What's in the initial WebRTC release includes the following:
Our implementation will support multiple independent PeerConnections, each capable of sending and receiving multiple independent media sources.
ICE / STUN / TURN
ICE and STUN are standardized methods for establishing a peer-to-peer connection on the Internet, even if the two end points are behind private network addresses (NAT). Chrome’s current stack deviates from the official current standards. We are working to fix this.
We will also support TURN servers to allow connections through tougher firewalls, where relaying and encapsulation are needed. Exactly what type of TURN will be supported is TBD.
Encryption will be mandatory for all usage of WebRTC in Chrome. For our first stable release, we will implement DTLS-SRTP.
VP8, iSAC, iLBC, G.711
The video codec support by Chrome will be VP8. We've made several major improvements inside and around VP8 to ensure it can deliver a great real time experience. On the audio side, we will initially support iSAC, iLBC, G.711, and DTMF, with iSAC being the default. It is a royalty free wideband codec optimized for speech, open sourced at webrtc.org.
I should mention WebRTC will also work on Firefox browsers with Mozilla doing some early demos of it in action and Microsoft is expected to support it in Internet Explorer. As for the future, Google announced some of their plans. The next release will include a Data API, screen sharing, PeerConnection proxying and recording - though the MediaRecorder specification hasn't been completed yet. The PeerConnection proxying is interesting since it will allow users to bypass restrictions that restrictive countries put in place. Think of WebRTC as Tor but for VoIP and video and more feature-rich collaboration. Exciting times ahead!
Tags: chrome, codec, collaboration, dtmf, google, isac, microsoft, skype, tor, video conferencing, voip, vp8, webrtc, wideband
Related tags: peerconnection proxying, multiple independent, should mention, webrtc roadmap, webrtc release, webrtc
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