When I first started writing HTML code, I was excited. Here I was typing letters into the keyboard, each one giving the computer some instructions that it knew what to do with. I was excited that I was creating something and that these instructions allowed me to express my creativity. When I finally saved my work, fired up my browser, and loaded my page, I was in awe. I saw my name in big bold letters with a moving picture, called a GIF, of an animated fire.
The Web has certainly come a long way since then. This is largely due to the fact that it is not just a place to create something new, but also a platform to share and show this creativity to others. This powerful platform for creative expression is what powers the Web and keeps it growing faster than ever. It has become so popular that we are moving our entire lives onto it. Websites power your e-mail, entertainment, bank accounts, legal documents, taxes, and even parts of this book were written using web tools. It is the want to move our lives to the Web that drives the development of powerful, easy-to-use APIs, such as WebRTC.
WebRTC is one of the most substantial additions to the Web platform. It brings about an entire suite of new technologies, such as cameras, streaming data, and even an entirely new network protocol stack. It is amazing to not only see the amount of work going into the WebRTC API, but also to know that this is all free for use by any application developer out there.
The aim of WebRTC is to democratize real-time communication. Earlier, building even a smaller video communication application used to take months and involved custom engineering to make even the smallest of applications. However, now we can do it in half the time or even less. This also brings the open source community into real-time communication. You can find other examples of WebRTC in the world and look at how these applications are built by searching through the source code.
It is this creative expression and freedom provided by the Web that drives the motivations for this book. I am glad to have the chance to bring it to more people through my writing, and I hope to inspire others just as I was inspired the first time I built a web page. Writing this book is one of the largest and toughest things I have done in my career, but I am grateful for all the help I have had along the way.
If you are looking for the easiest way to create a new real-time experience and share this with others, then read this book. This book, along with all technical books, is just a way of continuing to drive people to create something even better on the Web. You will learn not just about how to use WebRTC, but also what powers it under the hood. This book serves as not just a learning tool, but also as an inspiration for creating something truly amazing.
Chapter 1, Getting Started with WebRTC, covers how WebRTC enables audio and video communication for web-based applications. You will also begin by running an example of a WebRTC application inside your browser.
Chapter 2, Getting the User's Media, covers the first step when creating a communication application to get webcam and microphone input. This chapter also covers how to use the Media Capture and Streams API to capture this information from you. We also begin development by building the foundation of our communication example.
Chapter 3, Creating a Basic WebRTC Application, covers an introduction to the first WebRTC API—the RTCPeerConnection. This chapter also lays the groundwork for creating a WebRTC application by peeking inside the complex structure of WebRTC and what we can expect when we begin working with the API.
Chapter 4, Creating a Signaling Server, covers the steps in creating our very own signaling server to help our clients find each other on the Internet. This includes in-depth information on how signaling works in WebRTC and how we will utilize it in our example application.
Chapter 5, Connecting Clients Together, covers the actual usage of our signaling server. It also covers connecting two users successfully using the WebRTC API, Media Capture, and the signaling server that we created in the previous chapter to build our working example.
Chapter 6, Sending Data with WebRTC, covers an introduction to the RTCDataChannel and how it is used to send raw data between two peers. This chapter elaborates on our example by adding a text-based chat for our clients.
Chapter 7, File Sharing, elaborates on the concept of sending raw data by looking at how we can share files between two peers. This will demonstrate the many uses of WebRTC outside of audio and video sharing.
Chapter 8, Advanced Security and Large-scale Optimization, covers advanced topics when delivering a large-scale WebRTC application. We look at theoretical security and performance optimizations used by other companies in the industry.
Appendix, Answers to Self-test Questions, covers the answers to all the self-test questions that appear at the end of every chapter.
All the examples in this book are built on the Web using web standards. Since the WebRTC specification is fairly new, it is recommended to run the examples inside an updated browser. It is preferred to use the latest Firefox or Chrome web browser for these.
All of the server code is written using Node.js. The Node.js framework runs inside most Windows, Linux, and Mac OSX machines.
You can use any text editor that supports the writing of JavaScript and HTML code.
You should have some experience building web applications using HTML and JavaScript. Maybe, you are building an application right now, or have an idea of a new application that utilizes the power of audio and video communication between users. You might also need to deliver an application with high-performance data transfer between users.
You should have a firm grasp of programming concepts and web development, but the book is written for even a novice web engineer. The concepts are covered in-depth and taken one bit at a time, rather than charging ahead into the advanced topics. You may not know anything about WebRTC or have only heard a bit about it, and want to learn the inner workings of real-time communication.
In this book, you will find a number of text styles 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: "The one thing to notice is that when we get an ICE candidate from theirConnection, we are adding it to our connection, and vice versa."
A block of code is set as follows:
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="utf-8" />
<title>Learning WebRTC - Chapter 4: Creating a RTCPeerConnection</title>
</head>
<body>
<div id=""container"">
<video id=""yours"" autoplay></video>
<video id=""theirs"" autoplay></video>
</div>
<script src=""main.js""></script>
</body>
</html>Any command-line input or output is written as follows:
> 1 + 1 2 > var hello = "world"; undefined > "Hello" + hello; 'Helloworld'
New terms and important words are shown in bold. Words that you see on the screen, for example, in menus or dialog boxes, appear in the text like this: "Now, you should be able to click on the Capture button and capture one frame of the video feed on the canvas."
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