At the beginning of May, the creation of the Web Real-Time Communications Working Group at the W3C consortium and the RTC-WEB group at the IETF, which should develop peer-to-peer communications standards for the web,
was reported (
twice ) at Habré. In the future, it will be possible to transfer video and sound directly between browsers, so that the need for separate programs like Skype or plug-ins will completely disappear.
To provide such P2P communications, you need to develop a series of APIs, as well as choose codecs for audio and video processing, noise suppression, etc.
No wonder Google is actively involved in creating these standards. With the help of Mozilla, it developed and laid out for public use a set of open
WebRTC technologies for voice and video transmission through simple Javascript APIs (source codes, specifications and testing tools are available).
The WebRTC project contains codecs and protocols that went to Google with the purchase of the Norwegian company Global IP Solutions (GIPS)
in May 2010 . This company has been developing software for video conferencing and IP telephony, owning more than 20 patents in this field. GIPS engine is based on Yahoo Messenger, Cisco WebEx and Lotus Sametime. GIPS technologies are used by mobile phone manufacturers LG, Samsung and others to improve voice quality. It was GIPS that released the
first video chat for Android in April 2010. Now, thanks to Google, many GIPS technologies and intellectual property are actually becoming public domain: the code is published under the BSD license.
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The WebRTC technology allows the use of various signaling protocols, including SIP or XMPP / Jabber.
Of course, such actions by Google cannot be considered as a reciprocal step for the purchase of Skype by one of its main competitors. This project began in January 2011, and the creation of unified open standards for P2P communication between browsers is one of Google’s strategic priorities, regardless of who owns Skype.