webrtc Getting started with webrtc Introduction to WebRTC
Example
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.
These components, when implemented in a browser, can be accessed through a JavaScript API, enabling developers to easily implement their own RTC web app.
The WebRTC effort is being standardized on an API level at the W3C and at the protocol level at the IETF.
- A key factor in the success of the web 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 enables this.
- Already integrated with best-of-breed voice and video engines that have been deployed on millions of endpoints over the last 8+ years. Google does not charge royalties for WebRTC.
- 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).
Read more about WebRTC from here
