Last January, Chrome was the first major browser to preview WebRTC,
HTML5's new real time audio and video stack. Since then, we've been
hard at work keeping up with the evolving specification, fixing bugs and
listening to the web community’s feedback.
The main parts of the WebRTC specification are now stable and are coming
soon to all 200M+ Chrome users. With this blog post, we want to help
developers plan for what will be introduced in this first stable release
later this year.
What's in:
JSEP
JSEP
(Javascript Session Establishment Protocol) is an API for signaling that
allows for much more powerful apps and flexibility in choice of
signaling protocols. To abstract the complexity, we provide and maintain
a Javascript lib that makes browser to browser calls a few lines of Javascript.
Topologies
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.
DTLS-SRTP
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.
What’s next?
More functionality and features will appear in future versions of
Chrome. We’ll work on prioritizing them once we get the basics right:
- Data API. Implementation will start once the network stack is ready.
- Screen sharing.
- PeerConnection proxying. The ability to relay a stream to a third party will not make our first version.
- Recording. MediaRecorder specification work has not been completed yet.
Source : http://blog.chromium.org/2012/04/chromes-webrtc-roadmap.html
