--- 1/draft-ietf-rtcweb-overview-14.txt 2016-01-21 12:15:22.256872040 -0800 +++ 2/draft-ietf-rtcweb-overview-15.txt 2016-01-21 12:15:22.304873261 -0800 @@ -1,18 +1,18 @@ Network Working Group H. Alvestrand Internet-Draft Google -Intended status: Standards Track June 16, 2015 -Expires: December 18, 2015 +Intended status: Standards Track January 21, 2016 +Expires: July 24, 2016 Overview: Real Time Protocols for Browser-based Applications - draft-ietf-rtcweb-overview-14 + draft-ietf-rtcweb-overview-15 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 @@ -32,25 +32,25 @@ 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 December 18, 2015. + This Internet-Draft will expire on July 24, 2016. Copyright Notice - Copyright (c) 2015 IETF Trust and the persons identified as the + Copyright (c) 2016 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 @@ -92,20 +92,21 @@ 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 A.17. Changes from -12 to -13 . . . . . . . . . . . . . . . . . 22 A.18. Changes from -13 to -14 . . . . . . . . . . . . . . . . . 22 + A.19. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 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, @@ -240,21 +241,21 @@ A WebRTC non-browser may be capable of hosting applications in a similar way to the way in which a browser can host Javascript applications, typically by offering APIs in other languages. For instance it may be implemented as a library that offers a C++ API intended to be loaded into applications. In this case, similar security considerations as for Javascript may be needed; however, since such APIs are not defined or referenced here, this document cannot give any specific rules for those interfaces. WebRTC gateways are described in a separate document, - [I-D.alvestrand-rtcweb-gateways]. + [I-D.ietf-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: @@ -712,103 +713,103 @@ 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-08 (work in - progress), April 2015. + Requirements", draft-ietf-rtcweb-audio-05 (work in + progress), February 2014. [I-D.ietf-rtcweb-data-channel] Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data - Channels", draft-ietf-rtcweb-data-channel-13 (work in - progress), January 2015. + Channels", draft-ietf-rtcweb-data-channel-11 (work in + progress), July 2014. [I-D.ietf-rtcweb-data-protocol] Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel Establishment Protocol", draft-ietf-rtcweb-data- - protocol-09 (work in progress), January 2015. + protocol-07 (work in progress), July 2014. [I-D.ietf-rtcweb-jsep] Uberti, J., Jennings, C., and E. Rescorla, "Javascript - Session Establishment Protocol", draft-ietf-rtcweb-jsep-10 - (work in progress), June 2015. + 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-22 (work in progress), - February 2015. + draft-ietf-rtcweb-rtp-usage-16 (work in progress), July + 2014. [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for WebRTC", draft- - ietf-rtcweb-security-08 (work in progress), February 2015. + 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-11 (work in progress), March 2015. + rtcweb-security-arch-10 (work in progress), July 2014. [I-D.ietf-rtcweb-transports] Alvestrand, H., "Transports for WebRTC", draft-ietf- - rtcweb-transports-08 (work in progress), February 2015. + rtcweb-transports-06 (work in progress), August 2014. [I-D.ietf-rtcweb-video] Roach, A., "WebRTC Video Processing and Codec - Requirements", draft-ietf-rtcweb-video-06 (work in - progress), June 2015. + Requirements", draft-ietf-rtcweb-video-00 (work in + progress), July 2014. [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", 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. [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] + [I-D.ietf-rtcweb-gateways] Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways", - draft-alvestrand-rtcweb-gateways-02 (work in progress), - March 2015. + draft-ietf-rtcweb-gateways-01 (work in progress), July + 2015. [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-16 (work in - progress), January 2015. + 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 @@ -986,19 +987,23 @@ document. Added words about WebRTC APIs in languages other than Javascript. Referenced draft-ietf-rtcweb-video for video codecs to support. A.18. Changes from -13 to -14 None. This is a "keepalive" update. +A.19. Changes from -14 to -15 + + Changed "gateways" reference to point to the WG document. + Author's Address Harald T. Alvestrand Google Kungsbron 2 Stockholm 11122 Sweden Email: harald@alvestrand.no