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With WebRTC, real-time communications come to the browser

Chris Minnick and Ed Tittel | June 6, 2013
The WebRTC standard aims to make peer-to-peer communication over the Web as easy as picking up a phone. Here's what developers need to know about WebRTC, including how to set it up and what limitations the protocol currently faces.

More practical architectures for enabling peer-to-peer communications in larger groups include the star architecture, where one peer acts as the focus of the call and sends and receives to all other peers, or a server called a multipoint control unit (MCU), which relays all data between each of the peers.

Security, meanwhile, is built into WebRTC in a serious way. First, all camera and microphone access is explicitly opt-in. That is, the browser will ask the user for each session whether the application can access the camera and microphone and the user must click OK. Next, all data shared between peers is encrypted using AES encryption. Lastly, because WebRTC doesn't use any plug-ins, it runs within the browser sandbox and has only the same access to the user's computer as any Web application does.

How WebRTC Is Being Used Today
WebRTC is still in its infancy, but following the release of Firefox 22 and later versions, its available on about a billion devices and is being deployed. Two great examples of WebRTC being deployed on a large scale are Crunched, a tool for enabling instant live meetings in the browser, and the video collaboration tool Ten Hands.

Developer tools, meanwhile, include the PhonoSDK from Voxeo Labs, which can be used to build robust voice and messaging applications with WebRTC on any browser, and the WebRTC Internals testing and logging tool built into Google Chrome.


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