Overview

IMPORTANT NOTE: The third edition of this book is now available! ISBN-13: 978-0-9859788-6-0 Up to date with the latest changes in the APIs and protocols, the third edition includes a new chapter on data channels with running demo code. 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. 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. Written by experts involved in the standardization effort, this book introduces and explains the W3C APIs and the IETF protocols of WebRTC. Packed with figures, example code, and summary tables, this book makes complicated concepts and technologies such as peer-to-peer media and NAT and firewall traversal easy to understand. The 2nd edition has all new chapters on Signaling and Security & Privacy, as well as running demo code (client and server-side) and further details on NAT traversal with ICE, STUN, and TURN protocols. In addition the book contains the latest updates on the W3C and IETF standards documents. Chapters: 1 Introduction to Web Real-Time Communications 1.1 WebRTC Introduction 1.2 Multiple Media Streams in WebRTC 1.3 Multi-Party Sessions in WebRTC 1.4 WebRTC Standards 1.5 What is New in WebRTC 1.6 Important Terminology Notes 1.7 References 2 How to Use WebRTC 2.1 Setting Up a WebRTC Session 2.2 WebRTC Example Implementations 2.3 WebRTC Pseudo-Code Example 2.4 References 3 WebRTC Peer-to-Peer Media 3.1 WebRTC Media Flows 3.2 WebRTC and Network Address Translation (NAT) 3.3 Introduction to Hole Punching 3.4 Interactive Connectivity Establishment 3.5 WebRTC and Firewalls 3.6 References 4 WebRTC Signaling 4.1 The Role of Signaling 4.2 Signaling Transport 4.3 Signaling Protocol 4.4 Summary 4.5 References 5 W3C WebRTC Documents 5.1 WebRTC API Reference 5.2 WEBRTC Recommendations 5.3 WEBRTC Drafts 5.4 Related Work 5.5 References 6 WebRTC Protocols 6.1 Protocols 6.2 WebRTC Protocol Overview 6.3 References 7 Demo Application Code 7.1 Overview of Basic WebRTC Demo Code 7.2 Web Server 7.3 Signaling channel 7.4 Client WebRTC application 7.5 References 8 IETF WebRTC Documents 8.1 Request For Comments 8.2 Internet-Drafts 8.3 RTCWEB Working Group Internet-Drafts 8.4 Individual Internet-Drafts 8.5 RTCWEB Documents in Other Working Groups 8.6 References 9 IETF Related RFC Documents 9.1 Real-time Transport Protocol RFCs 9.2 Session Description Protocol RFCs 9.3 NAT Traversal RFCs 9.4 Codecs 9.5 References 10 Security and Privacy 10.1 Browser Security Model 10.2 New WebRTC Browser Attacks 10.3 Communication Security 10.4 Identity in WebRTC 10.5 Enterprise Issues 10.6 Privacy 10.7 Summary 10.8 References 11 WebRTC Implementations 11.1 Apple Safari 11.2 Google Chrome 11.3 Mozilla Firefox 11.4 Microsoft Internet Explorer 11.5 Opera 11.6 References

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Dr. Alan B. Johnston has over thirteen years of experience in SIP, VoIP (Voice over IP), and Internet Communications, having been a co-author of the SIP specification and a dozen other IETF RFCs, including the ZRTP media security protocol. He is the author of four best selling technical books on Internet Communications, SIP, and security, and a technothriller novel Counting from Zero that teaches the basics of Internet and computer security. He is on the board of directors of the SIP Forum. He holds Bachelors and Ph.D. degrees in electrical engineering. Alan is an active participant in the IETF RTCWEB working group. He is currently a Distinguished Engineer at Avaya, Inc. and an Adjunct Instructor at Washington University in St Louis. He owns and rides a number of motorcycles, and enjoys mentoring a robotics team. Dr. Daniel C. Burnett has more than a dozen years of experience in computer standards work, having been author and editor of the W3C standards underlying the majority of today's automated Interactive Voice Response (IVR) systems. He has twice received the prestigious Speech Luminary award from Speech Tech Magazine for his contributions to standards in the Automated Speech Recognition (Voice Recognition) field. As an editor of the PeerConnection and getUserMedia W3C WEBRTC specifications and a participant in the IETF, Dan has been involved from the beginning in this exciting new field. He is currently the Chief Scientist at Voxeo Labs and Director of Standards at Voxeo. When he can get away, Dan loves camping both with his family and with his son's Boy Scout Troop.

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