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This book is a step-by-step project-based guide that aims to teach you how to develop your own web applications and services with WebRTC in a concise, practical manner. This book will be perfect for you if you are a WebRTC developer and want to build complex WebRTC applications and projects, or if you want to gain practical experience in developing web applications, advanced WebRTC media handling, server and client signaling, call flows, or third-party integration. It is essential to have prior knowledge of building simple applications using WebRTC.
Deliver rich audio and video real-time communication and peer-to-peer data exchange right in the browser, without the need for proprietary plug-ins. This concise hands-on guide shows you how to use the emerging Web Real-Time Communication (WebRTC) technology to build a browser-to-browser application, piece by piece. The authors’ learn-by-example approach is perfect for web programmers looking to understand real-time communication, and telecommunications architects unfamiliar with HTML5 and JavaScript-based client-server web programming. You’ll use a ten-step recipe to create a complete WebRTC system, with exercises that you can apply to your own projects. Tour the WebRTC development cycle and trapezoid architectural model Understand how and why VoIP is shifting from standalone functionality to a browser component Use mechanisms that let client-side web apps interact with browsers through the WebRTC API Transfer streaming data between browser peers with the RTCPeerConnection API Create a signaling channel between peers for setting up a WebRTC session Put everything together to create a basic WebRTC system from scratch Learn about conferencing, authorization, and other advanced WebRTC features
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 6.3 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 14 Implementations and Uses INDEX ABOUT THE AUTHORS
Deliver rich audio and video real-time communication and peer-to-peer data exchange right in the browser, without the need for proprietary plug-ins. This concise hands-on guide shows you how to use the emerging Web Real-Time Communication (WebRTC) technology to build a browser-to-browser application, piece by piece. The authors’ learn-by-example approach is perfect for web programmers looking to understand real-time communication, and telecommunications architects unfamiliar with HTML5 and JavaScript-based client-server web programming. You’ll use a ten-step recipe to create a complete WebRTC system, with exercises that you can apply to your own projects. Tour the WebRTC development cycle and trapezoid architectural model Understand how and why VoIP is shifting from standalone functionality to a browser component Use mechanisms that let client-side web apps interact with browsers through the WebRTC API Transfer streaming data between browser peers with the RTCPeerConnection API Create a signaling channel between peers for setting up a WebRTC session Put everything together to create a basic WebRTC system from scratch Learn about conferencing, authorization, and other advanced WebRTC features
Learn how to use the WebRTC API to establish peer-to-peer communications. After reading this guide, you will know how to connect your users with each other, how to create a system to let your users perform video calls, and how to transfer data from one user to another. Table of Contents WEBRTC API Web Paradigms ICE Servers Peer Connection ICE Candidate Offer and Answer Session Description Media Streams Events Configuration Configuring the Signaling Server Configuring the ICE Servers Implementing WebRTC Data Channels QUICK REFERENCE Connection ICE Candidate Signals Streams Events Data Channels This guide assumes that you have a basic knowledge of HTML, CSS and JavaScript, and you know how to create files and upload them to a server. If you don't know how to program in HTML, CSS or JavaScript, you can download our guides Introduction to HTML, Introduction to CSS, and Introduction to JavaScript. For a complete course on web development, read our book HTML5 for Masterminds. This guide is a collection of excerpts from the book HTML5 for Masterminds. The information included in this guide will help you understand a particular aspect of web development, but it will not teach you everything you need to know to develop a website or a web application. If you need a complete course on web development, read our book HTML5 for Masterminds. For more information, visit our website at www.formasterminds.com.
How prepared are you to build fast and efficient web applications? This eloquent book provides what every web developer should know about the network, from fundamental limitations that affect performance to major innovations for building even more powerful browser applications—including HTTP 2.0 and XHR improvements, Server-Sent Events (SSE), WebSocket, and WebRTC. Author Ilya Grigorik, a web performance engineer at Google, demonstrates performance optimization best practices for TCP, UDP, and TLS protocols, and explains unique wireless and mobile network optimization requirements. You’ll then dive into performance characteristics of technologies such as HTTP 2.0, client-side network scripting with XHR, real-time streaming with SSE and WebSocket, and P2P communication with WebRTC. Deliver superlative TCP, UDP, and TLS performance Speed up network performance over 3G/4G mobile networks Develop fast and energy-efficient mobile applications Address bottlenecks in HTTP 1.x and other browser protocols Plan for and deliver the best HTTP 2.0 performance Enable efficient real-time streaming in the browser Create efficient peer-to-peer videoconferencing and low-latency applications with real-time WebRTC transports
The book begins by teaching you how to capture audio and video streams from the browser using the Media Capture and Streams API. You will then create your first WebRTC application capable of audio and video calling. The book will also give you in-depth knowledge about signaling and building a signaling server in Node.js. While being introduced to the RTCDataChannel object, you will learn how it relates to WebRTC and how to add text-based chat to your application. You will also learn to take your application further by supporting multiple users through different technologies and scale its performance and security. This book will also cover several theories using full mesh networks, partial mesh networks, and multipoint control units. By the end of this book, you will have an extensive understanding of real-time communication and the WebRTC protocol and APIs.
Build your own video chat application - but that's just the beginning. With WebRTC, you'll create real-time applications to stream any kind of user media and data directly from one browser to another, all built on familiar HTML, CSS, and JavaScript. Power real-time activities like text-based chats, secure peer-to-peer file transfers, collaborative brainstorming sessions - even multiplayer gaming. And you're not limited to two connected users: an entire chapter of the book is devoted to engineering multipeer WebRTC apps that let groups of people communicate in real time. You'll create your own video conferencing app. It's all here. WebRTC is an API exposed in all modern web browsers. After almost a decade of development, the WebRTC specification was finalized, and this book provides faithful coverage of that finalized specification. You'll start by building a basic but complete WebRTC application for video chatting. Chapter by chapter, you'll refine that app and its core logic to spin up new and exciting WebRTC-powered apps that will have your users sharing all manner of data with one another, all in real time. No third-party libraries or heavy downloads are required for you or your users: you'll be writing and strengthening your knowledge of vanilla JavaScript and native browser APIs. You'll learn how to directly connect multiple browsers over the open internet using a signaling channel. You will gain familiarity with a whole set of Web APIs whose features bring WebRTC to life: requesting access to users' cameras and microphones; accessing and manipulating arbitrary user files, right in the browser; and web storage for persisting shared data over the life of a WebRTC call. Like any Web API, WebRTC doesn't enjoy a perfect implementation in any browser. But this book will guide you in writing elegant code to the specification, with backward-compatible fallback code for use in almost all modern browsers. Use WebRTC to build the next generation of web applications that stream media and data in real time, directly from one user to another - all by working in the browser. What You Need: Readers need a text editor, an up-to-date copy of Chrome or Firefox, and a POSIX-style command-line shell. They'll also need to install a little bit of open-source software, especially Node.js. All necessary setup is covered in full in the book's introductory chapter.
IMPORTANT NOTE: The third edition of this book is now available ISBN-13: 978-0-9859788-6-0 Up to date with the latest changes in the APIs and protocols, the third edition includes a new chapter on data channels with running demo code. A new step-by-step approach introduces developers to WebRTC starting with getting access to media, establishing a signaling connection, then creating the peer connection. 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
The book will follow a step-by-step tutorial approach to construct an application that allows video conferencing and calls between two browsers and a system for sharing files among a group.This book is ideal for developers new to the WebRTC standards who are interested in adding sensor-driven, real-time, peer-to-peer communication to their web applications. You will only need basic experience with HTML and JavaScript.