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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 16, 2015 5 Expires: December 18, 2015 7 Overview: Real Time Protocols for Browser-based Applications 8 draft-ietf-rtcweb-overview-14 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 December 18, 2015. 44 Copyright Notice 46 Copyright (c) 2015 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 A.18. Changes from -13 to -14 . . . . . . . . . . . . . . . . . 22 102 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 22 104 1. Introduction 106 The Internet was, from very early in its lifetime, considered a 107 possible vehicle for the deployment of real-time, interactive 108 applications - with the most easily imaginable being audio 109 conversations (aka "Internet telephony") and video conferencing. 111 The first attempts to build this were dependent on special networks, 112 special hardware and custom-built software, often at very high prices 113 or at low quality, placing great demands on the infrastructure. 115 As the available bandwidth has increased, and as processors an other 116 hardware has become ever faster, the barriers to participation have 117 decreased, and it has become possible to deliver a satisfactory 118 experience on commonly available computing hardware. 120 Still, there are a number of barriers to the ability to communicate 121 universally - one of these is that there is, as of yet, no single set 122 of communication protocols that all agree should be made available 123 for communication; another is the sheer lack of universal 124 identification systems (such as is served by telephone numbers or 125 email addresses in other communications systems). 127 Development of The Universal Solution has proved hard, however, for 128 all the usual reasons. 130 The last few years have also seen a new platform rise for deployment 131 of services: The browser-embedded application, or "Web application". 132 It turns out that as long as the browser platform has the necessary 133 interfaces, it is possible to deliver almost any kind of service on 134 it. 136 Traditionally, these interfaces have been delivered by plugins, which 137 had to be downloaded and installed separately from the browser; in 138 the development of HTML5, application developers see much promise in 139 the possibility of making those interfaces available in a 140 standardized way within the browser. 142 This memo describes a set of building blocks that can be made 143 accessible and controllable through a Javascript API in a browser, 144 and which together form a sufficient set of functions to allow the 145 use of interactive audio and video in applications that communicate 146 directly between browsers across the Internet. The resulting 147 protocol suite is intended to enable all the applications that are 148 described as required scenarios in the use cases document 149 [I-D.ietf-rtcweb-use-cases-and-requirements]. 151 Other efforts, for instance the W3C WEBRTC, Web Applications and 152 Device API working groups, focus on making standardized APIs and 153 interfaces available, within or alongside the HTML5 effort, for those 154 functions; this memo concentrates on specifying the protocols and 155 subprotocols that are needed to specify the interactions that happen 156 across the network. 158 This memo uses the term "WebRTC" (note the case used) to refer to the 159 overall effort consisting of both IETF and W3C efforts. 161 2. Principles and Terminology 163 2.1. Goals of this document 165 The goal of the WebRTC protocol specification is to specify a set of 166 protocols that, if all are implemented, will allow an implementation 167 to communicate with another implementation using audio, video and 168 data sent along the most direct possible path between the 169 participants. 171 This document is intended to serve as the roadmap to the WebRTC 172 specifications. It defines terms used by other pieces of 173 specification, lists references to other specifications that don't 174 need further elaboration in the WebRTC context, and gives pointers to 175 other documents that form part of the WebRTC suite. 177 By reading this document and the documents it refers to, it should be 178 possible to have all information needed to implement an WebRTC 179 compatible implementation. 181 2.2. Relationship between API and protocol 183 The total WebRTC effort consists of two pieces: 185 o A protocol specification, done in the IETF 187 o A Javascript API specification, done in the W3C 188 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 190 Together, these two specifications aim to provide an environment 191 where Javascript embedded in any page, viewed in any compatible 192 browser, when suitably authorized by its user, is able to set up 193 communication using audio, video and auxiliary data, where the 194 browser environment does not constrain the types of application in 195 which this functionality can be used. 197 The protocol specification does not assume that all implementations 198 implement this API; it is not intended to be necessary for 199 interoperation to know whether the entity one is communicating with 200 is a browser or another device implementing this specification. 202 The goal of cooperation between the protocol specification and the 203 API specification is that for all options and features of the 204 protocol specification, it should be clear which API calls to make to 205 exercise that option or feature; similarly, for any sequence of API 206 calls, it should be clear which protocol options and features will be 207 invoked. Both subject to constraints of the implementation, of 208 course. 210 For the purpose of this document, we define the following terminology 211 to talk about WebRTC things: 213 o A WebRTC browser (also called a WebRTC User Agent or WebRTC UA) is 214 something that conforms to both the protocol specification and the 215 Javascript API defined above. 217 o A WebRTC non-browser is something that conforms to the protocol 218 specification, but does not claim to implement the Javascript API. 219 This can also be called a "WebRTC device" or "WebRTC native 220 application". 222 o A WebRTC endpoint is either a WebRTC browser or a WebRTC non- 223 browser. It conforms to the protocol specification. 225 o A WebRTC-compatible endpoint is an endpoint that is able to 226 successfully communicate with a WebRTC endpoint, but may fail to 227 meet some requirements of a WebRTC endpoint. This may limit where 228 in the network such an endpoint can be attached, or may limit the 229 security guarantees that it offers to others. It is not 230 constrained by this specification; when it is mentioned at all, it 231 is to note the implications on WebRTC-compatible endpoints of the 232 requirements placed on WebRTC endpoints. 234 o A WebRTC gateway is a WebRTC-compatible endpoint that mediates 235 media traffic to non-WebRTC entities. 237 All WebRTC browsers are WebRTC endpoints, so any requirement on a 238 WebRTC endpoint also applies to a WebRTC browser. 240 A WebRTC non-browser may be capable of hosting applications in a 241 similar way to the way in which a browser can host Javascript 242 applications, typically by offering APIs in other languages. For 243 instance it may be implemented as a library that offers a C++ API 244 intended to be loaded into applications. In this case, similar 245 security considerations as for Javascript may be needed; however, 246 since such APIs are not defined or referenced here, this document 247 cannot give any specific rules for those interfaces. 249 WebRTC gateways are described in a separate document, 250 [I-D.alvestrand-rtcweb-gateways]. 252 2.3. On interoperability and innovation 254 The "Mission statement of the IETF" [RFC3935] states that "The 255 benefit of a standard to the Internet is in interoperability - that 256 multiple products implementing a standard are able to work together 257 in order to deliver valuable functions to the Internet's users." 259 Communication on the Internet frequently occurs in two phases: 261 o Two parties communicate, through some mechanism, what 262 functionality they both are able to support 264 o They use that shared communicative functionality to communicate, 265 or, failing to find anything in common, give up on communication. 267 There are often many choices that can be made for communicative 268 functionality; the history of the Internet is rife with the proposal, 269 standardization, implementation, and success or failure of many types 270 of options, in all sorts of protocols. 272 The goal of having a mandatory to implement function set is to 273 prevent negotiation failure, not to preempt or prevent negotiation. 275 The presence of a mandatory to implement function set serves as a 276 strong changer of the marketplace of deployment - in that it gives a 277 guarantee that, as long as you conform to a specification, and the 278 other party is willing to accept communication at the base level of 279 that specification, you can communicate successfully. 281 The alternative - that of having no mandatory to implement - does not 282 mean that you cannot communicate, it merely means that in order to be 283 part of the communications partnership, you have to implement the 284 standard "and then some" - that "and then some" usually being called 285 a profile of some sort; in the version most antithetical to the 286 Internet ethos, that "and then some" consists of having to use a 287 specific vendor's product only. 289 2.4. Terminology 291 The following terms are used across the documents specifying the 292 WebRTC suite, in the specific meanings given here. Not all terms are 293 used in this document. Other terms are used in their commonly used 294 meaning. 296 The list is in alphabetical order. 298 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 300 API: Application Programming Interface - a specification of a set of 301 calls and events, usually tied to a programming language or an 302 abstract formal specification such as WebIDL, with its defined 303 semantics. 305 Browser: Used synonymously with "Interactive User Agent" as defined 306 in the HTML specification [W3C.WD-html5-20110525]. See also 307 "WebRTC User Agent". 309 ICE Agent: An implementation of the Interactive Connectivty 310 Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be 311 an SDP Agent, but there exist ICE Agents that do not use SDP (for 312 instance those that use Jingle). 314 Interactive: Communication between multiple parties, where the 315 expectation is that an action from one party can cause a reaction 316 by another party, and the reaction can be observed by the first 317 party, with the total time required for the action/reaction/ 318 observation is on the order of no more than hundreds of 319 milliseconds. 321 Media: Audio and video content. Not to be confused with 322 "transmission media" such as wires. 324 Media path: The path that media data follows from one WebRTC 325 endpoint to another. 327 Protocol: A specification of a set of data units, their 328 representation, and rules for their transmission, with their 329 defined semantics. A protocol is usually thought of as going 330 between systems. 332 Real-time media: Media where generation of content and display of 333 content are intended to occur closely together in time (on the 334 order of no more than hundreds of milliseconds). Real-time media 335 can be used to support interactive communication. 337 SDP Agent: The protocol implementation involved in the SDP offer/ 338 answer exchange, as defined in [RFC3264] section 3. 340 Signaling: Communication that happens in order to establish, manage 341 and control media paths. 343 Signaling Path: The communication channels used between entities 344 participating in signaling to transfer signaling. There may be 345 more entities in the signaling path than in the media path. 347 NOTE: Where common definitions exist for these terms, those 348 definitions should be used to the greatest extent possible. 350 3. Architecture and Functionality groups 352 The model of real-time support for browser-based applications does 353 not assume that the browser will contain all the functions that need 354 to be performed in order to have a function such as a telephone or a 355 video conferencing unit; the vision is that the browser will have the 356 functions that are needed for a Web application, working in 357 conjunction with its backend servers, to implement these functions. 359 This means that two vital interfaces need specification: The 360 protocols that browsers talk to each other, without any intervening 361 servers, and the APIs that are offered for a Javascript application 362 to take advantage of the browser's functionality. 364 +------------------------+ On-the-wire 365 | | Protocols 366 | Servers |---------> 367 | | 368 | | 369 +------------------------+ 370 ^ 371 | 372 | 373 | HTTP/ 374 | Websockets 375 | 376 | 377 +----------------------------+ 378 | Javascript/HTML/CSS | 379 +----------------------------+ 380 Other ^ ^RTC 381 APIs | |APIs 382 +---|-----------------|------+ 383 | | | | 384 | +---------+| 385 | | Browser || On-the-wire 386 | Browser | RTC || Protocols 387 | | Function|-----------> 388 | | || 389 | | || 390 | +---------+| 391 +---------------------|------+ 392 | 393 V 394 Native OS Services 396 Figure 1: Browser Model 398 Note that HTTP and Websockets are also offered to the Javascript 399 application through browser APIs. 401 As for all protocol and API specifications, there is no restriction 402 that the protocols can only be used to talk to another browser; since 403 they are fully specified, any endpoint that implements the protocols 404 faithfully should be able to interoperate with the application 405 running in the browser. 407 A commonly imagined model of deployment is the one depicted below. 409 +-----------+ +-----------+ 410 | Web | | Web | 411 | | Signaling | | 412 | |-------------| | 413 | Server | path | Server | 414 | | | | 415 +-----------+ +-----------+ 416 / \ 417 / \ Application-defined 418 / \ over 419 / \ HTTP/Websockets 420 / Application-defined over \ 421 / HTTP/Websockets \ 422 / \ 423 +-----------+ +-----------+ 424 |JS/HTML/CSS| |JS/HTML/CSS| 425 +-----------+ +-----------+ 426 +-----------+ +-----------+ 427 | | | | 428 | | | | 429 | Browser | ------------------------- | Browser | 430 | | Media path | | 431 | | | | 432 +-----------+ +-----------+ 434 Figure 2: Browser RTC Trapezoid 436 On this drawing, the critical part to note is that the media path 437 ("low path") goes directly between the browsers, so it has to be 438 conformant to the specifications of the WebRTC protocol suite; the 439 signaling path ("high path") goes via servers that can modify, 440 translate or massage the signals as needed. 442 If the two Web servers are operated by different entities, the inter- 443 server signaling mechanism needs to be agreed upon, either by 444 standardization or by other means of agreement. Existing protocols 445 (for example SIP [RFC3261] or XMPP [RFC6120]) could be used between 446 servers, while either a standards-based or proprietary protocol could 447 be used between the browser and the web server. 449 For example, if both operators' servers implement SIP, SIP could be 450 used for communication between servers, along with either a 451 standardized signaling mechanism (e.g. SIP over Websockets) or a 452 proprietary signaling mechanism used between the application running 453 in the browser and the web server. Similarly, if both operators' 454 servers implement XMPP, XMPP could be used for communication between 455 XMPP servers, with either a standardized signaling mechanism (e.g. 456 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 457 used between the application running in the browser and the web 458 server. 460 The choice of protocols, and definition of the translation between 461 them, is outside the scope of the WebRTC protocol suite described in 462 the document. 464 The functionality groups that are needed in the browser can be 465 specified, more or less from the bottom up, as: 467 o Data transport: TCP, UDP and the means to securely set up 468 connections between entities, as well as the functions for 469 deciding when to send data: Congestion management, bandwidth 470 estimation and so on. 472 o Data framing: RTP and other data formats that serve as containers, 473 and their functions for data confidentiality and integrity. 475 o Data formats: Codec specifications, format specifications and 476 functionality specifications for the data passed between systems. 477 Audio and video codecs, as well as formats for data and document 478 sharing, belong in this category. In order to make use of data 479 formats, a way to describe them, a session description, is needed. 481 o Connection management: Setting up connections, agreeing on data 482 formats, changing data formats during the duration of a call; SIP 483 and Jingle/XMPP belong in this category. 485 o Presentation and control: What needs to happen in order to ensure 486 that interactions behave in a non-surprising manner. This can 487 include floor control, screen layout, voice activated image 488 switching and other such functions - where part of the system 489 require the cooperation between parties. XCON and Cisco/ 490 Tandberg's TIP were some attempts at specifying this kind of 491 functionality; many applications have been built without 492 standardized interfaces to these functions. 494 o Local system support functions: These are things that need not be 495 specified uniformly, because each participant may choose to do 496 these in a way of the participant's choosing, without affecting 497 the bits on the wire in a way that others have to be cognizant of. 498 Examples in this category include echo cancellation (some forms of 499 it), local authentication and authorization mechanisms, OS access 500 control and the ability to do local recording of conversations. 502 Within each functionality group, it is important to preserve both 503 freedom to innovate and the ability for global communication. 504 Freedom to innovate is helped by doing the specification in terms of 505 interfaces, not implementation; any implementation able to 506 communicate according to the interfaces is a valid implementation. 507 Ability to communicate globally is helped both by having core 508 specifications be unencumbered by IPR issues and by having the 509 formats and protocols be fully enough specified to allow for 510 independent implementation. 512 One can think of the three first groups as forming a "media transport 513 infrastructure", and of the three last groups as forming a "media 514 service". In many contexts, it makes sense to use a common 515 specification for the media transport infrastructure, which can be 516 embedded in browsers and accessed using standard interfaces, and "let 517 a thousand flowers bloom" in the "media service" layer; to achieve 518 interoperable services, however, at least the first five of the six 519 groups need to be specified. 521 4. Data transport 523 Data transport refers to the sending and receiving of data over the 524 network interfaces, the choice of network-layer addresses at each end 525 of the communication, and the interaction with any intermediate 526 entities that handle the data, but do not modify it (such as TURN 527 relays). 529 It includes necessary functions for congestion control: When not to 530 send data. 532 WebRTC endpoints MUST implement the transport protocols described in 533 [I-D.ietf-rtcweb-transports]. 535 5. Data framing and securing 537 The format for media transport is RTP [RFC3550]. Implementation of 538 SRTP [RFC3711] is REQUIRED for all implementations. 540 The detailed considerations for usage of functions from RTP and SRTP 541 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 542 considerations for the WebRTC use case are in 543 [I-D.ietf-rtcweb-security], and the resulting security functions are 544 described in [I-D.ietf-rtcweb-security-arch]. 546 Considerations for the transfer of data that is not in RTP format is 547 described in [I-D.ietf-rtcweb-data-channel], and a supporting 548 protocol for establishing individual data channels is described in 549 [I-D.ietf-rtcweb-data-protocol]. WebRTC endpoints MUST implement 550 these two specifications. 552 WebRTC endpoints MUST implement [I-D.ietf-rtcweb-rtp-usage], 553 [I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the 554 requirements they include. 556 6. Data formats 558 The intent of this specification is to allow each communications 559 event to use the data formats that are best suited for that 560 particular instance, where a format is supported by both sides of the 561 connection. However, a minimum standard is greatly helpful in order 562 to ensure that communication can be achieved. This document 563 specifies a minimum baseline that will be supported by all 564 implementations of this specification, and leaves further codecs to 565 be included at the will of the implementor. 567 WebRTC endpoints that support audio and/or video MUST implement the 568 codecs and profiles required in [I-D.ietf-rtcweb-audio] and 569 [I-D.ietf-rtcweb-video]. 571 7. Connection management 573 The methods, mechanisms and requirements for setting up, negotiating 574 and tearing down connections is a large subject, and one where it is 575 desirable to have both interoperability and freedom to innovate. 577 The following principles apply: 579 1. The WebRTC media negotiations will be capable of representing the 580 same SDP offer/answer semantics that are used in SIP [RFC3264], 581 in such a way that it is possible to build a signaling gateway 582 between SIP and the WebRTC media negotiation. 584 2. It will be possible to gateway between legacy SIP devices that 585 support ICE and appropriate RTP / SDP mechanisms, codecs and 586 security mechanisms without using a media gateway. A signaling 587 gateway to convert between the signaling on the web side to the 588 SIP signaling may be needed. 590 3. When a new codec is specified, and the SDP for the new codec is 591 specified in the MMUSIC WG, no other standardization should be 592 required for it to be possible to use that in the web browsers. 593 Adding new codecs which might have new SDP parameters should not 594 change the APIs between the browser and Javascript application. 595 As soon as the browsers support the new codecs, old applications 596 written before the codecs were specified should automatically be 597 able to use the new codecs where appropriate with no changes to 598 the JS applications. 600 The particular choices made for WebRTC, and their implications for 601 the API offered by a browser implementing WebRTC, are described in 602 [I-D.ietf-rtcweb-jsep]. 604 WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. 606 WebRTC endpoints MUST implement the functions described in that 607 document that relate to the network layer (for example Bundle, RTCP- 608 mux and Trickle ICE), but do not need to support the API 609 functionality described there. 611 8. Presentation and control 613 The most important part of control is the user's control over the 614 browser's interaction with input/output devices and communications 615 channels. It is important that the user have some way of figuring 616 out where his audio, video or texting is being sent, for what 617 purported reason, and what guarantees are made by the parties that 618 form part of this control channel. This is largely a local function 619 between the browser, the underlying operating system and the user 620 interface; this is specified in the peer connection API 621 [W3C.WD-webrtc-20120209], and the media capture API 622 [W3C.WD-mediacapture-streams-20120628]. 624 WebRTC browsers MUST implement these two specifications. 626 9. Local system support functions 628 These are characterized by the fact that the quality of these 629 functions strongly influence the user experience, but the exact 630 algorithm does not need coordination. In some cases (for instance 631 echo cancellation, as described below), the overall system definition 632 may need to specify that the overall system needs to have some 633 characteristics for which these facilities are useful, without 634 requiring them to be implemented a certain way. 636 Local functions include echo cancellation, volume control, camera 637 management including focus, zoom, pan/tilt controls (if available), 638 and more. 640 Certain parts of the system SHOULD conform to certain properties, for 641 instance: 643 o Echo cancellation should be good enough to achieve the suppression 644 of acoustical feedback loops below a perceptually noticeable 645 level. 647 o Privacy concerns MUST be satisfied; for instance, if remote 648 control of camera is offered, the APIs should be available to let 649 the local participant figure out who's controlling the camera, and 650 possibly decide to revoke the permission for camera usage. 652 o Automatic gain control, if present, should normalize a speaking 653 voice into a reasonable dB range. 655 The requirements on WebRTC systems with regard to audio processing 656 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 657 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 659 WebRTC endpoints MUST implement the processing functions in 660 [I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6, 661 this means that WebRTC endpoints MUST implement the whole document.) 663 10. IANA Considerations 665 This document makes no request of IANA. 667 Note to RFC Editor: this section may be removed on publication as an 668 RFC. 670 11. Security Considerations 672 Security of the web-enabled real time communications comes in several 673 pieces: 675 o Security of the components: The browsers, and other servers 676 involved. The most target-rich environment here is probably the 677 browser; the aim here should be that the introduction of these 678 components introduces no additional vulnerability. 680 o Security of the communication channels: It should be easy for a 681 participant to reassure himself of the security of his 682 communication - by verifying the crypto parameters of the links he 683 himself participates in, and to get reassurances from the other 684 parties to the communication that they promise that appropriate 685 measures are taken. 687 o Security of the partners' identity: verifying that the 688 participants are who they say they are (when positive 689 identification is appropriate), or that their identity cannot be 690 uncovered (when anonymity is a goal of the application). 692 The security analysis, and the requirements derived from that 693 analysis, is contained in [I-D.ietf-rtcweb-security]. 695 It is also important to read the security sections of 696 [W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209]. 698 12. Acknowledgements 700 The number of people who have taken part in the discussions 701 surrounding this draft are too numerous to list, or even to identify. 702 The ones below have made special, identifiable contributions; this 703 does not mean that others' contributions are less important. 705 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 706 Westerlund and Joerg Ott, who offered technical contributions on 707 various versions of the draft. 709 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 710 the ASCII drawings in section 1. 712 Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric 713 Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage 714 and Simon Leinen for document review. 716 13. References 718 13.1. Normative References 720 [I-D.ietf-rtcweb-audio] 721 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 722 Requirements", draft-ietf-rtcweb-audio-08 (work in 723 progress), April 2015. 725 [I-D.ietf-rtcweb-data-channel] 726 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 727 Channels", draft-ietf-rtcweb-data-channel-13 (work in 728 progress), January 2015. 730 [I-D.ietf-rtcweb-data-protocol] 731 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 732 Establishment Protocol", draft-ietf-rtcweb-data- 733 protocol-09 (work in progress), January 2015. 735 [I-D.ietf-rtcweb-jsep] 736 Uberti, J., Jennings, C., and E. Rescorla, "Javascript 737 Session Establishment Protocol", draft-ietf-rtcweb-jsep-10 738 (work in progress), June 2015. 740 [I-D.ietf-rtcweb-rtp-usage] 741 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 742 Communication (WebRTC): Media Transport and Use of RTP", 743 draft-ietf-rtcweb-rtp-usage-22 (work in progress), 744 February 2015. 746 [I-D.ietf-rtcweb-security] 747 Rescorla, E., "Security Considerations for WebRTC", draft- 748 ietf-rtcweb-security-08 (work in progress), February 2015. 750 [I-D.ietf-rtcweb-security-arch] 751 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 752 rtcweb-security-arch-11 (work in progress), March 2015. 754 [I-D.ietf-rtcweb-transports] 755 Alvestrand, H., "Transports for WebRTC", draft-ietf- 756 rtcweb-transports-08 (work in progress), February 2015. 758 [I-D.ietf-rtcweb-video] 759 Roach, A., "WebRTC Video Processing and Codec 760 Requirements", draft-ietf-rtcweb-video-06 (work in 761 progress), June 2015. 763 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 764 with Session Description Protocol (SDP)", RFC 3264, June 765 2002. 767 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 768 Jacobson, "RTP: A Transport Protocol for Real-Time 769 Applications", STD 64, RFC 3550, July 2003. 771 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 772 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 773 RFC 3711, March 2004. 775 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 776 (ICE): A Protocol for Network Address Translator (NAT) 777 Traversal for Offer/Answer Protocols", RFC 5245, April 778 2010. 780 [W3C.WD-mediacapture-streams-20120628] 781 Burnett, D. and A. Narayanan, "Media Capture and Streams", 782 World Wide Web Consortium WD WD-mediacapture- 783 streams-20120628, June 2012, . 786 [W3C.WD-webrtc-20120209] 787 Bergkvist, A., Burnett, D., Jennings, C., and A. 788 Narayanan, "WebRTC 1.0: Real-time Communication Between 789 Browsers", World Wide Web Consortium WD WD- 790 webrtc-20120209, February 2012, 791 . 793 13.2. Informative References 795 [I-D.alvestrand-rtcweb-gateways] 796 Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways", 797 draft-alvestrand-rtcweb-gateways-02 (work in progress), 798 March 2015. 800 [I-D.ietf-rtcweb-use-cases-and-requirements] 801 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 802 Time Communication Use-cases and Requirements", draft- 803 ietf-rtcweb-use-cases-and-requirements-16 (work in 804 progress), January 2015. 806 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 807 A., Peterson, J., Sparks, R., Handley, M., and E. 808 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 809 June 2002. 811 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", BCP 812 95, RFC 3935, October 2004. 814 [RFC6120] Saint-Andre, P., "Extensible Messaging and Presence 815 Protocol (XMPP): Core", RFC 6120, March 2011. 817 [W3C.WD-html5-20110525] 818 Hickson, I., "HTML5", World Wide Web Consortium LastCall 819 WD-html5-20110525, May 2011, 820 . 822 Appendix A. Change log 824 This section may be deleted by the RFC Editor when preparing for 825 publication. 827 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 829 Added section "On interoperability and innovation" 831 Added data confidentiality and integrity to the "data framing" layer 832 Added congestion management requirements in the "data transport" 833 layer section 835 Changed need for non-media data from "question: do we need this?" to 836 "Open issue: How do we do this?" 838 Strengthened disclaimer that listed codecs are placeholders, not 839 decisions. 841 More details on why the "local system support functions" section is 842 there. 844 A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand- 845 rtcweb-overview-00 847 Added section on "Relationship between API and protocol" 849 Added terminology section 851 Mentioned congestion management as part of the "data transport" layer 852 in the layer list 854 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 856 Removed most technical content, and replaced with pointers to drafts 857 as requested and identified by the RTCWEB WG chairs. 859 Added content to acknowledgments section. 861 Added change log. 863 Spell-checked document. 865 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf- 866 rtcweb-overview-00 868 Changed draft name and document date. 870 Removed unused references 872 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 874 Added architecture figures to section 2. 876 Changed the description of "echo cancellation" under "local system 877 support functions". 879 Added a few more definitions. 881 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 883 Added pointers to use cases, security and rtp-usage drafts (now WG 884 drafts). 886 Changed description of SRTP from mandatory-to-use to mandatory-to- 887 implement. 889 Added the "3 principles of negotiation" to the connection management 890 section. 892 Added an explicit statement that ICE is required for both NAT and 893 consent-to-receive. 895 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 897 Added references to a number of new drafts. 899 Expanded the description text under the "trapezoid" drawing with some 900 more text discussed on the list. 902 Changed the "Connection management" sentence from "will be done using 903 SDP offer/answer" to "will be capable of representing SDP offer/ 904 answer" - this seems more consistent with JSEP. 906 Added "security mechanisms" to the things a non-gatewayed SIP devices 907 must support in order to not need a media gateway. 909 Added a definition for "browser". 911 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 913 Made introduction more normative. 915 Several wording changes in response to review comments from EKR 917 Added an appendix to hold references and notes that are not yet in a 918 separate document. 920 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 922 Minor grammatical fixes. This is mainly a "keepalive" refresh. 924 A.10. Changes from -05 to -06 926 Clarifications in response to Last Call review comments. Inserted 927 reference to draft-ietf-rtcweb-audio. 929 A.11. Changes from -06 to -07 931 Added a reference to the "unified plan" draft, and updated some 932 references. 934 Otherwise, it's a "keepalive" draft. 936 A.12. Changes from -07 to -08 938 Removed the appendix that detailed transports, and replaced it with a 939 reference to draft-ietf-rtcweb-transports. Removed now-unused 940 references. 942 A.13. Changes from -08 to -09 944 Added text to the Abstract indicating that the intended status is an 945 Applicability Statement. 947 A.14. Changes from -09 to -10 949 Defined "WebRTC Browser" and "WebRTC device" as things that do, or 950 don't, conform to the API. 952 Updated reference to data-protocol draft 954 Updated data formats to reference -rtcweb-audio- and not the expired 955 -cbran draft. 957 Deleted references to -unified-plan 959 Deleted reference to -generic-idp (draft expired) 961 Added notes on which referenced documents WebRTC browsers or devices 962 MUST conform to. 964 Added pointer to the security section of the API drafts. 966 A.15. Changes from -10 to -11 968 Added "WebRTC Gateway" as a third class of device, and referenced the 969 doc describing them. 971 Made a number of text clarifications in response to document reviews. 973 A.16. Changes from -11 to -12 975 Refined entity definitions to define "WebRTC endpoint" and "WebRTC- 976 compatible endpoint". 978 Changed remaining usage of the term "RTCWEB" to "WebRTC", including 979 in the page header. 981 A.17. Changes from -12 to -13 983 Changed "WebRTC device" to be "WebRTC non-browser", per decision at 984 IETF 91. This led to the need for "WebRTC endpoint" as the common 985 label for both, and the usage of that term in the rest of the 986 document. 988 Added words about WebRTC APIs in languages other than Javascript. 990 Referenced draft-ietf-rtcweb-video for video codecs to support. 992 A.18. Changes from -13 to -14 994 None. This is a "keepalive" update. 996 Author's Address 998 Harald T. Alvestrand 999 Google 1000 Kungsbron 2 1001 Stockholm 11122 1002 Sweden 1004 Email: harald@alvestrand.no