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Tropo’s Chief Scientist releases 3rd edition of definitive WebRTC reference guide

Posted on March 19, 2014 by Johnny Diggz

webrtc-book-cover”Tropo, the leader in real-time communications API’s, today announced the release of the 3rd edition of “WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web, co-authored by Tropo Chief Scientist, Dr. Daniel Burnett, and Dr. Alan Johnston.   The new edition is available from Amazon today in paperback and digital formats, and contains the latest updates on the W3C and IETF documents for the emerging WebRTC standard.

WebRTC, Web Real-Time Communications, is revolutionizing the way web users communicate, both in the consumer and enterprise worlds. WebRTC adds standard APIs (Application Programming Interfaces) and built-in real-time audio and video capabilities and codecs to browsers without a plug-in. With just a few lines of JavaScript, web developers can add high quality peer-to-peer voice, video, and data channel communications to their collaboration, conferencing, telephony, or even gaming site or application.

A new step-by-step approach introduces developers to WebRTC starting with getting access to media, establishing a signaling connection, then creating the peer connection.  All new in the 3rd edition is WebRTC data channels:  the latest APIs and protocols, including running data channel demo code.  Actual WebRTC session descriptions from Chrome and Firefox browsers are also explained.  Key topics such as signaling, security and privacy, and NAT traversal with ICE, STUN, and TURN protocols are covered.

“This is the most comprehensive edition to date,” said Burnett, who is also an editor of the two main W3C WebRTC specifications. “Our code samples are now presented in a build-as-you-go manner that developers will love, and the new details on SDP, along with real Wireshark captures, tell the whole story that others only hint at.”

headshot-Dan-Burnett-200x200In his role as Chief Scientist, Dan guides the standardization efforts of Tropo’s products including our Phono WebRTC Gateway and Client.  Developers who want to go beyond the simple demos included with the book can use the free Phono SDK by Tropo to build full-fledged WebRTC-enabled applications.  Dan has a long history of web-related telephony standards experience, being partly responsible for VoiceXML, SRGS, SSML, MRCP, and other such standards. You can reach him via Twitter at @DanielCBurnett.

The newest edition is immediately available to order or download from Amazon in both paperback and digital formats.

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