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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 October 13, 2014 5 Expires: April 16, 2015 7 Overview: Real Time Protocols for Browser-based Applications 8 draft-ietf-rtcweb-overview-12 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 April 16, 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 . . . . . . . . . . . 5 66 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 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 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 22 102 1. Introduction 104 The Internet was, from very early in its lifetime, considered a 105 possible vehicle for the deployment of real-time, interactive 106 applications - with the most easily imaginable being audio 107 conversations (aka "Internet telephony") and video conferencing. 109 The first attempts to build this were dependent on special networks, 110 special hardware and custom-built software, often at very high prices 111 or at low quality, placing great demands on the infrastructure. 113 As the available bandwidth has increased, and as processors an other 114 hardware has become ever faster, the barriers to participation have 115 decreased, and it has become possible to deliver a satisfactory 116 experience on commonly available computing hardware. 118 Still, there are a number of barriers to the ability to communicate 119 universally - one of these is that there is, as of yet, no single set 120 of communication protocols that all agree should be made available 121 for communication; another is the sheer lack of universal 122 identification systems (such as is served by telephone numbers or 123 email addresses in other communications systems). 125 Development of The Universal Solution has proved hard, however, for 126 all the usual reasons. 128 The last few years have also seen a new platform rise for deployment 129 of services: The browser-embedded application, or "Web application". 130 It turns out that as long as the browser platform has the necessary 131 interfaces, it is possible to deliver almost any kind of service on 132 it. 134 Traditionally, these interfaces have been delivered by plugins, which 135 had to be downloaded and installed separately from the browser; in 136 the development of HTML5, application developers see much promise in 137 the possibility of making those interfaces available in a 138 standardized way within the browser. 140 This memo describes a set of building blocks that can be made 141 accessible and controllable through a Javascript API in a browser, 142 and which together form a sufficient set of functions to allow the 143 use of interactive audio and video in applications that communicate 144 directly between browsers across the Internet. The resulting 145 protocol suite is intended to enable all the applications that are 146 described as required scenarios in the use cases document 147 [I-D.ietf-rtcweb-use-cases-and-requirements]. 149 Other efforts, for instance the W3C WEBRTC, Web Applications and 150 Device API working groups, focus on making standardized APIs and 151 interfaces available, within or alongside the HTML5 effort, for those 152 functions; this memo concentrates on specifying the protocols and 153 subprotocols that are needed to specify the interactions that happen 154 across the network. 156 This memo uses the term "WebRTC" (note the case used) to refer to the 157 overall effort consisting of both IETF and W3C efforts. 159 2. Principles and Terminology 161 2.1. Goals of this document 163 The goal of the WebRTC protocol specification is to specify a set of 164 protocols that, if all are implemented, will allow an implementation 165 to communicate with another implementation using audio, video and 166 data sent along the most direct possible path between the 167 participants. 169 This document is intended to serve as the roadmap to the WebRTC 170 specifications. It defines terms used by other pieces of 171 specification, lists references to other specifications that don't 172 need further elaboration in the WebRTC context, and gives pointers to 173 other documents that form part of the WebRTC suite. 175 By reading this document and the documents it refers to, it should be 176 possible to have all information needed to implement an RTCWEB 177 compatible implementation. 179 2.2. Relationship between API and protocol 181 The total WebRTC effort consists of two pieces: 183 o A protocol specification, done in the IETF 185 o A Javascript API specification, done in the W3C 186 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 188 Together, these two specifications aim to provide an environment 189 where Javascript embedded in any page, viewed in any compatible 190 browser, when suitably authorized by its user, is able to set up 191 communication using audio, video and auxiliary data, where the 192 browser environment does not constrain the types of application in 193 which this functionality can be used. 195 The protocol specification does not assume that all implementations 196 implement this API; it is not intended to be necessary for 197 interoperation to know whether the entity one is communicating with 198 is a browser or another device implementing this specification. 200 The goal of cooperation between the protocol specification and the 201 API specification is that for all options and features of the 202 protocol specification, it should be clear which API calls to make to 203 exercise that option or feature; similarly, for any sequence of API 204 calls, it should be clear which protocol options and features will be 205 invoked. Both subject to constraints of the implementation, of 206 course. 208 For the purpose of this document, five types of entities are defined: 210 o A WebRTC User Agent (also called a WebRTC UA or a WebRTC browser) 211 is something that conforms to both the protocol specification and 212 the Javascript API defined above. 214 o A WebRTC device is something that conforms to the protocol 215 specification, but does not claim to implement the Javascript API. 217 o A WebRTC endpoint is either a WebRTC User Agent or a WebRTC 218 device. 220 o A WebRTC-compatible endpoint is an endpoint that is capable of 221 successfully communicating with a WebRTC endpoint, but may fail to 222 meet some requirements of a WebRTC endpoint. This may limit where 223 in the network such an endpoint can be attached, or may limit the 224 security guarantees that it offers to others. 226 o A WebRTC gateway is a WebRTC-compatible endpoint that mediates 227 traffic to non-WebRTC entities. 229 All WebRTC browsers (UAs) are WebRTC devices, so any requirement on a 230 WebRTC device also applies to a WebRTC browser. 232 WebRTC gateways are described in a separate document 233 [I-D.alvestrand-rtcweb-gateways]. 235 2.3. On interoperability and innovation 237 The "Mission statement of the IETF" [RFC3935] states that "The 238 benefit of a standard to the Internet is in interoperability - that 239 multiple products implementing a standard are able to work together 240 in order to deliver valuable functions to the Internet's users." 242 Communication on the Internet frequently occurs in two phases: 244 o Two parties communicate, through some mechanism, what 245 functionality they both are able to support 247 o They use that shared communicative functionality to communicate, 248 or, failing to find anything in common, give up on communication. 250 There are often many choices that can be made for communicative 251 functionality; the history of the Internet is rife with the proposal, 252 standardization, implementation, and success or failure of many types 253 of options, in all sorts of protocols. 255 The goal of having a mandatory to implement function set is to 256 prevent negotiation failure, not to preempt or prevent negotiation. 258 The presence of a mandatory to implement function set serves as a 259 strong changer of the marketplace of deployment - in that it gives a 260 guarantee that, as long as you conform to a specification, and the 261 other party is willing to accept communication at the base level of 262 that specification, you can communicate successfully. 264 The alternative - that of having no mandatory to implement - does not 265 mean that you cannot communicate, it merely means that in order to be 266 part of the communications partnership, you have to implement the 267 standard "and then some" - that "and then some" usually being called 268 a profile of some sort; in the version most antithetical to the 269 Internet ethos, that "and then some" consists of having to use a 270 specific vendor's product only. 272 2.4. Terminology 274 The following terms are used across the documents specifying the 275 WebRTC suite, in the specific meanings given here. Not all terms are 276 used in this document. Other terms are used in their commonly used 277 meaning. 279 The list is in alphabetical order. 281 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 283 API: Application Programming Interface - a specification of a set of 284 calls and events, usually tied to a programming language or an 285 abstract formal specification such as WebIDL, with its defined 286 semantics. 288 Browser: Used synonymously with "Interactive User Agent" as defined 289 in the HTML specification [W3C.WD-html5-20110525]. See also 290 "WebRTC User Agent". 292 ICE Agent: An implementation of the Interactive Connectivty 293 Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be 294 an SDP Agent, but there exist ICE Agents that do not use SDP (for 295 instance those that use Jingle). 297 Interactive: Communication between multiple parties, where the 298 expectation is that an action from one party can cause a reaction 299 by another party, and the reaction can be observed by the first 300 party, with the total time required for the action/reaction/ 301 observation is on the order of no more than hundreds of 302 milliseconds. 304 Media: Audio and video content. Not to be confused with 305 "transmission media" such as wires. 307 Media path: The path that media data follows from one WebRTC device 308 to another. 310 Protocol: A specification of a set of data units, their 311 representation, and rules for their transmission, with their 312 defined semantics. A protocol is usually thought of as going 313 between systems. 315 Real-time media: Media where generation of content and display of 316 content are intended to occur closely together in time (on the 317 order of no more than hundreds of milliseconds). Real-time media 318 can be used to support interactive communication. 320 SDP Agent: The protocol implementation involved in the SDP offer/ 321 answer exchange, as defined in [RFC3264] section 3. 323 Signaling: Communication that happens in order to establish, manage 324 and control media paths. 326 Signaling Path: The communication channels used between entities 327 participating in signaling to transfer signaling. There may be 328 more entities in the signaling path than in the media path. 330 WebRTC User Agent: An entity that conforms to the WebRTC protocol 331 specifications and offer the WebRTC Javascript APIs. Also called 332 a WebRTC browser. 334 WebRTC Device: An unit (software, hardware or combinations) that 335 conforms to the WebRTC protocol specifications, but does not offer 336 the WebRTC Javascript APIs. 338 WebRTC Endpoint: Either a WebRTC browser or a WebRTC device. 340 NOTE: Where common definitions exist for these terms, those 341 definitions should be used to the greatest extent possible. 343 3. Architecture and Functionality groups 345 The model of real-time support for browser-based applications does 346 not assume that the browser will contain all the functions that need 347 to be performed in order to have a function such as a telephone or a 348 video conferencing unit; the vision is that the browser will have the 349 functions that are needed for a Web application, working in 350 conjunction with its backend servers, to implement these functions. 352 This means that two vital interfaces need specification: The 353 protocols that browsers talk to each other, without any intervening 354 servers, and the APIs that are offered for a Javascript application 355 to take advantage of the browser's functionality. 357 +------------------------+ On-the-wire 358 | | Protocols 359 | Servers |---------> 360 | | 361 | | 362 +------------------------+ 363 ^ 364 | 365 | 366 | HTTP/ 367 | Websockets 368 | 369 | 370 +----------------------------+ 371 | Javascript/HTML/CSS | 372 +----------------------------+ 373 Other ^ ^RTC 374 APIs | |APIs 375 +---|-----------------|------+ 376 | | | | 377 | +---------+| 378 | | Browser || On-the-wire 379 | Browser | RTC || Protocols 380 | | Function|-----------> 381 | | || 382 | | || 383 | +---------+| 384 +---------------------|------+ 385 | 386 V 387 Native OS Services 389 Figure 1: Browser Model 391 Note that HTTP and Websockets are also offered to the Javascript 392 application through browser APIs. 394 As for all protocol and API specifications, there is no restriction 395 that the protocols can only be used to talk to another browser; since 396 they are fully specified, any device that implements the protocols 397 faithfully should be able to interoperate with the application 398 running in the browser. 400 A commonly imagined model of deployment is the one depicted below. 402 +-----------+ +-----------+ 403 | Web | | Web | 404 | | Signaling | | 405 | |-------------| | 406 | Server | path | Server | 407 | | | | 408 +-----------+ +-----------+ 409 / \ 410 / \ Application-defined 411 / \ over 412 / \ HTTP/Websockets 413 / Application-defined over \ 414 / HTTP/Websockets \ 415 / \ 416 +-----------+ +-----------+ 417 |JS/HTML/CSS| |JS/HTML/CSS| 418 +-----------+ +-----------+ 419 +-----------+ +-----------+ 420 | | | | 421 | | | | 422 | Browser | ------------------------- | Browser | 423 | | Media path | | 424 | | | | 425 +-----------+ +-----------+ 427 Figure 2: Browser RTC Trapezoid 429 On this drawing, the critical part to note is that the media path 430 ("low path") goes directly between the browsers, so it has to be 431 conformant to the specifications of the WebRTC protocol suite; the 432 signaling path ("high path") goes via servers that can modify, 433 translate or massage the signals as needed. 435 If the two Web servers are operated by different entities, the inter- 436 server signaling mechanism needs to be agreed upon, either by 437 standardization or by other means of agreement. Existing protocols 438 (for example SIP [RFC3261] or XMPP [RFC6120]) could be used between 439 servers, while either a standards-based or proprietary protocol could 440 be used between the browser and the web server. 442 For example, if both operators' servers implement SIP, SIP could be 443 used for communication between servers, along with either a 444 standardized signaling mechanism (e.g. SIP over Websockets) or a 445 proprietary signaling mechanism used between the application running 446 in the browser and the web server. Similarly, if both operators' 447 servers implement XMPP, XMPP could be used for communication between 448 XMPP servers, with either a standardized signaling mechanism (e.g. 449 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 450 used between the application running in the browser and the web 451 server. 453 The choice of protocols, and definition of the translation between 454 them, is outside the scope of the WebRTC protocol suite described in 455 the document. 457 The functionality groups that are needed in the browser can be 458 specified, more or less from the bottom up, as: 460 o Data transport: TCP, UDP and the means to securely set up 461 connections between entities, as well as the functions for 462 deciding when to send data: Congestion management, bandwidth 463 estimation and so on. 465 o Data framing: RTP and other data formats that serve as containers, 466 and their functions for data confidentiality and integrity. 468 o Data formats: Codec specifications, format specifications and 469 functionality specifications for the data passed between systems. 470 Audio and video codecs, as well as formats for data and document 471 sharing, belong in this category. In order to make use of data 472 formats, a way to describe them, a session description, is needed. 474 o Connection management: Setting up connections, agreeing on data 475 formats, changing data formats during the duration of a call; SIP 476 and Jingle/XMPP belong in this category. 478 o Presentation and control: What needs to happen in order to ensure 479 that interactions behave in a non-surprising manner. This can 480 include floor control, screen layout, voice activated image 481 switching and other such functions - where part of the system 482 require the cooperation between parties. XCON and Cisco/ 483 Tandberg's TIP were some attempts at specifying this kind of 484 functionality; many applications have been built without 485 standardized interfaces to these functions. 487 o Local system support functions: These are things that need not be 488 specified uniformly, because each participant may choose to do 489 these in a way of the participant's choosing, without affecting 490 the bits on the wire in a way that others have to be cognizant of. 491 Examples in this category include echo cancellation (some forms of 492 it), local authentication and authorization mechanisms, OS access 493 control and the ability to do local recording of conversations. 495 Within each functionality group, it is important to preserve both 496 freedom to innovate and the ability for global communication. 497 Freedom to innovate is helped by doing the specification in terms of 498 interfaces, not implementation; any implementation able to 499 communicate according to the interfaces is a valid implementation. 500 Ability to communicate globally is helped both by having core 501 specifications be unencumbered by IPR issues and by having the 502 formats and protocols be fully enough specified to allow for 503 independent implementation. 505 One can think of the three first groups as forming a "media transport 506 infrastructure", and of the three last groups as forming a "media 507 service". In many contexts, it makes sense to use a common 508 specification for the media transport infrastructure, which can be 509 embedded in browsers and accessed using standard interfaces, and "let 510 a thousand flowers bloom" in the "media service" layer; to achieve 511 interoperable services, however, at least the first five of the six 512 groups need to be specified. 514 4. Data transport 516 Data transport refers to the sending and receiving of data over the 517 network interfaces, the choice of network-layer addresses at each end 518 of the communication, and the interaction with any intermediate 519 entities that handle the data, but do not modify it (such as TURN 520 relays). 522 It includes necessary functions for congestion control: When not to 523 send data. 525 WebRTC devices MUST implement the transport protocols described in 526 [I-D.ietf-rtcweb-transports]. 528 5. Data framing and securing 530 The format for media transport is RTP [RFC3550]. Implementation of 531 SRTP [RFC3711] is REQUIRED for all implementations. 533 The detailed considerations for usage of functions from RTP and SRTP 534 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 535 considerations for the WebRTC use case are in 536 [I-D.ietf-rtcweb-security], and the resulting security functions are 537 described in [I-D.ietf-rtcweb-security-arch]. 539 Considerations for the transfer of data that is not in RTP format is 540 described in [I-D.ietf-rtcweb-data-channel], and a supporting 541 protocol for establishing individual data channels is described in 542 [I-D.ietf-rtcweb-data-protocol]. Webrtc devices MUST implement these 543 two specifications. 545 WebRTC devices MUST implement [I-D.ietf-rtcweb-rtp-usage], 546 [I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the 547 requirements they include. 549 6. Data formats 551 The intent of this specification is to allow each communications 552 event to use the data formats that are best suited for that 553 particular instance, where a format is supported by both sides of the 554 connection. However, a minimum standard is greatly helpful in order 555 to ensure that communication can be achieved. This document 556 specifies a minimum baseline that will be supported by all 557 implementations of this specification, and leaves further codecs to 558 be included at the will of the implementor. 560 WebRTC devices MUST implement the codecs and profiles required in 561 [I-D.ietf-rtcweb-audio] 563 NOTE IN DRAFT: At this time (October 2014) there is no consensus on 564 what to say about video codecs in this section. 566 7. Connection management 568 The methods, mechanisms and requirements for setting up, negotiating 569 and tearing down connections is a large subject, and one where it is 570 desirable to have both interoperability and freedom to innovate. 572 The following principles apply: 574 1. The WebRTC media negotiations will be capable of representing the 575 same SDP offer/answer semantics that are used in SIP [RFC3264], 576 in such a way that it is possible to build a signaling gateway 577 between SIP and the WebRTC media negotiation. 579 2. It will be possible to gateway between legacy SIP devices that 580 support ICE and appropriate RTP / SDP mechanisms, codecs and 581 security mechanisms without using a media gateway. A signaling 582 gateway to convert between the signaling on the web side to the 583 SIP signaling may be needed. 585 3. When a new codec is specified, and the SDP for the new codec is 586 specified in the MMUSIC WG, no other standardization should be 587 required for it to be possible to use that in the web browsers. 588 Adding new codecs which might have new SDP parameters should not 589 change the APIs between the browser and Javascript application. 590 As soon as the browsers support the new codecs, old applications 591 written before the codecs were specified should automatically be 592 able to use the new codecs where appropriate with no changes to 593 the JS applications. 595 The particular choices made for WebRTC, and their implications for 596 the API offered by a WebRTC endpoint, are described in 597 [I-D.ietf-rtcweb-jsep]. 599 WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. 601 WebRTC devices MUST implement the functions described in that 602 document that relate to the network layer (for example Bundle, RTCP- 603 mux and Trickle ICE), but do not need to support the API 604 functionality described there. 606 8. Presentation and control 608 The most important part of control is the user's control over the 609 browser's interaction with input/output devices and communications 610 channels. It is important that the user have some way of figuring 611 out where his audio, video or texting is being sent, for what 612 purported reason, and what guarantees are made by the parties that 613 form part of this control channel. This is largely a local function 614 between the browser, the underlying operating system and the user 615 interface; this is specified in the peer connection API 616 [W3C.WD-webrtc-20120209], and the media capture API 617 [W3C.WD-mediacapture-streams-20120628]. 619 WebRTC browsers MUST implement these two specifications. 621 9. Local system support functions 623 These are characterized by the fact that the quality of these 624 functions strongly influence the user experience, but the exact 625 algorithm does not need coordination. In some cases (for instance 626 echo cancellation, as described below), the overall system definition 627 may need to specify that the overall system needs to have some 628 characteristics for which these facilities are useful, without 629 requiring them to be implemented a certain way. 631 Local functions include echo cancellation, volume control, camera 632 management including focus, zoom, pan/tilt controls (if available), 633 and more. 635 Certain parts of the system SHOULD conform to certain properties, for 636 instance: 638 o Echo cancellation should be good enough to achieve the suppression 639 of acoustical feedback loops below a perceptually noticeable 640 level. 642 o Privacy concerns MUST be satisfied; for instance, if remote 643 control of camera is offered, the APIs should be available to let 644 the local participant figure out who's controlling the camera, and 645 possibly decide to revoke the permission for camera usage. 647 o Automatic gain control, if present, should normalize a speaking 648 voice into a reasonable dB range. 650 The requirements on WebRTC devices with regard to audio processing 651 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 652 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 654 WebRTC devices MUST implement the processing functions in 655 [I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6, 656 this means that WebRTC devices MUST implement the whole document.) 658 10. IANA Considerations 660 This document makes no request of IANA. 662 Note to RFC Editor: this section may be removed on publication as an 663 RFC. 665 11. Security Considerations 667 Security of the web-enabled real time communications comes in several 668 pieces: 670 o Security of the components: The browsers, and other servers 671 involved. The most target-rich environment here is probably the 672 browser; the aim here should be that the introduction of these 673 components introduces no additional vulnerability. 675 o Security of the communication channels: It should be easy for a 676 participant to reassure himself of the security of his 677 communication - by verifying the crypto parameters of the links he 678 himself participates in, and to get reassurances from the other 679 parties to the communication that they promise that appropriate 680 measures are taken. 682 o Security of the partners' identity: verifying that the 683 participants are who they say they are (when positive 684 identification is appropriate), or that their identity cannot be 685 uncovered (when anonymity is a goal of the application). 687 The security analysis, and the requirements derived from that 688 analysis, is contained in [I-D.ietf-rtcweb-security]. 690 It is also important to read the security sections of 691 [W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209]. 693 12. Acknowledgements 695 The number of people who have taken part in the discussions 696 surrounding this draft are too numerous to list, or even to identify. 697 The ones below have made special, identifiable contributions; this 698 does not mean that others' contributions are less important. 700 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 701 Westerlund and Joerg Ott, who offered technical contributions on 702 various versions of the draft. 704 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 705 the ASCII drawings in section 1. 707 Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric 708 Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage 709 and Simon Leinen for document review. 711 13. References 713 13.1. Normative References 715 [I-D.alvestrand-rtcweb-gateways] 716 Alvestrand, H., "WebRTC Gateways", draft-alvestrand- 717 rtcweb-gateways-00 (work in progress), August 2014. 719 [I-D.ietf-rtcweb-audio] 720 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 721 Requirements", draft-ietf-rtcweb-audio-06 (work in 722 progress), September 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-12 (work in 727 progress), September 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-08 (work in progress), September 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-17 (work in progress), August 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 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 758 with Session Description Protocol (SDP)", RFC 3264, June 759 2002. 761 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 762 Jacobson, "RTP: A Transport Protocol for Real-Time 763 Applications", STD 64, RFC 3550, July 2003. 765 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 766 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 767 RFC 3711, March 2004. 769 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 770 (ICE): A Protocol for Network Address Translator (NAT) 771 Traversal for Offer/Answer Protocols", RFC 5245, April 772 2010. 774 [W3C.WD-mediacapture-streams-20120628] 775 Burnett, D. and A. Narayanan, "Media Capture and Streams", 776 World Wide Web Consortium WD WD-mediacapture- 777 streams-20120628, June 2012, . 780 [W3C.WD-webrtc-20120209] 781 Bergkvist, A., Burnett, D., Jennings, C., and A. 782 Narayanan, "WebRTC 1.0: Real-time Communication Between 783 Browsers", World Wide Web Consortium WD WD- 784 webrtc-20120209, February 2012, 785 . 787 13.2. Informative References 789 [I-D.ietf-rtcweb-use-cases-and-requirements] 790 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 791 Time Communication Use-cases and Requirements", draft- 792 ietf-rtcweb-use-cases-and-requirements-14 (work in 793 progress), February 2014. 795 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 796 A., Peterson, J., Sparks, R., Handley, M., and E. 797 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 798 June 2002. 800 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", BCP 801 95, RFC 3935, October 2004. 803 [RFC6120] Saint-Andre, P., "Extensible Messaging and Presence 804 Protocol (XMPP): Core", RFC 6120, March 2011. 806 [W3C.WD-html5-20110525] 807 Hickson, I., "HTML5", World Wide Web Consortium LastCall 808 WD-html5-20110525, May 2011, 809 . 811 Appendix A. Change log 813 This section may be deleted by the RFC Editor when preparing for 814 publication. 816 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 818 Added section "On interoperability and innovation" 820 Added data confidentiality and integrity to the "data framing" layer 821 Added congestion management requirements in the "data transport" 822 layer section 824 Changed need for non-media data from "question: do we need this?" to 825 "Open issue: How do we do this?" 827 Strengthened disclaimer that listed codecs are placeholders, not 828 decisions. 830 More details on why the "local system support functions" section is 831 there. 833 A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand- 834 rtcweb-overview-00 836 Added section on "Relationship between API and protocol" 838 Added terminology section 840 Mentioned congestion management as part of the "data transport" layer 841 in the layer list 843 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 845 Removed most technical content, and replaced with pointers to drafts 846 as requested and identified by the RTCWEB WG chairs. 848 Added content to acknowledgments section. 850 Added change log. 852 Spell-checked document. 854 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf- 855 rtcweb-overview-00 857 Changed draft name and document date. 859 Removed unused references 861 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 863 Added architecture figures to section 2. 865 Changed the description of "echo cancellation" under "local system 866 support functions". 868 Added a few more definitions. 870 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 872 Added pointers to use cases, security and rtp-usage drafts (now WG 873 drafts). 875 Changed description of SRTP from mandatory-to-use to mandatory-to- 876 implement. 878 Added the "3 principles of negotiation" to the connection management 879 section. 881 Added an explicit statement that ICE is required for both NAT and 882 consent-to-receive. 884 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 886 Added references to a number of new drafts. 888 Expanded the description text under the "trapezoid" drawing with some 889 more text discussed on the list. 891 Changed the "Connection management" sentence from "will be done using 892 SDP offer/answer" to "will be capable of representing SDP offer/ 893 answer" - this seems more consistent with JSEP. 895 Added "security mechanisms" to the things a non-gatewayed SIP devices 896 must support in order to not need a media gateway. 898 Added a definition for "browser". 900 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 902 Made introduction more normative. 904 Several wording changes in response to review comments from EKR 906 Added an appendix to hold references and notes that are not yet in a 907 separate document. 909 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 911 Minor grammatical fixes. This is mainly a "keepalive" refresh. 913 A.10. Changes from -05 to -06 915 Clarifications in response to Last Call review comments. Inserted 916 reference to draft-ietf-rtcweb-audio. 918 A.11. Changes from -06 to -07 920 Added a reference to the "unified plan" draft, and updated some 921 references. 923 Otherwise, it's a "keepalive" draft. 925 A.12. Changes from -07 to -08 927 Removed the appendix that detailed transports, and replaced it with a 928 reference to draft-ietf-rtcweb-transports. Removed now-unused 929 references. 931 A.13. Changes from -08 to -09 933 Added text to the Abstract indicating that the intended status is an 934 Applicability Statement. 936 A.14. Changes from -09 to -10 938 Defined "WebRTC Browser" and "WebRTC device" as things that do, or 939 don't, conform to the API. 941 Updated reference to data-protocol draft 943 Updated data formats to reference -rtcweb-audio- and not the expired 944 -cbran draft. 946 Deleted references to -unified-plan 948 Deleted reference to -generic-idp (draft expired) 950 Added notes on which referenced documents WebRTC browsers or devices 951 MUST conform to. 953 Added pointer to the security section of the API drafts. 955 A.15. Changes from -10 to -11 957 Added "WebRTC Gateway" as a third class of device, and referenced the 958 doc describing them. 960 Made a number of text clarifications, in response to document 961 reviews. 963 A.16. Changes from -11 to -12 965 Refined entity definitions to define "WebRTC endpoint" and "WebRTC- 966 compatible endpoint". 968 Changed remaining usage of the term "RTCWEB" to "WebRTC", including 969 in the page header. 971 Author's Address 973 Harald T. Alvestrand 974 Google 975 Kungsbron 2 976 Stockholm 11122 977 Sweden 979 Email: harald@alvestrand.no