (e.g. SIP over Websockets) or a 451 proprietary signaling mechanism used between the application running 452 in the browser and the web server. Similarly, if both operators' 453 servers implement XMPP, XMPP could be used for communication between 454 XMPP servers, with either a standardized signaling mechanism (e.g. 455 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 456 used between the application running in the browser and the web 457 server. 459 The choice of protocols, and definition of the translation between 460 them, is outside the scope of the WebRTC protocol suite described in 461 the document. 463 The functionality groups that are needed in the browser can be 464 specified, more or less from the bottom up, as: 466 o Data transport: TCP, UDP and the means to securely set up 467 connections between entities, as well as the functions for 468 deciding when to send data: Congestion management, bandwidth 469 estimation and so on. 471 o Data framing: RTP and other data formats that serve as containers, 472 and their functions for data confidentiality and integrity. 474 o Data formats: Codec specifications, format specifications and 475 functionality specifications for the data passed between systems. 476 Audio and video codecs, as well as formats for data and document 477 sharing, belong in this category. In order to make use of data 478 formats, a way to describe them, a session description, is needed. 480 o Connection management: Setting up connections, agreeing on data 481 formats, changing data formats during the duration of a call; SIP 482 and Jingle/XMPP belong in this category. 484 o Presentation and control: What needs to happen in order to ensure 485 that interactions behave in a non-surprising manner. This can 486 include floor control, screen layout, voice activated image 487 switching and other such functions - where part of the system 488 require the cooperation between parties. XCON and Cisco/ 489 Tandberg's TIP were some attempts at specifying this kind of 490 functionality; many applications have been built without 491 standardized interfaces to these functions. 493 o Local system support functions: These are things that need not be 494 specified uniformly, because each participant may choose to do 495 these in a way of the participant's choosing, without affecting 496 the bits on the wire in a way that others have to be cognizant of. 497 Examples in this category include echo cancellation (some forms of 498 it), local authentication and authorization mechanisms, OS access 499 control and the ability to do local recording of conversations. 501 Within each functionality group, it is important to preserve both 502 freedom to innovate and the ability for global communication. 503 Freedom to innovate is helped by doing the specification in terms of 504 interfaces, not implementation; any implementation able to 505 communicate according to the interfaces is a valid implementation. 506 Ability to communicate globally is helped both by having core 507 specifications be unencumbered by IPR issues and by having the 508 formats and protocols be fully enough specified to allow for 509 independent implementation. 511 One can think of the three first groups as forming a "media transport 512 infrastructure", and of the three last groups as forming a "media 513 service". In many contexts, it makes sense to use a common 514 specification for the media transport infrastructure, which can be 515 embedded in browsers and accessed using standard interfaces, and "let 516 a thousand flowers bloom" in the "media service" layer; to achieve 517 interoperable services, however, at least the first five of the six 518 groups need to be specified. 520 4. Data transport 522 Data transport refers to the sending and receiving of data over the 523 network interfaces, the choice of network-layer addresses at each end 524 of the communication, and the interaction with any intermediate 525 entities that handle the data, but do not modify it (such as TURN 526 relays). 528 It includes necessary functions for congestion control: When not to 529 send data. 531 WebRTC endpoints MUST implement the transport protocols described in 532 [I-D.ietf-rtcweb-transports]. 534 5. Data framing and securing 536 The format for media transport is RTP [RFC3550]. Implementation of 537 SRTP [RFC3711] is REQUIRED for all implementations. 539 The detailed considerations for usage of functions from RTP and SRTP 540 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 541 considerations for the WebRTC use case are in 542 [I-D.ietf-rtcweb-security], and the resulting security functions are 543 described in [I-D.ietf-rtcweb-security-arch]. 545 Considerations for the transfer of data that is not in RTP format is 546 described in [I-D.ietf-rtcweb-data-channel], and a supporting 547 protocol for establishing individual data channels is described in 548 [I-D.ietf-rtcweb-data-protocol]. WebRTC endpoints MUST implement 549 these two specifications. 551 WebRTC endpoints MUST implement [I-D.ietf-rtcweb-rtp-usage], 552 [I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the 553 requirements they include. 555 6. Data formats 557 The intent of this specification is to allow each communications 558 event to use the data formats that are best suited for that 559 particular instance, where a format is supported by both sides of the 560 connection. However, a minimum standard is greatly helpful in order 561 to ensure that communication can be achieved. This document 562 specifies a minimum baseline that will be supported by all 563 implementations of this specification, and leaves further codecs to 564 be included at the will of the implementor. 566 WebRTC endpoints that support audio and/or video MUST implement the 567 codecs and profiles required in [I-D.ietf-rtcweb-audio] and 568 [I-D.ietf-rtcweb-video]. 570 7. Connection management 572 The methods, mechanisms and requirements for setting up, negotiating 573 and tearing down connections is a large subject, and one where it is 574 desirable to have both interoperability and freedom to innovate. 576 The following principles apply: 578 1. The WebRTC media negotiations will be capable of representing the 579 same SDP offer/answer semantics that are used in SIP [RFC3264], 580 in such a way that it is possible to build a signaling gateway 581 between SIP and the WebRTC media negotiation. 583 2. It will be possible to gateway between legacy SIP devices that 584 support ICE and appropriate RTP / SDP mechanisms, codecs and 585 security mechanisms without using a media gateway. A signaling 586 gateway to convert between the signaling on the web side to the 587 SIP signaling may be needed. 589 3. When a new codec is specified, and the SDP for the new codec is 590 specified in the MMUSIC WG, no other standardization should be 591 required for it to be possible to use that in the web browsers. 592 Adding new codecs which might have new SDP parameters should not 593 change the APIs between the browser and Javascript application. 594 As soon as the browsers support the new codecs, old applications 595 written before the codecs were specified should automatically be 596 able to use the new codecs where appropriate with no changes to 597 the JS applications. 599 The particular choices made for WebRTC, and their implications for 600 the API offered by a browser implementing WebRTC, are described in 601 [I-D.ietf-rtcweb-jsep]. 603 WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. 605 WebRTC endpoints MUST implement the functions described in that 606 document that relate to the network layer (for example Bundle, RTCP- 607 mux and Trickle ICE), but do not need to support the API 608 functionality described there. 610 8. Presentation and control 612 The most important part of control is the user's control over the 613 browser's interaction with input/output devices and communications 614 channels. It is important that the user have some way of figuring 615 out where his audio, video or texting is being sent, for what 616 purported reason, and what guarantees are made by the parties that 617 form part of this control channel. This is largely a local function 618 between the browser, the underlying operating system and the user 619 interface; this is specified in the peer connection API 620 [W3C.WD-webrtc-20120209], and the media capture API 621 [W3C.WD-mediacapture-streams-20120628]. 623 WebRTC browsers MUST implement these two specifications. 625 9. Local system support functions 627 These are characterized by the fact that the quality of these 628 functions strongly influence the user experience, but the exact 629 algorithm does not need coordination. In some cases (for instance 630 echo cancellation, as described below), the overall system definition 631 may need to specify that the overall system needs to have some 632 characteristics for which these facilities are useful, without 633 requiring them to be implemented a certain way. 635 Local functions include echo cancellation, volume control, camera 636 management including focus, zoom, pan/tilt controls (if available), 637 and more. 639 Certain parts of the system SHOULD conform to certain properties, for 640 instance: 642 o Echo cancellation should be good enough to achieve the suppression 643 of acoustical feedback loops below a perceptually noticeable 644 level. 646 o Privacy concerns MUST be satisfied; for instance, if remote 647 control of camera is offered, the APIs should be available to let 648 the local participant figure out who's controlling the camera, and 649 possibly decide to revoke the permission for camera usage. 651 o Automatic gain control, if present, should normalize a speaking 652 voice into a reasonable dB range. 654 The requirements on WebRTC systems with regard to audio processing 655 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 656 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 658 WebRTC endpoints MUST implement the processing functions in 659 [I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6, 660 this means that WebRTC endpoints MUST implement the whole document.) 662 10. IANA Considerations 664 This document makes no request of IANA. 666 Note to RFC Editor: this section may be removed on publication as an 667 RFC. 669 11. Security Considerations 671 Security of the web-enabled real time communications comes in several 672 pieces: 674 o Security of the components: The browsers, and other servers 675 involved. The most target-rich environment here is probably the 676 browser; the aim here should be that the introduction of these 677 components introduces no additional vulnerability. 679 o Security of the communication channels: It should be easy for a 680 participant to reassure himself of the security of his 681 communication - by verifying the crypto parameters of the links he 682 himself participates in, and to get reassurances from the other 683 parties to the communication that they promise that appropriate 684 measures are taken. 686 o Security of the partners' identity: verifying that the 687 participants are who they say they are (when positive 688 identification is appropriate), or that their identity cannot be 689 uncovered (when anonymity is a goal of the application). 691 The security analysis, and the requirements derived from that 692 analysis, is contained in [I-D.ietf-rtcweb-security]. 694 It is also important to read the security sections of 695 [W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209]. 697 12. Acknowledgements 699 The number of people who have taken part in the discussions 700 surrounding this draft are too numerous to list, or even to identify. 701 The ones below have made special, identifiable contributions; this 702 does not mean that others' contributions are less important. 704 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 705 Westerlund and Joerg Ott, who offered technical contributions on 706 various versions of the draft. 708 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 709 the ASCII drawings in section 1. 711 Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric 712 Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage 713 and Simon Leinen for document review. 715 13. References 717 13.1. Normative References 719 [I-D.ietf-rtcweb-audio] 720 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 721 Requirements", draft-ietf-rtcweb-audio-05 (work in 722 progress), February 2014. 724 [I-D.ietf-rtcweb-data-channel] 725 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 726 Channels", draft-ietf-rtcweb-data-channel-11 (work in 727 progress), July 2014. 729 [I-D.ietf-rtcweb-data-protocol] 730 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 731 Establishment Protocol", draft-ietf-rtcweb-data- 732 protocol-07 (work in progress), July 2014. 734 [I-D.ietf-rtcweb-jsep] 735 Uberti, J., Jennings, C., and E. Rescorla, "Javascript 736 Session Establishment Protocol", draft-ietf-rtcweb-jsep-07 737 (work in progress), July 2014. 739 [I-D.ietf-rtcweb-rtp-usage] 740 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 741 Communication (WebRTC): Media Transport and Use of RTP", 742 draft-ietf-rtcweb-rtp-usage-16 (work in progress), July 743 2014. 745 [I-D.ietf-rtcweb-security] 746 Rescorla, E., "Security Considerations for WebRTC", draft- 747 ietf-rtcweb-security-07 (work in progress), July 2014. 749 [I-D.ietf-rtcweb-security-arch] 750 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 751 rtcweb-security-arch-10 (work in progress), July 2014. 753 [I-D.ietf-rtcweb-transports] 754 Alvestrand, H., "Transports for WebRTC", draft-ietf- 755 rtcweb-transports-06 (work in progress), August 2014. 757 [I-D.ietf-rtcweb-video] 758 Roach, A., "WebRTC Video Processing and Codec 759 Requirements", draft-ietf-rtcweb-video-00 (work in 760 progress), July 2014. 762 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 763 with Session Description Protocol (SDP)", RFC 3264, June 764 2002. 766 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 767 Jacobson, "RTP: A Transport Protocol for Real-Time 768 Applications", STD 64, RFC 3550, July 2003. 770 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 771 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 772 RFC 3711, March 2004. 774 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 775 (ICE): A Protocol for Network Address Translator (NAT) 776 Traversal for Offer/Answer Protocols", RFC 5245, April 777 2010. 779 [W3C.WD-mediacapture-streams-20120628] 780 Burnett, D. and A. Narayanan, "Media Capture and Streams", 781 World Wide Web Consortium WD WD-mediacapture-streams- 782 20120628, June 2012, . 785 [W3C.WD-webrtc-20120209] 786 Bergkvist, A., Burnett, D., Jennings, C., and A. 787 Narayanan, "WebRTC 1.0: Real-time Communication Between 788 Browsers", World Wide Web Consortium WD WD-webrtc- 789 20120209, February 2012, 790 . 792 13.2. Informative References 794 [I-D.alvestrand-rtcweb-gateways] 795 Alvestrand, H., "WebRTC Gateways", draft-alvestrand- 796 rtcweb-gateways-00 (work in progress), August 2014. 798 [I-D.ietf-rtcweb-use-cases-and-requirements] 799 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 800 Time Communication Use-cases and Requirements", draft- 801 ietf-rtcweb-use-cases-and-requirements-14 (work in 802 progress), February 2014. 804 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 805 A., Peterson, J., Sparks, R., Handley, M., and E. 806 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 807 June 2002. 809 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", BCP 810 95, RFC 3935, October 2004. 812 [RFC6120] Saint-Andre, P., "Extensible Messaging and Presence 813 Protocol (XMPP): Core", RFC 6120, March 2011. 815 [W3C.WD-html5-20110525] 816 Hickson, I., "HTML5", World Wide Web Consortium LastCall 817 WD-html5-20110525, May 2011, 818 . 820 Appendix A. Change log 822 This section may be deleted by the RFC Editor when preparing for 823 publication. 825 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 827 Added section "On interoperability and innovation" 829 Added data confidentiality and integrity to the "data framing" layer 831 Added congestion management requirements in the "data transport" 832 layer section 833 Changed need for non-media data from "question: do we need this?" to 834 "Open issue: How do we do this?" 836 Strengthened disclaimer that listed codecs are placeholders, not 837 decisions. 839 More details on why the "local system support functions" section is 840 there. 842 A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand- 843 rtcweb-overview-00 845 Added section on "Relationship between API and protocol" 847 Added terminology section 849 Mentioned congestion management as part of the "data transport" layer 850 in the layer list 852 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 854 Removed most technical content, and replaced with pointers to drafts 855 as requested and identified by the RTCWEB WG chairs. 857 Added content to acknowledgments section. 859 Added change log. 861 Spell-checked document. 863 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf- 864 rtcweb-overview-00 866 Changed draft name and document date. 868 Removed unused references 870 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 872 Added architecture figures to section 2. 874 Changed the description of "echo cancellation" under "local system 875 support functions". 877 Added a few more definitions. 879 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 881 Added pointers to use cases, security and rtp-usage drafts (now WG 882 drafts). 884 Changed description of SRTP from mandatory-to-use to mandatory-to- 885 implement. 887 Added the "3 principles of negotiation" to the connection management 888 section. 890 Added an explicit statement that ICE is required for both NAT and 891 consent-to-receive. 893 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 895 Added references to a number of new drafts. 897 Expanded the description text under the "trapezoid" drawing with some 898 more text discussed on the list. 900 Changed the "Connection management" sentence from "will be done using 901 SDP offer/answer" to "will be capable of representing SDP offer/ 902 answer" - this seems more consistent with JSEP. 904 Added "security mechanisms" to the things a non-gatewayed SIP devices 905 must support in order to not need a media gateway. 907 Added a definition for "browser". 909 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 911 Made introduction more normative. 913 Several wording changes in response to review comments from EKR 915 Added an appendix to hold references and notes that are not yet in a 916 separate document. 918 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 920 Minor grammatical fixes. This is mainly a "keepalive" refresh. 922 A.10. Changes from -05 to -06 924 Clarifications in response to Last Call review comments. Inserted 925 reference to draft-ietf-rtcweb-audio. 927 A.11. Changes from -06 to -07 929 Added a reference to the "unified plan" draft, and updated some 930 references. 932 Otherwise, it's a "keepalive" draft. 934 A.12. Changes from -07 to -08 936 Removed the appendix that detailed transports, and replaced it with a 937 reference to draft-ietf-rtcweb-transports. Removed now-unused 938 references. 940 A.13. Changes from -08 to -09 942 Added text to the Abstract indicating that the intended status is an 943 Applicability Statement. 945 A.14. Changes from -09 to -10 947 Defined "WebRTC Browser" and "WebRTC device" as things that do, or 948 don't, conform to the API. 950 Updated reference to data-protocol draft 952 Updated data formats to reference -rtcweb-audio- and not the expired 953 -cbran draft. 955 Deleted references to -unified-plan 957 Deleted reference to -generic-idp (draft expired) 959 Added notes on which referenced documents WebRTC browsers or devices 960 MUST conform to. 962 Added pointer to the security section of the API drafts. 964 A.15. Changes from -10 to -11 966 Added "WebRTC Gateway" as a third class of device, and referenced the 967 doc describing them. 969 Made a number of text clarifications in response to document reviews. 971 A.16. Changes from -11 to -12 973 Refined entity definitions to define "WebRTC endpoint" and "WebRTC- 974 compatible endpoint". 976 Changed remaining usage of the term "RTCWEB" to "WebRTC", including 977 in the page header. 979 A.17. Changes from -12 to -13 981 Changed "WebRTC device" to be "WebRTC non-browser", per decision at 982 IETF 91. This led to the need for "WebRTC endpoint" as the common 983 label for both, and the usage of that term in the rest of the 984 document. 986 Added words about WebRTC APIs in languages other than Javascript. 988 Referenced draft-ietf-rtcweb-video for video codecs to support. 990 Author's Address 992 Harald T. Alvestrand 993 Google 994 Kungsbron 2 995 Stockholm 11122 996 Sweden 998 Email: harald@alvestrand.no