WebRTC standards update (Jul 2014)
2 likes3,946 views
WebRTC standards update (July 2014) The document discusses updates to WebRTC standards in July 2014, including discussions around signaling, video codecs, browser support, and interworking with legacy VoIP/IMS deployments. It notes that each WebRTC deployment implements proprietary signaling, WebRTC signaling and media are not compatible with existing VoIP without gateways, and the WebRTC API could evolve in the future.
WebRTC standards update (Jul 2014)
- 1. WebRTC standards update (July 2014) Victor Pascual Avila Victor.pascual@quobis.com @victorpascual Amir Zmora amzmora@gmail.com @AmirZmora
- 2. Upperside WebRTC Conference, Dec 16-18 7/15/14 2
- 4. New WebRTC Blog 7/15/14 4 blog.uppersideconferences.com
- 5. Technology, Innovation & Strategy Consultant Main focus: help make WebRTC happen – involved in WebRTC standardization, development and first industry deployments (on-going RFX's, PoC's and field trials) Other activities: - Chief Strategy Officer (CSO) - IETF contributor (SIP, Diameter and WebRTC areas) - IETF STRAW WG co-chair - SIP Forum WebRTC Task Group co-chair - WebRTCHacks.com co-founder and blogger - Independent Expert at European Commission About Me
- 8. OeAng WebRTC is…..? • A media engine in the browser with JavaScript APIs • Open Source, no royalKes, license fees • Real-‐Kme voice, video & data • No plug-‐in • Secured • Standard interfaces
- 9. 周小逸 Ian
- 11. - Audio codecs – G.711, Opus - Video codecs – H.264 vs. VP8 - Media codecs are negotiated with SDP (for now at least) - Requires Secure RTP (SRTP) – DTLS-SRTP (SDES is prohibited) - Requires Peer-2-peer NAT traversal tools (STUN, TURN, ICE) – trickle ICE - Multiplexing: RTPs & RTP+RTCP - Tools for firewall traversal - DataChannel - Etc. NEW PROTOCOL PROFILE FOR MEDIA RTCWeb WG (and other)
- 12. RTCWeb WG
- 13. Don’t panic, it’s not a bad thing! WebRTC Doesn’t Define Signaling
- 14. Some discussion on the topic: http://webrtchacks.com/signalling-options-for-webrtc- applications/ Signaling Plane • WebRTC has no defined signaling method. JavaScript app downloaded from web server. Popular choices are: • SIP over Websockets - Standard mechanism (RFC7118) - Extend SIP directly into the browser by embedding a SIP stack directly into the webpage – typically based on JavaScript - WebSocket create a full-duplex channel right from the web browser - Popular examples are jsSIP, sip-js, QoffeeSIP, or sipML5 • Call Control API - proprietary signaling scheme based on more traditional web tools and techniques - “standard” APIs enhanced to include WebRTC support • Other alternatives based on XMPP, JSON or foobar
- 15. each deployment/vendor is implementing its own proprietary signaling mechanism (1/3)
- 16. Interworking Towards Legacy? • A browser-embedded media engine • Best-of-breed echo canceler • Video jitter buffer, image enhancer • Audio codecs – G.711, Opus are MTI • Video codecs – H.264 vs. VP8 (MTI TBD - IPR discussion) • Media codecs are negotiated with SDP (for now at least) • Requires Secure RTP (SRTP) – DTLS • Requires Peer-2-peer NAT traversal tools (STUN, TURN, ICE) – trickle ICE • Multiplexing: RTPs & RTP+RTCP • Yes, your favorite SIP client implementation is compatible with most of this. But, the vast majority of deployments • Use plain RTP (and SDES if encrypted at all) • Do not support STUN/TURN/ICE • Do not support multiplexing (ok, not really an issue) • Use different codecs that might not be supported on the WebRTC side
- 17. WebRTC signaling and media is NOT compatible with existing VoIP/IMS deployments – gateways are required to bridge the two worlds (2/3)
- 18. Some discussion on the topic: http://webrtchacks.com/cisco-openh264/ The Video Codec Battle
- 20. Poll Question
- 21. Room participants: 30/50 in favor of H.264 Remote participants (minority): 75/25 in favor of VP8 → No clear consensus No decisionSome discussion on the topic: http://webrtchacks.com/ietf-finally-made-decision- mandatory-implement-mti-video-codec-webrtc/ Result of The Discussion?
- 22. “The mission of the W3C WebRTC WG is to define client-side APIs to enable Real-Time Communications in Web-browsers. These APIs should enable building applications that can be run inside a browser, requiring no extra downloads or plugins, that allow communication between parties using audio, video and supplementary real-time communication, without having to use intervening servers (unless needed for firewall traversal).” Discussion: provides the current API in its form (e.g. based on SDP O/A) the flexibility Web developers need? Answer: well, not really but it's good enough for most of the use cases we have today Alternative proposals: Microsoft's CU- RTC-WEB (Aug'12), WebRTC Object API (ORTC) (Aug'13) Next step: “Done is better than perfect”, Let's finish WebRTC 1.0, Let the industry adopt it Future work: “fix/improve things in WebRTC 2.0”, Backward interoperability? Obtain local media Setup Peer Connection Attach media or Data Close Connection ← getUserMedia(), etc. ← RTCPeerConnection(), etc. ← addStream(), createOffer(), etc. WebRTC WG
- 24. Poll Question
- 25. Some discussion on the topic: http:// webrtchacks.com/why-the-webrtc-api-has-it- wrong-interview-with-webrtc-object-api-ortc-co- author-inaki-baz-3-2/ Browser API
- 28. Browser API
- 29. Plug-‐in free or free plug-‐in? 7/15/14 29
- 30. the WebRTC API can have different flavors (3/3)
- 32. Adding New Wheels to IMS with WebRTC
- 33. 3GPP TS 23.228 V12.5.0 (2014-06)
- 35. codec1 SRTP IP IP UDP IP UDP UDP UDP IP UE eIMS-AGW peer SRTP RTP codec1 codec2 RTP codec2 BFCP SCTP DTLS IP SCTP DTLS IP TCP IP UDP UDP BFCP TCP IP UE eIMS-AGW peer MSRP SCTP DTLS IP MSRP SCTP DTLS IP MSRP TCP IP UDP UDP MSRP TCP IP UE eIMS-AGW peer Interworking Towards Legacy IMS
- 36. “the initial focus of the Task Group is to determine what the needs are for successful interoperability of WebRTC-to-SIP deployments” covering both Enterprises and Service Providers “recommendations, Reference Architecture Documents, Certifications, and/or White Papers” SIP Forum WebRTC Task Group
- 37. Alliance for Telecom Solutions
- 38. “focuses on interoperability issues relating to the use of WebRTC” “the group is focused on enterprise WebRTC , interworking of WebRTC and other carrier technologies, and other existing videoconferencing systems” “develop an interoperability test framework and prepare for IOT events” WebRTC Interop Activity Group
- 39. How does WebRTC relate to VoLTE and RCS? GSMA
- 40. l each deployment/vendor is implementing its own proprietary signaling mechanism l WebRTC signaling and media is incompatible with existing VoIP deployments – gateways are required to bridge the two worlds l the WebRTC API can have different flavors Summary
- 41. Thank You! Victor Pascual Avila Victor.pascual@quobis.com @victorpascual Amir Zmora amzmora@gmail.com @AmirZmora