WebRTC

WebRTC (Web Real-Time Communication, German "Web Real-Time Communications") is the World Wide Web Consortium (W3C ) in- standardization of open standard for real -time communication (VoIP, instant messaging, video telephony ) within a Web browser and a corresponding software implementation. It is used for recording, coding and ( peer-to -peer ) transfer of multimedia content, and files between web browsers in real-time. The reference implementation is distributed as Free Software in source code under the terms of a BSD-style license. The standardization is largely operated and supported by Google, Inc., Mozilla Foundation, and Opera Software ASA.

History

In May 2010, bought by Google Inc. on the company Global IP Solutions ( GIPS), thus acquiring the ownership of the underlying technology. Since the spring of 2011, a working group of the W3C deals with the standardization. Another working group in the Internet Engineering Task Force ( IETF), which was formed in May 2011, supports the standardization work. On 1 June 2011, published by Google, Inc., the reference software framework as free software. The design of the programming interface is based on preparatory work by the Web Hypertext Application Technology Working Group ( WHATWG ).

WebRTC is already working in stable releases of Opera and Google Chrome and Mozilla Firefox. Microsoft working on an implementation of the programming interfaces in Internet Explorer. WebRTC is seen as an attack on the degree of monopoly in desktop Skype VoIP applications using Microsoft with Skype apparently even wanted to put on WebRTC. Now Microsoft wants to establish its own standard CU- RTC as a standard in the Internet standardization body W3C.

Technology

The framework is based on HTML5 and JavaScript. The transmission is carried out with (S ) RTP over a means of XMPP Jingle extension in conjunction with the newly introduced JavaScript Session Establishment Protocol ( JSEP ) at negotiated direct connection ( peer -to-peer ). In addition to a number at least supportive, ( license fee ) free multimedia codecs standards-compliant implementations may support any other method. For audio Opus as preferable method is provided next to the also required for compatibility with conventional telephone systems support for A-law and μ -law. The newer Internet standard Opus replaces the iLBC and iSAC previously used, developed by GIPS. Google sits down next to it for the self- ransomed video codec VP8 as prescribed video format that is presented as alternative. In addition to multimedia codecs included the reference implementation are further tools for, among other background noise suppression and the software library at libjingle.

  • Since WebRTC is open source, anyone can write his own WebRTC -based application, or simply use an Internet service to do so. For example, the software is available behind palava.tv open source software and GitHub.
  • The video and audio stream is directly transferred from browser to browser, without the interposition of a streaming server.
  • No additional software is required, since modern browser WebRTC already mastered. There are also needed no browser plugins or addons. In order for the application to no operating system is bound, therefore platform independent.
  • Video and audio stream is encrypted.
  • The WebRTC application or web page must be only allows access to camera and microphone.
  • In principle, no user accounts are needed and therefore no personal data must be give away.
  • The interlocutors offered special link only needs to be submitted in order to participate in the conference.

Alternatives

  • Microsoft: CU- RTC ( Customizable, Ubiquitous Real Time Communication)
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