idnits 2.17.1 draft-ietf-rtcweb-overview-17.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 542: '...WebRTC endpoints MUST implement the tr...' RFC 2119 keyword, line 548: '... SRTP [RFC3711] is REQUIRED for all implementations....' RFC 2119 keyword, line 559: '... [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(See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Outdated reference: A later version (-20) exists of draft-ietf-ice-rfc5245bis-08 == Outdated reference: A later version (-26) exists of draft-ietf-rtcweb-jsep-18 == Outdated reference: A later version (-12) exists of draft-ietf-rtcweb-security-08 == Outdated reference: A later version (-20) exists of draft-ietf-rtcweb-security-arch-12 Summary: 1 error (**), 0 flaws (~~), 5 warnings (==), 1 comment (--).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 February 17, 2017 5 Expires: August 21, 2017 7 Overview: Real Time Protocols for Browser-based Applications 8 draft-ietf-rtcweb-overview-17 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 August 21, 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 . . . . . 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 A.21. Changes from -16 to -17 . . . . . . . . . . . . . . . . . 23 105 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 23 107 1. Introduction 109 The Internet was, from very early in its lifetime, considered a 110 possible vehicle for the deployment of real-time, interactive 111 applications - with the most easily imaginable being audio 112 conversations (aka "Internet telephony") and video conferencing. 114 The first attempts to build this were dependent on special networks, 115 special hardware and custom-built software, often at very high prices 116 or at low quality, placing great demands on the infrastructure. 118 As the available bandwidth has increased, and as processors an other 119 hardware has become ever faster, the barriers to participation have 120 decreased, and it has become possible to deliver a satisfactory 121 experience on commonly available computing hardware. 123 Still, there are a number of barriers to the ability to communicate 124 universally - one of these is that there is, as of yet, no single set 125 of communication protocols that all agree should be made available 126 for communication; another is the sheer lack of universal 127 identification systems (such as is served by telephone numbers or 128 email addresses in other communications systems). 130 Development of The Universal Solution has proved hard, however, for 131 all the usual reasons. 133 The last few years have also seen a new platform rise for deployment 134 of services: The browser-embedded application, or "Web application". 135 It turns out that as long as the browser platform has the necessary 136 interfaces, it is possible to deliver almost any kind of service on 137 it. 139 Traditionally, these interfaces have been delivered by plugins, which 140 had to be downloaded and installed separately from the browser; in 141 the development of HTML5, application developers see much promise in 142 the possibility of making those interfaces available in a 143 standardized way within the browser. 145 This memo describes a set of building blocks that can be made 146 accessible and controllable through a Javascript API in a browser, 147 and which together form a sufficient set of functions to allow the 148 use of interactive audio and video in applications that communicate 149 directly between browsers across the Internet. The resulting 150 protocol suite is intended to enable all the applications that are 151 described as required scenarios in the use cases document 152 [I-D.ietf-rtcweb-use-cases-and-requirements]. 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 190 o A Javascript API specification, defined in a series of W3C 191 documents 192 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 194 Together, these two specifications aim to provide an environment 195 where Javascript embedded in any page, when suitably authorized by 196 its user, is able to set up communication using audio, video and 197 auxiliary data, as long as the browser supports this specification. 198 The browser environment does not constrain the types of application 199 in which this functionality can be used. 201 The protocol specification does not assume that all implementations 202 implement this API; it is not intended to be necessary for 203 interoperation to know whether the entity one is communicating with 204 is a browser or another device implementing this specification. 206 The goal of cooperation between the protocol specification and the 207 API specification is that for all options and features of the 208 protocol specification, it should be clear which API calls to make to 209 exercise that option or feature; similarly, for any sequence of API 210 calls, it should be clear which protocol options and features will be 211 invoked. Both subject to constraints of the implementation, of 212 course. 214 For the purpose of this document, we define the following terminology 215 to talk about WebRTC things: 217 o A WebRTC browser (also called a WebRTC User Agent or WebRTC UA) is 218 something that conforms to both the protocol specification and the 219 Javascript API defined above. 221 o A WebRTC non-browser is something that conforms to the protocol 222 specification, but does not claim to implement the Javascript API. 223 This can also be called a "WebRTC device" or "WebRTC native 224 application". 226 o A WebRTC endpoint is either a WebRTC browser or a WebRTC non- 227 browser. It conforms to the protocol specification. 229 o A WebRTC-compatible endpoint is an endpoint that is able to 230 successfully communicate with a WebRTC endpoint, but may fail to 231 meet some requirements of a WebRTC endpoint. This may limit where 232 in the network such an endpoint can be attached, or may limit the 233 security guarantees that it offers to others. It is not 234 constrained by this specification; when it is mentioned at all, it 235 is to note the implications on WebRTC-compatible endpoints of the 236 requirements placed on WebRTC endpoints. 238 o A WebRTC gateway is a WebRTC-compatible endpoint that mediates 239 media traffic to non-WebRTC entities. 241 All WebRTC browsers are WebRTC endpoints, so any requirement on a 242 WebRTC endpoint also applies to a WebRTC browser. 244 A WebRTC non-browser may be capable of hosting applications in a 245 similar way to the way in which a browser can host Javascript 246 applications, typically by offering APIs in other languages. For 247 instance it may be implemented as a library that offers a C++ API 248 intended to be loaded into applications. In this case, similar 249 security considerations as for Javascript may be needed; however, 250 since such APIs are not defined or referenced here, this document 251 cannot give any specific rules for those interfaces. 253 WebRTC gateways are described in a separate document, 254 [I-D.ietf-rtcweb-gateways]. 256 2.3. On interoperability and innovation 258 The "Mission statement of the IETF" [RFC3935] states that "The 259 benefit of a standard to the Internet is in interoperability - that 260 multiple products implementing a standard are able to work together 261 in order to deliver valuable functions to the Internet's users." 263 Communication on the Internet frequently occurs in two phases: 265 o Two parties communicate, through some mechanism, what 266 functionality they both are able to support 268 o They use that shared communicative functionality to communicate, 269 or, failing to find anything in common, give up on communication. 271 There are often many choices that can be made for communicative 272 functionality; the history of the Internet is rife with the proposal, 273 standardization, implementation, and success or failure of many types 274 of options, in all sorts of protocols. 276 The goal of having a mandatory to implement function set is to 277 prevent negotiation failure, not to preempt or prevent negotiation. 279 The presence of a mandatory to implement function set serves as a 280 strong changer of the marketplace of deployment - in that it gives a 281 guarantee that, as long as you conform to a specification, and the 282 other party is willing to accept communication at the base level of 283 that specification, you can communicate successfully. 285 The alternative - that of having no mandatory to implement - does not 286 mean that you cannot communicate, it merely means that in order to be 287 part of the communications partnership, you have to implement the 288 standard "and then some" - that "and then some" usually being called 289 a profile of some sort; in the version most antithetical to the 290 Internet ethos, that "and then some" consists of having to use a 291 specific vendor's product only. 293 2.4. Terminology 295 The following terms are used across the documents specifying the 296 WebRTC suite, in the specific meanings given here. Not all terms are 297 used in this document. Other terms are used in their commonly used 298 meaning. 300 The list is in alphabetical order. 302 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 304 API: Application Programming Interface - a specification of a set of 305 calls and events, usually tied to a programming language or an 306 abstract formal specification such as WebIDL, with its defined 307 semantics. 309 Browser: Used synonymously with "Interactive User Agent" as defined 310 in the HTML specification [W3C.WD-html5-20110525]. See also 311 "WebRTC User Agent". 313 Data channel: An abstraction that allows data to be sent between 314 WebRTC endpoints in the form of messages. Two endpoints can have 315 multiple data channels between them. 317 ICE Agent: An implementation of the Interactive Connectivty 318 Establishment (ICE) [I-D.ietf-ice-rfc5245bis] protocol. An ICE 319 Agent may also be an SDP Agent, but there exist ICE Agents that do 320 not use SDP (for instance those that use Jingle). 322 Interactive: Communication between multiple parties, where the 323 expectation is that an action from one party can cause a reaction 324 by another party, and the reaction can be observed by the first 325 party, with the total time required for the action/reaction/ 326 observation is on the order of no more than hundreds of 327 milliseconds. 329 Media: Audio and video content. Not to be confused with 330 "transmission media" such as wires. 332 Media path: The path that media data follows from one WebRTC 333 endpoint to another. 335 Protocol: A specification of a set of data units, their 336 representation, and rules for their transmission, with their 337 defined semantics. A protocol is usually thought of as going 338 between systems. 340 Real-time media: Media where generation of content and display of 341 content are intended to occur closely together in time (on the 342 order of no more than hundreds of milliseconds). Real-time media 343 can be used to support interactive communication. 345 SDP Agent: The protocol implementation involved in the SDP offer/ 346 answer exchange, as defined in [RFC3264] section 3. 348 Signaling: Communication that happens in order to establish, manage 349 and control media paths and data paths. 351 Signaling Path: The communication channels used between entities 352 participating in signaling to transfer signaling. There may be 353 more entities in the signaling path than in the media path. 355 NOTE: Where common definitions exist for these terms, those 356 definitions should be used to the greatest extent possible. 358 3. Architecture and Functionality groups 360 The model of real-time support for browser-based applications does 361 not assume that the browser will contain all the functions that need 362 to be performed in order to have a function such as a telephone or a 363 video conferencing unit; the vision is that the browser will have the 364 functions that are needed for a Web application, working in 365 conjunction with its backend servers, to implement these functions. 367 This means that two vital interfaces need specification: The 368 protocols that browsers use to talk to each other, without any 369 intervening servers, and the APIs that are offered for a Javascript 370 application to take advantage of the browser's functionality. 372 +------------------------+ On-the-wire 373 | | Protocols 374 | Servers |---------> 375 | | 376 | | 377 +------------------------+ 378 ^ 379 | 380 | 381 | HTTP/ 382 | WebSockets 383 | 384 | 385 +----------------------------+ 386 | Javascript/HTML/CSS | 387 +----------------------------+ 388 Other ^ ^RTC 389 APIs | |APIs 390 +---|-----------------|------+ 391 | | | | 392 | +---------+| 393 | | Browser || On-the-wire 394 | Browser | RTC || Protocols 395 | | Function|-----------> 396 | | || 397 | | || 398 | +---------+| 399 +---------------------|------+ 400 | 401 V 402 Native OS Services 404 Figure 1: Browser Model 406 Note that HTTP and WebSockets are also offered to the Javascript 407 application through browser APIs. 409 As for all protocol and API specifications, there is no restriction 410 that the protocols can only be used to talk to another browser; since 411 they are fully specified, any endpoint that implements the protocols 412 faithfully should be able to interoperate with the application 413 running in the browser. 415 A commonly imagined model of deployment is the one depicted below. 417 +-----------+ +-----------+ 418 | Web | | Web | 419 | | Signaling | | 420 | |-------------| | 421 | Server | path | Server | 422 | | | | 423 +-----------+ +-----------+ 424 / \ 425 / \ Application-defined 426 / \ over 427 / \ HTTP/WebSockets 428 / Application-defined over \ 429 / HTTP/WebSockets \ 430 / \ 431 +-----------+ +-----------+ 432 |JS/HTML/CSS| |JS/HTML/CSS| 433 +-----------+ +-----------+ 434 +-----------+ +-----------+ 435 | | | | 436 | | | | 437 | Browser | ------------------------- | Browser | 438 | | Media path | | 439 | | | | 440 +-----------+ +-----------+ 442 Figure 2: Browser RTC Trapezoid 444 On this drawing, the critical part to note is that the media path 445 ("low path") goes directly between the browsers, so it has to be 446 conformant to the specifications of the WebRTC protocol suite; the 447 signaling path ("high path") goes via servers that can modify, 448 translate or massage the signals as needed. 450 If the two Web servers are operated by different entities, the inter- 451 server signaling mechanism needs to be agreed upon, either by 452 standardization or by other means of agreement. Existing protocols 453 (for example SIP [RFC3261] or XMPP [RFC6120]) could be used between 454 servers, while either a standards-based or proprietary protocol could 455 be used between the browser and the web server. 457 For example, if both operators' servers implement SIP, SIP could be 458 used for communication between servers, along with either a 459 standardized signaling mechanism (e.g. SIP over WebSockets) or a 460 proprietary signaling mechanism used between the application running 461 in the browser and the web server. Similarly, if both operators' 462 servers implement XMPP, XMPP could be used for communication between 463 XMPP servers, with either a standardized signaling mechanism (e.g. 464 XMPP over WebSockets or BOSH) or a proprietary signaling mechanism 465 used between the application running in the browser and the web 466 server. 468 The choice of protocols for client-server and inter-server 469 signalling, and definition of the translation between them, is 470 outside the scope of the WebRTC protocol suite described in the 471 document. 473 The functionality groups that are needed in the browser can be 474 specified, more or less from the bottom up, as: 476 o Data transport: TCP, UDP and the means to securely set up 477 connections between entities, as well as the functions for 478 deciding when to send data: Congestion management, bandwidth 479 estimation and so on. 481 o Data framing: RTP, SCTP and other data formats that serve as 482 containers, and their functions for data confidentiality and 483 integrity. 485 o Data formats: Codec specifications, format specifications and 486 functionality specifications for the data passed between systems. 487 Audio and video codecs, as well as formats for data and document 488 sharing, belong in this category. In order to make use of data 489 formats, a way to describe them, a session description, is needed. 491 o Connection management: Setting up connections, agreeing on data 492 formats, changing data formats during the duration of a call; SIP 493 and Jingle/XMPP belong in this category. 495 o Presentation and control: What needs to happen in order to ensure 496 that interactions behave in a non-surprising manner. This can 497 include floor control, screen layout, voice activated image 498 switching and other such functions - where part of the system 499 require the cooperation between parties. XCON and Cisco/ 500 Tandberg's TIP were some attempts at specifying this kind of 501 functionality; many applications have been built without 502 standardized interfaces to these functions. 504 o Local system support functions: These are things that need not be 505 specified uniformly, because each participant may choose to do 506 these in a way of the participant's choosing, without affecting 507 the bits on the wire in a way that others have to be cognizant of. 508 Examples in this category include echo cancellation (some forms of 509 it), local authentication and authorization mechanisms, OS access 510 control and the ability to do local recording of conversations. 512 Within each functionality group, it is important to preserve both 513 freedom to innovate and the ability for global communication. 514 Freedom to innovate is helped by doing the specification in terms of 515 interfaces, not implementation; any implementation able to 516 communicate according to the interfaces is a valid implementation. 517 Ability to communicate globally is helped both by having core 518 specifications be unencumbered by IPR issues and by having the 519 formats and protocols be fully enough specified to allow for 520 independent implementation. 522 One can think of the three first groups as forming a "media transport 523 infrastructure", and of the three last groups as forming a "media 524 service". In many contexts, it makes sense to use a common 525 specification for the media transport infrastructure, which can be 526 embedded in browsers and accessed using standard interfaces, and "let 527 a thousand flowers bloom" in the "media service" layer; to achieve 528 interoperable services, however, at least the first five of the six 529 groups need to be specified. 531 4. Data transport 533 Data transport refers to the sending and receiving of data over the 534 network interfaces, the choice of network-layer addresses at each end 535 of the communication, and the interaction with any intermediate 536 entities that handle the data, but do not modify it (such as TURN 537 relays). 539 It includes necessary functions for congestion control: When not to 540 send data. 542 WebRTC endpoints MUST implement the transport protocols described in 543 [I-D.ietf-rtcweb-transports]. 545 5. Data framing and securing 547 The format for media transport is RTP [RFC3550]. Implementation of 548 SRTP [RFC3711] is REQUIRED for all implementations. 550 The detailed considerations for usage of functions from RTP and SRTP 551 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 552 considerations for the WebRTC use case are in 553 [I-D.ietf-rtcweb-security], and the resulting security functions are 554 described in [I-D.ietf-rtcweb-security-arch]. 556 Considerations for the transfer of data that is not in RTP format is 557 described in [I-D.ietf-rtcweb-data-channel], and a supporting 558 protocol for establishing individual data channels is described in 559 [I-D.ietf-rtcweb-data-protocol]. WebRTC endpoints MUST implement 560 these two specifications. 562 WebRTC endpoints MUST implement [I-D.ietf-rtcweb-rtp-usage], 563 [I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the 564 requirements they include. 566 6. Data formats 568 The intent of this specification is to allow each communications 569 event to use the data formats that are best suited for that 570 particular instance, where a format is supported by both sides of the 571 connection. However, a minimum standard is greatly helpful in order 572 to ensure that communication can be achieved. This document 573 specifies a minimum baseline that will be supported by all 574 implementations of this specification, and leaves further codecs to 575 be included at the will of the implementor. 577 WebRTC endpoints that support audio and/or video MUST implement the 578 codecs and profiles required in [I-D.ietf-rtcweb-audio] and 579 [I-D.ietf-rtcweb-video]. 581 7. Connection management 583 The methods, mechanisms and requirements for setting up, negotiating 584 and tearing down connections is a large subject, and one where it is 585 desirable to have both interoperability and freedom to innovate. 587 The following principles apply: 589 1. The WebRTC media negotiations will be capable of representing the 590 same SDP offer/answer semantics that are used in SIP [RFC3264], 591 in such a way that it is possible to build a signaling gateway 592 between SIP and the WebRTC media negotiation. 594 2. It will be possible to gateway between legacy SIP devices that 595 support ICE and appropriate RTP / SDP mechanisms, codecs and 596 security mechanisms without using a media gateway. A signaling 597 gateway to convert between the signaling on the web side to the 598 SIP signaling may be needed. 600 3. When a new codec is specified, and the SDP for the new codec is 601 specified in the MMUSIC WG, no other standardization should be 602 required for it to be possible to use that in the web browsers. 603 Adding new codecs which might have new SDP parameters should not 604 change the APIs between the browser and Javascript application. 605 As soon as the browsers support the new codecs, old applications 606 written before the codecs were specified should automatically be 607 able to use the new codecs where appropriate with no changes to 608 the JS applications. 610 The particular choices made for WebRTC, and their implications for 611 the API offered by a browser implementing WebRTC, are described in 612 [I-D.ietf-rtcweb-jsep]. 614 WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. 616 WebRTC endpoints MUST implement the functions described in that 617 document that relate to the network layer (for example Bundle, RTCP- 618 mux and Trickle ICE), but do not need to support the API 619 functionality described there. 621 8. Presentation and control 623 The most important part of control is the user's control over the 624 browser's interaction with input/output devices and communications 625 channels. It is important that the user have some way of figuring 626 out where his audio, video or texting is being sent, for what 627 purported reason, and what guarantees are made by the parties that 628 form part of this control channel. This is largely a local function 629 between the browser, the underlying operating system and the user 630 interface; this is specified in the peer connection API 631 [W3C.WD-webrtc-20120209], and the media capture API 632 [W3C.WD-mediacapture-streams-20120628]. 634 WebRTC browsers MUST implement these two specifications. 636 9. Local system support functions 638 These are characterized by the fact that the quality of these 639 functions strongly influence the user experience, but the exact 640 algorithm does not need coordination. In some cases (for instance 641 echo cancellation, as described below), the overall system definition 642 may need to specify that the overall system needs to have some 643 characteristics for which these facilities are useful, without 644 requiring them to be implemented a certain way. 646 Local functions include echo cancellation, volume control, camera 647 management including focus, zoom, pan/tilt controls (if available), 648 and more. 650 Certain parts of the system SHOULD conform to certain properties, for 651 instance: 653 o Echo cancellation should be good enough to achieve the suppression 654 of acoustical feedback loops below a perceptually noticeable 655 level. 657 o Privacy concerns MUST be satisfied; for instance, if remote 658 control of camera is offered, the APIs should be available to let 659 the local participant figure out who's controlling the camera, and 660 possibly decide to revoke the permission for camera usage. 662 o Automatic gain control, if present, should normalize a speaking 663 voice into a reasonable dB range. 665 The requirements on WebRTC systems with regard to audio processing 666 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 667 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 669 WebRTC endpoints MUST implement the processing functions in 670 [I-D.ietf-rtcweb-audio]. (Together with the requirement in 671 Section 6, this means that WebRTC endpoints MUST implement the whole 672 document.) 674 10. IANA Considerations 676 This document makes no request of IANA. 678 Note to RFC Editor: this section may be removed on publication as an 679 RFC. 681 11. Security Considerations 683 Security of the web-enabled real time communications comes in several 684 pieces: 686 o Security of the components: The browsers, and other servers 687 involved. The most target-rich environment here is probably the 688 browser; the aim here should be that the introduction of these 689 components introduces no additional vulnerability. 691 o Security of the communication channels: It should be easy for a 692 participant to reassure himself of the security of his 693 communication - by verifying the crypto parameters of the links he 694 himself participates in, and to get reassurances from the other 695 parties to the communication that they promise that appropriate 696 measures are taken. 698 o Security of the partners' identity: verifying that the 699 participants are who they say they are (when positive 700 identification is appropriate), or that their identity cannot be 701 uncovered (when anonymity is a goal of the application). 703 The security analysis, and the requirements derived from that 704 analysis, is contained in [I-D.ietf-rtcweb-security]. 706 It is also important to read the security sections of 707 [W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209]. 709 12. Acknowledgements 711 The number of people who have taken part in the discussions 712 surrounding this draft are too numerous to list, or even to identify. 713 The ones below have made special, identifiable contributions; this 714 does not mean that others' contributions are less important. 716 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 717 Westerlund and Joerg Ott, who offered technical contributions on 718 various versions of the draft. 720 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 721 the ASCII drawings in section 1. 723 Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric 724 Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage, 725 Magnus Westerlund, Olle E. Johansson and Simon Leinen for document 726 review. 728 13. References 730 13.1. Normative References 732 [I-D.ietf-ice-rfc5245bis] 733 Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive 734 Connectivity Establishment (ICE): A Protocol for Network 735 Address Translator (NAT) Traversal", draft-ietf-ice- 736 rfc5245bis-08 (work in progress), December 2016. 738 [I-D.ietf-rtcweb-audio] 739 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 740 Requirements", draft-ietf-rtcweb-audio-11 (work in 741 progress), April 2016. 743 [I-D.ietf-rtcweb-data-channel] 744 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 745 Channels", draft-ietf-rtcweb-data-channel-13 (work in 746 progress), January 2015. 748 [I-D.ietf-rtcweb-data-protocol] 749 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 750 Establishment Protocol", draft-ietf-rtcweb-data- 751 protocol-09 (work in progress), January 2015. 753 [I-D.ietf-rtcweb-jsep] 754 Uberti, J., Jennings, C., and E. Rescorla, "Javascript 755 Session Establishment Protocol", draft-ietf-rtcweb-jsep-18 756 (work in progress), January 2017. 758 [I-D.ietf-rtcweb-rtp-usage] 759 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 760 Communication (WebRTC): Media Transport and Use of RTP", 761 draft-ietf-rtcweb-rtp-usage-26 (work in progress), March 762 2016. 764 [I-D.ietf-rtcweb-security] 765 Rescorla, E., "Security Considerations for WebRTC", draft- 766 ietf-rtcweb-security-08 (work in progress), February 2015. 768 [I-D.ietf-rtcweb-security-arch] 769 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 770 rtcweb-security-arch-12 (work in progress), June 2016. 772 [I-D.ietf-rtcweb-transports] 773 Alvestrand, H., "Transports for WebRTC", draft-ietf- 774 rtcweb-transports-17 (work in progress), October 2016. 776 [I-D.ietf-rtcweb-video] 777 Roach, A., "WebRTC Video Processing and Codec 778 Requirements", draft-ietf-rtcweb-video-06 (work in 779 progress), June 2015. 781 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 782 with Session Description Protocol (SDP)", RFC 3264, 783 DOI 10.17487/RFC3264, June 2002, 784 . 786 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 787 Jacobson, "RTP: A Transport Protocol for Real-Time 788 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 789 July 2003, . 791 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 792 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 793 RFC 3711, DOI 10.17487/RFC3711, March 2004, 794 . 796 [W3C.WD-mediacapture-streams-20120628] 797 Burnett, D. and A. Narayanan, "Media Capture and Streams", 798 World Wide Web Consortium WD WD-mediacapture-streams- 799 20120628, June 2012, . 802 [W3C.WD-webrtc-20120209] 803 Bergkvist, A., Burnett, D., Jennings, C., and A. 804 Narayanan, "WebRTC 1.0: Real-time Communication Between 805 Browsers", World Wide Web Consortium WD WD-webrtc- 806 20120209, February 2012, 807 . 809 13.2. Informative References 811 [I-D.ietf-rtcweb-gateways] 812 Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways", 813 draft-ietf-rtcweb-gateways-02 (work in progress), January 814 2016. 816 [I-D.ietf-rtcweb-use-cases-and-requirements] 817 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 818 Time Communication Use-cases and Requirements", draft- 819 ietf-rtcweb-use-cases-and-requirements-16 (work in 820 progress), January 2015. 822 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 823 A., Peterson, J., Sparks, R., Handley, M., and E. 824 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 825 DOI 10.17487/RFC3261, June 2002, 826 . 828 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", 829 BCP 95, RFC 3935, DOI 10.17487/RFC3935, October 2004, 830 . 832 [RFC6120] Saint-Andre, P., "Extensible Messaging and Presence 833 Protocol (XMPP): Core", RFC 6120, DOI 10.17487/RFC6120, 834 March 2011, . 836 [W3C.WD-html5-20110525] 837 Hickson, I., "HTML5", World Wide Web Consortium LastCall 838 WD-html5-20110525, May 2011, 839 . 841 Appendix A. Change log 843 This section may be deleted by the RFC Editor when preparing for 844 publication. 846 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 848 Added section "On interoperability and innovation" 850 Added data confidentiality and integrity to the "data framing" layer 852 Added congestion management requirements in the "data transport" 853 layer section 855 Changed need for non-media data from "question: do we need this?" to 856 "Open issue: How do we do this?" 858 Strengthened disclaimer that listed codecs are placeholders, not 859 decisions. 861 More details on why the "local system support functions" section is 862 there. 864 A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand- 865 rtcweb-overview-00 867 Added section on "Relationship between API and protocol" 869 Added terminology section 871 Mentioned congestion management as part of the "data transport" layer 872 in the layer list 874 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 876 Removed most technical content, and replaced with pointers to drafts 877 as requested and identified by the RTCWEB WG chairs. 879 Added content to acknowledgments section. 881 Added change log. 883 Spell-checked document. 885 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf- 886 rtcweb-overview-00 888 Changed draft name and document date. 890 Removed unused references 892 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 894 Added architecture figures to section 2. 896 Changed the description of "echo cancellation" under "local system 897 support functions". 899 Added a few more definitions. 901 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 903 Added pointers to use cases, security and rtp-usage drafts (now WG 904 drafts). 906 Changed description of SRTP from mandatory-to-use to mandatory-to- 907 implement. 909 Added the "3 principles of negotiation" to the connection management 910 section. 912 Added an explicit statement that ICE is required for both NAT and 913 consent-to-receive. 915 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 917 Added references to a number of new drafts. 919 Expanded the description text under the "trapezoid" drawing with some 920 more text discussed on the list. 922 Changed the "Connection management" sentence from "will be done using 923 SDP offer/answer" to "will be capable of representing SDP offer/ 924 answer" - this seems more consistent with JSEP. 926 Added "security mechanisms" to the things a non-gatewayed SIP devices 927 must support in order to not need a media gateway. 929 Added a definition for "browser". 931 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 933 Made introduction more normative. 935 Several wording changes in response to review comments from EKR 937 Added an appendix to hold references and notes that are not yet in a 938 separate document. 940 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 942 Minor grammatical fixes. This is mainly a "keepalive" refresh. 944 A.10. Changes from -05 to -06 946 Clarifications in response to Last Call review comments. Inserted 947 reference to draft-ietf-rtcweb-audio. 949 A.11. Changes from -06 to -07 951 Added a reference to the "unified plan" draft, and updated some 952 references. 954 Otherwise, it's a "keepalive" draft. 956 A.12. Changes from -07 to -08 958 Removed the appendix that detailed transports, and replaced it with a 959 reference to draft-ietf-rtcweb-transports. Removed now-unused 960 references. 962 A.13. Changes from -08 to -09 964 Added text to the Abstract indicating that the intended status is an 965 Applicability Statement. 967 A.14. Changes from -09 to -10 969 Defined "WebRTC Browser" and "WebRTC device" as things that do, or 970 don't, conform to the API. 972 Updated reference to data-protocol draft 974 Updated data formats to reference -rtcweb-audio- and not the expired 975 -cbran draft. 977 Deleted references to -unified-plan 978 Deleted reference to -generic-idp (draft expired) 980 Added notes on which referenced documents WebRTC browsers or devices 981 MUST conform to. 983 Added pointer to the security section of the API drafts. 985 A.15. Changes from -10 to -11 987 Added "WebRTC Gateway" as a third class of device, and referenced the 988 doc describing them. 990 Made a number of text clarifications in response to document reviews. 992 A.16. Changes from -11 to -12 994 Refined entity definitions to define "WebRTC endpoint" and "WebRTC- 995 compatible endpoint". 997 Changed remaining usage of the term "RTCWEB" to "WebRTC", including 998 in the page header. 1000 A.17. Changes from -12 to -13 1002 Changed "WebRTC device" to be "WebRTC non-browser", per decision at 1003 IETF 91. This led to the need for "WebRTC endpoint" as the common 1004 label for both, and the usage of that term in the rest of the 1005 document. 1007 Added words about WebRTC APIs in languages other than Javascript. 1009 Referenced draft-ietf-rtcweb-video for video codecs to support. 1011 A.18. Changes from -13 to -14 1013 None. This is a "keepalive" update. 1015 A.19. Changes from -14 to -15 1017 Changed "gateways" reference to point to the WG document. 1019 A.20. Changes from -15 to -16 1021 None. This is a "keepalive" publication. 1023 A.21. Changes from -16 to -17 1025 Addressed review comments by Olle E. Johansson and Magnus Westerlund 1027 Author's Address 1029 Harald T. Alvestrand 1030 Google 1031 Kungsbron 2 1032 Stockholm 11122 1033 Sweden 1035 Email: harald@alvestrand.no