idnits 2.17.1 draft-ietf-rtcweb-overview-16.txt: Checking boilerplate required by RFC 5378 and the IETF Trust (see https://trustee.ietf.org/license-info): ---------------------------------------------------------------------------- No issues found here. Checking nits according to https://www.ietf.org/id-info/1id-guidelines.txt: ---------------------------------------------------------------------------- No issues found here. Checking nits according to https://www.ietf.org/id-info/checklist : ---------------------------------------------------------------------------- ** The document seems to lack a both a reference to RFC 2119 and the recommended RFC 2119 boilerplate, even if it appears to use RFC 2119 keywords. RFC 2119 keyword, line 534: '...WebRTC endpoints MUST implement the tr...' RFC 2119 keyword, line 540: '... SRTP [RFC3711] is REQUIRED for all implementations....' RFC 2119 keyword, line 551: '... [I-D.ietf-rtcweb-data-protocol]. WebRTC endpoints MUST implement...' 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Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Outdated reference: A later version (-26) exists of draft-ietf-rtcweb-jsep-17 == Outdated reference: A later version (-12) exists of draft-ietf-rtcweb-security-08 == Outdated reference: A later version (-20) exists of draft-ietf-rtcweb-security-arch-12 ** Obsolete normative reference: RFC 5245 (Obsoleted by RFC 8445, RFC 8839) Summary: 2 errors (**), 0 flaws (~~), 4 warnings (==), 1 comment (--). 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 14, 2016 5 Expires: May 18, 2017 7 Overview: Real Time Protocols for Browser-based Applications 8 draft-ietf-rtcweb-overview-16 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 May 18, 2017. 44 Copyright Notice 46 Copyright (c) 2016 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 . . . . . . . . . . . . . . . . . . . . . 19 81 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 82 to -01 . . . . . . . . . . . . . . . . . . . . . . . . . 19 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 . . . . . . . . . . . . . . 20 88 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 20 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 . . 21 92 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 21 93 A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 21 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 . . . . . . . . . . . . . . . . . 22 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 A.19. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 22 103 A.20. Changes from -15 to -16 . . . . . . . . . . . . . . . . . 22 104 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 23 106 1. Introduction 108 The Internet was, from very early in its lifetime, considered a 109 possible vehicle for the deployment of real-time, interactive 110 applications - with the most easily imaginable being audio 111 conversations (aka "Internet telephony") and video conferencing. 113 The first attempts to build this were dependent on special networks, 114 special hardware and custom-built software, often at very high prices 115 or at low quality, placing great demands on the infrastructure. 117 As the available bandwidth has increased, and as processors an other 118 hardware has become ever faster, the barriers to participation have 119 decreased, and it has become possible to deliver a satisfactory 120 experience on commonly available computing hardware. 122 Still, there are a number of barriers to the ability to communicate 123 universally - one of these is that there is, as of yet, no single set 124 of communication protocols that all agree should be made available 125 for communication; another is the sheer lack of universal 126 identification systems (such as is served by telephone numbers or 127 email addresses in other communications systems). 129 Development of The Universal Solution has proved hard, however, for 130 all the usual reasons. 132 The last few years have also seen a new platform rise for deployment 133 of services: The browser-embedded application, or "Web application". 134 It turns out that as long as the browser platform has the necessary 135 interfaces, it is possible to deliver almost any kind of service on 136 it. 138 Traditionally, these interfaces have been delivered by plugins, which 139 had to be downloaded and installed separately from the browser; in 140 the development of HTML5, application developers see much promise in 141 the possibility of making those interfaces available in a 142 standardized way within the browser. 144 This memo describes a set of building blocks that can be made 145 accessible and controllable through a Javascript API in a browser, 146 and which together form a sufficient set of functions to allow the 147 use of interactive audio and video in applications that communicate 148 directly between browsers across the Internet. The resulting 149 protocol suite is intended to enable all the applications that are 150 described as required scenarios in the use cases document 151 [I-D.ietf-rtcweb-use-cases-and-requirements]. 153 Other efforts, for instance the W3C WEBRTC, Web Applications and 154 Device API working groups, focus on making standardized APIs and 155 interfaces available, within or alongside the HTML5 effort, for those 156 functions; this memo concentrates on specifying the protocols and 157 subprotocols that are needed to specify the interactions that happen 158 across the network. 160 This memo uses the term "WebRTC" (note the case used) to refer to the 161 overall effort consisting of both IETF and W3C efforts. 163 2. Principles and Terminology 165 2.1. Goals of this document 167 The goal of the WebRTC protocol specification is to specify a set of 168 protocols that, if all are implemented, will allow an implementation 169 to communicate with another implementation using audio, video and 170 data sent along the most direct possible path between the 171 participants. 173 This document is intended to serve as the roadmap to the WebRTC 174 specifications. It defines terms used by other pieces of 175 specification, lists references to other specifications that don't 176 need further elaboration in the WebRTC context, and gives pointers to 177 other documents that form part of the WebRTC suite. 179 By reading this document and the documents it refers to, it should be 180 possible to have all information needed to implement an WebRTC 181 compatible implementation. 183 2.2. Relationship between API and protocol 185 The total WebRTC effort consists of two pieces: 187 o A protocol specification, done in the IETF 189 o A Javascript API specification, done in the W3C 190 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 192 Together, these two specifications aim to provide an environment 193 where Javascript embedded in any page, viewed in any compatible 194 browser, when suitably authorized by its user, is able to set up 195 communication using audio, video and auxiliary data, where the 196 browser environment does not constrain the types of application in 197 which this functionality can be used. 199 The protocol specification does not assume that all implementations 200 implement this API; it is not intended to be necessary for 201 interoperation to know whether the entity one is communicating with 202 is a browser or another device implementing this specification. 204 The goal of cooperation between the protocol specification and the 205 API specification is that for all options and features of the 206 protocol specification, it should be clear which API calls to make to 207 exercise that option or feature; similarly, for any sequence of API 208 calls, it should be clear which protocol options and features will be 209 invoked. Both subject to constraints of the implementation, of 210 course. 212 For the purpose of this document, we define the following terminology 213 to talk about WebRTC things: 215 o A WebRTC browser (also called a WebRTC User Agent or WebRTC UA) is 216 something that conforms to both the protocol specification and the 217 Javascript API defined above. 219 o A WebRTC non-browser is something that conforms to the protocol 220 specification, but does not claim to implement the Javascript API. 221 This can also be called a "WebRTC device" or "WebRTC native 222 application". 224 o A WebRTC endpoint is either a WebRTC browser or a WebRTC non- 225 browser. It conforms to the protocol specification. 227 o A WebRTC-compatible endpoint is an endpoint that is able to 228 successfully communicate with a WebRTC endpoint, but may fail to 229 meet some requirements of a WebRTC endpoint. This may limit where 230 in the network such an endpoint can be attached, or may limit the 231 security guarantees that it offers to others. It is not 232 constrained by this specification; when it is mentioned at all, it 233 is to note the implications on WebRTC-compatible endpoints of the 234 requirements placed on WebRTC endpoints. 236 o A WebRTC gateway is a WebRTC-compatible endpoint that mediates 237 media traffic to non-WebRTC entities. 239 All WebRTC browsers are WebRTC endpoints, so any requirement on a 240 WebRTC endpoint also applies to a WebRTC browser. 242 A WebRTC non-browser may be capable of hosting applications in a 243 similar way to the way in which a browser can host Javascript 244 applications, typically by offering APIs in other languages. For 245 instance it may be implemented as a library that offers a C++ API 246 intended to be loaded into applications. In this case, similar 247 security considerations as for Javascript may be needed; however, 248 since such APIs are not defined or referenced here, this document 249 cannot give any specific rules for those interfaces. 251 WebRTC gateways are described in a separate document, 252 [I-D.ietf-rtcweb-gateways]. 254 2.3. On interoperability and innovation 256 The "Mission statement of the IETF" [RFC3935] states that "The 257 benefit of a standard to the Internet is in interoperability - that 258 multiple products implementing a standard are able to work together 259 in order to deliver valuable functions to the Internet's users." 261 Communication on the Internet frequently occurs in two phases: 263 o Two parties communicate, through some mechanism, what 264 functionality they both are able to support 266 o They use that shared communicative functionality to communicate, 267 or, failing to find anything in common, give up on communication. 269 There are often many choices that can be made for communicative 270 functionality; the history of the Internet is rife with the proposal, 271 standardization, implementation, and success or failure of many types 272 of options, in all sorts of protocols. 274 The goal of having a mandatory to implement function set is to 275 prevent negotiation failure, not to preempt or prevent negotiation. 277 The presence of a mandatory to implement function set serves as a 278 strong changer of the marketplace of deployment - in that it gives a 279 guarantee that, as long as you conform to a specification, and the 280 other party is willing to accept communication at the base level of 281 that specification, you can communicate successfully. 283 The alternative - that of having no mandatory to implement - does not 284 mean that you cannot communicate, it merely means that in order to be 285 part of the communications partnership, you have to implement the 286 standard "and then some" - that "and then some" usually being called 287 a profile of some sort; in the version most antithetical to the 288 Internet ethos, that "and then some" consists of having to use a 289 specific vendor's product only. 291 2.4. Terminology 293 The following terms are used across the documents specifying the 294 WebRTC suite, in the specific meanings given here. Not all terms are 295 used in this document. Other terms are used in their commonly used 296 meaning. 298 The list is in alphabetical order. 300 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 302 API: Application Programming Interface - a specification of a set of 303 calls and events, usually tied to a programming language or an 304 abstract formal specification such as WebIDL, with its defined 305 semantics. 307 Browser: Used synonymously with "Interactive User Agent" as defined 308 in the HTML specification [W3C.WD-html5-20110525]. See also 309 "WebRTC User Agent". 311 ICE Agent: An implementation of the Interactive Connectivty 312 Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be 313 an SDP Agent, but there exist ICE Agents that do not use SDP (for 314 instance those that use Jingle). 316 Interactive: Communication between multiple parties, where the 317 expectation is that an action from one party can cause a reaction 318 by another party, and the reaction can be observed by the first 319 party, with the total time required for the action/reaction/ 320 observation is on the order of no more than hundreds of 321 milliseconds. 323 Media: Audio and video content. Not to be confused with 324 "transmission media" such as wires. 326 Media path: The path that media data follows from one WebRTC 327 endpoint to another. 329 Protocol: A specification of a set of data units, their 330 representation, and rules for their transmission, with their 331 defined semantics. A protocol is usually thought of as going 332 between systems. 334 Real-time media: Media where generation of content and display of 335 content are intended to occur closely together in time (on the 336 order of no more than hundreds of milliseconds). Real-time media 337 can be used to support interactive communication. 339 SDP Agent: The protocol implementation involved in the SDP offer/ 340 answer exchange, as defined in [RFC3264] section 3. 342 Signaling: Communication that happens in order to establish, manage 343 and control media paths. 345 Signaling Path: The communication channels used between entities 346 participating in signaling to transfer signaling. There may be 347 more entities in the signaling path than in the media path. 349 NOTE: Where common definitions exist for these terms, those 350 definitions should be used to the greatest extent possible. 352 3. Architecture and Functionality groups 354 The model of real-time support for browser-based applications does 355 not assume that the browser will contain all the functions that need 356 to be performed in order to have a function such as a telephone or a 357 video conferencing unit; the vision is that the browser will have the 358 functions that are needed for a Web application, working in 359 conjunction with its backend servers, to implement these functions. 361 This means that two vital interfaces need specification: The 362 protocols that browsers talk to each other, without any intervening 363 servers, and the APIs that are offered for a Javascript application 364 to take advantage of the browser's functionality. 366 +------------------------+ On-the-wire 367 | | Protocols 368 | Servers |---------> 369 | | 370 | | 371 +------------------------+ 372 ^ 373 | 374 | 375 | HTTP/ 376 | Websockets 377 | 378 | 379 +----------------------------+ 380 | Javascript/HTML/CSS | 381 +----------------------------+ 382 Other ^ ^RTC 383 APIs | |APIs 384 +---|-----------------|------+ 385 | | | | 386 | +---------+| 387 | | Browser || On-the-wire 388 | Browser | RTC || Protocols 389 | | Function|-----------> 390 | | || 391 | | || 392 | +---------+| 393 +---------------------|------+ 394 | 395 V 396 Native OS Services 398 Figure 1: Browser Model 400 Note that HTTP and Websockets are also offered to the Javascript 401 application through browser APIs. 403 As for all protocol and API specifications, there is no restriction 404 that the protocols can only be used to talk to another browser; since 405 they are fully specified, any endpoint that implements the protocols 406 faithfully should be able to interoperate with the application 407 running in the browser. 409 A commonly imagined model of deployment is the one depicted below. 411 +-----------+ +-----------+ 412 | Web | | Web | 413 | | Signaling | | 414 | |-------------| | 415 | Server | path | Server | 416 | | | | 417 +-----------+ +-----------+ 418 / \ 419 / \ Application-defined 420 / \ over 421 / \ HTTP/Websockets 422 / Application-defined over \ 423 / HTTP/Websockets \ 424 / \ 425 +-----------+ +-----------+ 426 |JS/HTML/CSS| |JS/HTML/CSS| 427 +-----------+ +-----------+ 428 +-----------+ +-----------+ 429 | | | | 430 | | | | 431 | Browser | ------------------------- | Browser | 432 | | Media path | | 433 | | | | 434 +-----------+ +-----------+ 436 Figure 2: Browser RTC Trapezoid 438 On this drawing, the critical part to note is that the media path 439 ("low path") goes directly between the browsers, so it has to be 440 conformant to the specifications of the WebRTC protocol suite; the 441 signaling path ("high path") goes via servers that can modify, 442 translate or massage the signals as needed. 444 If the two Web servers are operated by different entities, the inter- 445 server signaling mechanism needs to be agreed upon, either by 446 standardization or by other means of agreement. Existing protocols 447 (for example SIP [RFC3261] or XMPP [RFC6120]) could be used between 448 servers, while either a standards-based or proprietary protocol could 449 be used between the browser and the web server. 451 For example, if both operators' servers implement SIP, SIP could be 452 used for communication between servers, along with either a 453 standardized signaling mechanism (e.g. SIP over Websockets) or a 454 proprietary signaling mechanism used between the application running 455 in the browser and the web server. Similarly, if both operators' 456 servers implement XMPP, XMPP could be used for communication between 457 XMPP servers, with either a standardized signaling mechanism (e.g. 458 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 459 used between the application running in the browser and the web 460 server. 462 The choice of protocols, and definition of the translation between 463 them, is outside the scope of the WebRTC protocol suite described in 464 the document. 466 The functionality groups that are needed in the browser can be 467 specified, more or less from the bottom up, as: 469 o Data transport: TCP, UDP and the means to securely set up 470 connections between entities, as well as the functions for 471 deciding when to send data: Congestion management, bandwidth 472 estimation and so on. 474 o Data framing: RTP and other data formats that serve as containers, 475 and their functions for data confidentiality and integrity. 477 o Data formats: Codec specifications, format specifications and 478 functionality specifications for the data passed between systems. 479 Audio and video codecs, as well as formats for data and document 480 sharing, belong in this category. In order to make use of data 481 formats, a way to describe them, a session description, is needed. 483 o Connection management: Setting up connections, agreeing on data 484 formats, changing data formats during the duration of a call; SIP 485 and Jingle/XMPP belong in this category. 487 o Presentation and control: What needs to happen in order to ensure 488 that interactions behave in a non-surprising manner. This can 489 include floor control, screen layout, voice activated image 490 switching and other such functions - where part of the system 491 require the cooperation between parties. XCON and Cisco/ 492 Tandberg's TIP were some attempts at specifying this kind of 493 functionality; many applications have been built without 494 standardized interfaces to these functions. 496 o Local system support functions: These are things that need not be 497 specified uniformly, because each participant may choose to do 498 these in a way of the participant's choosing, without affecting 499 the bits on the wire in a way that others have to be cognizant of. 500 Examples in this category include echo cancellation (some forms of 501 it), local authentication and authorization mechanisms, OS access 502 control and the ability to do local recording of conversations. 504 Within each functionality group, it is important to preserve both 505 freedom to innovate and the ability for global communication. 506 Freedom to innovate is helped by doing the specification in terms of 507 interfaces, not implementation; any implementation able to 508 communicate according to the interfaces is a valid implementation. 509 Ability to communicate globally is helped both by having core 510 specifications be unencumbered by IPR issues and by having the 511 formats and protocols be fully enough specified to allow for 512 independent implementation. 514 One can think of the three first groups as forming a "media transport 515 infrastructure", and of the three last groups as forming a "media 516 service". In many contexts, it makes sense to use a common 517 specification for the media transport infrastructure, which can be 518 embedded in browsers and accessed using standard interfaces, and "let 519 a thousand flowers bloom" in the "media service" layer; to achieve 520 interoperable services, however, at least the first five of the six 521 groups need to be specified. 523 4. Data transport 525 Data transport refers to the sending and receiving of data over the 526 network interfaces, the choice of network-layer addresses at each end 527 of the communication, and the interaction with any intermediate 528 entities that handle the data, but do not modify it (such as TURN 529 relays). 531 It includes necessary functions for congestion control: When not to 532 send data. 534 WebRTC endpoints MUST implement the transport protocols described in 535 [I-D.ietf-rtcweb-transports]. 537 5. Data framing and securing 539 The format for media transport is RTP [RFC3550]. Implementation of 540 SRTP [RFC3711] is REQUIRED for all implementations. 542 The detailed considerations for usage of functions from RTP and SRTP 543 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 544 considerations for the WebRTC use case are in 545 [I-D.ietf-rtcweb-security], and the resulting security functions are 546 described in [I-D.ietf-rtcweb-security-arch]. 548 Considerations for the transfer of data that is not in RTP format is 549 described in [I-D.ietf-rtcweb-data-channel], and a supporting 550 protocol for establishing individual data channels is described in 551 [I-D.ietf-rtcweb-data-protocol]. WebRTC endpoints MUST implement 552 these two specifications. 554 WebRTC endpoints MUST implement [I-D.ietf-rtcweb-rtp-usage], 555 [I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the 556 requirements they include. 558 6. Data formats 560 The intent of this specification is to allow each communications 561 event to use the data formats that are best suited for that 562 particular instance, where a format is supported by both sides of the 563 connection. However, a minimum standard is greatly helpful in order 564 to ensure that communication can be achieved. This document 565 specifies a minimum baseline that will be supported by all 566 implementations of this specification, and leaves further codecs to 567 be included at the will of the implementor. 569 WebRTC endpoints that support audio and/or video MUST implement the 570 codecs and profiles required in [I-D.ietf-rtcweb-audio] and 571 [I-D.ietf-rtcweb-video]. 573 7. Connection management 575 The methods, mechanisms and requirements for setting up, negotiating 576 and tearing down connections is a large subject, and one where it is 577 desirable to have both interoperability and freedom to innovate. 579 The following principles apply: 581 1. The WebRTC media negotiations will be capable of representing the 582 same SDP offer/answer semantics that are used in SIP [RFC3264], 583 in such a way that it is possible to build a signaling gateway 584 between SIP and the WebRTC media negotiation. 586 2. It will be possible to gateway between legacy SIP devices that 587 support ICE and appropriate RTP / SDP mechanisms, codecs and 588 security mechanisms without using a media gateway. A signaling 589 gateway to convert between the signaling on the web side to the 590 SIP signaling may be needed. 592 3. When a new codec is specified, and the SDP for the new codec is 593 specified in the MMUSIC WG, no other standardization should be 594 required for it to be possible to use that in the web browsers. 595 Adding new codecs which might have new SDP parameters should not 596 change the APIs between the browser and Javascript application. 597 As soon as the browsers support the new codecs, old applications 598 written before the codecs were specified should automatically be 599 able to use the new codecs where appropriate with no changes to 600 the JS applications. 602 The particular choices made for WebRTC, and their implications for 603 the API offered by a browser implementing WebRTC, are described in 604 [I-D.ietf-rtcweb-jsep]. 606 WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. 608 WebRTC endpoints MUST implement the functions described in that 609 document that relate to the network layer (for example Bundle, RTCP- 610 mux and Trickle ICE), but do not need to support the API 611 functionality described there. 613 8. Presentation and control 615 The most important part of control is the user's control over the 616 browser's interaction with input/output devices and communications 617 channels. It is important that the user have some way of figuring 618 out where his audio, video or texting is being sent, for what 619 purported reason, and what guarantees are made by the parties that 620 form part of this control channel. This is largely a local function 621 between the browser, the underlying operating system and the user 622 interface; this is specified in the peer connection API 623 [W3C.WD-webrtc-20120209], and the media capture API 624 [W3C.WD-mediacapture-streams-20120628]. 626 WebRTC browsers MUST implement these two specifications. 628 9. Local system support functions 630 These are characterized by the fact that the quality of these 631 functions strongly influence the user experience, but the exact 632 algorithm does not need coordination. In some cases (for instance 633 echo cancellation, as described below), the overall system definition 634 may need to specify that the overall system needs to have some 635 characteristics for which these facilities are useful, without 636 requiring them to be implemented a certain way. 638 Local functions include echo cancellation, volume control, camera 639 management including focus, zoom, pan/tilt controls (if available), 640 and more. 642 Certain parts of the system SHOULD conform to certain properties, for 643 instance: 645 o Echo cancellation should be good enough to achieve the suppression 646 of acoustical feedback loops below a perceptually noticeable 647 level. 649 o Privacy concerns MUST be satisfied; for instance, if remote 650 control of camera is offered, the APIs should be available to let 651 the local participant figure out who's controlling the camera, and 652 possibly decide to revoke the permission for camera usage. 654 o Automatic gain control, if present, should normalize a speaking 655 voice into a reasonable dB range. 657 The requirements on WebRTC systems with regard to audio processing 658 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 659 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 661 WebRTC endpoints MUST implement the processing functions in 662 [I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6, 663 this means that WebRTC endpoints MUST implement the whole document.) 665 10. IANA Considerations 667 This document makes no request of IANA. 669 Note to RFC Editor: this section may be removed on publication as an 670 RFC. 672 11. Security Considerations 674 Security of the web-enabled real time communications comes in several 675 pieces: 677 o Security of the components: The browsers, and other servers 678 involved. The most target-rich environment here is probably the 679 browser; the aim here should be that the introduction of these 680 components introduces no additional vulnerability. 682 o Security of the communication channels: It should be easy for a 683 participant to reassure himself of the security of his 684 communication - by verifying the crypto parameters of the links he 685 himself participates in, and to get reassurances from the other 686 parties to the communication that they promise that appropriate 687 measures are taken. 689 o Security of the partners' identity: verifying that the 690 participants are who they say they are (when positive 691 identification is appropriate), or that their identity cannot be 692 uncovered (when anonymity is a goal of the application). 694 The security analysis, and the requirements derived from that 695 analysis, is contained in [I-D.ietf-rtcweb-security]. 697 It is also important to read the security sections of 698 [W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209]. 700 12. Acknowledgements 702 The number of people who have taken part in the discussions 703 surrounding this draft are too numerous to list, or even to identify. 704 The ones below have made special, identifiable contributions; this 705 does not mean that others' contributions are less important. 707 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 708 Westerlund and Joerg Ott, who offered technical contributions on 709 various versions of the draft. 711 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 712 the ASCII drawings in section 1. 714 Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric 715 Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage 716 and Simon Leinen for document review. 718 13. References 720 13.1. Normative References 722 [I-D.ietf-rtcweb-audio] 723 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 724 Requirements", draft-ietf-rtcweb-audio-11 (work in 725 progress), April 2016. 727 [I-D.ietf-rtcweb-data-channel] 728 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 729 Channels", draft-ietf-rtcweb-data-channel-13 (work in 730 progress), January 2015. 732 [I-D.ietf-rtcweb-data-protocol] 733 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 734 Establishment Protocol", draft-ietf-rtcweb-data- 735 protocol-09 (work in progress), January 2015. 737 [I-D.ietf-rtcweb-jsep] 738 Uberti, J., Jennings, C., and E. Rescorla, "Javascript 739 Session Establishment Protocol", draft-ietf-rtcweb-jsep-17 740 (work in progress), October 2016. 742 [I-D.ietf-rtcweb-rtp-usage] 743 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 744 Communication (WebRTC): Media Transport and Use of RTP", 745 draft-ietf-rtcweb-rtp-usage-26 (work in progress), March 746 2016. 748 [I-D.ietf-rtcweb-security] 749 Rescorla, E., "Security Considerations for WebRTC", draft- 750 ietf-rtcweb-security-08 (work in progress), February 2015. 752 [I-D.ietf-rtcweb-security-arch] 753 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 754 rtcweb-security-arch-12 (work in progress), June 2016. 756 [I-D.ietf-rtcweb-transports] 757 Alvestrand, H., "Transports for WebRTC", draft-ietf- 758 rtcweb-transports-17 (work in progress), October 2016. 760 [I-D.ietf-rtcweb-video] 761 Roach, A., "WebRTC Video Processing and Codec 762 Requirements", draft-ietf-rtcweb-video-06 (work in 763 progress), June 2015. 765 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 766 with Session Description Protocol (SDP)", RFC 3264, 767 DOI 10.17487/RFC3264, June 2002, 768 . 770 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 771 Jacobson, "RTP: A Transport Protocol for Real-Time 772 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 773 July 2003, . 775 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 776 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 777 RFC 3711, DOI 10.17487/RFC3711, March 2004, 778 . 780 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 781 (ICE): A Protocol for Network Address Translator (NAT) 782 Traversal for Offer/Answer Protocols", RFC 5245, 783 DOI 10.17487/RFC5245, April 2010, 784 . 786 [W3C.WD-mediacapture-streams-20120628] 787 Burnett, D. and A. Narayanan, "Media Capture and Streams", 788 World Wide Web Consortium WD WD-mediacapture-streams- 789 20120628, June 2012, . 792 [W3C.WD-webrtc-20120209] 793 Bergkvist, A., Burnett, D., Jennings, C., and A. 794 Narayanan, "WebRTC 1.0: Real-time Communication Between 795 Browsers", World Wide Web Consortium WD WD-webrtc- 796 20120209, February 2012, 797 . 799 13.2. Informative References 801 [I-D.ietf-rtcweb-gateways] 802 Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways", 803 draft-ietf-rtcweb-gateways-02 (work in progress), January 804 2016. 806 [I-D.ietf-rtcweb-use-cases-and-requirements] 807 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 808 Time Communication Use-cases and Requirements", draft- 809 ietf-rtcweb-use-cases-and-requirements-16 (work in 810 progress), January 2015. 812 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 813 A., Peterson, J., Sparks, R., Handley, M., and E. 814 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 815 DOI 10.17487/RFC3261, June 2002, 816 . 818 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", 819 BCP 95, RFC 3935, DOI 10.17487/RFC3935, October 2004, 820 . 822 [RFC6120] Saint-Andre, P., "Extensible Messaging and Presence 823 Protocol (XMPP): Core", RFC 6120, DOI 10.17487/RFC6120, 824 March 2011, . 826 [W3C.WD-html5-20110525] 827 Hickson, I., "HTML5", World Wide Web Consortium LastCall 828 WD-html5-20110525, May 2011, 829 . 831 Appendix A. Change log 833 This section may be deleted by the RFC Editor when preparing for 834 publication. 836 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 838 Added section "On interoperability and innovation" 840 Added data confidentiality and integrity to the "data framing" layer 842 Added congestion management requirements in the "data transport" 843 layer section 845 Changed need for non-media data from "question: do we need this?" to 846 "Open issue: How do we do this?" 848 Strengthened disclaimer that listed codecs are placeholders, not 849 decisions. 851 More details on why the "local system support functions" section is 852 there. 854 A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand- 855 rtcweb-overview-00 857 Added section on "Relationship between API and protocol" 859 Added terminology section 861 Mentioned congestion management as part of the "data transport" layer 862 in the layer list 864 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 866 Removed most technical content, and replaced with pointers to drafts 867 as requested and identified by the RTCWEB WG chairs. 869 Added content to acknowledgments section. 871 Added change log. 873 Spell-checked document. 875 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf- 876 rtcweb-overview-00 878 Changed draft name and document date. 880 Removed unused references 882 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 884 Added architecture figures to section 2. 886 Changed the description of "echo cancellation" under "local system 887 support functions". 889 Added a few more definitions. 891 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 893 Added pointers to use cases, security and rtp-usage drafts (now WG 894 drafts). 896 Changed description of SRTP from mandatory-to-use to mandatory-to- 897 implement. 899 Added the "3 principles of negotiation" to the connection management 900 section. 902 Added an explicit statement that ICE is required for both NAT and 903 consent-to-receive. 905 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 907 Added references to a number of new drafts. 909 Expanded the description text under the "trapezoid" drawing with some 910 more text discussed on the list. 912 Changed the "Connection management" sentence from "will be done using 913 SDP offer/answer" to "will be capable of representing SDP offer/ 914 answer" - this seems more consistent with JSEP. 916 Added "security mechanisms" to the things a non-gatewayed SIP devices 917 must support in order to not need a media gateway. 919 Added a definition for "browser". 921 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 923 Made introduction more normative. 925 Several wording changes in response to review comments from EKR 927 Added an appendix to hold references and notes that are not yet in a 928 separate document. 930 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 932 Minor grammatical fixes. This is mainly a "keepalive" refresh. 934 A.10. Changes from -05 to -06 936 Clarifications in response to Last Call review comments. Inserted 937 reference to draft-ietf-rtcweb-audio. 939 A.11. Changes from -06 to -07 941 Added a reference to the "unified plan" draft, and updated some 942 references. 944 Otherwise, it's a "keepalive" draft. 946 A.12. Changes from -07 to -08 948 Removed the appendix that detailed transports, and replaced it with a 949 reference to draft-ietf-rtcweb-transports. Removed now-unused 950 references. 952 A.13. Changes from -08 to -09 954 Added text to the Abstract indicating that the intended status is an 955 Applicability Statement. 957 A.14. Changes from -09 to -10 959 Defined "WebRTC Browser" and "WebRTC device" as things that do, or 960 don't, conform to the API. 962 Updated reference to data-protocol draft 964 Updated data formats to reference -rtcweb-audio- and not the expired 965 -cbran draft. 967 Deleted references to -unified-plan 968 Deleted reference to -generic-idp (draft expired) 970 Added notes on which referenced documents WebRTC browsers or devices 971 MUST conform to. 973 Added pointer to the security section of the API drafts. 975 A.15. Changes from -10 to -11 977 Added "WebRTC Gateway" as a third class of device, and referenced the 978 doc describing them. 980 Made a number of text clarifications in response to document reviews. 982 A.16. Changes from -11 to -12 984 Refined entity definitions to define "WebRTC endpoint" and "WebRTC- 985 compatible endpoint". 987 Changed remaining usage of the term "RTCWEB" to "WebRTC", including 988 in the page header. 990 A.17. Changes from -12 to -13 992 Changed "WebRTC device" to be "WebRTC non-browser", per decision at 993 IETF 91. This led to the need for "WebRTC endpoint" as the common 994 label for both, and the usage of that term in the rest of the 995 document. 997 Added words about WebRTC APIs in languages other than Javascript. 999 Referenced draft-ietf-rtcweb-video for video codecs to support. 1001 A.18. Changes from -13 to -14 1003 None. This is a "keepalive" update. 1005 A.19. Changes from -14 to -15 1007 Changed "gateways" reference to point to the WG document. 1009 A.20. Changes from -15 to -16 1011 None. This is a "keepalive" publication. 1013 Author's Address 1015 Harald T. Alvestrand 1016 Google 1017 Kungsbron 2 1018 Stockholm 11122 1019 Sweden 1021 Email: harald@alvestrand.no