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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 March 3, 2017 5 Expires: September 4, 2017 7 Overview: Real Time Protocols for Browser-based Applications 8 draft-ietf-rtcweb-overview-18 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 September 4, 2017. 44 Copyright Notice 46 Copyright (c) 2017 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 . . . . . 20 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 . . . . . . . . . . . . . . . . . 22 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 . . . . . . . . . . . . . . . . . 23 102 A.19. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 23 103 A.20. Changes from -15 to -16 . . . . . . . . . . . . . . . . . 23 104 A.21. Changes from -16 to -17 . . . . . . . . . . . . . . . . . 23 105 A.22. Changes from -17 to -18 . . . . . . . . . . . . . . . . . 23 106 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 23 108 1. Introduction 110 The Internet was, from very early in its lifetime, considered a 111 possible vehicle for the deployment of real-time, interactive 112 applications - with the most easily imaginable being audio 113 conversations (aka "Internet telephony") and video conferencing. 115 The first attempts to build this were dependent on special networks, 116 special hardware and custom-built software, often at very high prices 117 or at low quality, placing great demands on the infrastructure. 119 As the available bandwidth has increased, and as processors an other 120 hardware has become ever faster, the barriers to participation have 121 decreased, and it has become possible to deliver a satisfactory 122 experience on commonly available computing hardware. 124 Still, there are a number of barriers to the ability to communicate 125 universally - one of these is that there is, as of yet, no single set 126 of communication protocols that all agree should be made available 127 for communication; another is the sheer lack of universal 128 identification systems (such as is served by telephone numbers or 129 email addresses in other communications systems). 131 Development of The Universal Solution has proved hard, however, for 132 all the usual reasons. 134 The last few years have also seen a new platform rise for deployment 135 of services: The browser-embedded application, or "Web application". 136 It turns out that as long as the browser platform has the necessary 137 interfaces, it is possible to deliver almost any kind of service on 138 it. 140 Traditionally, these interfaces have been delivered by plugins, which 141 had to be downloaded and installed separately from the browser; in 142 the development of HTML5, application developers see much promise in 143 the possibility of making those interfaces available in a 144 standardized way within the browser. 146 This memo describes a set of building blocks that can be made 147 accessible and controllable through a Javascript API in a browser, 148 and which together form a sufficient set of functions to allow the 149 use of interactive audio and video in applications that communicate 150 directly between browsers across the Internet. The resulting 151 protocol suite is intended to enable all the applications that are 152 described as required scenarios in the use cases document [RFC7478]. 154 Other efforts, for instance the W3C WEBRTC, Web Applications and 155 Device API working groups, focus on making standardized APIs and 156 interfaces available, within or alongside the HTML5 effort, for those 157 functions; this memo concentrates on specifying the protocols and 158 subprotocols that are needed to specify the interactions that happen 159 across the network. 161 This memo uses the term "WebRTC" (note the case used) to refer to the 162 overall effort consisting of both IETF and W3C efforts. 164 2. Principles and Terminology 166 2.1. Goals of this document 168 The goal of the WebRTC protocol specification is to specify a set of 169 protocols that, if all are implemented, will allow an implementation 170 to communicate with another implementation using audio, video and 171 data sent along the most direct possible path between the 172 participants. 174 This document is intended to serve as the roadmap to the WebRTC 175 specifications. It defines terms used by other parts of the WebRTC 176 protocol specifications, lists references to other specifications 177 that don't need further elaboration in the WebRTC context, and gives 178 pointers to other documents that form part of the WebRTC suite. 180 By reading this document and the documents it refers to, it should be 181 possible to have all information needed to implement an WebRTC 182 compatible implementation. 184 2.2. Relationship between API and protocol 186 The total WebRTC effort consists of two major parts, each consisting 187 of multiple documents: 189 o A protocol specification, done in the IETF 191 o A Javascript API specification, defined in a series of W3C 192 documents 193 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 195 Together, these two specifications aim to provide an environment 196 where Javascript embedded in any page, when suitably authorized by 197 its user, is able to set up communication using audio, video and 198 auxiliary data, as long as the browser supports this specification. 199 The browser environment does not constrain the types of application 200 in which this functionality can be used. 202 The protocol specification does not assume that all implementations 203 implement this API; it is not intended to be necessary for 204 interoperation to know whether the entity one is communicating with 205 is a browser or another device implementing this specification. 207 The goal of cooperation between the protocol specification and the 208 API specification is that for all options and features of the 209 protocol specification, it should be clear which API calls to make to 210 exercise that option or feature; similarly, for any sequence of API 211 calls, it should be clear which protocol options and features will be 212 invoked. Both subject to constraints of the implementation, of 213 course. 215 For the purpose of this document, we define the following terminology 216 to talk about WebRTC things: 218 o A WebRTC browser (also called a WebRTC User Agent or WebRTC UA) is 219 something that conforms to both the protocol specification and the 220 Javascript API cited above. 222 o A WebRTC non-browser is something that conforms to the protocol 223 specification, but does not claim to implement the Javascript API. 224 This can also be called a "WebRTC device" or "WebRTC native 225 application". 227 o A WebRTC endpoint is either a WebRTC browser or a WebRTC non- 228 browser. It conforms to the protocol specification. 230 o A WebRTC-compatible endpoint is an endpoint that is able to 231 successfully communicate with a WebRTC endpoint, but may fail to 232 meet some requirements of a WebRTC endpoint. This may limit where 233 in the network such an endpoint can be attached, or may limit the 234 security guarantees that it offers to others. It is not 235 constrained by this specification; when it is mentioned at all, it 236 is to note the implications on WebRTC-compatible endpoints of the 237 requirements placed on WebRTC endpoints. 239 o A WebRTC gateway is a WebRTC-compatible endpoint that mediates 240 media traffic to non-WebRTC entities. 242 All WebRTC browsers are WebRTC endpoints, so any requirement on a 243 WebRTC endpoint also applies to a WebRTC browser. 245 A WebRTC non-browser may be capable of hosting applications in a 246 similar way to the way in which a browser can host Javascript 247 applications, typically by offering APIs in other languages. For 248 instance it may be implemented as a library that offers a C++ API 249 intended to be loaded into applications. In this case, similar 250 security considerations as for Javascript may be needed; however, 251 since such APIs are not defined or referenced here, this document 252 cannot give any specific rules for those interfaces. 254 WebRTC gateways are described in a separate document, 255 [I-D.ietf-rtcweb-gateways]. 257 2.3. On interoperability and innovation 259 The "Mission statement of the IETF" [RFC3935] states that "The 260 benefit of a standard to the Internet is in interoperability - that 261 multiple products implementing a standard are able to work together 262 in order to deliver valuable functions to the Internet's users." 264 Communication on the Internet frequently occurs in two phases: 266 o Two parties communicate, through some mechanism, what 267 functionality they both are able to support 269 o They use that shared communicative functionality to communicate, 270 or, failing to find anything in common, give up on communication. 272 There are often many choices that can be made for communicative 273 functionality; the history of the Internet is rife with the proposal, 274 standardization, implementation, and success or failure of many types 275 of options, in all sorts of protocols. 277 The goal of having a mandatory to implement function set is to 278 prevent negotiation failure, not to preempt or prevent negotiation. 280 The presence of a mandatory to implement function set serves as a 281 strong changer of the marketplace of deployment - in that it gives a 282 guarantee that, as long as you conform to a specification, and the 283 other party is willing to accept communication at the base level of 284 that specification, you can communicate successfully. 286 The alternative - that of having no mandatory to implement - does not 287 mean that you cannot communicate, it merely means that in order to be 288 part of the communications partnership, you have to implement the 289 standard "and then some" - that "and then some" usually being called 290 a profile of some sort; in the version most antithetical to the 291 Internet ethos, that "and then some" consists of having to use a 292 specific vendor's product only. 294 2.4. Terminology 296 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 297 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 298 document are to be interpreted as described in [RFC2119]. 300 The following terms are used across the documents specifying the 301 WebRTC suite, in the specific meanings given here. Not all terms are 302 used in this document. Other terms are used in their commonly used 303 meaning. 305 The list is in alphabetical order. 307 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 309 API: Application Programming Interface - a specification of a set of 310 calls and events, usually tied to a programming language or an 311 abstract formal specification such as WebIDL, with its defined 312 semantics. 314 Browser: Used synonymously with "Interactive User Agent" as defined 315 in the HTML specification [W3C.WD-html5-20110525]. See also 316 "WebRTC User Agent". 318 Data channel: An abstraction that allows data to be sent between 319 WebRTC endpoints in the form of messages. Two endpoints can have 320 multiple data channels between them. 322 ICE Agent: An implementation of the Interactive Connectivty 323 Establishment (ICE) [I-D.ietf-ice-rfc5245bis] protocol. An ICE 324 Agent may also be an SDP Agent, but there exist ICE Agents that do 325 not use SDP (for instance those that use Jingle [XEP-0166]). 327 Interactive: Communication between multiple parties, where the 328 expectation is that an action from one party can cause a reaction 329 by another party, and the reaction can be observed by the first 330 party, with the total time required for the action/reaction/ 331 observation is on the order of no more than hundreds of 332 milliseconds. 334 Media: Audio and video content. Not to be confused with 335 "transmission media" such as wires. 337 Media path: The path that media data follows from one WebRTC 338 endpoint to another. 340 Protocol: A specification of a set of data units, their 341 representation, and rules for their transmission, with their 342 defined semantics. A protocol is usually thought of as going 343 between systems. 345 Real-time media: Media where generation of content and display of 346 content are intended to occur closely together in time (on the 347 order of no more than hundreds of milliseconds). Real-time media 348 can be used to support interactive communication. 350 SDP Agent: The protocol implementation involved in the SDP offer/ 351 answer exchange, as defined in [RFC3264] section 3. 353 Signaling: Communication that happens in order to establish, manage 354 and control media paths and data paths. 356 Signaling Path: The communication channels used between entities 357 participating in signaling to transfer signaling. There may be 358 more entities in the signaling path than in the media path. 360 3. Architecture and Functionality groups 362 The model of real-time support for browser-based applications does 363 not assume that the browser will contain all the functions that need 364 to be performed in order to have a function such as a telephone or a 365 video conferencing unit; the vision is that the browser will have the 366 functions that are needed for a Web application, working in 367 conjunction with its backend servers, to implement these functions. 369 This means that two vital interfaces need specification: The 370 protocols that browsers use to talk to each other, without any 371 intervening servers, and the APIs that are offered for a Javascript 372 application to take advantage of the browser's functionality. 374 +------------------------+ On-the-wire 375 | | Protocols 376 | Servers |---------> 377 | | 378 | | 379 +------------------------+ 380 ^ 381 | 382 | 383 | HTTP/ 384 | WebSockets 385 | 386 | 387 +----------------------------+ 388 | Javascript/HTML/CSS | 389 +----------------------------+ 390 Other ^ ^RTC 391 APIs | |APIs 392 +---|-----------------|------+ 393 | | | | 394 | +---------+| 395 | | Browser || On-the-wire 396 | Browser | RTC || Protocols 397 | | Function|-----------> 398 | | || 399 | | || 400 | +---------+| 401 +---------------------|------+ 402 | 403 V 404 Native OS Services 406 Figure 1: Browser Model 408 Note that HTTP and WebSockets are also offered to the Javascript 409 application through browser APIs. 411 As for all protocol and API specifications, there is no restriction 412 that the protocols can only be used to talk to another browser; since 413 they are fully specified, any endpoint that implements the protocols 414 faithfully should be able to interoperate with the application 415 running in the browser. 417 A commonly imagined model of deployment is the one depicted below. 419 +-----------+ +-----------+ 420 | Web | | Web | 421 | | Signaling | | 422 | |-------------| | 423 | Server | path | Server | 424 | | | | 425 +-----------+ +-----------+ 426 / \ 427 / \ Application-defined 428 / \ over 429 / \ HTTP/WebSockets 430 / Application-defined over \ 431 / HTTP/WebSockets \ 432 / \ 433 +-----------+ +-----------+ 434 |JS/HTML/CSS| |JS/HTML/CSS| 435 +-----------+ +-----------+ 436 +-----------+ +-----------+ 437 | | | | 438 | | | | 439 | Browser | ------------------------- | Browser | 440 | | Media path | | 441 | | | | 442 +-----------+ +-----------+ 444 Figure 2: Browser RTC Trapezoid 446 On this drawing, the critical part to note is that the media path 447 ("low path") goes directly between the browsers, so it has to be 448 conformant to the specifications of the WebRTC protocol suite; the 449 signaling path ("high path") goes via servers that can modify, 450 translate or massage the signals as needed. 452 If the two Web servers are operated by different entities, the inter- 453 server signaling mechanism needs to be agreed upon, either by 454 standardization or by other means of agreement. Existing protocols 455 (for example SIP [RFC3261] or XMPP [RFC6120]) could be used between 456 servers, while either a standards-based or proprietary protocol could 457 be used between the browser and the web server. 459 For example, if both operators' servers implement SIP, SIP could be 460 used for communication between servers, along with either a 461 standardized signaling mechanism (e.g. SIP over WebSockets) or a 462 proprietary signaling mechanism used between the application running 463 in the browser and the web server. Similarly, if both operators' 464 servers implement XMPP, XMPP could be used for communication between 465 XMPP servers, with either a standardized signaling mechanism (e.g. 466 XMPP over WebSockets or BOSH) or a proprietary signaling mechanism 467 used between the application running in the browser and the web 468 server. 470 The choice of protocols for client-server and inter-server 471 signalling, and definition of the translation between them, is 472 outside the scope of the WebRTC protocol suite described in the 473 document. 475 The functionality groups that are needed in the browser can be 476 specified, more or less from the bottom up, as: 478 o Data transport: TCP, UDP and the means to securely set up 479 connections between entities, as well as the functions for 480 deciding when to send data: Congestion management, bandwidth 481 estimation and so on. 483 o Data framing: RTP, SCTP and other data formats that serve as 484 containers, and their functions for data confidentiality and 485 integrity. 487 o Data formats: Codec specifications, format specifications and 488 functionality specifications for the data passed between systems. 489 Audio and video codecs, as well as formats for data and document 490 sharing, belong in this category. In order to make use of data 491 formats, a way to describe them, a session description, is needed. 493 o Connection management: Setting up connections, agreeing on data 494 formats, changing data formats during the duration of a call; SIP 495 and Jingle/XMPP belong in this category. 497 o Presentation and control: What needs to happen in order to ensure 498 that interactions behave in a non-surprising manner. This can 499 include floor control, screen layout, voice activated image 500 switching and other such functions - where part of the system 501 require the cooperation between parties. XCON and Cisco/ 502 Tandberg's TIP were some attempts at specifying this kind of 503 functionality; many applications have been built without 504 standardized interfaces to these functions. 506 o Local system support functions: These are things that need not be 507 specified uniformly, because each participant may choose to do 508 these in a way of the participant's choosing, without affecting 509 the bits on the wire in a way that others have to be cognizant of. 510 Examples in this category include echo cancellation (some forms of 511 it), local authentication and authorization mechanisms, OS access 512 control and the ability to do local recording of conversations. 514 Within each functionality group, it is important to preserve both 515 freedom to innovate and the ability for global communication. 516 Freedom to innovate is helped by doing the specification in terms of 517 interfaces, not implementation; any implementation able to 518 communicate according to the interfaces is a valid implementation. 519 Ability to communicate globally is helped both by having core 520 specifications be unencumbered by IPR issues and by having the 521 formats and protocols be fully enough specified to allow for 522 independent implementation. 524 One can think of the three first groups as forming a "media transport 525 infrastructure", and of the three last groups as forming a "media 526 service". In many contexts, it makes sense to use a common 527 specification for the media transport infrastructure, which can be 528 embedded in browsers and accessed using standard interfaces, and "let 529 a thousand flowers bloom" in the "media service" layer; to achieve 530 interoperable services, however, at least the first five of the six 531 groups need to be specified. 533 4. Data transport 535 Data transport refers to the sending and receiving of data over the 536 network interfaces, the choice of network-layer addresses at each end 537 of the communication, and the interaction with any intermediate 538 entities that handle the data, but do not modify it (such as TURN 539 relays). 541 It includes necessary functions for congestion control: When not to 542 send data. 544 WebRTC endpoints MUST implement the transport protocols described in 545 [I-D.ietf-rtcweb-transports]. 547 5. Data framing and securing 549 The format for media transport is RTP [RFC3550]. Implementation of 550 SRTP [RFC3711] is REQUIRED for all implementations. 552 The detailed considerations for usage of functions from RTP and SRTP 553 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 554 considerations for the WebRTC use case are in 555 [I-D.ietf-rtcweb-security], and the resulting security functions are 556 described in [I-D.ietf-rtcweb-security-arch]. 558 Considerations for the transfer of data that is not in RTP format is 559 described in [I-D.ietf-rtcweb-data-channel], and a supporting 560 protocol for establishing individual data channels is described in 561 [I-D.ietf-rtcweb-data-protocol]. WebRTC endpoints MUST implement 562 these two specifications. 564 WebRTC endpoints MUST implement [I-D.ietf-rtcweb-rtp-usage], 565 [I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the 566 requirements they include. 568 6. Data formats 570 The intent of this specification is to allow each communications 571 event to use the data formats that are best suited for that 572 particular instance, where a format is supported by both sides of the 573 connection. However, a minimum standard is greatly helpful in order 574 to ensure that communication can be achieved. This document 575 specifies a minimum baseline that will be supported by all 576 implementations of this specification, and leaves further codecs to 577 be included at the will of the implementor. 579 WebRTC endpoints that support audio and/or video MUST implement the 580 codecs and profiles required in [RFC7874] and [RFC7742]. 582 7. Connection management 584 The methods, mechanisms and requirements for setting up, negotiating 585 and tearing down connections is a large subject, and one where it is 586 desirable to have both interoperability and freedom to innovate. 588 The following principles apply: 590 1. The WebRTC media negotiations will be capable of representing the 591 same SDP offer/answer semantics that are used in SIP [RFC3264], 592 in such a way that it is possible to build a signaling gateway 593 between SIP and the WebRTC media negotiation. 595 2. It will be possible to gateway between legacy SIP devices that 596 support ICE and appropriate RTP / SDP mechanisms, codecs and 597 security mechanisms without using a media gateway. A signaling 598 gateway to convert between the signaling on the web side to the 599 SIP signaling may be needed. 601 3. When a new codec is specified, and the SDP for the new codec is 602 specified in the MMUSIC WG, no other standardization should be 603 required for it to be possible to use that in the web browsers. 604 Adding new codecs which might have new SDP parameters should not 605 change the APIs between the browser and Javascript application. 607 As soon as the browsers support the new codecs, old applications 608 written before the codecs were specified should automatically be 609 able to use the new codecs where appropriate with no changes to 610 the JS applications. 612 The particular choices made for WebRTC, and their implications for 613 the API offered by a browser implementing WebRTC, are described in 614 [I-D.ietf-rtcweb-jsep]. 616 WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. 618 WebRTC endpoints MUST implement the functions described in that 619 document that relate to the network layer (for example Bundle 620 [I-D.ietf-mmusic-sdp-bundle-negotiation], RTCP-mux [RFC5761] and 621 Trickle ICE [I-D.ietf-ice-trickle]), but do not need to support the 622 API functionality described there. 624 8. Presentation and control 626 The most important part of control is the user's control over the 627 browser's interaction with input/output devices and communications 628 channels. It is important that the user have some way of figuring 629 out where his audio, video or texting is being sent, for what 630 purported reason, and what guarantees are made by the parties that 631 form part of this control channel. This is largely a local function 632 between the browser, the underlying operating system and the user 633 interface; this is specified in the peer connection API 634 [W3C.WD-webrtc-20120209], and the media capture API 635 [W3C.WD-mediacapture-streams-20120628]. 637 WebRTC browsers MUST implement these two specifications. 639 9. Local system support functions 641 These are characterized by the fact that the quality of these 642 functions strongly influence the user experience, but the exact 643 algorithm does not need coordination. In some cases (for instance 644 echo cancellation, as described below), the overall system definition 645 may need to specify that the overall system needs to have some 646 characteristics for which these facilities are useful, without 647 requiring them to be implemented a certain way. 649 Local functions include echo cancellation, volume control, camera 650 management including focus, zoom, pan/tilt controls (if available), 651 and more. 653 One would want to see certain parts of the system conform to certain 654 properties, for instance: 656 o Echo cancellation should be good enough to achieve the suppression 657 of acoustical feedback loops below a perceptually noticeable 658 level. 660 o Privacy concerns MUST be satisfied; for instance, if remote 661 control of camera is offered, the APIs should be available to let 662 the local participant figure out who's controlling the camera, and 663 possibly decide to revoke the permission for camera usage. 665 o Automatic gain control, if present, should normalize a speaking 666 voice into a reasonable dB range. 668 The requirements on WebRTC systems with regard to audio processing 669 are found in [RFC7874] and includes more guidance about echo 670 cancellation and AGC; the proposed API for control of local devices 671 are found in [W3C.WD-mediacapture-streams-20120628]. 673 WebRTC endpoints MUST implement the processing functions in 674 [RFC7874]. (Together with the requirement in Section 6, this means 675 that WebRTC endpoints MUST implement the whole document.) 677 10. IANA Considerations 679 This document makes no request of IANA. 681 Note to RFC Editor: this section may be removed on publication as an 682 RFC. 684 11. Security Considerations 686 Security of the web-enabled real time communications comes in several 687 pieces: 689 o Security of the components: The browsers, and other servers 690 involved. The most target-rich environment here is probably the 691 browser; the aim here should be that the introduction of these 692 components introduces no additional vulnerability. 694 o Security of the communication channels: It should be easy for a 695 participant to reassure himself of the security of his 696 communication - by verifying the crypto parameters of the links he 697 himself participates in, and to get reassurances from the other 698 parties to the communication that they promise that appropriate 699 measures are taken. 701 o Security of the partners' identity: verifying that the 702 participants are who they say they are (when positive 703 identification is appropriate), or that their identity cannot be 704 uncovered (when anonymity is a goal of the application). 706 The security analysis, and the requirements derived from that 707 analysis, is contained in [I-D.ietf-rtcweb-security]. 709 It is also important to read the security sections of 710 [W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209]. 712 12. Acknowledgements 714 The number of people who have taken part in the discussions 715 surrounding this draft are too numerous to list, or even to identify. 716 The ones below have made special, identifiable contributions; this 717 does not mean that others' contributions are less important. 719 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 720 Westerlund and Joerg Ott, who offered technical contributions on 721 various versions of the draft. 723 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 724 the ASCII drawings in section 1. 726 Thanks to Alissa Cooper, Bjoern Hoehrmann, Colin Perkins, Colton 727 Shields, Eric Rescorla, Heath Matlock, Henry Sinnreich, Justin 728 Uberti, Keith Drage, Magnus Westerlund, Olle E. Johansson, Sean 729 Turner and Simon Leinen for document review. 731 13. References 733 13.1. Normative References 735 [I-D.ietf-ice-rfc5245bis] 736 Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive 737 Connectivity Establishment (ICE): A Protocol for Network 738 Address Translator (NAT) Traversal", draft-ietf-ice- 739 rfc5245bis-08 (work in progress), December 2016. 741 [I-D.ietf-rtcweb-data-channel] 742 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 743 Channels", draft-ietf-rtcweb-data-channel-11 (work in 744 progress), July 2014. 746 [I-D.ietf-rtcweb-data-protocol] 747 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 748 Establishment Protocol", draft-ietf-rtcweb-data- 749 protocol-07 (work in progress), July 2014. 751 [I-D.ietf-rtcweb-jsep] 752 Uberti, J., Jennings, C., and E. Rescorla, "Javascript 753 Session Establishment Protocol", draft-ietf-rtcweb-jsep-07 754 (work in progress), July 2014. 756 [I-D.ietf-rtcweb-rtp-usage] 757 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 758 Communication (WebRTC): Media Transport and Use of RTP", 759 draft-ietf-rtcweb-rtp-usage-16 (work in progress), July 760 2014. 762 [I-D.ietf-rtcweb-security] 763 Rescorla, E., "Security Considerations for WebRTC", draft- 764 ietf-rtcweb-security-07 (work in progress), July 2014. 766 [I-D.ietf-rtcweb-security-arch] 767 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 768 rtcweb-security-arch-10 (work in progress), July 2014. 770 [I-D.ietf-rtcweb-transports] 771 Alvestrand, H., "Transports for WebRTC", draft-ietf- 772 rtcweb-transports-06 (work in progress), August 2014. 774 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 775 Requirement Levels", BCP 14, RFC 2119, March 1997. 777 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 778 with Session Description Protocol (SDP)", RFC 3264, June 779 2002. 781 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 782 Jacobson, "RTP: A Transport Protocol for Real-Time 783 Applications", STD 64, RFC 3550, July 2003. 785 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 786 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 787 RFC 3711, March 2004. 789 [RFC7742] Roach, A., "WebRTC Video Processing and Codec 790 Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016, 791 . 793 [RFC7874] Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing 794 Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016, 795 . 797 [W3C.WD-mediacapture-streams-20120628] 798 Burnett, D. and A. Narayanan, "Media Capture and Streams", 799 World Wide Web Consortium WD WD-mediacapture-streams- 800 20120628, June 2012, . 803 [W3C.WD-webrtc-20120209] 804 Bergkvist, A., Burnett, D., Jennings, C., and A. 805 Narayanan, "WebRTC 1.0: Real-time Communication Between 806 Browsers", World Wide Web Consortium WD WD-webrtc- 807 20120209, February 2012, 808 . 810 13.2. Informative References 812 [I-D.ietf-ice-trickle] 813 Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre, 814 "Trickle ICE: Incremental Provisioning of Candidates for 815 the Interactive Connectivity Establishment (ICE) 816 Protocol", draft-ietf-ice-trickle-07 (work in progress), 817 February 2017. 819 [I-D.ietf-mmusic-sdp-bundle-negotiation] 820 Holmberg, C., Alvestrand, H., and C. Jennings, 821 "Negotiating Media Multiplexing Using the Session 822 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 823 negotiation-07 (work in progress), April 2014. 825 [I-D.ietf-rtcweb-gateways] 826 Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways", 827 draft-ietf-rtcweb-gateways-02 (work in progress), January 828 2016. 830 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 831 A., Peterson, J., Sparks, R., Handley, M., and E. 832 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 833 June 2002. 835 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", BCP 836 95, RFC 3935, October 2004. 838 [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and 839 Control Packets on a Single Port", RFC 5761, April 2010. 841 [RFC6120] Saint-Andre, P., "Extensible Messaging and Presence 842 Protocol (XMPP): Core", RFC 6120, March 2011. 844 [RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 845 Time Communication Use Cases and Requirements", RFC 7478, 846 DOI 10.17487/RFC7478, March 2015, 847 . 849 [W3C.WD-html5-20110525] 850 Hickson, I., "HTML5", World Wide Web Consortium LastCall 851 WD-html5-20110525, May 2011, 852 . 854 [XEP-0166] 855 Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan, 856 S., and J. Hildebrand, "Jingle", XSF XEP 0166, June 2007. 858 Appendix A. Change log 860 This section may be deleted by the RFC Editor when preparing for 861 publication. 863 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 865 Added section "On interoperability and innovation" 867 Added data confidentiality and integrity to the "data framing" layer 869 Added congestion management requirements in the "data transport" 870 layer section 872 Changed need for non-media data from "question: do we need this?" to 873 "Open issue: How do we do this?" 875 Strengthened disclaimer that listed codecs are placeholders, not 876 decisions. 878 More details on why the "local system support functions" section is 879 there. 881 A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand- 882 rtcweb-overview-00 884 Added section on "Relationship between API and protocol" 886 Added terminology section 888 Mentioned congestion management as part of the "data transport" layer 889 in the layer list 891 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 893 Removed most technical content, and replaced with pointers to drafts 894 as requested and identified by the RTCWEB WG chairs. 896 Added content to acknowledgments section. 898 Added change log. 900 Spell-checked document. 902 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf- 903 rtcweb-overview-00 905 Changed draft name and document date. 907 Removed unused references 909 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 911 Added architecture figures to section 2. 913 Changed the description of "echo cancellation" under "local system 914 support functions". 916 Added a few more definitions. 918 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 920 Added pointers to use cases, security and rtp-usage drafts (now WG 921 drafts). 923 Changed description of SRTP from mandatory-to-use to mandatory-to- 924 implement. 926 Added the "3 principles of negotiation" to the connection management 927 section. 929 Added an explicit statement that ICE is required for both NAT and 930 consent-to-receive. 932 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 934 Added references to a number of new drafts. 936 Expanded the description text under the "trapezoid" drawing with some 937 more text discussed on the list. 939 Changed the "Connection management" sentence from "will be done using 940 SDP offer/answer" to "will be capable of representing SDP offer/ 941 answer" - this seems more consistent with JSEP. 943 Added "security mechanisms" to the things a non-gatewayed SIP devices 944 must support in order to not need a media gateway. 946 Added a definition for "browser". 948 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 950 Made introduction more normative. 952 Several wording changes in response to review comments from EKR 954 Added an appendix to hold references and notes that are not yet in a 955 separate document. 957 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 959 Minor grammatical fixes. This is mainly a "keepalive" refresh. 961 A.10. Changes from -05 to -06 963 Clarifications in response to Last Call review comments. Inserted 964 reference to draft-ietf-rtcweb-audio. 966 A.11. Changes from -06 to -07 968 Added a reference to the "unified plan" draft, and updated some 969 references. 971 Otherwise, it's a "keepalive" draft. 973 A.12. Changes from -07 to -08 975 Removed the appendix that detailed transports, and replaced it with a 976 reference to draft-ietf-rtcweb-transports. Removed now-unused 977 references. 979 A.13. Changes from -08 to -09 981 Added text to the Abstract indicating that the intended status is an 982 Applicability Statement. 984 A.14. Changes from -09 to -10 986 Defined "WebRTC Browser" and "WebRTC device" as things that do, or 987 don't, conform to the API. 989 Updated reference to data-protocol draft 991 Updated data formats to reference -rtcweb-audio- and not the expired 992 -cbran draft. 994 Deleted references to -unified-plan 996 Deleted reference to -generic-idp (draft expired) 998 Added notes on which referenced documents WebRTC browsers or devices 999 MUST conform to. 1001 Added pointer to the security section of the API drafts. 1003 A.15. Changes from -10 to -11 1005 Added "WebRTC Gateway" as a third class of device, and referenced the 1006 doc describing them. 1008 Made a number of text clarifications in response to document reviews. 1010 A.16. Changes from -11 to -12 1012 Refined entity definitions to define "WebRTC endpoint" and "WebRTC- 1013 compatible endpoint". 1015 Changed remaining usage of the term "RTCWEB" to "WebRTC", including 1016 in the page header. 1018 A.17. Changes from -12 to -13 1020 Changed "WebRTC device" to be "WebRTC non-browser", per decision at 1021 IETF 91. This led to the need for "WebRTC endpoint" as the common 1022 label for both, and the usage of that term in the rest of the 1023 document. 1025 Added words about WebRTC APIs in languages other than Javascript. 1027 Referenced draft-ietf-rtcweb-video for video codecs to support. 1029 A.18. Changes from -13 to -14 1031 None. This is a "keepalive" update. 1033 A.19. Changes from -14 to -15 1035 Changed "gateways" reference to point to the WG document. 1037 A.20. Changes from -15 to -16 1039 None. This is a "keepalive" publication. 1041 A.21. Changes from -16 to -17 1043 Addressed review comments by Olle E. Johansson and Magnus Westerlund 1045 A.22. Changes from -17 to -18 1047 Addressed review comments from Sean Turner and Alissa Cooper 1049 Author's Address 1051 Harald T. Alvestrand 1052 Google 1053 Kungsbron 2 1054 Stockholm 11122 1055 Sweden 1057 Email: harald@alvestrand.no