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WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web (English Edition) [Format Kindle]

Alan B. Johnston , Daniel C. Burnett

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  • Longueur : 350 pages
  • Langue : Anglais
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Descriptions du produit

Présentation de l'éditeur

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. Key topics such as signaling, security and privacy, and NAT traversal with ICE, STUN, and TURN protocols are covered.

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
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
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

Appendix A – The W3C Standards Process
A.1 Introduction to the Wo

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.

Détails sur le produit

  • Format : Format Kindle
  • Taille du fichier : 20698 KB
  • Nombre de pages de l'édition imprimée : 350 pages
  • Utilisation simultanée de l'appareil : Illimité
  • Editeur : Digital Codex LLC; Édition : 3 (12 mars 2014)
  • Vendu par : Amazon Media EU S.à r.l.
  • Langue : Anglais
  • ASIN: B00IZNUP22
  • Synthèse vocale : Activée
  • X-Ray :
  • Word Wise: Non activé
  • Composition améliorée: Activé
  • Classement des meilleures ventes d'Amazon: n°260.571 dans la Boutique Kindle (Voir le Top 100 dans la Boutique Kindle)
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Amazon.com: 5.0 étoiles sur 5  4 commentaires
3 internautes sur 3 ont trouvé ce commentaire utile 
5.0 étoiles sur 5 I was quickly humbled by just how much WebRTC technology has progressed and carry the Kindle book along on 3 devices. 27 juillet 2014
Par Henry Sinnreich - Publié sur Amazon.com
Format:Format Kindle|Achat vérifié
Video chat with text and various application data from browser to browser is emerging as a new feature for mobile and fixed devices, actually on most anything equipped with a browser, even though at present not all but some leading browsers are supporting such interoperability. Innovators discover how their apps and back-ends can be improved and updated using WebRTC for such as privacy and security, features, and performance.
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.
3 internautes sur 3 ont trouvé ce commentaire utile 
5.0 étoiles sur 5 Best WebRTC Technical Book Available 10 mai 2014
Par Philip K Edholm - Publié sur Amazon.com
This is the best technical WebRTC book available. It is absolutely mandatory for any web or telecom developer looking to understand how to develop and deploy WebRTC. In this Third Edition, Dan and Alan continue to expand the overall content and keep up with the emerging standards. WebRTC is an exciting technology that will change communication int he same way the original web changed information and will create huge opportunities, both in technology, start-ups and business (think Google, Amazon, Facebook, etc.). For any developer or company with a web presence, WebRTC is a big part of your future and this book is the clear way to understand how to develop. The book includes clear code examples and easy to implement initial code. With this book, your website will be managing real time HD voice, HD video,and data interactions between your users or subscribers in a matter of days, or even hours. Highly Recommended!!
1 internautes sur 1 ont trouvé ce commentaire utile 
5.0 étoiles sur 5 The Definitive Source for WebRTC Technical Insight 2 juillet 2014
Par John - Publié sur Amazon.com
Alan and Dan provide comprehensive WebRTC information in an easy to understand format while bringing WebRTC to life! This WebRTC technical reference not only includes example source code, it has a companion website where anyone with a browser can experiment with a fully functional WebRTC system and see the various message flows involved with WebRTC media and data flows. Each new addition of this book brings updates on the progress of the WebRTC related standards and the rapid evolution of this emerging disruptive technology. This economical reference is all most web developers need to WebRTC enable their applications with voice, video, or interactive real-time data flows.
1 internautes sur 1 ont trouvé ce commentaire utile 
5.0 étoiles sur 5 Mandatory reading for those working with WebRTC 3 avril 2014
Par Tsahi Levent-Levi - Publié sur Amazon.com
Format:Format Kindle
This is the third edition of the book by Alan and Dan.
While the first one was excellent, this fills in all the missing "holes" that existed in the first edition: it gives more information about the use of the data channel, it covers more ground in explaining the SDP and it provides some "debugging" by showing Wireshark captures and explaining them.
For people who need to get up to speed with WebRTC - understand the standard, how it operates in the network itself, how to use its APIs and how to debug it - this is an invaluable resource.
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