Web Real-Time Communication (WebRTC) is a new (still under development) open framework for the Web to enable browser-to-browser applications for audio/video calling, video chats, and peer-to-peer file sharing without any additional third-party software/plugins.

It was open sourced by Google in 2011 and includes the fundamental building components for high-quality communication on the Web. These components, when implemented in a browser, can be accessed through a JavaScript API, enabling developers to build their own rich media web applications. Google, Mozilla, and Opera support WebRTC and are involved in the development process.

The following are the major components of the WebRTC API:

The WebRTC technology contains a complete stack for voice communications, which includes all the necessary codecs. In addition, it provides software-based AEC (acoustic echo cancellation), AGC (automatic gain control), and noise reduction/suppression.

WebRTC also provides the following modern and powerful possibilities to utilize video communications:

Peer-to-peer connections can be established using the built-in key NAT/firewall traversal technology using ICE/STUN/TURN and support proxies. So, you don't need to use any special third-party technologies to build a peer-to-peer communication between the customers located behind a firewall or NAT.

To help mitigate the effects of packet loss and unreliable networks, WebRTC supports dynamic jitter buffers and error concealment techniques for audio or video communications.

Chapter 1, Developing a WebRTC Application, covers the basics of the technology and building a completed audio/video conference real-life web application. We will also learn about SDP (Session Description Protocol), signaling, client-server sides' interoperation, and configuring the STUN and TURN servers.

Chapter 2, Using the WebRTC Data API, explains how to build a peer-to-peer, cross-platform file sharing web service using the WebRTC Data API.

Chapter 3, The Media Streaming and Screen Casting Services, introduces you to streaming prerecorded, peer-to-peer media content and desktop sharing. In this chapter, you will build a simple application that provides such functionality.

Chapter 4, Security and Authentication, covers security questions and why you shouldn't forget them while developing your applications. So, in this chapter, you will learn how to make your WebRTC solutions secure, why authentication may be very important, and how you can implement this functionality in your products.

Chapter 5, Mobile Platforms, covers how to make your interactive application work great on mobile devices. You will learn about the aspects that will help you develop great WebRTC products keeping mobile devices in mind.

You don't need any special software to read this book and to make applications. We will use JavaScript for the client side, so you will need a web browser (Chrome is preferable). We will use Erlang for server-side code, so you need an OS compatible with Erlang; for example, you can use MS Windows, but any Linux distribution or Mac OS X would be preferable.

This book will introduce you to creating interactive media web applications and services using WebRTC. In this book, we will create several web-based applications using WebRTC, including ones for the client side and the server side.

Client-side code will be presented in JavaScript. This is an optimal and simple way to utilize a browser WebRTC API.

Server-side code for applications in the book is written in Erlang. Why Erlang? It is a small language, good for quick prototyping and writing production code in an elegant and powerful way.

Thus, you don't have to be a professional software developer, but you are expected to have programming knowledge in Java, JavaScript, Ruby, Python or other mainstream languages, or at least have typical web development skills.

In this book, you will find a number of styles of text that distinguish between different kinds of information. Here are some examples of these styles, and an explanation of their meaning.

Code words in text, database table names, folder names, filenames, file extensions, pathnames, dummy URLs, user input, and Twitter handles are shown as follows: "We can declare our module through the use of the module directive."

A block of code is set as follows:

{ok, Port} = application:get_env(port),
{ok, Timeout} = application:get_env(timeout),
{ok, Workers} = application:get_env(workers),

When we wish to draw your attention to a particular part of a code block, the relevant lines or items are set in bold:

{ok, Port} = application:get_env(port),
{ok, Timeout} = application:get_env(timeout),
{ok, Workers} = application:get_env(workers),

Any command-line input or output is written as follows:

# ln –s /usr/local/www/index.html /usr/local/www/default.html

New terms and important words are shown in bold. Words that you see on the screen, in menus or dialog boxes for example, appear in the text like this: "Now, you can right-click on the link and select the Download to... option."

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