--- 1/draft-ietf-rtcweb-overview-09.txt 2014-06-17 06:14:24.827924195 -0700 +++ 2/draft-ietf-rtcweb-overview-10.txt 2014-06-17 06:14:24.867925175 -0700 @@ -1,108 +1,108 @@ Network Working Group H. Alvestrand Internet-Draft Google -Intended status: Standards Track February 14, 2014 -Expires: August 18, 2014 +Intended status: Standards Track June 17, 2014 +Expires: December 19, 2014 - Overview: Real Time Protocols for Brower-based Applications - draft-ietf-rtcweb-overview-09 + Overview: Real Time Protocols for Browser-based Applications + draft-ietf-rtcweb-overview-10 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. - The document will be publishd as an Applicability Statement - it does - not itself specify any protocol, but specifies which other - specifications RTCWEB compliant implementations are supposed to - follow. + This document is an Applicability Statement - it does not itself + specify any protocol, but specifies which other specifications RTCWEB + compliant implementations are supposed to follow. This document is a work item of the RTCWEB working group. -Status of this Memo +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 August 18, 2014. + This Internet-Draft will expire on December 19, 2014. 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 carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents - 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 - 2. Principles and Terminology . . . . . . . . . . . . . . . . . . 5 - 2.1. Goals of this document . . . . . . . . . . . . . . . . . . 5 - 2.2. Relationship between API and protocol . . . . . . . . . . 5 - 2.3. On interoperability and innovation . . . . . . . . . . . . 6 - 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 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 - 13.1. Normative References . . . . . . . . . . . . . . . . . . . 16 - 13.2. Informative References . . . . . . . . . . . . . . . . . . 17 - Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 18 - A.1. Changes from - draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 18 - A.2. Changes from draft-alvestrand-dispatch-01 to - draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 19 - A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 19 + 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 + 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . 11 + 5. Data framing and securing . . . . . . . . . . . . . . . . . . 11 + 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . 12 + 7. Connection management . . . . . . . . . . . . . . . . . . . . 12 + 8. Presentation and control . . . . . . . . . . . . . . . . . . 13 + 9. Local system support functions . . . . . . . . . . . . . . . 13 + 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14 + 11. Security Considerations . . . . . . . . . . . . . . . . . . . 14 + 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 15 + 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 15 + 13.1. Normative References . . . . . . . . . . . . . . . . . . 15 + 13.2. Informative References . . . . . . . . . . . . . . . . . 17 + Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 17 + A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 + to -01 . . . . . . . . . . . . . . . . . . . . . . . . . 17 + A.2. Changes from draft-alvestrand-dispatch-01 to draft- + alvestrand-rtcweb-overview-00 . . . . . . . . . . . . . . 18 + A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . 18 A.4. Changes from draft-alvestrand-rtcweb-overview-01 to - draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 19 - A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 19 - A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 19 - 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 . . . . . . . . . . . . . . . . . 20 - A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 21 - A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 21 - Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21 + draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 18 + A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 18 + A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 18 + A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 19 + A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 19 + A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 19 + A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 19 + A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 19 + A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 20 + A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 20 + A.14. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 20 + Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 20 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, special hardware and custom-built software, often at very high prices @@ -137,27 +137,30 @@ 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 [I-D.ietf-rtcweb-use-cases-and-requirements]. - Other efforts, for instance the W3C WebRTC, Web Applications and + 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 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. @@ -166,21 +169,21 @@ 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. 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 RTCWEB/WEBRTC effort consists of two pieces: + The total WebRTC effort consists of two pieces: o A protocol specification, done in the IETF o A Javascript API specification, done in the W3C [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] Together, these two specifications aim to provide an environment where Javascript embedded in any page, viewed in any compatible browser, when suitably authorized by its user, is able to set up communication using audio, video and auxiliary data, where the @@ -193,20 +196,32 @@ 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, two classes of things that can + claim conformance are defined: + + o 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. + + All WebRTC browsers are WebRTC devices, so any requirement on a + WebRTC device also applies to a WebRTC browser. + 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: o Two parties communicate, through some mechanism, what @@ -290,29 +305,27 @@ 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. NOTE: Where common definitions exist for these terms, those definitions should be used to the greatest extent possible. - TODO: Extend this list with other terms that might prove slippery. - 3. Architecture and Functionality groups The model of real-time support for browser-based applications does - not envisage 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 + 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 conjunction with its backend servers, to implement these functions. This means that two vital interfaces need specification: The protocols that browsers talk to each other, without any intervening servers, and the APIs that are offered for a Javascript application to take advantage of the browser's functionality. +------------------------+ On-the-wire | | Protocols | Servers |---------> @@ -387,23 +400,23 @@ 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 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 or XMPP) could be used between servers, while either - a standards-based or proprietary protocol could be used between the - browser and the web server. + (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. For example, if both operators' servers implement SIP, SIP could be used for communication between servers, along with either a 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 @@ -474,54 +487,59 @@ Data transport refers to the sending and receiving of data over the network interfaces, the choice of network-layer addresses at each end of the communication, and the interaction with any intermediate entities that handle the data, but do not modify it (such as TURN relays). It includes necessary functions for congestion control: When not to send data. - The data transport protocols used by RTCWEB are described in + 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. + 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 [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 the resulting - protocol is described in [I-D.jesup-rtcweb-data-protocol] (not yet a - WG document) + described in [I-D.ietf-rtcweb-data-channel], and a supporting + protocol 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], + [I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the + requirements they include. 6. Data formats The intent of this specification is to allow each communications event to use the data formats that are best suited for that 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. - The mandatory to implement codecs, as well as any profiling - requirements for both mandatory and optional codecs, is described in - (candidate draft: - [I-D.cbran-rtcweb-codec]. + 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. 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 @@ -531,47 +549,51 @@ 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 - [I-D.ietf-rtcweb-jsep]. This document in turn implements the - solutions described in [I-D.roach-mmusic-unified-plan]. + [I-D.ietf-rtcweb-jsep]. + + WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. + + NOTE IN DRAFT: Is there any part of -jsep that WebRTC devices need to + be required to implement, and are not also required via other paths? 8. Presentation and control The most important part of control is the user's control over the browser's interaction with input/output devices and communications channels. It is important that the user have some way of figuring out where his audio, video or texting is being sent, for what purported reason, and what guarantees are made by the parties that form part of this control channel. This is largely a local function between the browser, the underlying operating system and the user - interface; this is being worked on as part of the W3C API effort, and - will be part of the peer connection API [W3C.WD-webrtc-20120209], and - the media capture API [W3C.WD-mediacapture-streams-20120628]. - Considerations for the implications of wanting to identify - correspondents are described in [I-D.rescorla-rtcweb-generic-idp] - (not a WG item). + interface; this is specified in the peer connection API + [W3C.WD-webrtc-20120209], and the media capture API + [W3C.WD-mediacapture-streams-20120628]. + + WebRTC browsers MUST implement these two specifications. 9. Local system support functions These are characterized by the fact that the quality of these functions strongly influence the user experience, but the exact algorithm does not need coordination. In some cases (for instance echo cancellation, as described below), the overall system definition may need to specify that the overall system needs to have some characteristics for which these facilities are useful, without requiring them to be implemented a certain way. @@ -580,32 +602,36 @@ management including focus, zoom, pan/tilt controls (if available), and more. Certain parts of the system SHOULD conform to certain properties, for instance: o Echo cancellation should be good enough to achieve the suppression of acoustical feedback loops below a perceptually noticeable level. - o Privacy concerns must be satisfied; for instance, if remote + 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 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 + [I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6, + this means that browsers 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 Security of the web-enabled real time communications comes in several @@ -624,146 +650,137 @@ measures are taken. o Security of the partners' identity: verifying that the participants are who they say they are (when positive identification is appropriate), or that their identity cannot be uncovered (when anonymity is a goal of the application). The security analysis, and the requirements derived from that analysis, is contained in [I-D.ietf-rtcweb-security]. + It is also important to read the security sections of + [W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209]. + 12. Acknowledgements The number of people who have taken part in the discussions surrounding this draft are too numerous to list, or even to identify. The ones below have made special, identifiable contributions; this does not mean that others' contributions are less important. Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus Westerlund and Joerg Ott, who offered technical contributions on various versions of the draft. Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for the ASCII drawings in section 1. - Thanks to Eric Rescorla, Justin Uberti, Henry Sinnreich, Colin - Perkins, Bjoern Hoehrmann and Simon Leinen for document review, and - to Heath Matlock for grammatical review. + 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.ietf-rtcweb-audio] Valin, J. and C. Bran, "WebRTC Audio Codec and Processing - Requirements", draft-ietf-rtcweb-audio-02 (work in - progress), August 2013. + Requirements", draft-ietf-rtcweb-audio-05 (work in + progress), February 2014. [I-D.ietf-rtcweb-data-channel] - Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Data - Channels", draft-ietf-rtcweb-data-channel-05 (work in - progress), July 2013. + Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data + Channels", draft-ietf-rtcweb-data-channel-10 (work in + progress), June 2014. + + [I-D.ietf-rtcweb-data-protocol] + Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel + Establishment Protocol", draft-ietf-rtcweb-data- + protocol-06 (work in progress), June 2014. [I-D.ietf-rtcweb-jsep] Uberti, J. and C. Jennings, "Javascript Session - Establishment Protocol", draft-ietf-rtcweb-jsep-04 (work - in progress), September 2013. + Establishment Protocol", draft-ietf-rtcweb-jsep-06 (work + in progress), February 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-09 (work in progress), - September 2013. + draft-ietf-rtcweb-rtp-usage-15 (work in progress), May + 2014. [I-D.ietf-rtcweb-security] - Rescorla, E., "Security Considerations for WebRTC", - draft-ietf-rtcweb-security-05 (work in progress), - July 2013. + Rescorla, E., "Security Considerations for WebRTC", draft- + ietf-rtcweb-security-06 (work in progress), January 2014. [I-D.ietf-rtcweb-security-arch] - Rescorla, E., "WebRTC Security Architecture", - draft-ietf-rtcweb-security-arch-07 (work in progress), - July 2013. + Rescorla, E., "WebRTC Security Architecture", draft-ietf- + rtcweb-security-arch-09 (work in progress), February 2014. [I-D.ietf-rtcweb-transports] - Alvestrand, H., "Transports for RTCWEB", - draft-ietf-rtcweb-transports-01 (work in progress), - September 2013. - - [I-D.roach-mmusic-unified-plan] - Roach, A., Uberti, J., and M. Thomson, "A Unified Plan for - Using SDP with Large Numbers of Media Flows", - draft-roach-mmusic-unified-plan-00 (work in progress), - July 2013. + Alvestrand, H., "Transports for RTCWEB", draft-ietf- + rtcweb-transports-05 (work in progress), June 2014. [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model - with Session Description Protocol (SDP)", RFC 3264, - June 2002. + with Session Description Protocol (SDP)", RFC 3264, June + 2002. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 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. - -13.2. Informative References - - [I-D.cbran-rtcweb-codec] - Bran, C., Jennings, C., and J. Valin, "WebRTC Codec and - Media Processing Requirements", - draft-cbran-rtcweb-codec-02 (work in progress), - March 2012. - - [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-11 (work in - progress), June 2013. - - [I-D.jesup-rtcweb-data-protocol] - Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel - Protocol", draft-jesup-rtcweb-data-protocol-04 (work in - progress), February 2013. - - [I-D.rescorla-rtcweb-generic-idp] - Rescorla, E., "RTCWEB Generic Identity Provider - Interface", draft-rescorla-rtcweb-generic-idp-01 (work in - progress), March 2012. - - [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", - BCP 95, RFC 3935, October 2004. - - [W3C.WD-html5-20110525] - Hickson, I., "HTML5", World Wide Web Consortium - LastCall WD-html5-20110525, May 2011, - . + 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, . +13.2. Informative References + + [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. + + [RFC3935] Alvestrand, H., "A Mission Statement for the IETF", BCP + 95, RFC 3935, October 2004. + + [RFC6120] Saint-Andre, P., "Extensible Messaging and Presence + Protocol (XMPP): Core", RFC 6120, March 2011. + + [W3C.WD-html5-20110525] + Hickson, I., "HTML5", World Wide Web Consortium LastCall + WD-html5-20110525, May 2011, + . + Appendix A. Change log This section may be deleted by the RFC Editor when preparing for publication. A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 Added section "On interoperability and innovation" Added data confidentiality and integrity to the "data framing" layer @@ -766,49 +783,50 @@ Added section "On interoperability and innovation" Added data confidentiality and integrity to the "data framing" layer Added congestion management requirements in the "data transport" layer section Changed need for non-media data from "question: do we need this?" to "Open issue: How do we do this?" + Strengthened disclaimer that listed codecs are placeholders, not decisions. More details on why the "local system support functions" section is there. -A.2. Changes from draft-alvestrand-dispatch-01 to - draft-alvestrand-rtcweb-overview-00 +A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand- + rtcweb-overview-00 Added section on "Relationship between API and protocol" Added terminology section Mentioned congestion management as part of the "data transport" layer in the layer list A.3. Changes from draft-alvestrand-rtcweb-00 to -01 Removed most technical content, and replaced with pointers to drafts as requested and identified by the RTCWEB WG chairs. Added content to acknowledgments section. Added change log. Spell-checked document. -A.4. Changes from draft-alvestrand-rtcweb-overview-01 to - draft-ietf-rtcweb-overview-00 +A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf- + rtcweb-overview-00 Changed draft name and document date. Removed unused references A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview Added architecture figures to section 2. Changed the description of "echo cancellation" under "local system @@ -859,35 +877,54 @@ Minor grammatical fixes. This is mainly a "keepalive" refresh. A.10. Changes from -05 to -06 Clarifications in response to Last Call review comments. Inserted reference to draft-ietf-rtcweb-audio. A.11. Changes from -06 to -07 - Added a refereence to the "unified plan" draft, and updated some + Added a reference to the "unified plan" draft, and updated some references. Otherwise, it's a "keepalive" draft. A.12. Changes from -07 to -08 Removed the appendix that detailed transports, and replaced it with a reference to draft-ietf-rtcweb-transports. Removed now-unused references. A.13. Changes from -08 to -09 Added text to the Abstract indicating that the intended status is an Applicability Statement. +A.14. Changes from -09 to -10 + + Defined "WebRTC Browser" and "WebRTC device" as things that do, or + don't, conform to the API. + + Updated reference to data-protocol draft + + Updated data formats to reference -rtcweb-audio- and not the expired + -cbran draft. + + Deleted references to -unified-plan + + Deleted reference to -generic-idp (draft expired) + + Added notes on which referenced documents WebRTC browsers or devices + MUST conform to. + + Added pointer to the security section of the API drafts. + Author's Address Harald T. Alvestrand Google Kungsbron 2 - Stockholm, 11122 + Stockholm 11122 Sweden Email: harald@alvestrand.no