WebRTC is an open framework for the web that enables Real Time Communications in the 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.
The WebRTC effort is being standardized on a API level at the W3C and at the protocol level at the IETF.
Why should I use WebRTC?
We think you’ll want to build your next video chat style application using WebRTC. Here’s why:
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…).