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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(See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Outdated reference: A later version (-26) exists of draft-ietf-mmusic-sctp-sdp-01 == Outdated reference: A later version (-13) exists of draft-ietf-rtcweb-data-channel-00 == Outdated reference: A later version (-26) exists of draft-ietf-rtcweb-jsep-01 == Outdated reference: A later version (-26) exists of draft-ietf-rtcweb-rtp-usage-04 == Outdated reference: A later version (-12) exists of draft-ietf-rtcweb-security-03 == Outdated reference: A later version (-20) exists of draft-ietf-rtcweb-security-arch-03 == Outdated reference: A later version (-08) exists of draft-nandakumar-rtcweb-stun-uri-01 ** Obsolete normative reference: RFC 5245 (Obsoleted by RFC 8445, RFC 8839) ** Obsolete normative reference: RFC 5766 (Obsoleted by RFC 8656) == Outdated reference: A later version (-16) exists of draft-ietf-rtcweb-use-cases-and-requirements-09 == Outdated reference: A later version (-04) exists of draft-jesup-rtcweb-data-protocol-02 Summary: 2 errors (**), 0 flaws (~~), 10 warnings (==), 1 comment (--).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 December 14, 2012 5 Expires: June 17, 2013 7 Overview: Real Time Protocols for Brower-based Applications 8 draft-ietf-rtcweb-overview-05 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 June 17, 2013. 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 B.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 22 99 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 22 101 1. Introduction 103 The Internet was, from very early in its lifetime, considered a 104 possible vehicle for the deployment of real-time, interactive 105 applications - with the most easily imaginable being audio 106 conversations (aka "Internet telephony") and video conferencing. 108 The first attempts to build this were dependent on special networks, 109 special hardware and custom-built software, often at very high prices 110 or at low quality, placing great demands on the infrastructure. 112 As the available bandwidth has increased, and as processors and other 113 hardware has become ever faster, the barriers to participation have 114 decreased, and it has become possible to deliver a satisfactory 115 experience on commonly available computing hardware. 117 Still, there are a number of barriers to the ability to communicate 118 universally - one of these is that there is, as of yet, no single set 119 of communication protocols that all agree should be made available 120 for communication; another is the sheer lack of universal 121 identification systems (such as is served by telephone numbers or 122 email addresses in other communications systems). 124 Development of The Universal Solution has proved hard, however, for 125 all the usual reasons. 127 The last few years have also seen a new platform rise for deployment 128 of services: The browser-embedded application, or "Web application". 129 It turns out that as long as the browser platform has the necessary 130 interfaces, it is possible to deliver almost any kind of service on 131 it. 133 Traditionally, these interfaces have been delivered by plugins, which 134 had to be downloaded and installed separately from the browser; in 135 the development of HTML5, application developers see much promise in 136 the possibility of making those interfaces available in a 137 standardized way within the browser. 139 This memo describes a set of building blocks that can be made 140 accessible and controllable through a Javascript API in a browser, 141 and which together form a sufficient set of functions to allow the 142 use of interactive audio and video in applications that communicate 143 directly between browsers across the Internet. The resulting 144 protocol suite is intended to enable all the applications that are 145 described as required scenarios in the RTCWEB use cases document 146 [I-D.ietf-rtcweb-use-cases-and-requirements]. 148 Other efforts, for instance the W3C WebRTC, Web Applications and 149 Device API working groups, focus on making standardized APIs and 150 interfaces available, within or alongside the HTML5 effort, for those 151 functions; this memo concentrates on specifying the protocols and 152 subprotocols that are needed to specify the interactions that happen 153 across the network. 155 2. Principles and Terminology 157 2.1. Goals of this document 159 The goal of the RTCWEB protocol specification is to specify a set of 160 protocols that, if all are implemented, will allow an implementation 161 to communicate with another implementation using audio, video and 162 data sent along the most direct possible path between the 163 participants. 165 This document is intended to serve as the roadmap to the RTCWEB 166 specifications. It defines terms used by other pieces of 167 specification, lists references to other specifications that don't 168 need further elaboration in the RTCWEB context, and gives pointers to 169 other documents that form part of the RTCWEB suite. 171 By reading this document and the documents it refers to, it should be 172 possible to have all information needed to implement an RTCWEB 173 compatible implementation. 175 2.2. Relationship between API and protocol 177 The total RTCWEB/WEBRTC effort consists of two pieces: 179 o A protocol specification, done in the IETF 181 o A Javascript API specification, done in the W3C 182 [W3C.WD-webrtc-20120209] 184 Together, these two specifications aim to provide an environment 185 where Javascript embedded in any page, viewed in any compatible 186 browser, when suitably authorized by its user, is able to set up 187 communication using audio, video and auxiliary data, where the 188 browser environment does not constrain the types of application in 189 which this functionality can be used. 191 The protocol specification does not assume that all implementations 192 implement this API; it is not intended to be necessary for 193 interoperation to know whether the entity one is communicating with 194 is a browser or another device implementing this specification. 196 The goal of cooperation between the protocol specification and the 197 API specification is that for all options and features of the 198 protocol specification, it should be clear which API calls to make to 199 exercise that option or feature; similarly, for any sequence of API 200 calls, it should be clear which protocol options and features will be 201 invoked. Both subject to constraints of the implementation, of 202 course. 204 2.3. On interoperability and innovation 206 The "Mission statement of the IETF" [RFC3935] states that "The 207 benefit of a standard to the Internet is in interoperability - that 208 multiple products implementing a standard are able to work together 209 in order to deliver valuable functions to the Internet's users." 211 Communication on the Internet frequently occurs in two phases: 213 o Two parties communicate, through some mechanism, what 214 functionality they both are able to support 216 o They use that shared communicative functionality to communicate, 217 or, failing to find anything in common, give up on communication. 219 There are often many choices that can be made for communicative 220 functionality; the history of the Internet is rife with the proposal, 221 standardization, implementation, and success or failure of many types 222 of options, in all sorts of protocols. 224 The goal of having a mandatory to implement function set is to 225 prevent negotiation failure, not to preempt or prevent negotiation. 227 The presence of a mandatory to implement function set serves as a 228 strong changer of the marketplace of deployment - in that it gives a 229 guarantee that, as long as you conform to a specification, and the 230 other party is willing to accept communication at the base level of 231 that specification, you can communicate successfully. 233 The alternative - that of having no mandatory to implement - does not 234 mean that you cannot communicate, it merely means that in order to be 235 part of the communications partnership, you have to implement the 236 standard "and then some" - that "and then some" usually being called 237 a profile of some sort; in the version most antithetical to the 238 Internet ethos, that "and then some" consists of having to use a 239 specific vendor's product only. 241 2.4. Terminology 243 The following terms are used in this document, and as far as possible 244 across the documents specifying the RTCWEB suite, in the specific 245 meanings given here. Not all terms are used in this document. Other 246 terms are used in their commonly used meaning. 248 The list is in alphabetical order. 250 Agent: Undefined term. See "SDP Agent" and "ICE Agent". 252 API: Application Programming Interface - a specification of a set of 253 calls and events, usually tied to a programming language or an 254 abstract formal specification such as WebIDL, with its defined 255 semantics. 257 Browser: Used synonymously with "Interactive User Agent" as defined 258 in the HTML specification [W3C.WD-html5-20110525]. 260 ICE Agent: An implementation of the ICE [RFC5245] protocol. An ICE 261 Agent may also be an SDP Agent, but there exist ICE Agents that do 262 not use SDP (for instance those that use Jingle). 264 Interactive: Communication between multiple parties, where the 265 expectation is that an action from one party can cause a reaction 266 by another party, and the reaction can be observed by the first 267 party, with the total time required for the action/reaction/ 268 observation is on the order of no more than hundreds of 269 milliseconds. 271 Media: Audio and video content. Not to be confused with 272 "transmission media" such as wires. 274 Media path: The path that media data follows from one browser to 275 another. 277 Protocol: A specification of a set of data units, their 278 representation, and rules for their transmission, with their 279 defined semantics. A protocol is usually thought of as going 280 between systems. 282 Real-time media: Media where generation of content and display of 283 content are intended to occur closely together in time (on the 284 order of no more than hundreds of milliseconds). Real-time media 285 can be used to support interactive communication. 287 SDP Agent: The protocol implementation involved in the SDP offer/ 288 answer exchange, as defined in [RFC3264] section 3. 290 Signaling: Communication that happens in order to establish, manage 291 and control media paths. 293 Signaling Path: The communication channels used between entities 294 participating in signaling to transfer signaling. There may be 295 more entities in the signaling path than in the media path. 297 NOTE: Where common definitions exist for these terms, those 298 definitions should be used to the greatest extent possible. 300 TODO: Extend this list with other terms that might prove slippery. 302 3. Architecture and Functionality groups 304 The model of real-time support for browser-based applications does 305 not envisage that the browser will contain all the functions that 306 need to be performed in order to have a function such as a telephone 307 or a video conferencing unit; the vision is that the browser will 308 have the functions that are needed for a Web application, working in 309 conjunction with its backend servers, to implement these functions. 311 This means that two vital interfaces need specification: The 312 protocols that browsers talk to each other, without any intervening 313 servers, and the APIs that are offered for a Javascript application 314 to take advantage of the browser's functionality. 316 +------------------------+ On-the-wire 317 | | Protocols 318 | Servers |---------> 319 | | 320 | | 321 +------------------------+ 322 ^ 323 | 324 | 325 | HTTP/ 326 | Websockets 327 | 328 | 329 +----------------------------+ 330 | Javascript/HTML/CSS | 331 +----------------------------+ 332 Other ^ ^RTC 333 APIs | |APIs 334 +---|-----------------|------+ 335 | | | | 336 | +---------+| 337 | | Browser || On-the-wire 338 | Browser | RTC || Protocols 339 | | Function|-----------> 340 | | || 341 | | || 342 | +---------+| 343 +---------------------|------+ 344 | 345 V 346 Native OS Services 348 Figure 1: Browser Model 350 Note that HTTP and Websockets are also offered to the Javascript 351 application through browser APIs. 353 As for all protocol and API specifications, there is no restriction 354 that the protocols can only be used to talk to another browser; since 355 they are fully specified, any device that implements the protocols 356 faithfully should be able to interoperate with the application 357 running in the browser. 359 A commonly imagined model of deployment is the one depicted below. 361 +-----------+ +-----------+ 362 | Web | | Web | 363 | | Signaling | | 364 | |-------------| | 365 | Server | path | Server | 366 | | | | 367 +-----------+ +-----------+ 368 / \ 369 / \ Application-defined 370 / \ over 371 / \ HTTP/Websockets 372 / Application-defined over \ 373 / HTTP/Websockets \ 374 / \ 375 +-----------+ +-----------+ 376 |JS/HTML/CSS| |JS/HTML/CSS| 377 +-----------+ +-----------+ 378 +-----------+ +-----------+ 379 | | | | 380 | | | | 381 | Browser | ------------------------- | Browser | 382 | | Media path | | 383 | | | | 384 +-----------+ +-----------+ 386 Figure 2: Browser RTC Trapezoid 388 On this drawing, the critical part to note is that the media path 389 ("low path") goes directly between the browsers, so it has to be 390 conformant to the specifications of the RTCWEB protocol suite; the 391 signaling path ("high path") goes via servers that can modify, 392 translate or massage the signals as needed. 394 If the two Web servers are operated by different entities, the inter- 395 server signaling mechanism needs to be agreed upon, either by 396 standardization or by other means of agreement. Existing protocols 397 (for example SIP or XMPP) could be used between servers, while either 398 a standards-based or proprietary protocol could be used between the 399 browser and the web server. 401 For example, if both operators' servers implement SIP, SIP could be 402 used for communication between servers, along with either a 403 standardized signaling mechanism (e.g. SIP over Websockets) or a 404 proprietary signaling mechanism used between the application running 405 in the browser and the web server. Similarly, if both operators' 406 servers implement XMPP, XMPP could be used for communication between 407 XMPP servers, with either a standardized signaling mechanism (e.g. 408 XMPP over Websockets or BOSH) or a proprietary signaling mechanism 409 used between the application running in the browser and the web 410 server. 412 The choice of protocols, and definition of the translation between 413 them, is outside the scope of the RTCWEB standards suite described in 414 the document. 416 The functionality groups that are needed in the browser can be 417 specified, more or less from the bottom up, as: 419 o Data transport: TCP, UDP and the means to securely set up 420 connections between entities, as well as the functions for 421 deciding when to send data: Congestion management, bandwidth 422 estimation and so on. 424 o Data framing: RTP and other data formats that serve as containers, 425 and their functions for data confidentiality and integrity. 427 o Data formats: Codec specifications, format specifications and 428 functionality specifications for the data passed between systems. 429 Audio and video codecs, as well as formats for data and document 430 sharing, belong in this category. In order to make use of data 431 formats, a way to describe them, a session description, is needed. 433 o Connection management: Setting up connections, agreeing on data 434 formats, changing data formats during the duration of a call; SIP 435 and Jingle/XMPP belong in this category. 437 o Presentation and control: What needs to happen in order to ensure 438 that interactions behave in a non-surprising manner. This can 439 include floor control, screen layout, voice activated image 440 switching and other such functions - where part of the system 441 require the cooperation between parties. XCON and Cisco/ 442 Tandberg's TIP were some attempts at specifying this kind of 443 functionality; many applications have been built without 444 standardized interfaces to these functions. 446 o Local system support functions: These are things that need not be 447 specified uniformly, because each participant may choose to do 448 these in a way of the participant's choosing, without affecting 449 the bits on the wire in a way that others have to be cognizant of. 450 Examples in this category include echo cancellation (some forms of 451 it), local authentication and authorization mechanisms, OS access 452 control and the ability to do local recording of conversations. 454 Within each functionality group, it is important to preserve both 455 freedom to innovate and the ability for global communication. 456 Freedom to innovate is helped by doing the specification in terms of 457 interfaces, not implementation; any implementation able to 458 communicate according to the interfaces is a valid implementation. 459 Ability to communicate globally is helped both by having core 460 specifications be unencumbered by IPR issues and by having the 461 formats and protocols be fully enough specified to allow for 462 independent implementation. 464 One can think of the three first groups as forming a "media transport 465 infrastructure", and of the three last groups as forming a "media 466 service". In many contexts, it makes sense to use a common 467 specification for the media transport infrastructure, which can be 468 embedded in browsers and accessed using standard interfaces, and "let 469 a thousand flowers bloom" in the "media service" layer; to achieve 470 interoperable services, however, at least the first five of the six 471 groups need to be specified. 473 4. Data transport 475 Data transport refers to the sending and receiving of data over the 476 network interfaces, the choice of network-layer addresses at each end 477 of the communication, and the interaction with any intermediate 478 entities that handle the data, but do not modify it (such as TURN 479 relays). 481 It includes necessary functions for congestion control: When not to 482 send data. 484 T are described in . 486 ICE is required for all media paths that use UDP; in addition to the 487 ability to pass NAT boxes, ICE fulfills the need for guaranteeing 488 that the media path is going to a UDP port that is willing to receive 489 the data. 491 The data transport protocols used by RTCWEB, as well as the details 492 of interactions with intermediate boxes, such as firewalls, relays 493 and NAT boxes, are intended to be described in a separate document; 494 for now, notes are gathered in Appendix A. 496 5. Data framing and securing 498 The format for media transport is RTP [RFC3550]. Implementation of 499 SRTP [RFC3711] is required for all implementations. 501 The detailed considerations for usage of functions from RTP and SRTP 502 are given in [I-D.ietf-rtcweb-rtp-usage]. The security 503 considerations for the RTCWEB use case are in 504 [I-D.ietf-rtcweb-security], and the resulting security functions are 505 described in [I-D.ietf-rtcweb-security-arch]. 507 Considerations for the transfer of data that is not in RTP format is 508 described in [I-D.ietf-rtcweb-data-channel], and the resulting 509 protocol is described in [I-D.jesup-rtcweb-data-protocol] (not yet a 510 WG document) 512 6. Data formats 514 The intent of this specification is to allow each communications 515 event to use the data formats that are best suited for that 516 particular instance, where a format is supported by both sides of the 517 connection. However, a minimum standard is greatly helpful in order 518 to ensure that communication can be achieved. This document 519 specifies a minimum baseline that will be supported by all 520 implementations of this specification, and leaves further codecs to 521 be included at the will of the implementor. 523 The mandatory to implement codecs, as well as any profiling 524 requirements for both mandatory and optional codecs, is described in 525 (candidate draft: 526 [I-D.cbran-rtcweb-codec]. 528 7. Connection management 530 The methods, mechanisms and requirements for setting up, negotiating 531 and tearing down connections is a large subject, and one where it is 532 desirable to have both interoperability and freedom to innovate. 534 The following principles apply: 536 1. The RTCWEB media negotiations will be capable of representing the 537 same SDP offer/answer semantics that are used in SIP [RFC3264], 538 in such a way that it is possible to build a signaling gateway 539 between SIP and the RTCWEB media negotiation. 541 2. It will be possible to gateway between legacy SIP devices that 542 support ICE and appropriate RTP / SDP mechanisms, codecs and 543 security mechanisms without using a media gateway. A signaling 544 gateway to convert between the signaling on the web side to the 545 SIP signaling may be needed. 547 3. When a new codec is specified, and the SDP for the new codec is 548 specified in the MMUSIC WG, no other standardization should be 549 required for it to be possible to use that in the web browsers. 550 Adding new codecs which might have new SDP parameters should not 551 change the APIs between the browser and Javascript application. 552 As soon as the browsers support the new codecs, old applications 553 written before the codecs were specified should automatically be 554 able to use the new codecs where appropriate with no changes to 555 the JS applications. 557 The particular choices made for RTCWEB, and their implications for 558 the API offered by a browser implementing RTCWEB, are described in 559 [I-D.ietf-rtcweb-jsep] 561 8. Presentation and control 563 The most important part of control is the user's control over the 564 browser's interaction with input/output devices and communications 565 channels. It is important that the user have some way of figuring 566 out where his audio, video or texting is being sent, for what 567 purported reason, and what guarantees are made by the parties that 568 form part of this control channel. This is largely a local function 569 between the browser, the underlying operating system and the user 570 interface; this is being worked on as part of the W3C API effort, and 571 will be part of the peer connection API [W3C.WD-webrtc-20120209], and 572 the device control API [getusermedia]. Considerations for the 573 implications of wanting to identify correspondents are described in 574 [I-D.rescorla-rtcweb-generic-idp] (not a WG item). 576 9. Local system support functions 578 These are characterized by the fact that the quality of these 579 functions strongly influence the user experience, but the exact 580 algorithm does not need coordination. In some cases (for instance 581 echo cancellation, as described below), the overall system definition 582 may need to specify that the overall system needs to have some 583 characteristics for which these facilities are useful, without 584 requiring them to be implemented a certain way. 586 Local functions include echo cancellation, volume control, camera 587 management including focus, zoom, pan/tilt controls (if available), 588 and more. 590 Certain parts of the system SHOULD conform to certain properties, for 591 instance: 593 o Echo cancellation should be good enough to achieve the suppression 594 of acoustical feedback loops below a perceptually noticeable 595 level. 597 o Privacy concerns must be satisfied; for instance, if remote 598 control of camera is offered, the APIs should be available to let 599 the local participant figure out who's controlling the camera, and 600 possibly decide to revoke the permission for camera usage. 602 o Automatic gain control, if present, should normalize a speaking 603 voice into 606 The requirements on RTCWEB systems in this category are found in 607 ; the proposed API for 608 control of local devices are found in [getusermedia]. 610 10. IANA Considerations 612 This document makes no request of IANA. 614 Note to RFC Editor: this section may be removed on publication as an 615 RFC. 617 11. Security Considerations 619 Security of the web-enabled real time communications comes in several 620 pieces: 622 o Security of the components: The browsers, and other servers 623 involved. The most target-rich environment here is probably the 624 browser; the aim here should be that the introduction of these 625 components introduces no additional vulnerability. 627 o Security of the communication channels: It should be easy for a 628 participant to reassure himself of the security of his 629 communication - by verifying the crypto parameters of the links he 630 himself participates in, and to get reassurances from the other 631 parties to the communication that they promise that appropriate 632 measures are taken. 634 o Security of the partners' identity: verifying that the 635 participants are who they say they are (when positive 636 identification is appropriate), or that their identity cannot be 637 uncovered (when anonymity is a goal of the application). 639 The security analysis, and the requirements derived from that 640 analysis, is contained in [I-D.ietf-rtcweb-security]. 642 12. Acknowledgements 644 The number of people who have taken part in the discussions 645 surrounding this draft are too numerous to list, or even to identify. 646 The ones below have made special, identifiable contributions; this 647 does not mean that others' contributions are less important. 649 Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus 650 Westerlund and Joerg Ott, who offered technical contributions on 651 various versions of the draft. 653 Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for 654 the ASCII drawings in section 1. 656 Thanks to Eric Rescorla, Justin Uberti, Henry Sinnreich, Colin 657 Perkins and Simon Leinen for document review, ad to Heath Matlock for 658 grammatical review. 660 13. References 662 13.1. Normative References 664 [I-D.ietf-mmusic-sctp-sdp] 665 Loreto, S. and G. Camarillo, "Stream Control Transmission 666 Protocol (SCTP)-Based Media Transport in the Session 667 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-01 668 (work in progress), March 2012. 670 [I-D.ietf-rtcweb-data-channel] 671 Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Datagram 672 Connection", draft-ietf-rtcweb-data-channel-00 (work in 673 progress), March 2012. 675 [I-D.ietf-rtcweb-jsep] 676 Uberti, J. and C. Jennings, "Javascript Session 677 Establishment Protocol", draft-ietf-rtcweb-jsep-01 (work 678 in progress), June 2012. 680 [I-D.ietf-rtcweb-rtp-usage] 681 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 682 Communication (WebRTC): Media Transport and Use of RTP", 683 draft-ietf-rtcweb-rtp-usage-04 (work in progress), 684 July 2012. 686 [I-D.ietf-rtcweb-security] 687 Rescorla, E., "Security Considerations for RTC-Web", 688 draft-ietf-rtcweb-security-03 (work in progress), 689 June 2012. 691 [I-D.ietf-rtcweb-security-arch] 692 Rescorla, E., "RTCWEB Security Architecture", 693 draft-ietf-rtcweb-security-arch-03 (work in progress), 694 July 2012. 696 [I-D.nandakumar-rtcweb-stun-uri] 697 Nandakumar, S., Salgueiro, G., Jones, P., and M. Petit- 698 Huguenin, "URI Scheme for Session Traversal Utilities for 699 NAT (STUN) Protocol", draft-nandakumar-rtcweb-stun-uri-01 700 (work in progress), March 2012. 702 [I-D.tuexen-tsvwg-sctp-dtls-encaps] 703 Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS 704 Encapsulation of SCTP Packets for RTCWEB", 705 draft-tuexen-tsvwg-sctp-dtls-encaps-01 (work in progress), 706 July 2012. 708 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 709 Requirement Levels", BCP 14, RFC 2119, March 1997. 711 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 712 with Session Description Protocol (SDP)", RFC 3264, 713 June 2002. 715 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 716 Jacobson, "RTP: A Transport Protocol for Real-Time 717 Applications", STD 64, RFC 3550, July 2003. 719 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 720 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 721 RFC 3711, March 2004. 723 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 724 (ICE): A Protocol for Network Address Translator (NAT) 725 Traversal for Offer/Answer Protocols", RFC 5245, 726 April 2010. 728 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 729 Relays around NAT (TURN): Relay Extensions to Session 730 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 732 13.2. Informative References 734 [I-D.cbran-rtcweb-codec] 735 Bran, C., Jennings, C., and J. Valin, "WebRTC Codec and 736 Media Processing Requirements", 737 draft-cbran-rtcweb-codec-02 (work in progress), 738 March 2012. 740 [I-D.ietf-rtcweb-use-cases-and-requirements] 741 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 742 Time Communication Use-cases and Requirements", 743 draft-ietf-rtcweb-use-cases-and-requirements-09 (work in 744 progress), June 2012. 746 [I-D.jesup-rtcweb-data-protocol] 747 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel 748 Protocol", draft-jesup-rtcweb-data-protocol-02 (work in 749 progress), July 2012. 751 [I-D.rescorla-rtcweb-generic-idp] 752 Rescorla, E., "RTCWEB Generic Identity Provider 753 Interface", draft-rescorla-rtcweb-generic-idp-01 (work in 754 progress), March 2012. 756 [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", 757 BCP 95, RFC 3935, October 2004. 759 [W3C.WD-html5-20110525] 760 Hickson, I., "HTML5", World Wide Web Consortium 761 LastCall WD-html5-20110525, May 2011, 762 . 764 [W3C.WD-webrtc-20120209] 765 Bergkvist, A., Burnett, D., Narayanan, A., and C. 766 Jennings, "WebRTC 1.0: Real-time Communication Between 767 Browsers", World Wide Web Consortium WD WD-webrtc- 768 20120209, February 2012, 769 . 771 [getusermedia] 772 Burnett, D. and A. Narayanan, "getusermedia: Getting 773 access to local devices that can generate multimedia 774 streams", December 2011, 775 . 777 Appendix A. Transport and Middlebox specification 779 The draft referred to as "transport and middle boxes" in Section 4 780 has not been written yet. This appendix contains some keywords to 781 what it should say; this also serves the purpose of linking to the 782 drafts-in-progress that are relevant to this specification. 784 A.1. System-provided interfaces 786 The protocol specifications used here assume that the following 787 protocols are available as system-level interfaces: 789 o UDP. This is the protocol assumed by most protocol elements 790 described. 792 o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL 793 and ICE-TCP. 795 For both protocols, we assume the ability to set the DSCP code point 796 of the sockets opened. We do not assume that the DSCP codepoints 797 will be honored, and we do assume that they may be zeroed or changed, 798 since this is a local configuration issue. 800 We do not assume that the implementation will have access to ICMP or 801 raw IP. 803 A.2. Middle box related functions 805 The primary mechanism to deal with middle boxes is ICE, which is an 806 appropriate way to deal with NAT boxes and firewalls that accept 807 traffic from the inside, but only from the outside if it's in 808 response to inside traffic (simple stateful firewalls). 810 In order to deal with symmetric NATs, TURN MUST be supported. 812 In order to deal with firewalls that block all UDP traffic, TURN over 813 TCP MUST be supported. (QUESTION: What about ICE-TCP?) 815 The following specifications MUST be supported: 817 o ICE [RFC5245] 819 o TURN, including TURN over TCP [[QUESTION: and TURN over TLS]], 820 [RFC5766]. 822 For referring to ICE servers, we use the STUN URI, 823 [I-D.nandakumar-rtcweb-stun-uri]. 825 A.3. Transport protocols implemented 827 For data transport, we implement SCTP over DTLS over ICE. This is 828 specified in [I-D.tuexen-tsvwg-sctp-dtls-encaps]. Negotiation of 829 this transport in SCTP is defined in [I-D.ietf-mmusic-sctp-sdp]. 831 Appendix B. Change log 833 This section may be deleted by the RFC Editor when preparing for 834 publication. 836 B.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 838 Added section "On interoperability and innovation" 840 Added data confidentiality and integrity to the "data framing" layer 842 Added congestion management requirements in the "data transport" 843 layer section 845 Changed need for non-media data from "question: do we need this?" to 846 "Open issue: How do we do this?" 848 Strengthened disclaimer that listed codecs are placeholders, not 849 decisions. 851 More details on why the "local system support functions" section is 852 there. 854 B.2. Changes from draft-alvestrand-dispatch-01 to 855 draft-alvestrand-rtcweb-overview-00 857 Added section on "Relationship between API and protocol" 859 Added terminology section 861 Mentioned congestion management as part of the "data transport" layer 862 in the layer list 864 B.3. Changes from draft-alvestrand-rtcweb-00 to -01 866 Removed most technical content, and replaced with pointers to drafts 867 as requested and identified by the RTCWEB WG chairs. 869 Added content to acknowledgments section. 871 Added change log. 873 Spell-checked document. 875 B.4. Changes from draft-alvestrand-rtcweb-overview-01 to 876 draft-ietf-rtcweb-overview-00 878 Changed draft name and document date. 880 Removed unused references 882 B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview 884 Added architecture figures to section 2. 886 Changed the description of "echo cancellation" under "local system 887 support functions". 889 Added a few more definitions. 891 B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview 893 Added pointers to use cases, security and rtp-usage drafts (now WG 894 drafts). 896 Changed description of SRTP from mandatory-to-use to mandatory-to- 897 implement. 899 Added the "3 principles of negotiation" to the connection management 900 section. 902 Added an explicit statement that ICE is required for both NAT and 903 consent-to-receive. 905 B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview 907 Added references to a number of new drafts. 909 Expanded the description text under the "trapezoid" drawing with some 910 more text discussed on the list. 912 Changed the "Connection management" sentence from "will be done using 913 SDP offer/answer" to "will be capable of representing SDP offer/ 914 answer" - this seems more consistent with JSEP. 916 Added "security mechanisms" to the things a non-gatewayed SIP devices 917 must support in order to not need a media gateway. 919 Added a definition for "browser". 921 B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview 923 Made introduction more normative. 925 Several wording changes in response to review comments from EKR 927 Added Appendix A to hold references and notes that are not yet in a 928 separate document. 930 B.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview 932 Minor grammatical fixes. This is mainly a "keepalive" refresh. 934 Author's Address 936 Harald T. Alvestrand 937 Google 938 Kungsbron 2 939 Stockholm, 11122 940 Sweden 942 Email: harald@alvestrand.no