14 DAY TRIAL // WebRTC, the real-time communications API for the web, is evolving rapidly. It was only introduced last year, but there's already support for it in Chrome and Mozilla is working on adding support in Firefox as well.
The team behind the WebRTC API believes that the main components are now stable enough for regular use and Google plans to enable support by default in Chrome soon.
It looks like Google plans to build WebRTC into Chrome 20, which is already in the dev channel.
For now, it's signaling to developers that the technology is almost ready for actual use and detailing the components that are stable as well as the components that will come later.
"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," Google explained.
"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," it added.
JSEP, the Javascript Session Establishment Protocol, is an API designed to make it easier for developers to implement the signaling part of the connection in the form of a JavaScript library that can be built into apps.
Chrome will soon support ICE and STUN for establishing connections between the two devices at each end of the communication channel, and will later support TURN servers for non peer-to-peer connections.
WebRTC communication is always encrypted in Chrome. The browser is using DTLS-SRTP. Finally, Chrome will be supporting the VP8 open source video codec and iSAC, iLBC as well as G.711 and DTMF for audio.
What all of this means is that you'll soon be able to create a real-time communications app that is guaranteed to work in any browser that supports the proposed WebRTC standard, Chrome and Firefox for now.