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What Is WebRTC and How It Works

The world of web-based communication is changing at a speed you might not even imagine.  Now, for making a video conferencing call, sharing a file—all that you need is a URL, thanks to the power of WebRTC for opening the door of infinite possibilities for developers and businesses.     

Developed by Google in 2011 as an open-source framework comprising standards, protocols, and JavaScript APIs, WebRTC or Web Real-Time Communications facilitates the peer-to-peer transfer of audio, video, and data without plugins, native applications, or any third-party proprietary software.  

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What Is WebRTC?

WebRTC stands for web real-time communications. It is a very exciting, powerful, and highly disruptive cutting-edge technology and standard. WebRTC leverages a set of plugin-free APIs that can be used in both desktop and mobile browsers and is progressively becoming supported by all major modern browser vendors. Previously, external plugins were required in order to achieve similar functionality as is offered by WebRTC.

WebRTC leverages multiple standards and protocols, most of which will be discussed in this article. These include data streams, STUN/TURN servers, signaling, JSEP, ICE, SIP, SDP, NAT, UDP/TCP, network sockets, and more.