WebRTC Framework for Real Time Audio and Video Apps in Browser

WebRTC is an open framework for the web that enables Real Time Communications (RTC) in the browser. It is an open technology for voice and video on the web. The WebRTC initiative is supported by Google, Mozilla and Opera.

WebRTC offers web application developers the ability to write rich, realtime multimedia applications (like video chat) on the web, without requiring plugins, downloads or installs. It’s purpose is to help build a strong RTC platform that works across multiple web browsers, across multiple platforms.

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. WebRTC code is now about to land in Chromium as third party software at src/third_party/webrtc. This means that the crucial media processing capabilities necessary to implement real-time communication are available as a part of Chromium.

Google is working to complete the remaining steps for a full integration in Chromium and Chrome including JavaScript APIs in WebKit and handling of the native audio and video capture and rendering. When it is done, any web developer shall be able to create RTC applications, like the Google Talk client in Gmail, without using any plugins but only WebRTC components that runs in the sandbox. The overall architecture looks something like this: