--- 1/draft-ietf-rtcweb-overview-11.txt 2014-10-13 03:14:47.480663260 -0700 +++ 2/draft-ietf-rtcweb-overview-12.txt 2014-10-13 03:14:47.528664437 -0700 @@ -1,52 +1,52 @@ Network Working Group H. Alvestrand Internet-Draft Google -Intended status: Standards Track August 18, 2014 -Expires: February 19, 2015 +Intended status: Standards Track October 13, 2014 +Expires: April 16, 2015 Overview: Real Time Protocols for Browser-based Applications - draft-ietf-rtcweb-overview-11 + draft-ietf-rtcweb-overview-12 Abstract This document gives an overview and context of a protocol suite intended for use with real-time applications that can be deployed in browsers - "real time communication on the Web". It intends to serve as a starting and coordination point to make sure all the parts that are needed to achieve this goal are findable, and that the parts that belong in the Internet protocol suite are fully specified and on the right publication track. This document is an Applicability Statement - it does not itself - specify any protocol, but specifies which other specifications RTCWEB + specify any protocol, but specifies which other specifications WebRTC compliant implementations are supposed to follow. This document is a work item of the RTCWEB working group. Status of This Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on February 19, 2015. + This Internet-Draft will expire on April 16, 2015. Copyright Notice Copyright (c) 2014 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents @@ -57,21 +57,21 @@ described in the Simplified BSD License. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Principles and Terminology . . . . . . . . . . . . . . . . . 4 2.1. Goals of this document . . . . . . . . . . . . . . . . . 4 2.2. Relationship between API and protocol . . . . . . . . . . 4 2.3. On interoperability and innovation . . . . . . . . . . . 5 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 - 3. Architecture and Functionality groups . . . . . . . . . . . . 7 + 3. Architecture and Functionality groups . . . . . . . . . . . . 8 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . 12 5. Data framing and securing . . . . . . . . . . . . . . . . . . 12 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . 13 7. Connection management . . . . . . . . . . . . . . . . . . . . 13 8. Presentation and control . . . . . . . . . . . . . . . . . . 14 9. Local system support functions . . . . . . . . . . . . . . . 14 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 11. Security Considerations . . . . . . . . . . . . . . . . . . . 15 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 16 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 16 @@ -89,20 +89,21 @@ A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 20 A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 20 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 20 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 20 A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 20 A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 21 A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 21 A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 21 A.14. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 21 A.15. Changes from -10 to -11 . . . . . . . . . . . . . . . . . 21 + A.16. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 22 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 22 1. Introduction The Internet was, from very early in its lifetime, considered a possible vehicle for the deployment of real-time, interactive applications - with the most easily imaginable being audio conversations (aka "Internet telephony") and video conferencing. The first attempts to build this were dependent on special networks, @@ -135,48 +136,48 @@ the development of HTML5, application developers see much promise in the possibility of making those interfaces available in a standardized way within the browser. This memo describes a set of building blocks that can be made accessible and controllable through a Javascript API in a browser, and which together form a sufficient set of functions to allow the use of interactive audio and video in applications that communicate directly between browsers across the Internet. The resulting protocol suite is intended to enable all the applications that are - described as required scenarios in the RTCWEB use cases document + described as required scenarios in the use cases document [I-D.ietf-rtcweb-use-cases-and-requirements]. Other efforts, for instance the W3C WEBRTC, Web Applications and Device API working groups, focus on making standardized APIs and interfaces available, within or alongside the HTML5 effort, for those functions; this memo concentrates on specifying the protocols and subprotocols that are needed to specify the interactions that happen across the network. This memo uses the term "WebRTC" (note the case used) to refer to the overall effort consisting of both IETF and W3C efforts. 2. Principles and Terminology 2.1. Goals of this document - The goal of the RTCWEB protocol specification is to specify a set of + The goal of the WebRTC protocol specification is to specify a set of protocols that, if all are implemented, will allow an implementation to communicate with another implementation using audio, video and data sent along the most direct possible path between the participants. - This document is intended to serve as the roadmap to the RTCWEB + This document is intended to serve as the roadmap to the WebRTC specifications. It defines terms used by other pieces of specification, lists references to other specifications that don't - need further elaboration in the RTCWEB context, and gives pointers to - other documents that form part of the RTCWEB suite. + need further elaboration in the WebRTC context, and gives pointers to + other documents that form part of the WebRTC suite. By reading this document and the documents it refers to, it should be possible to have all information needed to implement an RTCWEB compatible implementation. 2.2. Relationship between API and protocol The total WebRTC effort consists of two pieces: o A protocol specification, done in the IETF @@ -197,38 +198,45 @@ is a browser or another device implementing this specification. The goal of cooperation between the protocol specification and the API specification is that for all options and features of the protocol specification, it should be clear which API calls to make to exercise that option or feature; similarly, for any sequence of API calls, it should be clear which protocol options and features will be invoked. Both subject to constraints of the implementation, of course. - For the purpose of this document, three classes of things that can - claim conformance are defined: + For the purpose of this document, five types of entities are defined: - o A WebRTC browser is something that conforms to both the protocol - specification and the Javascript API defined above. + o A WebRTC User Agent (also called a WebRTC UA or a WebRTC browser) + is something that conforms to both the protocol specification and + the Javascript API defined above. o A WebRTC device is something that conforms to the protocol specification, but does not claim to implement the Javascript API. - o A WebRTC gateway is something that mediates media traffic to non- - WebRTC entities. It is like a device, but has certain - restrictiions on where it can operate, which means that some of - the requirements can be relaxed. + o A WebRTC endpoint is either a WebRTC User Agent or a WebRTC + device. - All WebRTC browsers are WebRTC devices, so any requirement on a + o A WebRTC-compatible endpoint is an endpoint that is capable of + successfully communicating with a WebRTC endpoint, but may fail to + meet some requirements of a WebRTC endpoint. This may limit where + in the network such an endpoint can be attached, or may limit the + security guarantees that it offers to others. + + o A WebRTC gateway is a WebRTC-compatible endpoint that mediates + traffic to non-WebRTC entities. + + All WebRTC browsers (UAs) are WebRTC devices, so any requirement on a WebRTC device also applies to a WebRTC browser. - WebRTC gateways are described in a separate document, + WebRTC gateways are described in a separate document [I-D.alvestrand-rtcweb-gateways]. 2.3. On interoperability and innovation The "Mission statement of the IETF" [RFC3935] states that "The benefit of a standard to the Internet is in interoperability - that multiple products implementing a standard are able to work together in order to deliver valuable functions to the Internet's users." Communication on the Internet frequently occurs in two phases: @@ -256,36 +264,37 @@ The alternative - that of having no mandatory to implement - does not mean that you cannot communicate, it merely means that in order to be part of the communications partnership, you have to implement the standard "and then some" - that "and then some" usually being called a profile of some sort; in the version most antithetical to the Internet ethos, that "and then some" consists of having to use a specific vendor's product only. 2.4. Terminology - The following terms are used in this document, and as far as possible - across the documents specifying the RTCWEB suite, in the specific - meanings given here. Not all terms are used in this document. Other - terms are used in their commonly used meaning. + The following terms are used across the documents specifying the + WebRTC suite, in the specific meanings given here. Not all terms are + used in this document. Other terms are used in their commonly used + meaning. The list is in alphabetical order. Agent: Undefined term. See "SDP Agent" and "ICE Agent". API: Application Programming Interface - a specification of a set of calls and events, usually tied to a programming language or an abstract formal specification such as WebIDL, with its defined semantics. Browser: Used synonymously with "Interactive User Agent" as defined - in the HTML specification [W3C.WD-html5-20110525]. + in the HTML specification [W3C.WD-html5-20110525]. See also + "WebRTC User Agent". ICE Agent: An implementation of the Interactive Connectivty Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be an SDP Agent, but there exist ICE Agents that do not use SDP (for instance those that use Jingle). Interactive: Communication between multiple parties, where the expectation is that an action from one party can cause a reaction by another party, and the reaction can be observed by the first party, with the total time required for the action/reaction/ @@ -311,27 +320,30 @@ SDP Agent: The protocol implementation involved in the SDP offer/ answer exchange, as defined in [RFC3264] section 3. Signaling: Communication that happens in order to establish, manage and control media paths. Signaling Path: The communication channels used between entities participating in signaling to transfer signaling. There may be more entities in the signaling path than in the media path. - WebRTC Browser: Browser that conforms to the WebRTC protocol - specifications and offer the WebRTC Javascript APIs. + WebRTC User Agent: An entity that conforms to the WebRTC protocol + specifications and offer the WebRTC Javascript APIs. Also called + a WebRTC browser. WebRTC Device: An unit (software, hardware or combinations) that conforms to the WebRTC protocol specifications, but does not offer the WebRTC Javascript APIs. + WebRTC Endpoint: Either a WebRTC browser or a WebRTC device. + NOTE: Where common definitions exist for these terms, those definitions should be used to the greatest extent possible. 3. Architecture and Functionality groups The model of real-time support for browser-based applications does not assume that the browser will contain all the functions that need to be performed in order to have a function such as a telephone or a video conferencing unit; the vision is that the browser will have the functions that are needed for a Web application, working in @@ -409,21 +421,21 @@ | | | | | Browser | ------------------------- | Browser | | | Media path | | | | | | +-----------+ +-----------+ Figure 2: Browser RTC Trapezoid On this drawing, the critical part to note is that the media path ("low path") goes directly between the browsers, so it has to be - conformant to the specifications of the RTCWEB protocol suite; the + conformant to the specifications of the WebRTC protocol suite; the signaling path ("high path") goes via servers that can modify, translate or massage the signals as needed. If the two Web servers are operated by different entities, the inter- server signaling mechanism needs to be agreed upon, either by standardization or by other means of agreement. Existing protocols (for example SIP [RFC3261] or XMPP [RFC6120]) could be used between servers, while either a standards-based or proprietary protocol could be used between the browser and the web server. @@ -432,21 +444,21 @@ standardized signaling mechanism (e.g. SIP over Websockets) or a proprietary signaling mechanism used between the application running in the browser and the web server. Similarly, if both operators' servers implement XMPP, XMPP could be used for communication between XMPP servers, with either a standardized signaling mechanism (e.g. XMPP over Websockets or BOSH) or a proprietary signaling mechanism used between the application running in the browser and the web server. The choice of protocols, and definition of the translation between - them, is outside the scope of the RTCWEB standards suite described in + them, is outside the scope of the WebRTC protocol suite described in the document. The functionality groups that are needed in the browser can be specified, more or less from the bottom up, as: o Data transport: TCP, UDP and the means to securely set up connections between entities, as well as the functions for deciding when to send data: Congestion management, bandwidth estimation and so on. @@ -513,21 +525,21 @@ WebRTC devices MUST implement the transport protocols described in [I-D.ietf-rtcweb-transports]. 5. Data framing and securing The format for media transport is RTP [RFC3550]. Implementation of SRTP [RFC3711] is REQUIRED for all implementations. The detailed considerations for usage of functions from RTP and SRTP are given in [I-D.ietf-rtcweb-rtp-usage]. The security - considerations for the RTCWEB use case are in + considerations for the WebRTC use case are in [I-D.ietf-rtcweb-security], and the resulting security functions are described in [I-D.ietf-rtcweb-security-arch]. Considerations for the transfer of data that is not in RTP format is described in [I-D.ietf-rtcweb-data-channel], and a supporting protocol for establishing individual data channels is described in [I-D.ietf-rtcweb-data-protocol]. Webrtc devices MUST implement these two specifications. WebRTC devices MUST implement [I-D.ietf-rtcweb-rtp-usage], @@ -541,54 +553,54 @@ particular instance, where a format is supported by both sides of the connection. However, a minimum standard is greatly helpful in order to ensure that communication can be achieved. This document specifies a minimum baseline that will be supported by all implementations of this specification, and leaves further codecs to be included at the will of the implementor. WebRTC devices MUST implement the codecs and profiles required in [I-D.ietf-rtcweb-audio] - NOTE IN DRAFT: At this time (June 2014) there is no consensus on what - to say about video codecs in this section. + NOTE IN DRAFT: At this time (October 2014) there is no consensus on + what to say about video codecs in this section. 7. Connection management The methods, mechanisms and requirements for setting up, negotiating and tearing down connections is a large subject, and one where it is desirable to have both interoperability and freedom to innovate. The following principles apply: - 1. The RTCWEB media negotiations will be capable of representing the + 1. The WebRTC media negotiations will be capable of representing the same SDP offer/answer semantics that are used in SIP [RFC3264], in such a way that it is possible to build a signaling gateway - between SIP and the RTCWEB media negotiation. + between SIP and the WebRTC media negotiation. 2. It will be possible to gateway between legacy SIP devices that support ICE and appropriate RTP / SDP mechanisms, codecs and security mechanisms without using a media gateway. A signaling gateway to convert between the signaling on the web side to the SIP signaling may be needed. 3. When a new codec is specified, and the SDP for the new codec is specified in the MMUSIC WG, no other standardization should be required for it to be possible to use that in the web browsers. Adding new codecs which might have new SDP parameters should not change the APIs between the browser and Javascript application. As soon as the browsers support the new codecs, old applications written before the codecs were specified should automatically be able to use the new codecs where appropriate with no changes to the JS applications. - The particular choices made for RTCWEB, and their implications for - the API offered by a browser implementing RTCWEB, are described in + The particular choices made for WebRTC, and their implications for + the API offered by a WebRTC endpoint, are described in [I-D.ietf-rtcweb-jsep]. WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. WebRTC devices MUST implement the functions described in that document that relate to the network layer (for example Bundle, RTCP- mux and Trickle ICE), but do not need to support the API functionality described there. 8. Presentation and control @@ -628,27 +640,27 @@ level. o Privacy concerns MUST be satisfied; for instance, if remote control of camera is offered, the APIs should be available to let the local participant figure out who's controlling the camera, and possibly decide to revoke the permission for camera usage. o Automatic gain control, if present, should normalize a speaking voice into a reasonable dB range. - The requirements on RTCWEB systems with regard to audio processing + The requirements on WebRTC devices with regard to audio processing are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of local devices are found in [W3C.WD-mediacapture-streams-20120628]. - WebRTC browsers MUST implement the processing functions in + WebRTC devices MUST implement the processing functions in [I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6, - this means that browsers MUST implement the whole document.) + this means that WebRTC devices MUST implement the whole document.) 10. IANA Considerations This document makes no request of IANA. Note to RFC Editor: this section may be removed on publication as an RFC. 11. Security Considerations @@ -693,44 +705,48 @@ the ASCII drawings in section 1. Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage and Simon Leinen for document review. 13. References 13.1. Normative References + [I-D.alvestrand-rtcweb-gateways] + Alvestrand, H., "WebRTC Gateways", draft-alvestrand- + rtcweb-gateways-00 (work in progress), August 2014. + [I-D.ietf-rtcweb-audio] Valin, J. and C. Bran, "WebRTC Audio Codec and Processing - Requirements", draft-ietf-rtcweb-audio-05 (work in - progress), February 2014. + Requirements", draft-ietf-rtcweb-audio-06 (work in + progress), September 2014. [I-D.ietf-rtcweb-data-channel] Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data - Channels", draft-ietf-rtcweb-data-channel-11 (work in - progress), July 2014. + Channels", draft-ietf-rtcweb-data-channel-12 (work in + progress), September 2014. [I-D.ietf-rtcweb-data-protocol] Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel Establishment Protocol", draft-ietf-rtcweb-data- - protocol-07 (work in progress), July 2014. + protocol-08 (work in progress), September 2014. [I-D.ietf-rtcweb-jsep] Uberti, J., Jennings, C., and E. Rescorla, "Javascript Session Establishment Protocol", draft-ietf-rtcweb-jsep-07 (work in progress), July 2014. [I-D.ietf-rtcweb-rtp-usage] Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Communication (WebRTC): Media Transport and Use of RTP", - draft-ietf-rtcweb-rtp-usage-16 (work in progress), July + draft-ietf-rtcweb-rtp-usage-17 (work in progress), August 2014. [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for WebRTC", draft- ietf-rtcweb-security-07 (work in progress), July 2014. [I-D.ietf-rtcweb-security-arch] Rescorla, E., "WebRTC Security Architecture", draft-ietf- rtcweb-security-arch-10 (work in progress), July 2014. @@ -750,37 +766,33 @@ Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004. [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", RFC 5245, April 2010. [W3C.WD-mediacapture-streams-20120628] Burnett, D. and A. Narayanan, "Media Capture and Streams", - World Wide Web Consortium WD WD-mediacapture-streams- - 20120628, June 2012, . [W3C.WD-webrtc-20120209] Bergkvist, A., Burnett, D., Jennings, C., and A. Narayanan, "WebRTC 1.0: Real-time Communication Between - Browsers", World Wide Web Consortium WD WD-webrtc- - 20120209, February 2012, + Browsers", World Wide Web Consortium WD WD- + webrtc-20120209, February 2012, . 13.2. Informative References - [I-D.alvestrand-rtcweb-gateways] - Alvestrand, H., "WebRTC Gateways", draft-alvestrand- - rtcweb-gateways-00 (work in progress), August 2014. - [I-D.ietf-rtcweb-use-cases-and-requirements] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- Time Communication Use-cases and Requirements", draft- ietf-rtcweb-use-cases-and-requirements-14 (work in progress), February 2014. [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002. @@ -938,21 +950,30 @@ Added notes on which referenced documents WebRTC browsers or devices MUST conform to. Added pointer to the security section of the API drafts. A.15. Changes from -10 to -11 Added "WebRTC Gateway" as a third class of device, and referenced the doc describing them. - Made a number of text clarifications in response to document reviews. + Made a number of text clarifications, in response to document + reviews. + +A.16. Changes from -11 to -12 + + Refined entity definitions to define "WebRTC endpoint" and "WebRTC- + compatible endpoint". + + Changed remaining usage of the term "RTCWEB" to "WebRTC", including + in the page header. Author's Address Harald T. Alvestrand Google Kungsbron 2 Stockholm 11122 Sweden Email: harald@alvestrand.no