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'I-D.roach-mmusic-unified-plan') ** Obsolete normative reference: RFC 5245 (Obsoleted by RFC 8445, RFC 8839) ** Obsolete normative reference: RFC 5766 (Obsoleted by RFC 8656) == Outdated reference: A later version (-16) exists of draft-ietf-rtcweb-use-cases-and-requirements-10 == Outdated reference: A later version (-04) exists of draft-jesup-rtcweb-data-protocol-03 Summary: 4 errors (**), 0 flaws (~~), 12 warnings (==), 1 comment (--).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 August 14, 2013 5 Expires: February 15, 2014 7 Overview: Real Time Protocols for Brower-based Applications 8 draft-ietf-rtcweb-overview-07 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 a work item of the RTCWEB working group. 23 Status of this Memo 25 This Internet-Draft is submitted in full conformance with the 26 provisions of BCP 78 and BCP 79. 28 Internet-Drafts are working documents of the Internet Engineering 29 Task Force (IETF). Note that other groups may also distribute 30 working documents as Internet-Drafts. The list of current Internet- 31 Drafts is at http://datatracker.ietf.org/drafts/current/. 33 Internet-Drafts are draft documents valid for a maximum of six months 34 and may be updated, replaced, or obsoleted by other documents at any 35 time. It is inappropriate to use Internet-Drafts as reference 36 material or to cite them other than as "work in progress." 38 This Internet-Draft will expire on February 15, 2014. 40 Copyright Notice 42 Copyright (c) 2013 IETF Trust and the persons identified as the 43 document authors. All rights reserved. 45 This document is subject to BCP 78 and the IETF Trust's Legal 46 Provisions Relating to IETF Documents 47 (http://trustee.ietf.org/license-info) in effect on the date of 48 publication of this document. Please review these documents 49 carefully, as they describe your rights and restrictions with respect 50 to this document. Code Components extracted from this document must 51 include Simplified BSD License text as described in Section 4.e of 52 the Trust Legal Provisions and are provided without warranty as 53 described in the Simplified BSD License. 55 Table of Contents 57 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 58 2. Principles and Terminology . . . . . . . . . . . . . . . . . . 4 59 2.1. Goals of this document . . . . . . . . . . . . . . . . . . 4 60 2.2. Relationship between API and protocol . . . . . . . . . . 4 61 2.3. On interoperability and innovation . . . . . . . . . . . . 5 62 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 63 3. Architecture and Functionality groups . . . . . . . . . . . . 7 64 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . . 11 65 5. Data framing and securing . . . . . . . . . . . . . . . . . . 12 66 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . . 12 67 7. Connection management . . . . . . . . . . . . . . . . . . . . 12 68 8. Presentation and control . . . . . . . . . . . . . . . . . . . 13 69 9. Local system support functions . . . . . . . . . . . . . . . . 13 70 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14 71 11. Security Considerations . . . . . . . . . . . . . . . . . . . 14 72 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 15 73 13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15 74 13.1. Normative References . . . . . . . . . . . . . . . . . . . 15 75 13.2. Informative References . . . . . . . . . . . . . . . . . . 17 76 Appendix A. Transport and Middlebox specification . . . . . . . . 18 77 A.1. System-provided interfaces . . . . . . . . . . . . . . . . 18 78 A.2. Middle box related functions . . . . . . . . . . . . . . . 18 79 A.3. Transport protocols implemented . . . . . . . . . . . . . 19 80 Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 19 81 B.1. Changes from 82 draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 19 83 B.2. Changes from draft-alvestrand-dispatch-01 to 84 draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 19 85 B.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 20 86 B.4. Changes from draft-alvestrand-rtcweb-overview-01 to 87 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 20 88 B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 20 89 B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 20 90 B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 20 91 B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 21 92 B.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 21 93 B.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 21 94 B.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 21 95 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21 97 1. Introduction 99 The Internet was, from very early in its lifetime, considered a 100 possible vehicle for the deployment of real-time, interactive 101 applications - with the most easily imaginable being audio 102 conversations (aka "Internet telephony") and video conferencing. 104 The first attempts to build this were dependent on special networks, 105 special hardware and custom-built software, often at very high prices 106 or at low quality, placing great demands on the infrastructure. 108 As the available bandwidth has increased, and as processors and other 109 hardware has become ever faster, the barriers to participation have 110 decreased, and it has become possible to deliver a satisfactory 111 experience on commonly available computing hardware. 113 Still, there are a number of barriers to the ability to communicate 114 universally - one of these is that there is, as of yet, no single set 115 of communication protocols that all agree should be made available 116 for communication; another is the sheer lack of universal 117 identification systems (such as is served by telephone numbers or 118 email addresses in other communications systems). 120 Development of The Universal Solution has proved hard, however, for 121 all the usual reasons. 123 The last few years have also seen a new platform rise for deployment 124 of services: The browser-embedded application, or "Web application". 125 It turns out that as long as the browser platform has the necessary 126 interfaces, it is possible to deliver almost any kind of service on 127 it. 129 Traditionally, these interfaces have been delivered by plugins, which 130 had to be downloaded and installed separately from the browser; in 131 the development of HTML5, application developers see much promise in 132 the possibility of making those interfaces available in a 133 standardized way within the browser. 135 This memo describes a set of building blocks that can be made 136 accessible and controllable through a Javascript API in a browser, 137 and which together form a sufficient set of functions to allow the 138 use of interactive audio and video in applications that communicate 139 directly between browsers across the Internet. The resulting 140 protocol suite is intended to enable all the applications that are 141 described as required scenarios in the RTCWEB use cases document 142 [I-D.ietf-rtcweb-use-cases-and-requirements]. 144 Other efforts, for instance the W3C WebRTC, Web Applications and 145 Device API working groups, focus on making standardized APIs and 146 interfaces available, within or alongside the HTML5 effort, for those 147 functions; this memo concentrates on specifying the protocols and 148 subprotocols that are needed to specify the interactions that happen 149 across the network. 151 2. Principles and Terminology 153 2.1. Goals of this document 155 The goal of the RTCWEB protocol specification is to specify a set of 156 protocols that, if all are implemented, will allow an implementation 157 to communicate with another implementation using audio, video and 158 data sent along the most direct possible path between the 159 participants. 161 This document is intended to serve as the roadmap to the RTCWEB 162 specifications. It defines terms used by other pieces of 163 specification, lists references to other specifications that don't 164 need further elaboration in the RTCWEB context, and gives pointers to 165 other documents that form part of the RTCWEB suite. 167 By reading this document and the documents it refers to, it should be 168 possible to have all information needed to implement an RTCWEB 169 compatible implementation. 171 2.2. Relationship between API and protocol 173 The total RTCWEB/WEBRTC effort consists of two pieces: 175 o A protocol specification, done in the IETF 177 o A Javascript API specification, done in the W3C 178 [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] 180 Together, these two specifications aim to provide an environment 181 where Javascript embedded in any page, viewed in any compatible 182 browser, when suitably authorized by its user, is able to set up 183 communication using audio, video and auxiliary data, where the 184 browser environment does not constrain the types of application in 185 which this functionality can be used. 187 The protocol specification does not assume that all implementations 188 implement this API; it is not intended to be necessary for 189 interoperation to know whether the entity one is communicating with 190 is a browser or another device implementing this specification. 192 The goal of cooperation between the protocol specification and the 193 API specification is that for all options and features of the 194 protocol specification, it should be clear which API calls to make to 195 exercise that option or feature; similarly, for any sequence of API 196 calls, it should be clear which protocol options and features will be 197 invoked. Both subject to constraints of the implementation, of 198 course. 200 2.3. On interoperability and innovation 202 The "Mission statement of the IETF" [RFC3935] states that "The 203 benefit of a standard to the Internet is in interoperability - that 204 multiple products implementing a standard are able to work together 205 in order to deliver valuable functions to the Internet's users." 207 Communication on the Internet frequently occurs in two phases: 209 o Two parties communicate, through some mechanism, what 210 functionality they both are able to support 212 o They use that shared communicative functionality to communicate, 213 or, failing to find anything in common, give up on communication. 215 There are often many choices that can be made for communicative 216 functionality; the history of the Internet is rife with the proposal, 217 standardization, implementation, and success or failure of many types 218 of options, in all sorts of protocols. 220 The goal of having a mandatory to implement function set is to 221 prevent negotiation failure, not to preempt or prevent negotiation. 223 The presence of a mandatory to implement function set serves as a 224 strong changer of the marketplace of deployment - in that it gives a 225 guarantee that, as long as you conform to a specification, and the 226 other party is willing to accept communication at the base level of 227 that specification, you can communicate successfully. 229 The alternative - that of having no mandatory to implement - does not 230 mean that you cannot communicate, it merely means that in order to be 231 part of the communications partnership, you have to implement the 232 standard "and then some" - that "and then some" usually being called 233 a profile of some sort; in the version most antithetical to the 234 Internet ethos, that "and then some" consists of having to use a 235 specific vendor's product only. 237 2.4. Terminology 239 The following terms are used in this document, and as far as possible 240 across the documents specifying the RTCWEB suite, in the specific 241 meanings given here. Not all terms are used in this document. Other 242 terms are used in their commonly used meaning. 244 The list is in alphabetical order. 246 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 248 API: Application Programming Interface - a specification of a set of 249 calls and events, usually tied to a programming language or an 250 abstract formal specification such as WebIDL, with its defined 251 semantics. 253 Browser: Used synonymously with "Interactive User Agent" as defined 254 in the HTML specification [W3C.WD-html5-20110525]. 256 ICE Agent: An implementation of the ICE [RFC5245] protocol. An ICE 257 Agent may also be an SDP Agent, but there exist ICE Agents that do 258 not use SDP (for instance those that use Jingle). 260 Interactive: Communication between multiple parties, where the 261 expectation is that an action from one party can cause a reaction 262 by another party, and the reaction can be observed by the first 263 party, with the total time required for the action/reaction/ 264 observation is on the order of no more than hundreds of 265 milliseconds. 267 Media: Audio and video content. Not to be confused with 268 "transmission media" such as wires. 270 Media path: The path that media data follows from one browser to 271 another. 273 Protocol: A specification of a set of data units, their 274 representation, and rules for their transmission, with their 275 defined semantics. A protocol is usually thought of as going 276 between systems. 278 Real-time media: Media where generation of content and display of 279 content are intended to occur closely together in time (on the 280 order of no more than hundreds of milliseconds). Real-time media 281 can be used to support interactive communication. 283 SDP Agent: The protocol implementation involved in the SDP offer/ 284 answer exchange, as defined in [RFC3264] section 3. 286 Signaling: Communication that happens in order to establish, manage 287 and control media paths. 289 Signaling Path: The communication channels used between entities 290 participating in signaling to transfer signaling. There may be 291 more entities in the signaling path than in the media path. 293 NOTE: Where common definitions exist for these terms, those 294 definitions should be used to the greatest extent possible. 296 TODO: Extend this list with other terms that might prove slippery. 298 3. Architecture and Functionality groups 300 The model of real-time support for browser-based applications does 301 not envisage that the browser will contain all the functions that 302 need to be performed in order to have a function such as a telephone 303 or a video conferencing unit; the vision is that the browser will 304 have the functions that are needed for a Web application, working in 305 conjunction with its backend servers, to implement these functions. 307 This means that two vital interfaces need specification: The 308 protocols that browsers talk to each other, without any intervening 309 servers, and the APIs that are offered for a Javascript application 310 to take advantage of the browser's functionality. 312 +------------------------+ On-the-wire 313 | | Protocols 314 | Servers |---------> 315 | | 316 | | 317 +------------------------+ 318 ^ 319 | 320 | 321 | HTTP/ 322 | Websockets 323 | 324 | 325 +----------------------------+ 326 | Javascript/HTML/CSS | 327 +----------------------------+ 328 Other ^ ^RTC 329 APIs | |APIs 330 +---|-----------------|------+ 331 | | | | 332 | +---------+| 333 | | Browser || On-the-wire 334 | Browser | RTC || Protocols 335 | | Function|-----------> 336 | | || 337 | | || 338 | +---------+| 339 +---------------------|------+ 340 | 341 V 342 Native OS Services 344 Figure 1: Browser Model 346 Note that HTTP and Websockets are also offered to the Javascript 347 application through browser APIs. 349 As for all protocol and API specifications, there is no restriction 350 that the protocols can only be used to talk to another browser; since 351 they are fully specified, any device that implements the protocols 352 faithfully should be able to interoperate with the application 353 running in the browser. 355 A commonly imagined model of deployment is the one depicted below. 357 +-----------+ +-----------+ 358 | Web | | Web | 359 | | Signaling | | 360 | |-------------| | 361 | Server | path | Server | 362 | | | | 363 +-----------+ +-----------+ 364 / \ 365 / \ Application-defined 366 / \ over 367 / \ HTTP/Websockets 368 / Application-defined over \ 369 / HTTP/Websockets \ 370 / \ 371 +-----------+ +-----------+ 372 |JS/HTML/CSS| |JS/HTML/CSS| 373 +-----------+ +-----------+ 374 +-----------+ +-----------+ 375 | | | | 376 | | | | 377 | Browser | ------------------------- | Browser | 378 | | Media path | | 379 | | | | 380 +-----------+ +-----------+ 382 Figure 2: Browser RTC Trapezoid 384 On this drawing, the critical part to note is that the media path 385 ("low path") goes directly between the browsers, so it has to be 386 conformant to the specifications of the RTCWEB protocol suite; the 387 signaling path ("high path") goes via servers that can modify, 388 translate or massage the signals as needed. 390 If the two Web servers are operated by different entities, the inter- 391 server signaling mechanism needs to be agreed upon, either by 392 standardization or by other means of agreement. Existing protocols 393 (for example SIP or XMPP) could be used between servers, while either 394 a standards-based or proprietary protocol could be used between the 395 browser and the web server. 397 For example, if both operators' servers implement SIP, SIP could be 398 used for communication between servers, along with either a 399 standardized signaling mechanism (e.g. SIP over Websockets) or a 400 proprietary signaling mechanism used between the application running 401 in the browser and the web server. Similarly, if both operators' 402 servers implement XMPP, XMPP could be used for communication between 403 XMPP servers, with either a standardized signaling mechanism (e.g. 404 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 405 used between the application running in the browser and the web 406 server. 408 The choice of protocols, and definition of the translation between 409 them, is outside the scope of the RTCWEB standards suite described in 410 the document. 412 The functionality groups that are needed in the browser can be 413 specified, more or less from the bottom up, as: 415 o Data transport: TCP, UDP and the means to securely set up 416 connections between entities, as well as the functions for 417 deciding when to send data: Congestion management, bandwidth 418 estimation and so on. 420 o Data framing: RTP and other data formats that serve as containers, 421 and their functions for data confidentiality and integrity. 423 o Data formats: Codec specifications, format specifications and 424 functionality specifications for the data passed between systems. 425 Audio and video codecs, as well as formats for data and document 426 sharing, belong in this category. In order to make use of data 427 formats, a way to describe them, a session description, is needed. 429 o Connection management: Setting up connections, agreeing on data 430 formats, changing data formats during the duration of a call; SIP 431 and Jingle/XMPP belong in this category. 433 o Presentation and control: What needs to happen in order to ensure 434 that interactions behave in a non-surprising manner. This can 435 include floor control, screen layout, voice activated image 436 switching and other such functions - where part of the system 437 require the cooperation between parties. XCON and Cisco/ 438 Tandberg's TIP were some attempts at specifying this kind of 439 functionality; many applications have been built without 440 standardized interfaces to these functions. 442 o Local system support functions: These are things that need not be 443 specified uniformly, because each participant may choose to do 444 these in a way of the participant's choosing, without affecting 445 the bits on the wire in a way that others have to be cognizant of. 446 Examples in this category include echo cancellation (some forms of 447 it), local authentication and authorization mechanisms, OS access 448 control and the ability to do local recording of conversations. 450 Within each functionality group, it is important to preserve both 451 freedom to innovate and the ability for global communication. 452 Freedom to innovate is helped by doing the specification in terms of 453 interfaces, not implementation; any implementation able to 454 communicate according to the interfaces is a valid implementation. 455 Ability to communicate globally is helped both by having core 456 specifications be unencumbered by IPR issues and by having the 457 formats and protocols be fully enough specified to allow for 458 independent implementation. 460 One can think of the three first groups as forming a "media transport 461 infrastructure", and of the three last groups as forming a "media 462 service". In many contexts, it makes sense to use a common 463 specification for the media transport infrastructure, which can be 464 embedded in browsers and accessed using standard interfaces, and "let 465 a thousand flowers bloom" in the "media service" layer; to achieve 466 interoperable services, however, at least the first five of the six 467 groups need to be specified. 469 4. Data transport 471 Data transport refers to the sending and receiving of data over the 472 network interfaces, the choice of network-layer addresses at each end 473 of the communication, and the interaction with any intermediate 474 entities that handle the data, but do not modify it (such as TURN 475 relays). 477 It includes necessary functions for congestion control: When not to 478 send data. 480 The data transport protocols used by RTCWEB are described in . 483 ICE is required for all media paths that use UDP; in addition to the 484 ability to pass NAT boxes, ICE fulfills the need for guaranteeing 485 that the media path is going to a UDP port that is willing to receive 486 the data. 488 The data transport protocols used by RTCWEB, as well as the details 489 of interactions with intermediate boxes, such as firewalls, relays 490 and NAT boxes, are intended to be described in a separate document; 491 for now, notes are gathered in Appendix A. 493 5. Data framing and securing 495 The format for media transport is RTP [RFC3550]. Implementation of 496 SRTP [RFC3711] is required for all implementations. 498 The detailed considerations for usage of functions from RTP and SRTP 499 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 500 considerations for the RTCWEB use case are in 501 [I-D.ietf-rtcweb-security], and the resulting security functions are 502 described in [I-D.ietf-rtcweb-security-arch]. 504 Considerations for the transfer of data that is not in RTP format is 505 described in [I-D.ietf-rtcweb-data-channel], and the resulting 506 protocol is described in [I-D.jesup-rtcweb-data-protocol] (not yet a 507 WG document) 509 6. Data formats 511 The intent of this specification is to allow each communications 512 event to use the data formats that are best suited for that 513 particular instance, where a format is supported by both sides of the 514 connection. However, a minimum standard is greatly helpful in order 515 to ensure that communication can be achieved. This document 516 specifies a minimum baseline that will be supported by all 517 implementations of this specification, and leaves further codecs to 518 be included at the will of the implementor. 520 The mandatory to implement codecs, as well as any profiling 521 requirements for both mandatory and optional codecs, is described in 522 (candidate draft: 523 [I-D.cbran-rtcweb-codec]. 525 7. Connection management 527 The methods, mechanisms and requirements for setting up, negotiating 528 and tearing down connections is a large subject, and one where it is 529 desirable to have both interoperability and freedom to innovate. 531 The following principles apply: 533 1. The RTCWEB media negotiations will be capable of representing the 534 same SDP offer/answer semantics that are used in SIP [RFC3264], 535 in such a way that it is possible to build a signaling gateway 536 between SIP and the RTCWEB media negotiation. 538 2. It will be possible to gateway between legacy SIP devices that 539 support ICE and appropriate RTP / SDP mechanisms, codecs and 540 security mechanisms without using a media gateway. A signaling 541 gateway to convert between the signaling on the web side to the 542 SIP signaling may be needed. 544 3. When a new codec is specified, and the SDP for the new codec is 545 specified in the MMUSIC WG, no other standardization should be 546 required for it to be possible to use that in the web browsers. 547 Adding new codecs which might have new SDP parameters should not 548 change the APIs between the browser and Javascript application. 549 As soon as the browsers support the new codecs, old applications 550 written before the codecs were specified should automatically be 551 able to use the new codecs where appropriate with no changes to 552 the JS applications. 554 The particular choices made for RTCWEB, and their implications for 555 the API offered by a browser implementing RTCWEB, are described in 556 [I-D.ietf-rtcweb-jsep]. This document in turn implements the 557 solutions described in [I-D.roach-mmusic-unified-plan]. 559 8. Presentation and control 561 The most important part of control is the user's control over the 562 browser's interaction with input/output devices and communications 563 channels. It is important that the user have some way of figuring 564 out where his audio, video or texting is being sent, for what 565 purported reason, and what guarantees are made by the parties that 566 form part of this control channel. This is largely a local function 567 between the browser, the underlying operating system and the user 568 interface; this is being worked on as part of the W3C API effort, and 569 will be part of the peer connection API [W3C.WD-webrtc-20120209], and 570 the media capture API [W3C.WD-mediacapture-streams-20120628]. 571 Considerations for the implications of wanting to identify 572 correspondents are described in [I-D.rescorla-rtcweb-generic-idp] 573 (not a WG item). 575 9. Local system support functions 577 These are characterized by the fact that the quality of these 578 functions strongly influence the user experience, but the exact 579 algorithm does not need coordination. In some cases (for instance 580 echo cancellation, as described below), the overall system definition 581 may need to specify that the overall system needs to have some 582 characteristics for which these facilities are useful, without 583 requiring them to be implemented a certain way. 585 Local functions include echo cancellation, volume control, camera 586 management including focus, zoom, pan/tilt controls (if available), 587 and more. 589 Certain parts of the system SHOULD conform to certain properties, for 590 instance: 592 o Echo cancellation should be good enough to achieve the suppression 593 of acoustical feedback loops below a perceptually noticeable 594 level. 596 o Privacy concerns must be satisfied; for instance, if remote 597 control of camera is offered, the APIs should be available to let 598 the local participant figure out who's controlling the camera, and 599 possibly decide to revoke the permission for camera usage. 601 o Automatic gain control, if present, should normalize a speaking 602 voice into a reasonable dB range. 604 The requirements on RTCWEB systems with regard to audio processing 605 are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of 606 local devices are found in [W3C.WD-mediacapture-streams-20120628]. 608 10. IANA Considerations 610 This document makes no request of IANA. 612 Note to RFC Editor: this section may be removed on publication as an 613 RFC. 615 11. Security Considerations 617 Security of the web-enabled real time communications comes in several 618 pieces: 620 o Security of the components: The browsers, and other servers 621 involved. The most target-rich environment here is probably the 622 browser; the aim here should be that the introduction of these 623 components introduces no additional vulnerability. 625 o Security of the communication channels: It should be easy for a 626 participant to reassure himself of the security of his 627 communication - by verifying the crypto parameters of the links he 628 himself participates in, and to get reassurances from the other 629 parties to the communication that they promise that appropriate 630 measures are taken. 632 o Security of the partners' identity: verifying that the 633 participants are who they say they are (when positive 634 identification is appropriate), or that their identity cannot be 635 uncovered (when anonymity is a goal of the application). 637 The security analysis, and the requirements derived from that 638 analysis, is contained in [I-D.ietf-rtcweb-security]. 640 12. Acknowledgements 642 The number of people who have taken part in the discussions 643 surrounding this draft are too numerous to list, or even to identify. 644 The ones below have made special, identifiable contributions; this 645 does not mean that others' contributions are less important. 647 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 648 Westerlund and Joerg Ott, who offered technical contributions on 649 various versions of the draft. 651 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 652 the ASCII drawings in section 1. 654 Thanks to Eric Rescorla, Justin Uberti, Henry Sinnreich, Colin 655 Perkins, Bjoern Hoehrmann and Simon Leinen for document review, and 656 to Heath Matlock for grammatical review. 658 13. References 660 13.1. Normative References 662 [I-D.ietf-mmusic-sctp-sdp] 663 Loreto, S. and G. Camarillo, "Stream Control Transmission 664 Protocol (SCTP)-Based Media Transport in the Session 665 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-03 666 (work in progress), January 2013. 668 [I-D.ietf-rtcweb-audio] 669 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 670 Requirements", draft-ietf-rtcweb-audio-01 (work in 671 progress), November 2012. 673 [I-D.ietf-rtcweb-data-channel] 674 Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Datagram 675 Connection", draft-ietf-rtcweb-data-channel-02 (work in 676 progress), October 2012. 678 [I-D.ietf-rtcweb-jsep] 679 Uberti, J. and C. Jennings, "Javascript Session 680 Establishment Protocol", draft-ietf-rtcweb-jsep-02 (work 681 in progress), October 2012. 683 [I-D.ietf-rtcweb-rtp-usage] 684 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 685 Communication (WebRTC): Media Transport and Use of RTP", 686 draft-ietf-rtcweb-rtp-usage-05 (work in progress), 687 October 2012. 689 [I-D.ietf-rtcweb-security] 690 Rescorla, E., "Security Considerations for RTC-Web", 691 draft-ietf-rtcweb-security-04 (work in progress), 692 January 2013. 694 [I-D.ietf-rtcweb-security-arch] 695 Rescorla, E., "RTCWEB Security Architecture", 696 draft-ietf-rtcweb-security-arch-06 (work in progress), 697 January 2013. 699 [I-D.ietf-tsvwg-sctp-dtls-encaps] 700 Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS 701 Encapsulation of SCTP Packets for RTCWEB", 702 draft-ietf-tsvwg-sctp-dtls-encaps-00 (work in progress), 703 February 2013. 705 [I-D.nandakumar-rtcweb-stun-uri] 706 Nandakumar, S., Salgueiro, G., Jones, P., and M. Petit- 707 Huguenin, "URI Scheme for Session Traversal Utilities for 708 NAT (STUN) Protocol", draft-nandakumar-rtcweb-stun-uri-03 709 (work in progress), January 2013. 711 [I-D.roach-mmusic-unified-plan] 712 Roach, A., Uberti, J., and M. Thomson, "A Unified Plan for 713 Using SDP with Large Numbers of Media Flows", 714 draft-roach-mmusic-unified-plan-00 (work in progress), 715 July 2013. 717 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 718 with Session Description Protocol (SDP)", RFC 3264, 719 June 2002. 721 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 722 Jacobson, "RTP: A Transport Protocol for Real-Time 723 Applications", RFC 3550, July 2003. 725 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 727 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 728 RFC 3711, March 2004. 730 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 731 (ICE): A Protocol for Network Address Translator (NAT) 732 Traversal for Offer/Answer Protocols", RFC 5245, 733 April 2010. 735 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 736 Relays around NAT (TURN): Relay Extensions to Session 737 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 739 13.2. Informative References 741 [I-D.cbran-rtcweb-codec] 742 Bran, C., Jennings, C., and J. Valin, "WebRTC Codec and 743 Media Processing Requirements", 744 draft-cbran-rtcweb-codec-02 (work in progress), 745 March 2012. 747 [I-D.ietf-rtcweb-use-cases-and-requirements] 748 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 749 Time Communication Use-cases and Requirements", 750 draft-ietf-rtcweb-use-cases-and-requirements-10 (work in 751 progress), December 2012. 753 [I-D.jesup-rtcweb-data-protocol] 754 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 755 Protocol", draft-jesup-rtcweb-data-protocol-03 (work in 756 progress), September 2012. 758 [I-D.rescorla-rtcweb-generic-idp] 759 Rescorla, E., "RTCWEB Generic Identity Provider 760 Interface", draft-rescorla-rtcweb-generic-idp-01 (work in 761 progress), March 2012. 763 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", 764 BCP 95, RFC 3935, October 2004. 766 [W3C.WD-html5-20110525] 767 Hickson, I., "HTML5", World Wide Web Consortium 768 LastCall WD-html5-20110525, May 2011, 769 . 771 [W3C.WD-mediacapture-streams-20120628] 772 Burnett, D. and A. Narayanan, "Media Capture and Streams", 773 World Wide Web Consortium WD WD-mediacapture-streams- 774 20120628, June 2012, . 777 [W3C.WD-webrtc-20120209] 778 Bergkvist, A., Burnett, D., Narayanan, A., and C. 779 Jennings, "WebRTC 1.0: Real-time Communication Between 780 Browsers", World Wide Web Consortium WD WD-webrtc- 781 20120209, February 2012, 782 . 784 Appendix A. Transport and Middlebox specification 786 The draft referred to as "transport and middle boxes" in Section 4 787 has not been written yet. This appendix contains some keywords to 788 what it should say; this also serves the purpose of linking to the 789 drafts-in-progress that are relevant to this specification. 791 A.1. System-provided interfaces 793 The protocol specifications used here assume that the following 794 protocols are available as system-level interfaces: 796 o UDP. This is the protocol assumed by most protocol elements 797 described. 799 o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL 800 and ICE-TCP. 802 For both protocols, this specification assumes the ability to set the 803 DSCP code point of the sockets opened. It does not assume that the 804 DSCP codepoints will be honored, and does assume that they may be 805 zeroed or changed, since this is a local configuration issue. 807 This specification does not assume that the implementation will have 808 access to ICMP or raw IP. 810 A.2. Middle box related functions 812 The primary mechanism to deal with middle boxes is ICE, which is an 813 appropriate way to deal with NAT boxes and firewalls that accept 814 traffic from the inside, but only from the outside if it's in 815 response to inside traffic (simple stateful firewalls). 817 In order to deal with symmetric NATs, TURN MUST be supported. 819 In order to deal with firewalls that block all UDP traffic, TURN over 820 TCP MUST be supported. (QUESTION: What about ICE-TCP?) 821 The following specifications MUST be supported: 823 o ICE [RFC5245] 825 o TURN, including TURN over TCP [[QUESTION: and TURN over TLS]], 826 [RFC5766]. 828 For referring to STUN and TURN servers, this specification depends on 829 the STUN URI, [I-D.nandakumar-rtcweb-stun-uri]. 831 A.3. Transport protocols implemented 833 For data transport, RTCWEB implementations support SCTP over DTLS 834 over ICE. This is specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. 835 Negotiation of this transport in SCTP is defined in 836 [I-D.ietf-mmusic-sctp-sdp]. 838 Appendix B. Change log 840 This section may be deleted by the RFC Editor when preparing for 841 publication. 843 B.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 845 Added section "On interoperability and innovation" 847 Added data confidentiality and integrity to the "data framing" layer 849 Added congestion management requirements in the "data transport" 850 layer section 852 Changed need for non-media data from "question: do we need this?" to 853 "Open issue: How do we do this?" 855 Strengthened disclaimer that listed codecs are placeholders, not 856 decisions. 858 More details on why the "local system support functions" section is 859 there. 861 B.2. Changes from draft-alvestrand-dispatch-01 to 862 draft-alvestrand-rtcweb-overview-00 864 Added section on "Relationship between API and protocol" 866 Added terminology section 867 Mentioned congestion management as part of the "data transport" layer 868 in the layer list 870 B.3. Changes from draft-alvestrand-rtcweb-00 to -01 872 Removed most technical content, and replaced with pointers to drafts 873 as requested and identified by the RTCWEB WG chairs. 875 Added content to acknowledgments section. 877 Added change log. 879 Spell-checked document. 881 B.4. Changes from draft-alvestrand-rtcweb-overview-01 to 882 draft-ietf-rtcweb-overview-00 884 Changed draft name and document date. 886 Removed unused references 888 B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 890 Added architecture figures to section 2. 892 Changed the description of "echo cancellation" under "local system 893 support functions". 895 Added a few more definitions. 897 B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 899 Added pointers to use cases, security and rtp-usage drafts (now WG 900 drafts). 902 Changed description of SRTP from mandatory-to-use to mandatory-to- 903 implement. 905 Added the "3 principles of negotiation" to the connection management 906 section. 908 Added an explicit statement that ICE is required for both NAT and 909 consent-to-receive. 911 B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 913 Added references to a number of new drafts. 915 Expanded the description text under the "trapezoid" drawing with some 916 more text discussed on the list. 918 Changed the "Connection management" sentence from "will be done using 919 SDP offer/answer" to "will be capable of representing SDP offer/ 920 answer" - this seems more consistent with JSEP. 922 Added "security mechanisms" to the things a non-gatewayed SIP devices 923 must support in order to not need a media gateway. 925 Added a definition for "browser". 927 B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 929 Made introduction more normative. 931 Several wording changes in response to review comments from EKR 933 Added Appendix A to hold references and notes that are not yet in a 934 separate document. 936 B.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 938 Minor grammatical fixes. This is mainly a "keepalive" refresh. 940 B.10. Changes from -05 to -06 942 Clarifications in response to Last Call review comments. Inserted 943 reference to draft-ietf-rtcweb-audio. 945 B.11. Changes from -06 to -07 947 Added a refereence to the "unified plan" draft, and updated some 948 references. 950 Otherwise, it's a "keepalive" draft. 952 Author's Address 954 Harald T. Alvestrand 955 Google 956 Kungsbron 2 957 Stockholm, 11122 958 Sweden 960 Email: harald@alvestrand.no