WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web, Third Edition (Anglais) Broché – 11 mars 2014
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Description du produit
Présentation de l'éditeur
New for the 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.
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
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
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
4.1 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
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
6 Peer Connection and Offer/Answer Negotiation
6.1 Peer Connections
6.2 Offer/Answer Negotiation
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
7.3.1 Client WebRTC Application
8 W3C Documents
8.1 WebRTC API Reference
8.2 WEBRTC Recommendations
8.3 WEBRTC Drafts
8.4 Related Work
9 NAT and Firewall Traversal
9.1 Introduction to Hole Punching
9.3 WebRTC and Firewalls
9.3.1 WebRTC Firewall Traversal
10.2 WebRTC Protocol Overview
11 IETF Documents
11.1 Request For Comments
11.3 RTCWEB Working Group Internet-Drafts
11.4 Individual Internet-Drafts
11.5 RTCWEB Documents in Other Working Groups
12 IETF Related RFC Documents
12.1 Real-time Transport Protocol
12.2 Session Description Protocol
12.3 NAT Traversal RFCs
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
14 Implementations and Uses
ABOUT THE AUTHORS
Biographie de l'auteur
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 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.
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.
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Commentaires client les plus utiles sur Amazon.com
Though I am quite familiar with both the topic and the 1st edition of this book, I was quickly humbled by just how much WebRTC technology has progressed. The 2014, 3rd edition of the book has not only kept up but has also been completely restructured to make the underlying wealth of technical knowledge digestible as befits both university educators and known industry experts - the authors of this book. The usefulness for a broad audience of web and communication developers, IT managers and various business decision makers has been well preserved and actually improved. Following the text and the extremely clear diagrams makes reading the respective chapters attractive, all the way to understand the pseudo code and full code examples. A sample browser based video conferencing application developed by the authors is also included.
The book is an excellent tutorial on WebRTC as it is being developed by the W3C and IETF standards organizations; still one would have wished as well to be enlightened with some examples or comments on specifically how this technology impacts the perennial enterprise and institutional IT markets and the public telephony providers. How will WebRTC impact over the top (OTT) Internet platforms and social networks that already provide rich, state of the art video chats, multiparty conferencing, collaboration and may include streaming media along various data apps. However, given the complexities of closed, real life commercial platforms, it is understandable why this book just sticks to the basics and makes readers better understand the news from the technology industry.
I got really carried away by reading this new edition and keep the Kindle e-book on three devices to have it anywhere along.
So I am really pleased to now see this Chinese edition, making WebRTC technology much more accessible to web developers and technologists in China, as well as making it easier for Mandarin speakers in the U.S., Europe or anywhere in the world to read up on WebRTC. I am not a Chinese speaker so I cannot comment on the translation - but knowing the authors I expect this to be good. So I look forward to seeing reviews by Chinese speakers - my 5 stars are to acknowledge the importance of the first Chinese translation of this seminal WebRTC book.
A critical decision factor for me was the very recent Publication Date of this title, important for newer and evolving standards.
If you are looking for a book that is very practical, with a lot of example code, and a lot of "here are a lot of small exercises to build you up to the more complicated things" - then this not the book for you.
While incredibly informative, this book is very, very, very heavy on the theory before it even gets into a sample application (which you can't download the code to, which is a point of frustration).
That being said, it is incredibly thorough, and attempts to explain almost every aspect of how a WebRTC system does (and could) work - which has value in it's own right.
All in all - I don't think you can get this information condensed into one package anywhere else, and they have done a very good job in attempting to put it all together, especially knowing the state of flux of WebRTC at the time of publishing.