--- 1/draft-ietf-rtcweb-overview-07.txt 2013-09-03 06:14:23.732603652 -0700 +++ 2/draft-ietf-rtcweb-overview-08.txt 2013-09-03 06:14:23.776604767 -0700 @@ -1,18 +1,18 @@ Network Working Group H. Alvestrand Internet-Draft Google -Intended status: Standards Track August 14, 2013 -Expires: February 15, 2014 +Intended status: Standards Track September 3, 2013 +Expires: March 7, 2014 Overview: Real Time Protocols for Brower-based Applications - draft-ietf-rtcweb-overview-07 + draft-ietf-rtcweb-overview-08 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 @@ -28,21 +28,21 @@ 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 15, 2014. + This Internet-Draft will expire on March 7, 2014. Copyright Notice Copyright (c) 2013 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 @@ -55,51 +55,48 @@ 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 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . . 11 - 5. Data framing and securing . . . . . . . . . . . . . . . . . . 12 + 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. Transport and Middlebox specification . . . . . . . . 18 - A.1. System-provided interfaces . . . . . . . . . . . . . . . . 18 - A.2. Middle box related functions . . . . . . . . . . . . . . . 18 - A.3. Transport protocols implemented . . . . . . . . . . . . . 19 - Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 19 - B.1. Changes from - draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 19 - B.2. Changes from draft-alvestrand-dispatch-01 to - draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 19 - B.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 20 - B.4. Changes from draft-alvestrand-rtcweb-overview-01 to - draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 20 - B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 20 - B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 20 - B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 20 - B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 21 - B.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 21 - B.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 21 - B.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 21 - Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21 + 13.2. Informative References . . . . . . . . . . . . . . . . . . 16 + 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 . . . . . . . . . . . . . . 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 + 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 @@ -246,23 +243,24 @@ 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]. - ICE Agent: An implementation of the 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). + 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/ observation is on the order of no more than hundreds of milliseconds. Media: Audio and video content. Not to be confused with "transmission media" such as wires. @@ -470,32 +468,22 @@ 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 . - - ICE is required for all media paths that use UDP; in addition to the - ability to pass NAT boxes, ICE fulfills the need for guaranteeing - that the media path is going to a UDP port that is willing to receive - the data. - - The data transport protocols used by RTCWEB, as well as the details - of interactions with intermediate boxes, such as firewalls, relays - and NAT boxes, are intended to be described in a separate document; - for now, notes are gathered in Appendix A. + The data transport protocols used by RTCWEB are 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 [I-D.ietf-rtcweb-security], and the resulting security functions are @@ -652,115 +640,97 @@ 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. 13. References 13.1. Normative References - [I-D.ietf-mmusic-sctp-sdp] - Loreto, S. and G. Camarillo, "Stream Control Transmission - Protocol (SCTP)-Based Media Transport in the Session - Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-03 - (work in progress), January 2013. - [I-D.ietf-rtcweb-audio] Valin, J. and C. Bran, "WebRTC Audio Codec and Processing - Requirements", draft-ietf-rtcweb-audio-01 (work in - progress), November 2012. + Requirements", draft-ietf-rtcweb-audio-02 (work in + progress), August 2013. [I-D.ietf-rtcweb-data-channel] - Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Datagram - Connection", draft-ietf-rtcweb-data-channel-02 (work in - progress), October 2012. + Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Data + Channels", draft-ietf-rtcweb-data-channel-05 (work in + progress), July 2013. [I-D.ietf-rtcweb-jsep] Uberti, J. and C. Jennings, "Javascript Session - Establishment Protocol", draft-ietf-rtcweb-jsep-02 (work - in progress), October 2012. + Establishment Protocol", draft-ietf-rtcweb-jsep-03 (work + in progress), February 2013. [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-05 (work in progress), - October 2012. + draft-ietf-rtcweb-rtp-usage-07 (work in progress), + July 2013. [I-D.ietf-rtcweb-security] - Rescorla, E., "Security Considerations for RTC-Web", - draft-ietf-rtcweb-security-04 (work in progress), - January 2013. + Rescorla, E., "Security Considerations for WebRTC", + draft-ietf-rtcweb-security-05 (work in progress), + July 2013. [I-D.ietf-rtcweb-security-arch] - Rescorla, E., "RTCWEB Security Architecture", - draft-ietf-rtcweb-security-arch-06 (work in progress), - January 2013. - - [I-D.ietf-tsvwg-sctp-dtls-encaps] - Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS - Encapsulation of SCTP Packets for RTCWEB", - draft-ietf-tsvwg-sctp-dtls-encaps-00 (work in progress), - February 2013. + Rescorla, E., "WebRTC Security Architecture", + draft-ietf-rtcweb-security-arch-07 (work in progress), + July 2013. - [I-D.nandakumar-rtcweb-stun-uri] - Nandakumar, S., Salgueiro, G., Jones, P., and M. Petit- - Huguenin, "URI Scheme for Session Traversal Utilities for - NAT (STUN) Protocol", draft-nandakumar-rtcweb-stun-uri-03 - (work in progress), January 2013. + [I-D.ietf-rtcweb-transports] + Alvestrand, H., "Transports for RTCWEB", + draft-ietf-rtcweb-transports-00 (work in progress), + August 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. [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 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", RFC 3550, July 2003. + 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. - [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using - Relays around NAT (TURN): Relay Extensions to Session - Traversal Utilities for NAT (STUN)", RFC 5766, 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-10 (work in - progress), December 2012. + 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-03 (work in - progress), September 2012. + 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] @@ -768,86 +738,32 @@ LastCall WD-html5-20110525, May 2011, . [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., Narayanan, A., and C. - Jennings, "WebRTC 1.0: Real-time Communication Between + 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, . -Appendix A. Transport and Middlebox specification - - The draft referred to as "transport and middle boxes" in Section 4 - has not been written yet. This appendix contains some keywords to - what it should say; this also serves the purpose of linking to the - drafts-in-progress that are relevant to this specification. - -A.1. System-provided interfaces - - The protocol specifications used here assume that the following - protocols are available as system-level interfaces: - - o UDP. This is the protocol assumed by most protocol elements - described. - - o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL - and ICE-TCP. - - For both protocols, this specification assumes the ability to set the - DSCP code point of the sockets opened. It does not assume that the - DSCP codepoints will be honored, and does assume that they may be - zeroed or changed, since this is a local configuration issue. - - This specification does not assume that the implementation will have - access to ICMP or raw IP. - -A.2. Middle box related functions - - The primary mechanism to deal with middle boxes is ICE, which is an - appropriate way to deal with NAT boxes and firewalls that accept - traffic from the inside, but only from the outside if it's in - response to inside traffic (simple stateful firewalls). - - In order to deal with symmetric NATs, TURN MUST be supported. - - In order to deal with firewalls that block all UDP traffic, TURN over - TCP MUST be supported. (QUESTION: What about ICE-TCP?) - The following specifications MUST be supported: - - o ICE [RFC5245] - - o TURN, including TURN over TCP [[QUESTION: and TURN over TLS]], - [RFC5766]. - - For referring to STUN and TURN servers, this specification depends on - the STUN URI, [I-D.nandakumar-rtcweb-stun-uri]. - -A.3. Transport protocols implemented - - For data transport, RTCWEB implementations support SCTP over DTLS - over ICE. This is specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. - Negotiation of this transport in SCTP is defined in - [I-D.ietf-mmusic-sctp-sdp]. - -Appendix B. Change log +Appendix A. Change log This section may be deleted by the RFC Editor when preparing for publication. -B.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 +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 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?" @@ -844,117 +760,123 @@ 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. -B.2. Changes from draft-alvestrand-dispatch-01 to +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 -B.3. Changes from draft-alvestrand-rtcweb-00 to -01 +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. -B.4. Changes from draft-alvestrand-rtcweb-overview-01 to +A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf-rtcweb-overview-00 Changed draft name and document date. Removed unused references -B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview +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 support functions". Added a few more definitions. -B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview +A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview Added pointers to use cases, security and rtp-usage drafts (now WG drafts). Changed description of SRTP from mandatory-to-use to mandatory-to- implement. Added the "3 principles of negotiation" to the connection management section. Added an explicit statement that ICE is required for both NAT and consent-to-receive. -B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview +A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview Added references to a number of new drafts. Expanded the description text under the "trapezoid" drawing with some more text discussed on the list. Changed the "Connection management" sentence from "will be done using SDP offer/answer" to "will be capable of representing SDP offer/ answer" - this seems more consistent with JSEP. Added "security mechanisms" to the things a non-gatewayed SIP devices must support in order to not need a media gateway. Added a definition for "browser". -B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview +A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview Made introduction more normative. Several wording changes in response to review comments from EKR - Added Appendix A to hold references and notes that are not yet in a + Added an appendix to hold references and notes that are not yet in a separate document. -B.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview +A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview Minor grammatical fixes. This is mainly a "keepalive" refresh. -B.10. Changes from -05 to -06 +A.10. Changes from -05 to -06 Clarifications in response to Last Call review comments. Inserted reference to draft-ietf-rtcweb-audio. -B.11. Changes from -06 to -07 +A.11. Changes from -06 to -07 Added a refereence 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. + Author's Address Harald T. Alvestrand Google Kungsbron 2 Stockholm, 11122 Sweden Email: harald@alvestrand.no