idnits 2.17.1 draft-ietf-rtcweb-overview-15.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 533: '...WebRTC endpoints MUST implement the tr...' RFC 2119 keyword, line 539: '... SRTP [RFC3711] is REQUIRED for all implementations....' RFC 2119 keyword, line 550: '... [I-D.ietf-rtcweb-data-protocol]. WebRTC endpoints MUST implement...' RFC 2119 keyword, line 553: '... WebRTC endpoints MUST implement [I-D.ietf-rtcweb-rtp-usage],...' RFC 2119 keyword, line 568: '... audio and/or video MUST implement the...' (8 more instances...) Miscellaneous warnings: ---------------------------------------------------------------------------- == The copyright year in the IETF Trust and authors Copyright Line does not match the current year -- The document date (January 21, 2016) is 3015 days in the past. Is this intentional? 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(See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Outdated reference: A later version (-11) exists of draft-ietf-rtcweb-audio-05 == Outdated reference: A later version (-13) exists of draft-ietf-rtcweb-data-channel-11 == Outdated reference: A later version (-09) exists of draft-ietf-rtcweb-data-protocol-07 == Outdated reference: A later version (-26) exists of draft-ietf-rtcweb-jsep-07 == Outdated reference: A later version (-26) exists of draft-ietf-rtcweb-rtp-usage-16 == Outdated reference: A later version (-12) exists of draft-ietf-rtcweb-security-07 == Outdated reference: A later version (-20) exists of draft-ietf-rtcweb-security-arch-10 == Outdated reference: A later version (-17) exists of draft-ietf-rtcweb-transports-06 == Outdated reference: A later version (-06) exists of draft-ietf-rtcweb-video-00 ** Obsolete normative reference: RFC 5245 (Obsoleted by RFC 8445, RFC 8839) == Outdated reference: A later version (-02) exists of draft-ietf-rtcweb-gateways-01 == Outdated reference: A later version (-16) exists of draft-ietf-rtcweb-use-cases-and-requirements-14 Summary: 2 errors (**), 0 flaws (~~), 12 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 January 21, 2016 5 Expires: July 24, 2016 7 Overview: Real Time Protocols for Browser-based Applications 8 draft-ietf-rtcweb-overview-15 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 July 24, 2016. 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 . . . . . . . . . . . . . . . . . . . . . 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 A.19. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 22 103 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 22 105 1. Introduction 107 The Internet was, from very early in its lifetime, considered a 108 possible vehicle for the deployment of real-time, interactive 109 applications - with the most easily imaginable being audio 110 conversations (aka "Internet telephony") and video conferencing. 112 The first attempts to build this were dependent on special networks, 113 special hardware and custom-built software, often at very high prices 114 or at low quality, placing great demands on the infrastructure. 116 As the available bandwidth has increased, and as processors an other 117 hardware has become ever faster, the barriers to participation have 118 decreased, and it has become possible to deliver a satisfactory 119 experience on commonly available computing hardware. 121 Still, there are a number of barriers to the ability to communicate 122 universally - one of these is that there is, as of yet, no single set 123 of communication protocols that all agree should be made available 124 for communication; another is the sheer lack of universal 125 identification systems (such as is served by telephone numbers or 126 email addresses in other communications systems). 128 Development of The Universal Solution has proved hard, however, for 129 all the usual reasons. 131 The last few years have also seen a new platform rise for deployment 132 of services: The browser-embedded application, or "Web application". 133 It turns out that as long as the browser platform has the necessary 134 interfaces, it is possible to deliver almost any kind of service on 135 it. 137 Traditionally, these interfaces have been delivered by plugins, which 138 had to be downloaded and installed separately from the browser; in 139 the development of HTML5, application developers see much promise in 140 the possibility of making those interfaces available in a 141 standardized way within the browser. 143 This memo describes a set of building blocks that can be made 144 accessible and controllable through a Javascript API in a browser, 145 and which together form a sufficient set of functions to allow the 146 use of interactive audio and video in applications that communicate 147 directly between browsers across the Internet. The resulting 148 protocol suite is intended to enable all the applications that are 149 described as required scenarios in the use cases document 150 [I-D.ietf-rtcweb-use-cases-and-requirements]. 152 Other efforts, for instance the W3C WEBRTC, Web Applications and 153 Device API working groups, focus on making standardized APIs and 154 interfaces available, within or alongside the HTML5 effort, for those 155 functions; this memo concentrates on specifying the protocols and 156 subprotocols that are needed to specify the interactions that happen 157 across the network. 159 This memo uses the term "WebRTC" (note the case used) to refer to the 160 overall effort consisting of both IETF and W3C efforts. 162 2. Principles and Terminology 164 2.1. Goals of this document 166 The goal of the WebRTC protocol specification is to specify a set of 167 protocols that, if all are implemented, will allow an implementation 168 to communicate with another implementation using audio, video and 169 data sent along the most direct possible path between the 170 participants. 172 This document is intended to serve as the roadmap to the WebRTC 173 specifications. It defines terms used by other pieces of 174 specification, lists references to other specifications that don't 175 need further elaboration in the WebRTC context, and gives pointers to 176 other documents that form part of the WebRTC suite. 178 By reading this document and the documents it refers to, it should be 179 possible to have all information needed to implement an WebRTC 180 compatible implementation. 182 2.2. Relationship between API and protocol 184 The total WebRTC effort consists of two pieces: 186 o A protocol specification, done in the IETF 188 o A Javascript API specification, done in the W3C 189 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 191 Together, these two specifications aim to provide an environment 192 where Javascript embedded in any page, viewed in any compatible 193 browser, when suitably authorized by its user, is able to set up 194 communication using audio, video and auxiliary data, where the 195 browser environment does not constrain the types of application in 196 which this functionality can be used. 198 The protocol specification does not assume that all implementations 199 implement this API; it is not intended to be necessary for 200 interoperation to know whether the entity one is communicating with 201 is a browser or another device implementing this specification. 203 The goal of cooperation between the protocol specification and the 204 API specification is that for all options and features of the 205 protocol specification, it should be clear which API calls to make to 206 exercise that option or feature; similarly, for any sequence of API 207 calls, it should be clear which protocol options and features will be 208 invoked. Both subject to constraints of the implementation, of 209 course. 211 For the purpose of this document, we define the following terminology 212 to talk about WebRTC things: 214 o A WebRTC browser (also called a WebRTC User Agent or WebRTC UA) is 215 something that conforms to both the protocol specification and the 216 Javascript API defined above. 218 o A WebRTC non-browser is something that conforms to the protocol 219 specification, but does not claim to implement the Javascript API. 220 This can also be called a "WebRTC device" or "WebRTC native 221 application". 223 o A WebRTC endpoint is either a WebRTC browser or a WebRTC non- 224 browser. It conforms to the protocol specification. 226 o A WebRTC-compatible endpoint is an endpoint that is able to 227 successfully communicate with a WebRTC endpoint, but may fail to 228 meet some requirements of a WebRTC endpoint. This may limit where 229 in the network such an endpoint can be attached, or may limit the 230 security guarantees that it offers to others. It is not 231 constrained by this specification; when it is mentioned at all, it 232 is to note the implications on WebRTC-compatible endpoints of the 233 requirements placed on WebRTC endpoints. 235 o A WebRTC gateway is a WebRTC-compatible endpoint that mediates 236 media traffic to non-WebRTC entities. 238 All WebRTC browsers are WebRTC endpoints, so any requirement on a 239 WebRTC endpoint also applies to a WebRTC browser. 241 A WebRTC non-browser may be capable of hosting applications in a 242 similar way to the way in which a browser can host Javascript 243 applications, typically by offering APIs in other languages. For 244 instance it may be implemented as a library that offers a C++ API 245 intended to be loaded into applications. In this case, similar 246 security considerations as for Javascript may be needed; however, 247 since such APIs are not defined or referenced here, this document 248 cannot give any specific rules for those interfaces. 250 WebRTC gateways are described in a separate document, 251 [I-D.ietf-rtcweb-gateways]. 253 2.3. On interoperability and innovation 255 The "Mission statement of the IETF" [RFC3935] states that "The 256 benefit of a standard to the Internet is in interoperability - that 257 multiple products implementing a standard are able to work together 258 in order to deliver valuable functions to the Internet's users." 260 Communication on the Internet frequently occurs in two phases: 262 o Two parties communicate, through some mechanism, what 263 functionality they both are able to support 265 o They use that shared communicative functionality to communicate, 266 or, failing to find anything in common, give up on communication. 268 There are often many choices that can be made for communicative 269 functionality; the history of the Internet is rife with the proposal, 270 standardization, implementation, and success or failure of many types 271 of options, in all sorts of protocols. 273 The goal of having a mandatory to implement function set is to 274 prevent negotiation failure, not to preempt or prevent negotiation. 276 The presence of a mandatory to implement function set serves as a 277 strong changer of the marketplace of deployment - in that it gives a 278 guarantee that, as long as you conform to a specification, and the 279 other party is willing to accept communication at the base level of 280 that specification, you can communicate successfully. 282 The alternative - that of having no mandatory to implement - does not 283 mean that you cannot communicate, it merely means that in order to be 284 part of the communications partnership, you have to implement the 285 standard "and then some" - that "and then some" usually being called 286 a profile of some sort; in the version most antithetical to the 287 Internet ethos, that "and then some" consists of having to use a 288 specific vendor's product only. 290 2.4. Terminology 292 The following terms are used across the documents specifying the 293 WebRTC suite, in the specific meanings given here. Not all terms are 294 used in this document. Other terms are used in their commonly used 295 meaning. 297 The list is in alphabetical order. 299 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 301 API: Application Programming Interface - a specification of a set of 302 calls and events, usually tied to a programming language or an 303 abstract formal specification such as WebIDL, with its defined 304 semantics. 306 Browser: Used synonymously with "Interactive User Agent" as defined 307 in the HTML specification [W3C.WD-html5-20110525]. See also 308 "WebRTC User Agent". 310 ICE Agent: An implementation of the Interactive Connectivty 311 Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be 312 an SDP Agent, but there exist ICE Agents that do not use SDP (for 313 instance those that use Jingle). 315 Interactive: Communication between multiple parties, where the 316 expectation is that an action from one party can cause a reaction 317 by another party, and the reaction can be observed by the first 318 party, with the total time required for the action/reaction/ 319 observation is on the order of no more than hundreds of 320 milliseconds. 322 Media: Audio and video content. Not to be confused with 323 "transmission media" such as wires. 325 Media path: The path that media data follows from one WebRTC 326 endpoint to another. 328 Protocol: A specification of a set of data units, their 329 representation, and rules for their transmission, with their 330 defined semantics. A protocol is usually thought of as going 331 between systems. 333 Real-time media: Media where generation of content and display of 334 content are intended to occur closely together in time (on the 335 order of no more than hundreds of milliseconds). Real-time media 336 can be used to support interactive communication. 338 SDP Agent: The protocol implementation involved in the SDP offer/ 339 answer exchange, as defined in [RFC3264] section 3. 341 Signaling: Communication that happens in order to establish, manage 342 and control media paths. 344 Signaling Path: The communication channels used between entities 345 participating in signaling to transfer signaling. There may be 346 more entities in the signaling path than in the media path. 348 NOTE: Where common definitions exist for these terms, those 349 definitions should be used to the greatest extent possible. 351 3. Architecture and Functionality groups 353 The model of real-time support for browser-based applications does 354 not assume that the browser will contain all the functions that need 355 to be performed in order to have a function such as a telephone or a 356 video conferencing unit; the vision is that the browser will have the 357 functions that are needed for a Web application, working in 358 conjunction with its backend servers, to implement these functions. 360 This means that two vital interfaces need specification: The 361 protocols that browsers talk to each other, without any intervening 362 servers, and the APIs that are offered for a Javascript application 363 to take advantage of the browser's functionality. 365 +------------------------+ On-the-wire 366 | | Protocols 367 | Servers |---------> 368 | | 369 | | 370 +------------------------+ 371 ^ 372 | 373 | 374 | HTTP/ 375 | Websockets 376 | 377 | 378 +----------------------------+ 379 | Javascript/HTML/CSS | 380 +----------------------------+ 381 Other ^ ^RTC 382 APIs | |APIs 383 +---|-----------------|------+ 384 | | | | 385 | +---------+| 386 | | Browser || On-the-wire 387 | Browser | RTC || Protocols 388 | | Function|-----------> 389 | | || 390 | | || 391 | +---------+| 392 +---------------------|------+ 393 | 394 V 395 Native OS Services 397 Figure 1: Browser Model 399 Note that HTTP and Websockets are also offered to the Javascript 400 application through browser APIs. 402 As for all protocol and API specifications, there is no restriction 403 that the protocols can only be used to talk to another browser; since 404 they are fully specified, any endpoint that implements the protocols 405 faithfully should be able to interoperate with the application 406 running in the browser. 408 A commonly imagined model of deployment is the one depicted below. 410 +-----------+ +-----------+ 411 | Web | | Web | 412 | | Signaling | | 413 | |-------------| | 414 | Server | path | Server | 415 | | | | 416 +-----------+ +-----------+ 417 / \ 418 / \ Application-defined 419 / \ over 420 / \ HTTP/Websockets 421 / Application-defined over \ 422 / HTTP/Websockets \ 423 / \ 424 +-----------+ +-----------+ 425 |JS/HTML/CSS| |JS/HTML/CSS| 426 +-----------+ +-----------+ 427 +-----------+ +-----------+ 428 | | | | 429 | | | | 430 | Browser | ------------------------- | Browser | 431 | | Media path | | 432 | | | | 433 +-----------+ +-----------+ 435 Figure 2: Browser RTC Trapezoid 437 On this drawing, the critical part to note is that the media path 438 ("low path") goes directly between the browsers, so it has to be 439 conformant to the specifications of the WebRTC protocol suite; the 440 signaling path ("high path") goes via servers that can modify, 441 translate or massage the signals as needed. 443 If the two Web servers are operated by different entities, the inter- 444 server signaling mechanism needs to be agreed upon, either by 445 standardization or by other means of agreement. Existing protocols 446 (for example SIP [RFC3261] or XMPP [RFC6120]) could be used between 447 servers, while either a standards-based or proprietary protocol could 448 be used between the browser and the web server. 450 For example, if both operators' servers implement SIP, SIP could be 451 used for communication between servers, along with either a 452 standardized signaling mechanism (e.g. SIP over Websockets) or a 453 proprietary signaling mechanism used between the application running 454 in the browser and the web server. Similarly, if both operators' 455 servers implement XMPP, XMPP could be used for communication between 456 XMPP servers, with either a standardized signaling mechanism (e.g. 457 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 458 used between the application running in the browser and the web 459 server. 461 The choice of protocols, and definition of the translation between 462 them, is outside the scope of the WebRTC protocol suite described in 463 the document. 465 The functionality groups that are needed in the browser can be 466 specified, more or less from the bottom up, as: 468 o Data transport: TCP, UDP and the means to securely set up 469 connections between entities, as well as the functions for 470 deciding when to send data: Congestion management, bandwidth 471 estimation and so on. 473 o Data framing: RTP and other data formats that serve as containers, 474 and their functions for data confidentiality and integrity. 476 o Data formats: Codec specifications, format specifications and 477 functionality specifications for the data passed between systems. 478 Audio and video codecs, as well as formats for data and document 479 sharing, belong in this category. In order to make use of data 480 formats, a way to describe them, a session description, is needed. 482 o Connection management: Setting up connections, agreeing on data 483 formats, changing data formats during the duration of a call; SIP 484 and Jingle/XMPP belong in this category. 486 o Presentation and control: What needs to happen in order to ensure 487 that interactions behave in a non-surprising manner. This can 488 include floor control, screen layout, voice activated image 489 switching and other such functions - where part of the system 490 require the cooperation between parties. XCON and Cisco/ 491 Tandberg's TIP were some attempts at specifying this kind of 492 functionality; many applications have been built without 493 standardized interfaces to these functions. 495 o Local system support functions: These are things that need not be 496 specified uniformly, because each participant may choose to do 497 these in a way of the participant's choosing, without affecting 498 the bits on the wire in a way that others have to be cognizant of. 499 Examples in this category include echo cancellation (some forms of 500 it), local authentication and authorization mechanisms, OS access 501 control and the ability to do local recording of conversations. 503 Within each functionality group, it is important to preserve both 504 freedom to innovate and the ability for global communication. 505 Freedom to innovate is helped by doing the specification in terms of 506 interfaces, not implementation; any implementation able to 507 communicate according to the interfaces is a valid implementation. 508 Ability to communicate globally is helped both by having core 509 specifications be unencumbered by IPR issues and by having the 510 formats and protocols be fully enough specified to allow for 511 independent implementation. 513 One can think of the three first groups as forming a "media transport 514 infrastructure", and of the three last groups as forming a "media 515 service". In many contexts, it makes sense to use a common 516 specification for the media transport infrastructure, which can be 517 embedded in browsers and accessed using standard interfaces, and "let 518 a thousand flowers bloom" in the "media service" layer; to achieve 519 interoperable services, however, at least the first five of the six 520 groups need to be specified. 522 4. Data transport 524 Data transport refers to the sending and receiving of data over the 525 network interfaces, the choice of network-layer addresses at each end 526 of the communication, and the interaction with any intermediate 527 entities that handle the data, but do not modify it (such as TURN 528 relays). 530 It includes necessary functions for congestion control: When not to 531 send data. 533 WebRTC endpoints MUST implement the transport protocols described in 534 [I-D.ietf-rtcweb-transports]. 536 5. Data framing and securing 538 The format for media transport is RTP [RFC3550]. Implementation of 539 SRTP [RFC3711] is REQUIRED for all implementations. 541 The detailed considerations for usage of functions from RTP and SRTP 542 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 543 considerations for the WebRTC use case are in 544 [I-D.ietf-rtcweb-security], and the resulting security functions are 545 described in [I-D.ietf-rtcweb-security-arch]. 547 Considerations for the transfer of data that is not in RTP format is 548 described in [I-D.ietf-rtcweb-data-channel], and a supporting 549 protocol for establishing individual data channels is described in 550 [I-D.ietf-rtcweb-data-protocol]. WebRTC endpoints MUST implement 551 these two specifications. 553 WebRTC endpoints MUST implement [I-D.ietf-rtcweb-rtp-usage], 554 [I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the 555 requirements they include. 557 6. Data formats 559 The intent of this specification is to allow each communications 560 event to use the data formats that are best suited for that 561 particular instance, where a format is supported by both sides of the 562 connection. However, a minimum standard is greatly helpful in order 563 to ensure that communication can be achieved. This document 564 specifies a minimum baseline that will be supported by all 565 implementations of this specification, and leaves further codecs to 566 be included at the will of the implementor. 568 WebRTC endpoints that support audio and/or video MUST implement the 569 codecs and profiles required in [I-D.ietf-rtcweb-audio] and 570 [I-D.ietf-rtcweb-video]. 572 7. Connection management 574 The methods, mechanisms and requirements for setting up, negotiating 575 and tearing down connections is a large subject, and one where it is 576 desirable to have both interoperability and freedom to innovate. 578 The following principles apply: 580 1. The WebRTC media negotiations will be capable of representing the 581 same SDP offer/answer semantics that are used in SIP [RFC3264], 582 in such a way that it is possible to build a signaling gateway 583 between SIP and the WebRTC media negotiation. 585 2. It will be possible to gateway between legacy SIP devices that 586 support ICE and appropriate RTP / SDP mechanisms, codecs and 587 security mechanisms without using a media gateway. A signaling 588 gateway to convert between the signaling on the web side to the 589 SIP signaling may be needed. 591 3. When a new codec is specified, and the SDP for the new codec is 592 specified in the MMUSIC WG, no other standardization should be 593 required for it to be possible to use that in the web browsers. 594 Adding new codecs which might have new SDP parameters should not 595 change the APIs between the browser and Javascript application. 596 As soon as the browsers support the new codecs, old applications 597 written before the codecs were specified should automatically be 598 able to use the new codecs where appropriate with no changes to 599 the JS applications. 601 The particular choices made for WebRTC, and their implications for 602 the API offered by a browser implementing WebRTC, are described in 603 [I-D.ietf-rtcweb-jsep]. 605 WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. 607 WebRTC endpoints MUST implement the functions described in that 608 document that relate to the network layer (for example Bundle, RTCP- 609 mux and Trickle ICE), but do not need to support the API 610 functionality described there. 612 8. Presentation and control 614 The most important part of control is the user's control over the 615 browser's interaction with input/output devices and communications 616 channels. It is important that the user have some way of figuring 617 out where his audio, video or texting is being sent, for what 618 purported reason, and what guarantees are made by the parties that 619 form part of this control channel. This is largely a local function 620 between the browser, the underlying operating system and the user 621 interface; this is specified in the peer connection API 622 [W3C.WD-webrtc-20120209], and the media capture API 623 [W3C.WD-mediacapture-streams-20120628]. 625 WebRTC browsers MUST implement these two specifications. 627 9. Local system support functions 629 These are characterized by the fact that the quality of these 630 functions strongly influence the user experience, but the exact 631 algorithm does not need coordination. In some cases (for instance 632 echo cancellation, as described below), the overall system definition 633 may need to specify that the overall system needs to have some 634 characteristics for which these facilities are useful, without 635 requiring them to be implemented a certain way. 637 Local functions include echo cancellation, volume control, camera 638 management including focus, zoom, pan/tilt controls (if available), 639 and more. 641 Certain parts of the system SHOULD conform to certain properties, for 642 instance: 644 o Echo cancellation should be good enough to achieve the suppression 645 of acoustical feedback loops below a perceptually noticeable 646 level. 648 o Privacy concerns MUST be satisfied; for instance, if remote 649 control of camera is offered, the APIs should be available to let 650 the local participant figure out who's controlling the camera, and 651 possibly decide to revoke the permission for camera usage. 653 o Automatic gain control, if present, should normalize a speaking 654 voice into a reasonable dB range. 656 The requirements on WebRTC systems with regard to audio processing 657 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 658 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 660 WebRTC endpoints MUST implement the processing functions in 661 [I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6, 662 this means that WebRTC endpoints MUST implement the whole document.) 664 10. IANA Considerations 666 This document makes no request of IANA. 668 Note to RFC Editor: this section may be removed on publication as an 669 RFC. 671 11. Security Considerations 673 Security of the web-enabled real time communications comes in several 674 pieces: 676 o Security of the components: The browsers, and other servers 677 involved. The most target-rich environment here is probably the 678 browser; the aim here should be that the introduction of these 679 components introduces no additional vulnerability. 681 o Security of the communication channels: It should be easy for a 682 participant to reassure himself of the security of his 683 communication - by verifying the crypto parameters of the links he 684 himself participates in, and to get reassurances from the other 685 parties to the communication that they promise that appropriate 686 measures are taken. 688 o Security of the partners' identity: verifying that the 689 participants are who they say they are (when positive 690 identification is appropriate), or that their identity cannot be 691 uncovered (when anonymity is a goal of the application). 693 The security analysis, and the requirements derived from that 694 analysis, is contained in [I-D.ietf-rtcweb-security]. 696 It is also important to read the security sections of 697 [W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209]. 699 12. Acknowledgements 701 The number of people who have taken part in the discussions 702 surrounding this draft are too numerous to list, or even to identify. 703 The ones below have made special, identifiable contributions; this 704 does not mean that others' contributions are less important. 706 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 707 Westerlund and Joerg Ott, who offered technical contributions on 708 various versions of the draft. 710 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 711 the ASCII drawings in section 1. 713 Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric 714 Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage 715 and Simon Leinen for document review. 717 13. References 719 13.1. Normative References 721 [I-D.ietf-rtcweb-audio] 722 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 723 Requirements", draft-ietf-rtcweb-audio-05 (work in 724 progress), February 2014. 726 [I-D.ietf-rtcweb-data-channel] 727 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 728 Channels", draft-ietf-rtcweb-data-channel-11 (work in 729 progress), July 2014. 731 [I-D.ietf-rtcweb-data-protocol] 732 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 733 Establishment Protocol", draft-ietf-rtcweb-data- 734 protocol-07 (work in progress), July 2014. 736 [I-D.ietf-rtcweb-jsep] 737 Uberti, J., Jennings, C., and E. Rescorla, "Javascript 738 Session Establishment Protocol", draft-ietf-rtcweb-jsep-07 739 (work in progress), July 2014. 741 [I-D.ietf-rtcweb-rtp-usage] 742 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 743 Communication (WebRTC): Media Transport and Use of RTP", 744 draft-ietf-rtcweb-rtp-usage-16 (work in progress), July 745 2014. 747 [I-D.ietf-rtcweb-security] 748 Rescorla, E., "Security Considerations for WebRTC", draft- 749 ietf-rtcweb-security-07 (work in progress), July 2014. 751 [I-D.ietf-rtcweb-security-arch] 752 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 753 rtcweb-security-arch-10 (work in progress), July 2014. 755 [I-D.ietf-rtcweb-transports] 756 Alvestrand, H., "Transports for WebRTC", draft-ietf- 757 rtcweb-transports-06 (work in progress), August 2014. 759 [I-D.ietf-rtcweb-video] 760 Roach, A., "WebRTC Video Processing and Codec 761 Requirements", draft-ietf-rtcweb-video-00 (work in 762 progress), July 2014. 764 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 765 with Session Description Protocol (SDP)", RFC 3264, June 766 2002. 768 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 769 Jacobson, "RTP: A Transport Protocol for Real-Time 770 Applications", STD 64, RFC 3550, July 2003. 772 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 773 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 774 RFC 3711, March 2004. 776 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 777 (ICE): A Protocol for Network Address Translator (NAT) 778 Traversal for Offer/Answer Protocols", RFC 5245, April 779 2010. 781 [W3C.WD-mediacapture-streams-20120628] 782 Burnett, D. and A. Narayanan, "Media Capture and Streams", 783 World Wide Web Consortium WD WD-mediacapture-streams- 784 20120628, June 2012, . 787 [W3C.WD-webrtc-20120209] 788 Bergkvist, A., Burnett, D., Jennings, C., and A. 789 Narayanan, "WebRTC 1.0: Real-time Communication Between 790 Browsers", World Wide Web Consortium WD WD-webrtc- 791 20120209, February 2012, 792 . 794 13.2. Informative References 796 [I-D.ietf-rtcweb-gateways] 797 Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways", 798 draft-ietf-rtcweb-gateways-01 (work in progress), July 799 2015. 801 [I-D.ietf-rtcweb-use-cases-and-requirements] 802 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 803 Time Communication Use-cases and Requirements", draft- 804 ietf-rtcweb-use-cases-and-requirements-14 (work in 805 progress), February 2014. 807 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 808 A., Peterson, J., Sparks, R., Handley, M., and E. 809 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 810 June 2002. 812 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", BCP 813 95, RFC 3935, October 2004. 815 [RFC6120] Saint-Andre, P., "Extensible Messaging and Presence 816 Protocol (XMPP): Core", RFC 6120, March 2011. 818 [W3C.WD-html5-20110525] 819 Hickson, I., "HTML5", World Wide Web Consortium LastCall 820 WD-html5-20110525, May 2011, 821 . 823 Appendix A. Change log 825 This section may be deleted by the RFC Editor when preparing for 826 publication. 828 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 830 Added section "On interoperability and innovation" 832 Added data confidentiality and integrity to the "data framing" layer 833 Added congestion management requirements in the "data transport" 834 layer section 836 Changed need for non-media data from "question: do we need this?" to 837 "Open issue: How do we do this?" 839 Strengthened disclaimer that listed codecs are placeholders, not 840 decisions. 842 More details on why the "local system support functions" section is 843 there. 845 A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand- 846 rtcweb-overview-00 848 Added section on "Relationship between API and protocol" 850 Added terminology section 852 Mentioned congestion management as part of the "data transport" layer 853 in the layer list 855 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 857 Removed most technical content, and replaced with pointers to drafts 858 as requested and identified by the RTCWEB WG chairs. 860 Added content to acknowledgments section. 862 Added change log. 864 Spell-checked document. 866 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf- 867 rtcweb-overview-00 869 Changed draft name and document date. 871 Removed unused references 873 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 875 Added architecture figures to section 2. 877 Changed the description of "echo cancellation" under "local system 878 support functions". 880 Added a few more definitions. 882 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 884 Added pointers to use cases, security and rtp-usage drafts (now WG 885 drafts). 887 Changed description of SRTP from mandatory-to-use to mandatory-to- 888 implement. 890 Added the "3 principles of negotiation" to the connection management 891 section. 893 Added an explicit statement that ICE is required for both NAT and 894 consent-to-receive. 896 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 898 Added references to a number of new drafts. 900 Expanded the description text under the "trapezoid" drawing with some 901 more text discussed on the list. 903 Changed the "Connection management" sentence from "will be done using 904 SDP offer/answer" to "will be capable of representing SDP offer/ 905 answer" - this seems more consistent with JSEP. 907 Added "security mechanisms" to the things a non-gatewayed SIP devices 908 must support in order to not need a media gateway. 910 Added a definition for "browser". 912 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 914 Made introduction more normative. 916 Several wording changes in response to review comments from EKR 918 Added an appendix to hold references and notes that are not yet in a 919 separate document. 921 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 923 Minor grammatical fixes. This is mainly a "keepalive" refresh. 925 A.10. Changes from -05 to -06 927 Clarifications in response to Last Call review comments. Inserted 928 reference to draft-ietf-rtcweb-audio. 930 A.11. Changes from -06 to -07 932 Added a reference to the "unified plan" draft, and updated some 933 references. 935 Otherwise, it's a "keepalive" draft. 937 A.12. Changes from -07 to -08 939 Removed the appendix that detailed transports, and replaced it with a 940 reference to draft-ietf-rtcweb-transports. Removed now-unused 941 references. 943 A.13. Changes from -08 to -09 945 Added text to the Abstract indicating that the intended status is an 946 Applicability Statement. 948 A.14. Changes from -09 to -10 950 Defined "WebRTC Browser" and "WebRTC device" as things that do, or 951 don't, conform to the API. 953 Updated reference to data-protocol draft 955 Updated data formats to reference -rtcweb-audio- and not the expired 956 -cbran draft. 958 Deleted references to -unified-plan 960 Deleted reference to -generic-idp (draft expired) 962 Added notes on which referenced documents WebRTC browsers or devices 963 MUST conform to. 965 Added pointer to the security section of the API drafts. 967 A.15. Changes from -10 to -11 969 Added "WebRTC Gateway" as a third class of device, and referenced the 970 doc describing them. 972 Made a number of text clarifications in response to document reviews. 974 A.16. Changes from -11 to -12 976 Refined entity definitions to define "WebRTC endpoint" and "WebRTC- 977 compatible endpoint". 979 Changed remaining usage of the term "RTCWEB" to "WebRTC", including 980 in the page header. 982 A.17. Changes from -12 to -13 984 Changed "WebRTC device" to be "WebRTC non-browser", per decision at 985 IETF 91. This led to the need for "WebRTC endpoint" as the common 986 label for both, and the usage of that term in the rest of the 987 document. 989 Added words about WebRTC APIs in languages other than Javascript. 991 Referenced draft-ietf-rtcweb-video for video codecs to support. 993 A.18. Changes from -13 to -14 995 None. This is a "keepalive" update. 997 A.19. Changes from -14 to -15 999 Changed "gateways" reference to point to the WG document. 1001 Author's Address 1003 Harald T. Alvestrand 1004 Google 1005 Kungsbron 2 1006 Stockholm 11122 1007 Sweden 1009 Email: harald@alvestrand.no