WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web, Third Edition by Alan B. Johnston

WebRTC, Web Real-Time Communications, is revolutionizing the way web users communicate, both in the consumer and enterprise worlds. WebRTC adds standard APIs 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 to their sites.

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.

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.

Table of Contents

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 Networking and Interworking Examples

2.3 WebRTC Pseudo-Code Example

2.4 References

3 Local Media

3.1 Media in WebRTC

3.2 Capturing Local Media

3.3 Media Selection and Control

3.4 Media Streams Example

3.5 Local Media Runnable Code Example

4 Signaling

4.1 The Role of Signaling

4.2 Signaling Transport

4.3 Signaling Protocols

4.4 Summary of Signaling Choices

4.5 Signaling Channel Runnable Code Example

4.6 References

5 Peer-to-Peer Media

5.1 WebRTC Media Flows

5.2 WebRTC and Network Address Translation (NAT)

5.3 STUN Servers

5.4 TURN Servers

5.5 Candidates

6 Peer Connection and Offer/Answer Negotiation

6.1 Peer Connections

6.2 Offer/Answer Negotiation

6.3 JavaScript Offer/Answer Control

6.4 Runnable Code Example: Peer Connection and Offer/Answer Negotiation

7 Data Channel

7.1 Introduction to the Data Channel

7.2 Using Data Channels

7.3 Data Channel Runnable Code Example

8 W3C Documents

8.1 WebRTC API Reference

8.2 WEBRTC Recommendations

8.3 WEBRTC Drafts

8.4 Related Work

8.5 References

9 NAT and Firewall Traversal

9.1 Introduction to Hole Punching

9.3 WebRTC and Firewalls

9.3.1 WebRTC Firewall Traversal

9.4 References

10 Protocols

10.1 Protocols

10.2 WebRTC Protocol Overview

10.3 References

11 IETF Documents

11.1 Request For Comments

11.2 Internet-Drafts

11.3 RTCWEB Working Group Internet-Drafts

11.4 Individual Internet-Drafts

11.5 RTCWEB Documents in Other Working Groups

11.6 References

12 IETF Related RFC Documents

12.1 Real-time Transport Protocol

12.2 Session Description Protocol

12.3 NAT Traversal RFCs

12.4 Codecs

12.5 Signaling

12.6 References

13 Security and Privacy

13.1 Browser Security Model

13.2 New WebRTC Browser Attacks

13.3 Communication Security

13.4 Identity in WebRTC

13.5 Enterprise Issues

13.6 Privacy

13.7 ZRTP over Data Channel

13.8 Summary

13.9 References

14 Implementations and Uses

14.1 Browsers

14.2 Other Use Cases

14.3 STUN and TURN Server Implementations

14.4 References

INDEX

ABOUT THE AUTHORS

About 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. 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.

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