1. 1 Carol Davids © 2010 WebRTC Standards Summary

2. 2 What is WebRTC? WebRTC refers to protocols as well as Javascript APIs used to enable realtime communications within Web browsers, without requiring plugins. IETF RTCWEB WG is standardizing the protocols used in WebRTC. W3C WEBRTC WG is standardizing the Javascript APIs implemented in Web browsers. –getUserMedia API provides access to media streams from webcams and microphones. –RTCPeerConnection API enables mediastreams to be sent peer-to-peer.

3. 3 Source: http://samdutton.com/webrtc.pdf (Google Chrome Developer Relations)http://samdutton.com/webrtc.pdf WebRTC Protocols and APIs (P2P Scenario)

4. 4 What Protocol Functionality is Covered by IETF RTCWEB? Overview: draft-ietf-rtcweb-overview Codecs (Opus & G.711 MTI for audio) –draft-ietf-rtcweb-audio Security (SRTP and DTLS/SRTP key management) –draft-ietf-rtcweb-security, draft-ietf-rtcweb-security-arch NAT traversal (STUN/TURN/ICE) –RFC 5245 (ICE), 5389 (STUN), 5766 (TURN) –draft-muthu-behave-consent-freshness Data Channel (SCTP over DTLS over UDP) –draft-ietf-rtcweb-data-channel, draft-ietf-rtcweb-data-protocol RTP/RTCP (RTP/SAVPF profile) –draft-ietf-rtcweb-rtp-usage Congestion Control (Circuit Breakers, RMCAT WG) –draft-ietf-avtcore-rtp-circuit-breakers

5. 5 IETF Standards Status IETF RTCWEB WG documents likely to complete WG last call by end of CY 2013. –Major remaining issue is MTI video codec selection (VP8 vs. H.264). –Basic interoperability (audio, video, security) demonstrated between Chrome and Mozilla. –Core functionality implemented and being used in production. Focus now largely on enhancements and optimizations. –Screen sharing: security issues. –Security: DTLS/SRTP-EKT (optimization for conferencing scenarios) –NAT traversal: Trickle ICE –A/V Multiplexing: Signaling (BUNDLE), and RTP multiplexing. –Multi-stream support: Support for simulcast and layered codecs, RTCP reporting. –Congestion control (RMCAT).

6. 6 W3C Javascript APIs WebRTC Core APIs –getUserMedia: http://dev.w3.org/2011/webrtc/editor/getusermedia.html http://dev.w3.org/2011/webrtc/editor/getusermedia.html –RTCPeerConnection: http://dev.w3.org/2011/webrtc/editor/webrtc.html http://dev.w3.org/2011/webrtc/editor/webrtc.html –IETF API draft: draft-ietf-rtcweb-jsep Ancillary APIs (not required for every WebRTC application) –Canvas: http://www.w3.org/TR/2012/CR-2dcontext- 20121217/http://www.w3.org/TR/2012/CR-2dcontext- 20121217/ –Websockets: http://www.w3.org/TR/websockets/http://www.w3.org/TR/websockets/ –WebGL: http://www.khronos.org/registry/webgl/specs/latest/ http://www.khronos.org/registry/webgl/specs/latest/ –Web Telephony: http://www.w3.org/2012/sysapps/telephony/ http://www.w3.org/2012/sysapps/telephony/

7. 7 Implementation Status WebRTC Core APIs –getUserMedia: Chrome 23 (December 2012), Mozilla (March 2013), going to WG last call in 4Q2013. –RTCPeerConnection: Chrome 25 (February 2013), Mozilla (September 2013). Ancillary APIs (not required for every WebRTC application) –Canvas: Supported in all major browsers –Websockets: Supported in all major browsers –WebGL: Supported in all major browsers (in IE 11) –Web Telephony: for non-browser use (e.g. Firefox OS and Chrome OS).