WebRTC, according to Wikipedia, is “an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat and P2P file sharing without the need of either internal or external plug-ins”.

It is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.  The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Originally released as open source technology by Google Inc. in 2011, WebRTC lessens dependence on plug-ins such as Adobe Flash (and possible licensing fees) and allows those developers who only use HTML5 and JavaScript to initiate calls from browsers and to then transfer them to their smartphones without losing connections.

WebRTC has now implemented open standards for real-time, plugin-free video, audio and data communication.

WebRTC Architecture (Source: WebRTC.org)

The need is real:
  • Many web services already use RTC, but need downloads, native apps or plugins.
    These includes Skype, Facebook (which uses Skype) and Google Hangouts (which use the Google Talk plugin).
  • Downloading, installing and updating plugins can be complex, error prone and annoying.
  • Plugins can be difficult to deploy, debug, troubleshoot, test and maintain—and may require licensing and integration with complex, expensive technology.

The guiding principles of the WebRTC project are that its APIs should be open source, free, standardized, built into web browsers and more efficient than existing technologies.

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WebRTC is going to dominate internet telecoms development in 2015

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What is a WebRTC Application?