webrtc standards update (april 2014)
TRANSCRIPT
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WebRTC standards update (April 2014)
Victor Pascual Ávila – Technology, Innovation & Strategy Consultant
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About Me
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
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@victorpascual
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What is WebRTC?
A browser-embedded media engine
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WebRTC standards
(Media)
(Signaling)
(Signaling)
“Set or RTC APIs for Web Browsers”
“New protocol profile”
Some discussion on the topic:http://webrtchacks.com/a-hitchhikers-guide-to-webrtc-standardization/
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RTCWeb WG (and others)
- 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
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WebRTC does not define signaling
Don’t panic, it’s not a bad thing!
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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
Some discussion on the topic: http://webrtchacks.com/signalling-options-for-webrtc-applications/
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each deployment/vendor is implementing its own proprietary signaling mechanism
(1/3)
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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
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WebRTC signaling and media is NOT compatible with existing VoIP/IMS
deployments – gateways are required to bridge the two worlds
(2/3)
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The video codec battle
Some discussion on the topic: http://webrtchacks.com/cisco-openh264/
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Result of the discussion?
Room participants: 30/50 in favor of H.264Remote participants (minority): 75/25 in favor of VP8→ No clear consensus
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No decision
Some discussion on the topic: http://webrtchacks.com/ietf-finally-made-decision-mandatory-implement-mti-video-codec-webrtc/
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WebRTC WG
“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
Competing 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 localmedia
Setup PeerConnection
Attach mediaor Data
CloseConnection
← getUserMedia(), etc.
← RTCPeerConnection(), etc.
← addStream(), createOffer(),etc.
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Browser support
iswebrtcreadyyet.com
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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/
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http://status.modern.ie/
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the WebRTC API can have different flavors
(3/3)
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WebRTC Access to IMS (r12)
http://webrtchacks.com/ims-approach-webrtc/
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Adding new wheels to IMS with WebRTC
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SA1 (requirements): reusing IMS client security credentials and/orpublic identities/credentials; how IMS clients communicate withWebRTC clients connected to IMS; IMS services to the WebRTCclient; regulatory functions and charging; offer IMS services to users interacting with a 3rd party website, etc.
SA2 (architecture): expand the IMS architecture and stage 2 procedures as required by the support of WebRTC clients access to IMS; media plane aspects; PBX emulation; signalling; only UNI covered, NNI out of scope.
SA3 (security): WebRTC client authentication mechanisms, media plane security
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SIP Forum WebRTC Task Group
“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”
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Alliance for Telecom Solutions
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WebRTC interop Activity Group
“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”
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GSMA
How does WebRTC relate to VoLTE and RCS?
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Summary
● each deployment/vendor is implementing its own proprietary signaling mechanism
● WebRTC signaling and media is incompatible with existing VoIP deployments – gateways are required to bridge the two worlds
● the WebRTC API can have different flavors
![Page 29: WebRTC standards update (April 2014)](https://reader033.vdocuments.site/reader033/viewer/2022052321/554be46fb4c9055a368b47b7/html5/thumbnails/29.jpg)
MORE INFORMATION
Thank you!
webrtchacks.com
@victorpascual