Google: WebRTC For Chrome Now Stable, Launches Later This Year

Developer & Design

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We've covered WebRTC a few weeks ago when Firefox showed off a demo of the technology at a conference. For those new to WebRTC, it's essentially video chat in browser using open Web standards like HTML5 and Javascript. Google is equally serious about implementing WebRTC when they announced their plans last year.

The Chrome team updated their WebRTC roadmap to say that the main components of WebRTC are now stable. They will releasing the technology to stable channel users later this year, but they want developers to get a sneak preview of the software that's going into the first WebRTC release.

The software going into WebRTC for Chrome is JSEP, Topologies, ICE/STUN/TURN, DTLS-SRTP and VP8, iSAC, iLBC and G.711. All of these programs work together to provide the live video feed functionality that WebRTC brings to browsers. Here's what each one does to help bring WebRTC to life.

JSEP, or Javascript Session Establishment Protocol, is an API for signaling "that allows for much more powerful apps and flexibility in choice of signaling protocols." This allows them to keep browser-to-browser calls down to only a few lines of Javascript.

Topologies implementation supports "multiple independent PeerConnections" which is the key framework behind the ability for browsers to send and receive video feeds.

ICE and STUN are being used for their "standardized methods for establishing a peer-to-peer connection on the Internet." It will even work if the two browsers communicating are behind private network addresses. They will also be using TURN to support those computers that are behind really tough firewalls.

The DTLS-SRTP will be used to encrypt the communications being sent for WebRTC in Chrome. The first stable release will include this encryption, but Google doesn't say if later versions will drop it.

Chrome will be using VP8 as its video codec of choice. As for audio codecs, users will have a choice between iSAC, iLBC, G.711 and DTMF. iSAC will be the default since it's a "royalty free wideband codec optimized for speech."

All of this is just for the first stable release of WebRTC for Chrome. If that seems like a lot for a first release, that's because it is. It's necessary though since WebRTC requires a lot of different technologies to make it work. WebRTC is going to be the future of communication, especially for those in countries where access to video chat services like Skype are blocked.

Speaking of the future, Google already has a few plans for what they would like to add next. This includes a Data API, screen sharing, PeerConnection proxying and recording. The PeerConnection proxying will be especially useful for those people in countries with Internet restrictions.

Google hasn't provided a date for when WebRTC will roll out, but it will be later this year. Mozilla already has a working prototype up for their version of WebRTC, so Google had better get cracking if they want to beat Mozilla to the punch.

Regardless of who gets there first, the most important thing is that WebRTC is happening. It has the potential to be the next great game changer in online communications.