The biggest accomplishment of WebRTC is bringing high-quality audio and video to the open the Web without the need for third-party software or plugins. Currently, there are no high-quality, well-built, freely available solutions that enable real-time communication in the browser. The success of the Internet is largely due to the high availability and open use of technologies, such as HTML, HTTP, and TCP/IP. To move the Internet forward, we want to continue building on top of these technologies. This is where WebRTC comes into play.

To build a real-time communication application from scratch, we would need to bring in a wealth of libraries and frameworks to deal with the many issues faced when developing these types of applications. These typically include software to handle connection dropping, data loss, and NAT traversal. The great thing about WebRTC is that all of this comes built-in to the browser API. Google has open sourced much of the technology involved in accomplishing this communication in a high-quality and complete manner.

With WebRTC, the heavy lifting is all done for you. The API brings a host of technologies into the browser to make implementation details easy. This includes camera and microphone capture, video and audio encoding and decoding, transportation layers, and session management.

The great thing about the Web is that it moves so fast. New standards are changed or created everyday and it is always improving. Browsers have further improved on this concept by allowing updates to be downloaded and installed without the user ever knowing. This makes the web developer's job an easier one, but it does mean that you have to keep up with what is going on in the world of the Web, and this includes WebRTC.

The way these changes are implemented across browsers is through standardized bodies. These are groups of individuals who work through a common organization to democratize the changing of browser APIs. The two organizations that control the standards for WebRTC are the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETC).

Unlike many other standardized organizations, the W3C allows much of its information to be freely available to the public. This means that anyone can go online and view information about the implementation details of an API. The one for WebRTC is located at http://www.w3.org/TR/webrtc/. This is one way to refer to and learn more about how WebRTC works.

Getting involved in these organizations is one way to not only keep up-to-date on the latest technologies, but also to help shape the future of the Web. Participating in these communities makes browsers the fastest growing development stack out there. If you would like to learn more, visit http://www.w3.org/participate/ to find different ways to participate in the discussions.

Now that you know which browser to use, we will jump right in and try out WebRTC right now! Navigate your browser to the demo application available at https://apprtc.appspot.com/. If you use Chrome, Firefox, or Opera, you should see a drop-down notification that looks similar to this:

Click on Allow to start streaming your audio and video input to the web page. You might have to configure your microphone or web camera settings to get them to work. Once you allow browser access to your camera and microphone, you should see a video feed of yourself from your camera.

The page should generate a custom ID for your current session. You should see this reflected in the URL of the page, such as https://apprtc.appspot.com/r/359323927. Simply copy and paste this URL into another browser window, either on your own computer or another one, and load the web page. Now, if everything works correctly, you should see two video feeds—one from your first client and another from the second. It should start to make sense why WebRTC is a powerful solution. This is how easy WebRTC makes real-time communication in the browser.

Under the hood, WebRTC enables a basic peer-to-peer connection between two browsers. This is the heart of everything that happens with WebRTC. It is the first truly peer-to-peer connection inside a browser. This also means that anything you can do with peer connections can be easily extended to WebRTC. Many applications today use peer-to-peer capabilities, such as file sharing, text chat, multiplayer gaming, and even currencies. There are already hundreds of great examples of these types of applications working right inside the browser.

Most of these applications have one thing in common—they need a low-latency, high-performance connection between two users. WebRTC makes use of low-level protocols to deliver high-speed performance that could not be achieved otherwise. This speeds up data flow across the network, enabling large amounts of data to be transferred in a short amount of time.

WebRTC also enables a secure connection between two users to enable a higher level of privacy between them. Traffic traveling across a peer connection will not only be encrypted, but will also take a direct route to the other user. This means that packets sent in different connections might take entirely different routes over the Internet. This gives anonymity to users of WebRTC applications that is otherwise hard to guarantee when connecting to an application server.

This is just a subset of the types of applications enabled by WebRTC. Since WebRTC is built on the foundations of JavaScript and the Web, it can benefit many existing applications today. After reading this book, you should have the knowledge you need to create innovative WebRTC applications on your own!

Q1. The goal of WebRTC is to provide easy access to real-time communication with no plugins or licensing fees. True or false?

Q2. Which of the following is not a feature that the browser provides through WebRTC?

Q3. Participating in the W3C and IETC is only for big corporations with lots of money. True or false?

Q4. Which of the following is not a type of application that could benefit from using WebRTC?

In this chapter, we gave you a glimpse of the features and technology behind WebRTC. You should have a firm grasp of what WebRTC aims to achieve and how this affects web applications today. You should also have an idea of what types of applications can be built with WebRTC and you should have tried out WebRTC already in your browser.

There was a wealth of information in this chapter, though if you did not take it all in, do not worry. We will go back and cover many of the topics presented here in detail over the course of the book. Feel free to explore some of the resources already on the Web today to get an even better understanding of what WebRTC is all about.

Next, we will start exploring camera and microphone capture using the getUserMedia API.

Then, we will start building our WebRTC application to handle a full one on one video and audio call directly in the browser.

Later on, we will start exploring how to extend this to multiple users, add data transfer through text-based chat and file sharing, and learn about the best security practices that are in place when using WebRTC.