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

WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web, Third Edition - Daniel C. Burnett

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. 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
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.
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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. 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
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.
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