WebRTC (Web Real-Time Communication) is an API definition being drafted by the World Wide Web Consortium (W3C) to enable Web Browser-to-browser Real Time applications for voice calling (VoIP), Video Chat, and P2P File Sharing without plugins. https://en.wikipedia.org/wiki/WebRTC


Web R T C technology was first developed by Global IP Solutions (or GIPS), a company founded around 1999 in Sweden. In 2011 GIPS was acquired by Google and the W3C started to work on a standard for Web R T C... The framework is based on HTML5 and JavaScript but does not utilise SIP or H.323 like other Unified Communication solutions. The signalling and transfer of data works over RTP and with an XMPP extension called “Jingle”. With the newly introduced JavaScript Session Establishment Protocol (JSEP) direct connections can be established without the need of a piece of hardware in the middle. Depending on the deployment, audio, video and / or other data can now be exchanged to allow Web R T C to happen in your browser. And for our FireWall/NAT traversal specialists: Web R T C can do STUN, ICE, TURN, RTP-over-TCP and supports proxies. What else do you wish for? http://www.telepresence24.com/?p=1804

Already (Aug'2013) supported by Google Chrome, FireFox, Opera.

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