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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group H. Alvestrand 3 Internet-Draft Google 4 Intended status: Standards Track June 20, 2012 5 Expires: December 22, 2012 7 Overview: Real Time Protocols for Brower-based Applications 8 draft-ietf-rtcweb-overview-04 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 Requirements Language 25 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 26 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 27 document are to be interpreted as described in RFC 2119 [RFC2119]. 29 Status of this Memo 31 This Internet-Draft is submitted in full conformance with the 32 provisions of BCP 78 and BCP 79. 34 Internet-Drafts are working documents of the Internet Engineering 35 Task Force (IETF). Note that other groups may also distribute 36 working documents as Internet-Drafts. The list of current Internet- 37 Drafts is at http://datatracker.ietf.org/drafts/current/. 39 Internet-Drafts are draft documents valid for a maximum of six months 40 and may be updated, replaced, or obsoleted by other documents at any 41 time. It is inappropriate to use Internet-Drafts as reference 42 material or to cite them other than as "work in progress." 44 This Internet-Draft will expire on December 22, 2012. 46 Copyright Notice 48 Copyright (c) 2012 IETF Trust and the persons identified as the 49 document authors. All rights reserved. 51 This document is subject to BCP 78 and the IETF Trust's Legal 52 Provisions Relating to IETF Documents 53 (http://trustee.ietf.org/license-info) in effect on the date of 54 publication of this document. Please review these documents 55 carefully, as they describe your rights and restrictions with respect 56 to this document. Code Components extracted from this document must 57 include Simplified BSD License text as described in Section 4.e of 58 the Trust Legal Provisions and are provided without warranty as 59 described in the Simplified BSD License. 61 Table of Contents 63 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 64 2. Principles and Terminology . . . . . . . . . . . . . . . . . . 5 65 2.1. Goals of this document . . . . . . . . . . . . . . . . . . 5 66 2.2. Relationship between API and protocol . . . . . . . . . . 5 67 2.3. On interoperability and innovation . . . . . . . . . . . . 6 68 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 7 69 3. Architecture and Functionality groups . . . . . . . . . . . . 8 70 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . . 12 71 5. Data framing and securing . . . . . . . . . . . . . . . . . . 13 72 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . . 13 73 7. Connection management . . . . . . . . . . . . . . . . . . . . 13 74 8. Presentation and control . . . . . . . . . . . . . . . . . . . 14 75 9. Local system support functions . . . . . . . . . . . . . . . . 14 76 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 77 11. Security Considerations . . . . . . . . . . . . . . . . . . . 15 78 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 16 79 13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 16 80 13.1. Normative References . . . . . . . . . . . . . . . . . . . 16 81 13.2. Informative References . . . . . . . . . . . . . . . . . . 18 82 Appendix A. Transport and Middlebox specification . . . . . . . . 19 83 A.1. System-provided interfaces . . . . . . . . . . . . . . . . 19 84 A.2. Middle box related functions . . . . . . . . . . . . . . . 19 85 A.3. Transport protocols implemented . . . . . . . . . . . . . 20 86 Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 20 87 B.1. Changes from 88 draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 20 89 B.2. Changes from draft-alvestrand-dispatch-01 to 90 draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 20 91 B.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 20 92 B.4. Changes from draft-alvestrand-rtcweb-overview-01 to 93 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 21 94 B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 21 95 B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 21 96 B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 21 97 B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 22 98 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 22 100 1. Introduction 102 The Internet was, from very early in its lifetime, considered a 103 possible vehicle for the deployment of real-time, interactive 104 applications - with the most easily imaginable being audio 105 conversations (aka "Internet telephony") and videoconferencing. 107 The first attempts to build this were dependent on special networks, 108 special hardware and custom-built software, often at very high prices 109 or at low quality, placing great demands on the infrastructure. 111 As the available bandwidth has increased, and as processors and other 112 hardware has become ever faster, the barriers to participation have 113 decreased, and it has become possible to deliver a satisfactory 114 experience on commonly available computing hardware. 116 Still, there are a number of barriers to the ability to communicate 117 universally - one of these is that there is, as of yet, no single set 118 of communication protocols that all agree should be made available 119 for communication; another is the sheer lack of universal 120 identification systems (such as is served by telephone numbers or 121 email addresses in other communications systems). 123 Development of The Universal Solution has proved hard, however, for 124 all the usual reasons. 126 The last few years have also seen a new platform rise for deployment 127 of services: The browser-embedded application, or "Web application". 128 It turns out that as long as the browser platform has the necessary 129 interfaces, it is possible to deliver almost any kind of service on 130 it. 132 Traditionally, these interfaces have been delivered by plugins, which 133 had to be downloaded and installed separately from the browser; in 134 the development of HTML5, application developers see much promise in 135 the possibility of making those interfaces available in a 136 standardized way within the browser. 138 This memo describes a set of building blocks that can be made 139 accessible and controllable through a Javascript API in a browser, 140 and which together form a sufficient set of functions to allow the 141 use of interactive audio and video in applications that communicate 142 directly between browsers across the Internet. The resulting 143 protocol suite is intended to enable all the applications that are 144 described as required scenarios in the RTCWEB use cases document 145 [I-D.ietf-rtcweb-use-cases-and-requirements]. 147 Other efforts, for instance the W3C WebRTC, Web Applications and 148 Device API working groups, focus on making standardized APIs and 149 interfaces available, within or alongside the HTML5 effort, for those 150 functions; this memo concentrates on specifying the protocols and 151 subprotocols that are needed to specify the interactions that happen 152 across the network. 154 2. Principles and Terminology 156 2.1. Goals of this document 158 The goal of the RTCWEB protocol specification is to specify a set of 159 protocols that, if all are implemented, will allow an implementation 160 to communicate with another implementation using audio, video and 161 data sent along the most direct possible path between the 162 participants. 164 This document is intended to serve as the roadmap to the RTCWEB 165 specifications. It defines terms used by other pieces of 166 specification, lists references to other specifications that don't 167 need further elaboration in the RTCWEB context, and gives pointers to 168 other documents that form part of the RTCWEB suite. 170 By reading this document and the documents it refers to, it should be 171 possible to have all information needed to implement an RTCWEB 172 compatible implementation. 174 2.2. Relationship between API and protocol 176 The total RTCWEB/WEBRTC effort consists of two pieces: 178 o A protocol specification, done in the IETF 180 o A Javascript API specification, done in the W3C 181 [W3C.WD-webrtc-20120209] 183 Together, these two specifications aim to provide an environment 184 where Javascript embedded in any page, viewed in any compatible 185 browser, when suitably authorized by its user, is able to set up 186 communication using audio, video and auxiliary data, where the 187 browser environment does not constrain the types of application in 188 which this functionality can be used. 190 The protocol specification does not assume that all implementations 191 implement this API; it is not intended to be necessary for 192 interoperation to know whether the entity one is communicating with 193 is a browser or another device implementing this specification. 195 The goal of cooperation between the protocol specification and the 196 API specification is that for all options and features of the 197 protocol specification, it should be clear which API calls to make to 198 exercise that option or feature; similarly, for any sequence of API 199 calls, it should be clear which protocol options and features will be 200 invoked. Both subject to constraints of the implementation, of 201 course. 203 2.3. On interoperability and innovation 205 The "Mission statement of the IETF" [RFC3935] states that "The 206 benefit of a standard to the Internet is in interoperability - that 207 multiple products implementing a standard are able to work together 208 in order to deliver valuable functions to the Internet's users." 210 Communication on the Internet frequently occurs in two phases: 212 o Two parties communicate, through some mechanism, what 213 functionality they both are able to support 215 o They use that shared communicative functionality to communicate, 216 or, failing to find anything in common, give up on communication. 218 There are often many choices that can be made for communicative 219 functionality; the history of the Internet is rife with the proposal, 220 standardization, implementation, and success or failure of many types 221 of options, in all sorts of protocols. 223 The goal of having a mandatory to implement function set is to 224 prevent negotiation failure, not to preempt or prevent negotiation. 226 The presence of a mandatory to implement function set serves as a 227 strong changer of the marketplace of deployment - in that it gives a 228 guarantee that, as long as you conform to a specification, and the 229 other party is willing to accept communication at the base level of 230 that specification, you can communicate successfully. 232 The alternative - that of having no mandatory to implement - does not 233 mean that you cannot communicate, it merely means that in order to be 234 part of the communications partnership, you have to implement the 235 standard "and then some" - that "and then some" usually being called 236 a profile of some sort; in the version most antithetical to the 237 Internet ethos, that "and then some" consists of having to use a 238 specific vendor's product only. 240 2.4. Terminology 242 The following terms are used in this document, and as far as possible 243 across the documents specifying the RTCWEB suite, in the specific 244 meanings given here. Not all terms are used in this document. Other 245 terms are used in their commonly used meaning. 247 The list is in alphabetical order. 249 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 251 API: Application Programming Interface - a specification of a set of 252 calls and events, usually tied to a programming language or an 253 abstract formal specification such as WebIDL, with its defined 254 semantics. 256 Browser: Used synonymously with "Interactive User Agent" as defined 257 in the HTML specification [W3C.WD-html5-20110525]. 259 ICE Agent: An implementation of the ICE [RFC5245] protocol. An ICE 260 Agent may also be an SDP Agent, but there exist ICE Agents that do 261 not use SDP (for instance those that use Jingle). 263 Interactive: Communication between multiple parties, where the 264 expectation is that an action from one party can cause a reaction 265 by another party, and the reaction can be observed by the first 266 party, with the total time required for the action/reaction/ 267 observation is on the order of no more than hundreds of 268 milliseconds. 270 Media: Audio and video content. Not to be confused with 271 "transmission media" such as wires. 273 Media path: The path that media data follows from one browser to 274 another. 276 Protocol: A specification of a set of data units, their 277 representation, and rules for their transmission, with their 278 defined semantics. A protocol is usually thought of as going 279 between systems. 281 Real-time media: Media where generation of content and display of 282 content are intended to occur closely together in time (on the 283 order of no more than hundreds of milliseconds). Real-time media 284 can be used to support interactive communication. 286 SDP Agent: The protocol implementation involved in the SDP offer/ 287 answer exchange, as defined in [RFC3264] section 3. 289 Signaling: Communication that happens in order to establish, manage 290 and control media paths. 292 Signaling Path: The communication channels used between entities 293 participating in signalling to transfer signaling. There may be 294 more entities in the signaling path than in the media path. 296 NOTE: Where common definitions exist for these terms, those 297 definitions should be used to the greatest extent possible. 299 TODO: Extend this list with other terms that might prove slippery. 301 3. Architecture and Functionality groups 303 The model of real-time support for browser-based applications does 304 not envisage that the browser will contain all the functions that 305 need to be performed in order to have a function such as a telephone 306 or a videoconferencing unit; the vision is that the browser will have 307 the functions that are needed for a Web application, working in 308 conjunction with its backend servers, to implement these functions. 310 This means that two vital interfaces need specification: The 311 protocols that browsers talk to each other, without any intervening 312 servers, and the APIs that are offered for a Javascript application 313 to take advantage of the browser's functionality. 315 +------------------------+ On-the-wire 316 | | Protocols 317 | Servers |---------> 318 | | 319 | | 320 +------------------------+ 321 ^ 322 | 323 | 324 | HTTP/ 325 | Websockets 326 | 327 | 328 +----------------------------+ 329 | Javascript/HTML/CSS | 330 +----------------------------+ 331 Other ^ ^RTC 332 APIs | |APIs 333 +---|-----------------|------+ 334 | | | | 335 | +---------+| 336 | | Browser || On-the-wire 337 | Browser | RTC || Protocols 338 | | Function|-----------> 339 | | || 340 | | || 341 | +---------+| 342 +---------------------|------+ 343 | 344 V 345 Native OS Services 347 Figure 1: Browser Model 349 Note that HTTP and Websockets are also offered to the Javascript 350 application through browser APIs. 352 As for all protocol and API specifications, there is no restriction 353 that the protocols can only be used to talk to another browser; since 354 they are fully specified, any device that implements the protocols 355 faithfully should be able to interoperate with the application 356 running in the browser. 358 A commonly imagined model of deployment is the one depicted below. 360 +-----------+ +-----------+ 361 | Web | | Web | 362 | | Signalling | | 363 | |-------------| | 364 | Server | path | Server | 365 | | | | 366 +-----------+ +-----------+ 367 / \ 368 / \ Application-defined 369 / \ over 370 / \ HTTP/Websockets 371 / Application-defined over \ 372 / HTTP/Websockets \ 373 / \ 374 +-----------+ +-----------+ 375 |JS/HTML/CSS| |JS/HTML/CSS| 376 +-----------+ +-----------+ 377 +-----------+ +-----------+ 378 | | | | 379 | | | | 380 | Browser | ------------------------- | Browser | 381 | | Media path | | 382 | | | | 383 +-----------+ +-----------+ 385 Figure 2: Browser RTC Trapezoid 387 On this drawing, the critical part to note is that the media path 388 ("low path") goes directly between the browsers, so it has to be 389 conformant to the specifications of the RTCWEB protocol suite; the 390 signalling path ("high path") goes via servers that can modify, 391 translate or massage the signals as needed. 393 If the two Web servers are operated by different entities, the inter- 394 server signalling mechanism needs to be agreed upon, either by 395 standardization or by other means of agreement. Existing protocols 396 (for example SIP or XMPP) could be used between servers, while either 397 a standards-based or proprietary protocol could be used between the 398 browser and the web server. 400 For example, if both operators' servers implement SIP, SIP could be 401 used for communication between servers, along with either a 402 standardized signaling mechanism (e.g. SIP over Websockets) or a 403 proprietary signaling mechanism used between the application running 404 in the browser and the web server. Similarly, if both operators' 405 servers implement XMPP, XMPP couild be used for communication between 406 XMPP servers, with either a standardized signaling mechanism (e.g. 407 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 408 used between the application running in the browser and the web 409 server. 411 The choice of protocols, and definition of the translation between 412 them, is outside the scope of the RTCWEB standards suite described in 413 the document. 415 The functionality groups that are needed in the browser can be 416 specified, more or less from the bottom up, as: 418 o Data transport: TCP, UDP and the means to securely set up 419 connections between entities, as well as the functions for 420 deciding when to send data: Congestion management, bandwidth 421 estimation and so on. 423 o Data framing: RTP and other data formats that serve as containers, 424 and their functions for data confidentiality and integrity. 426 o Data formats: Codec specifications, format specifications and 427 functionality specifications for the data passed between systems. 428 Audio and video codecs, as well as formats for data and document 429 sharing, belong in this category. In order to make use of data 430 formats, a way to describe them, a session description, is needed. 432 o Connection management: Setting up connections, agreeing on data 433 formats, changing data formats during the duration of a call; SIP 434 and Jingle/XMPP belong in this category. 436 o Presentation and control: What needs to happen in order to ensure 437 that interactions behave in a non-surprising manner. This can 438 include floor control, screen layout, voice activated image 439 switching and other such functions - where part of the system 440 require the cooperation between parties. XCON and Cisco/ 441 Tandberg's TIP were some attempts at specifying this kind of 442 functionality; many applications have been built without 443 standardized interfaces to these functions. 445 o Local system support functions: These are things that need not be 446 specified uniformly, because each participant may choose to do 447 these in a way of the participant's choosing, without affecting 448 the bits on the wire in a way that others have to be cognizant of. 449 Examples in this category include echo cancellation (some forms of 450 it), local authentication and authorization mechanisms, OS access 451 control and the ability to do local recording of conversations. 453 Within each functionality group, it is important to preserve both 454 freedom to innovate and the ability for global communication. 455 Freedom to innovate is helped by doing the specification in terms of 456 interfaces, not implementation; any implementation able to 457 communicate according to the interfaces is a valid implementation. 458 Ability to communicate globally is helped both by having core 459 specifications be unencumbered by IPR issues and by having the 460 formats and protocols be fully enough specified to allow for 461 independent implementation. 463 One can think of the three first groups as forming a "media transport 464 infrastructure", and of the three last groups as forming a "media 465 service". In many contexts, it makes sense to use a common 466 specification for the media transport infrastructure, which can be 467 embedded in browsers and accessed using standard interfaces, and "let 468 a thousand flowers bloom" in the "media service" layer; to achieve 469 interoperable services, however, at least the first five of the six 470 groups need to be specified. 472 4. Data transport 474 Data transport refers to the sending and receiving of data over the 475 network interfaces, the choice of network-layer addresses at each end 476 of the communication, and the interaction with any intermediate 477 entities that handle the data, but do not modify it (such as TURN 478 relays). 480 It includes necessary functions for congestion control: When not to 481 send data. 483 T are described in . 485 ICE is required for all media paths that use UDP; in addition to the 486 ability to pass NAT boxes, ICE fulfils the need for guaranteeing that 487 the media path is going to an UDP port that is willing to receive the 488 data. 490 The data transport protocols used by RTCWEB, as well as the details 491 of interactions with intermediate boxes, such as firewalls, relays 492 and NAT boxes, are intended to be described in a separate document; 493 for now, notes are gathered in Appendix A. 495 5. Data framing and securing 497 The format for media transport is RTP [RFC3550]. Implementation of 498 SRTP [RFC3711] is required for all implementations. 500 The detailed considerations for usage of functions from RTP and SRTP 501 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 502 considerations for the RTCWEB use case are in 503 [I-D.ietf-rtcweb-security], and the resulting security functions are 504 described in [I-D.ietf-rtcweb-security-arch]. 506 Considerations for the transfer of data that is not in RTP format is 507 described in [I-D.ietf-rtcweb-data-channel], and the resulting 508 protocol is described in [I-D.jesup-rtcweb-data-protocol] (not yet a 509 WG document) 511 6. Data formats 513 The intent of this specification is to allow each communications 514 event to use the data formats that are best suited for that 515 particular instance, where a format is supported by both sides of the 516 connection. However, a minimum standard is greatly helpful in order 517 to ensure that communication can be achieved. This document 518 specifies a minimum baseline that will be supported by all 519 implementations of this specification, and leaves further codecs to 520 be included at the will of the implementor. 522 The mandatory to implement codecs, as well as any profiling 523 requirements for both mandatory and optional codecs, is described in 524 (candidate draft: 525 [I-D.cbran-rtcweb-codec]. 527 7. Connection management 529 The methods, mechanisms and requirements for setting up, negotiating 530 and tearing down connections is a large subject, and one where it is 531 desirable to have both interoperability and freedom to innovate. 533 The following principles apply: 535 1. The RTCWEB media negotiations will be capable of representing the 536 same SDP offer/answer semantics that are used in SIP [RFC3264], 537 in such a way that it is possible to build a signalling gateway 538 between SIP and the RTCWEB media negotiation. 540 2. It will be possible to gateway between legacy SIP devices that 541 support ICE and appropriate RTP / SDP mechanisms, codecs and 542 security mechanisms without using a media gateway. A signaling 543 gateway to convert between the signaling on the web side to the 544 SIP signaling may be needed. 546 3. When a new codec is specified, and the SDP for the new codec is 547 specified in the MMUSIC WG, no other standardization would should 548 be required for it to be possible to use that in the web 549 browsers. Adding new codecs which might have new SDP parameters 550 should not change the APIs between the browser and javascript 551 application. As soon as the browsers support the new codecs, old 552 applications written before the codecs were specified should 553 automatically be able to use the new codecs where appropriate 554 with no changes to the JS applications. 556 The particular choices made for RTCWEB, and their implications for 557 the API offered by a browser implementing RTCWEB, are described in 558 [I-D.ietf-rtcweb-jsep] 560 8. Presentation and control 562 The most important part of control is the user's control over the 563 browser's interaction with input/output devices and communications 564 channels. It is important that the user have some way of figuring 565 out where his audio, video or texting is being sent, for what 566 purported reason, and what guarantees are made by the parties that 567 form part of this control channel. This is largely a local function 568 between the browser, the underlying operating system and the user 569 interface; this is being worked on as part of the W3C API effort, and 570 will be part of the peer connection API [W3C.WD-webrtc-20120209], and 571 the device control API [getusermedia]. Considerations for the 572 implications of wanting to identify correspondents are described in 573 [I-D.rescorla-rtcweb-generic-idp] (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 influences 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 to figure out who's controlling the camera, 599 and possibly decide to revoke the permission for camera usage. 601 o Automatic gain control, if present, should normalize a speaking 602 voice into 605 The requirements on RTCWEB systems in this category are found in 606 ; the proposed API for 607 control of local devices are found in [getusermedia]. 609 10. IANA Considerations 611 This document makes no request of IANA. 613 Note to RFC Editor: this section may be removed on publication as an 614 RFC. 616 11. Security Considerations 618 Security of the web-enabled real time communications comes in several 619 pieces: 621 o Security of the components: The browsers, and other servers 622 involved. The most target-rich environment here is probably the 623 browser; the aim here should be that the introduction of these 624 components introduces no additional vulnerability. 626 o Security of the communication channels: It should be easy for a 627 participant to reassure himself of the security of his 628 communication - by verifying the crypto parameters of the links he 629 himself participates in, and to get reassurances from the other 630 parties to the communication that they promise that appropriate 631 measures are taken. 633 o Security of the partners' identity: verifying that the 634 participants are who they say they are (when positive 635 identification is appropriate), or that their identity cannot be 636 uncovered (when anonymity is a goal of the application). 638 The security analysis, and the requirements derived from that 639 analysis, is contained in [I-D.ietf-rtcweb-security]. 641 12. Acknowledgements 643 The number of people who have taken part in the discussions 644 surrounding this draft are too numerous to list, or even to identify. 645 The ones below have made special, identifiable contributions; this 646 does not mean that others' contributions are less important. 648 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 649 Westerlund and Joerg Ott, who offered technical contributions on 650 various versions of the draft. 652 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 653 the ASCII drawings in section 1. 655 Thanks to Eric Rescorla, Justin Uberti, Henry Sinnreich, Colin 656 Perkins and Simon Leinen for document 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-00 666 (work in progress), July 2011. 668 [I-D.ietf-rtcweb-data-channel] 669 Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Datagram 670 Connection", draft-ietf-rtcweb-data-channel-00 (work in 671 progress), March 2012. 673 [I-D.ietf-rtcweb-jsep] 674 Uberti, J. and C. Jennings, "Javascript Session 675 Establishment Protocol", draft-ietf-rtcweb-jsep-00 (work 676 in progress), March 2012. 678 [I-D.ietf-rtcweb-rtp-usage] 679 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 680 Communication (WebRTC): Media Transport and Use of RTP", 681 draft-ietf-rtcweb-rtp-usage-01 (work in progress), 682 October 2011. 684 [I-D.ietf-rtcweb-security] 685 Rescorla, E., "Security Considerations for RTC-Web", 686 draft-ietf-rtcweb-security-01 (work in progress), 687 October 2011. 689 [I-D.ietf-rtcweb-security-arch] 690 Rescorla, E., "RTCWEB Security Architecture", 691 draft-ietf-rtcweb-security-arch-00 (work in progress), 692 January 2012. 694 [I-D.nandakumar-rtcweb-stun-uri] 695 Nandakumar, S., Salgueiro, G., and P. Jones, "URI Scheme 696 for Session Traversal Utilities for NAT (STUN) Protocol", 697 draft-nandakumar-rtcweb-stun-uri-00 (work in progress), 698 October 2011. 700 [I-D.tuexen-tsvwg-sctp-dtls-encaps] 701 Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS 702 Encapsulation of SCTP Packets for RTCWEB", 703 draft-tuexen-tsvwg-sctp-dtls-encaps-00 (work in progress), 704 March 2012. 706 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 707 Requirement Levels", BCP 14, RFC 2119, March 1997. 709 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 710 with Session Description Protocol (SDP)", RFC 3264, 711 June 2002. 713 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 714 Jacobson, "RTP: A Transport Protocol for Real-Time 715 Applications", STD 64, RFC 3550, July 2003. 717 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 718 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 719 RFC 3711, March 2004. 721 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 722 (ICE): A Protocol for Network Address Translator (NAT) 723 Traversal for Offer/Answer Protocols", RFC 5245, 724 April 2010. 726 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 727 Relays around NAT (TURN): Relay Extensions to Session 728 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 730 13.2. Informative References 732 [I-D.cbran-rtcweb-codec] 733 Bran, C. and C. Jennings, "WebRTC Codec and Media 734 Processing Requirements", draft-cbran-rtcweb-codec-01 735 (work in progress), October 2011. 737 [I-D.ietf-rtcweb-use-cases-and-requirements] 738 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 739 Time Communication Use-cases and Requirements", 740 draft-ietf-rtcweb-use-cases-and-requirements-06 (work in 741 progress), October 2011. 743 [I-D.jesup-rtcweb-data-protocol] 744 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 745 Protocol", draft-jesup-rtcweb-data-protocol-00 (work in 746 progress), March 2012. 748 [I-D.rescorla-rtcweb-generic-idp] 749 Rescorla, E., "RTCWEB Generic Identity Provider 750 Interface", draft-rescorla-rtcweb-generic-idp-00 (work in 751 progress), January 2012. 753 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", 754 BCP 95, RFC 3935, October 2004. 756 [W3C.WD-html5-20110525] 757 Hickson, I., "HTML5", World Wide Web Consortium 758 LastCall WD-html5-20110525, May 2011, 759 . 761 [W3C.WD-webrtc-20120209] 762 Bergkvist, A., Burnett, D., Narayanan, A., and C. 763 Jennings, "WebRTC 1.0: Real-time Communication Between 764 Browsers", World Wide Web Consortium WD WD-webrtc- 765 20120209, February 2012, 766 . 768 [getusermedia] 769 Burnett, D. and A. Narayanan, "getusermedia: Getting 770 access to local devices that can generate multimedia 771 streams", December 2011, 772 . 774 Appendix A. Transport and Middlebox specification 776 The draft referred to as "transport and middle boxes" in Section 4 777 has not been written yet. This appendix contains some keywords to 778 what it should say; this also serves the purpose of linking to the 779 drafts-in-progress that are relevant to this specification. 781 A.1. System-provided interfaces 783 The protocol specifications used here assume that the following 784 protocols are available as system-level interfaces: 786 o UDP. This is the protocol assumed by most protocol elements 787 described. 789 o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL 790 and ICE-TCP. 792 For both protocols, we assume the ability to set the DSCP code point 793 of the sockets opened. We do not assume that the DSCP codepoints 794 will be honored, and we do assume that they may be zeroed or changed, 795 since this is a local configuration issue. 797 We do not assume that the implementation will have access to ICMP or 798 raw IP. 800 A.2. Middle box related functions 802 The primary mechanism to deal with middle boxes is ICE, which is an 803 appropriate way to deal with NAT boxes and firewalls that accept 804 traffic from the inside, but only from the outside if it's in 805 response to inside traffic (simple stateful firewalls). 807 In order to deal with symmetric NATs, TURN MUST be supported. 809 In order to deal with firewalls that block all UDP traffic, TURN over 810 TCP MUST be supported. (QUESTION: What about ICE-TCP?) 812 The following specifications MUST be supported: 814 o ICE [RFC5245] 816 o TURN, including TURN over TCP [[QUESTION: and TURN over TLS]], 817 [RFC5766]. 819 For referring to ICE servers, we use the STUN URI, 820 [I-D.nandakumar-rtcweb-stun-uri]. 822 A.3. Transport protocols implemented 824 For data transport, we implement SCTP over DTLS over ICE. This is 825 specified in [I-D.tuexen-tsvwg-sctp-dtls-encaps]. Negotiation of 826 this transport in SCTP is defined in [I-D.ietf-mmusic-sctp-sdp]. 828 Appendix B. Change log 830 This section may be deleted by the RFC Editor when preparing for 831 publication. 833 B.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 835 Added section "On interoperability and innovation" 837 Added data confidentiality and integrity to the "data framing" layer 839 Added congestion management requirements in the "data transport" 840 layer section 842 Changed need for non-media data from "question: do we need this?" to 843 "Open issue: How do we do this?" 845 Strengthened disclaimer that listed codecs are placeholders, not 846 decisions. 848 More details on why the "local system support functions" section is 849 there. 851 B.2. Changes from draft-alvestrand-dispatch-01 to 852 draft-alvestrand-rtcweb-overview-00 854 Added section on "Relationship between API and protocol" 856 Added terminology section 858 Mentioned congestion management as part of the "data transport" layer 859 in the layer list 861 B.3. Changes from draft-alvestrand-rtcweb-00 to -01 863 Removed most technical content, and replaced with pointers to drafts 864 as requested and identified by the RTCWEB WG chairs. 866 Added content to acknowledgements section. 868 Added change log. 870 Spell-checked document. 872 B.4. Changes from draft-alvestrand-rtcweb-overview-01 to 873 draft-ietf-rtcweb-overview-00 875 Changed draft name and document date. 877 Removed unused references 879 B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 881 Added architecture figures to section 2. 883 Changed the description of "echo cancellation" under "local system 884 support functions". 886 Added a few more definitions. 888 B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 890 Added pointers to use cases, security and rtp-usage drafts (now WG 891 drafts). 893 Changed description of SRTP from mandatory-to-use to mandatory-to- 894 implement. 896 Added the "3 principles of negotiation" to the connection management 897 section. 899 Added an explicit statement that ICE is required for both NAT and 900 consent-to-receive. 902 B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 904 Added references to a number of new drafts. 906 Expanded the description text under the "trapezoid" drawing with some 907 more text discussed on the list. 909 Changed the "Connection management" sentence from "will be done using 910 SDP offer/answer" to "will be capable of representing SDP offer/ 911 answer" - this seems more consistent with JSEP. 913 Added "security mechanisms" to the things a non-gatewayed SIP devices 914 must support in order to not need a media gateway. 916 Added a definition for "browser". 918 B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 920 Made introduction more normative. 922 Several wording changes in response to review comments from EKR 924 Added Appendix A to hold references and notes that are not yet in a 925 separate document. 927 Author's Address 929 Harald T. Alvestrand 930 Google 931 Kungsbron 2 932 Stockholm, 11122 933 Sweden 935 Email: harald@alvestrand.no