WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web, 2/e (Paperback)

Alan B. Johnston, Daniel C. Burnett

  • 出版商: Digital Codex LLC
  • 出版日期: 2013-06-19
  • 售價: $892
  • 貴賓價: 9.5$847
  • 語言: 英文
  • 頁數: 274
  • 裝訂: Paperback
  • ISBN: 098597883X
  • ISBN-13: 9780985978839
  • 相關分類: HTML
  • 無法訂購

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商品描述

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

Appendix A – The W3C Standards Process
A.1 Introduction to the World Wide Web Consortium
A.2 The W3C WEBRTC Working Group
A.3 How WEBRTC relates to other W3C Working Groups
A.4 References

Appendix B – The IETF Standards Process
B.1 Introduction to the Internet Engineering Task Force
B.2 The IETF RTCWEB Working Group
B.3 How RTCWEB relates to other IETF Working Groups
B.4 References

Appendix C – Glossary

Appendix D – Supplementary Reading and Sources