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WebRTC

APIs and RTCWEB Protocols of the HTML5 Real-Time Web, Third Edition

New for Third Edition
The third edition has an enhanced demo application which now shows the use of the data channel for real-time text sent directly between browsers. Also, a full description of the browser media negotiation process including actual SDP session descriptions from Firefox and Chrome. Hints on how to use Wireshark to monitor WebRTC protocols, and example captures are also included. TURN server support for NAT and firewall traversal is also new.

This edition also features a step-by-step introduction to WebRTC, with concepts such as local media, signaling, and the Peer Connection introduced through separate runnable demos.

Written by experts involved in the standardization effort, this book contains the most up to date discussion of WebRTC standards in W3C and IETF. Packed with figures, example code, and summary tables, this book is the ultimate WebRTC reference.

The Third Edition is available now as an eBook on Amazon Kindle, Smashwords (a variety of DRM-free eBook formats including EPUB, PDF, and MOBI) .

Available now as a paperback at Amazon and Barnes & Noble!

Try using a voice, video, and real-time text chat WebRTC demo with a friend!

Try a slightly updated demo version that works on Chrome, Firefox, and Edge browsers!

Read an interview with Alan B. Johnston by Tsahi Levent-Levi at BlogGeek.

A Review of the First Edition of the book by Alan Quayle.

WebRTC

APIs and RTCWEB Protocols of the HTML5 Real-Time Web, Third Edition

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 second edition has all new chapters on signaling, security and privacy, and a running demo code application! The book is also updated with additional descriptions of NAT traversal, including details of ICE, STUN, and TURN protocols. The latest references and descriptions of the W3C API specifications and the IETF protocol documents are also included.

The Third Edition is available now as an eBook on Amazon Kindle, Smashwords (a variety of DRM-free eBook formats including EPUB, PDF, and MOBI) .

Available now as a paperback at Amazon and Barnes & Noble!

Preface
1. Introduction to Web Real-Time Communications
2. How to Use WebRTC
3. Local Media
4. WebRTC Signaling
5. WebRTC Peer-to-Peer Media
6 Peer Connection and Offer Answer Negotiation
7. Data Channel
8. W3C Documents
9. NAT and Firewall Traversal
10. Protocols
11. IETF Documents
12. Related RFCs
13. Security and Privacy
14. Implementations and Uses
Appendix A - The W3C Standards Process
Appendix B - The IETF Standards Process

Appendix C - Glossary

Appendix D - Supplementary Reading and Sources

About the Authors

Alan B. Johnston

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 co-authored with Phil Zimmermann ZRTP. He is the author of four best selling technical books on Internet Communications, SIP, and security, and a techno thriller 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.

Daniel C. Burnett

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 Tropo and Director of Standards at Voxeo, an Aspect Company. When he can get away, Dan loves camping both with his family and with his son's Boy Scout Troop.

Other Books by Alan B. Johnston

Johnston, Alan B., Counting from Zero. 2011.
Johnston, Alan B., Understanding the Session Initiation Protocol. 3rd Edition Norwood: Artech House, 2009.
Sinnreich, Henry, and Alan B. Johnston, Internet Communications using SIP. 2nd Edition, New York: Wiley, 2006.
Johnston, Alan B., and Dave Piscitello, Understanding Voice over IP Security. Norwood: Artech House, 2006.
Sinnreich, Henry, Alan B. Johnston, and Robert J. Sparks, SIP Beyond VoIP. Melville: VON Publishing, 2005.