AMT Blog

Chrome dev channel now has WebRTC API

Today, WebRTC.org made an announcement on their blog, indicating that WebRTC API and its underlying components would now be available on the Chrome dev channel. This release implements a slightly older version of the W3C spec. Although the spec is evolving rapidly, WebRTC plans to catch up to it quickly as possible.

Here are some additional details for those developers who’d love to try it out.

Who should use this?

WebRTC must be exciting to developers who want to add real time communication capabilities to their apps.

How do I enable WebRTC?

Switch to the Chrome Dev Channel by following the instructions on this page. Then start Chrome with this switch: "--enable-media-stream"

How can I try it out?

The source code at webrtc.org includes a sample application. Follow these instructions to download and run it.

WebRTC – An Open FrameWork enabling Real Time Communication in the browser

WebRTC is an open framework for the web that allows Real Time Communications in your browser. It includes the fundamental building blocks for high quality communications on the web such as network, audio and video components used in voice and video chat applications.

These components, when implemented in a browser, can be accessed through a Javascript API, enabling developers to easily implement their own RTC web app.

Why WebRTC could be interesting for Developers?

A key factor in the success of the Internet is that its core technologies such as HTML, HTTP, and TCP/IP are open and freely implementable. Currently, there is no free, high quality, complete solution available that enables communication in the browser. WebRTC is a package that enables this.
Already integrated with best-of-breed voice and video engines that have been deployed on millions of end points over the last 8+ years. Google is not charging royalties for this technology.
Includes and abstracts key NAT and firewall traversal technology using STUN, ICE, TURN, RTP-over-TCP and support for proxies.
Builds on the strength of the web browser: WebRTC abstracts signaling by offering a signaling state machine that maps directly to PeerConnection. Web developers can therefore choose the protocol of choice for their usage scenario (for example, but not limited to: SIP, XMPP/Jingle, etc...).