WebRTC - A Revolution in Real-Time Communication

In today’s Web-based world, a new technology has been in the limelight for quite some time now. The name of this new technology is WebRTC, short for Web-based real-time communication. It is a new open-source project from the house of Google. As the name suggests, it provides a new level of flexible communication on a real-time basis without any kind of time lag. And it does all of this using standard Web browsers. It takes the help of simple HTML5 and JavaScript programs to create this real-time communication service. Best of all, users do not have to go through the hassle of downloading and running different kinds applications, as this technology only needs a browser to work. The prime mission of WebRTC is to create standardization for a richly featured application in a browser. This Google initiative has inspired many other organizations to build this kind of product as well.

A Closer Look

WebRTC is an open-source framework for Web technology that facilitates a real-time communication process in the browser. It includes some of the very basic building blocks for the creation of exceptional real-time communication through a Web browser. These blocks are components of audio, video, video chat and networking. They can be easily accessed by the developers using a JavaScript API when they are employed in a browser. This, in turn, allows developers to create their very own Web app for the real-time contact process. It is standardized on two different levels. On the API level, it is standardized by the W3C, while on the protocol level, it is standardized by the IETF. (For more on open source, see Open Source: Is It Too Good to Be True?)

Reasons for Using WebRTC

Now you must be thinking that with so many different communication technologies available today, why should we choose this particular technology for application-based video chat communication? Well, here are a few reasons:

This framework uses all open and free Internet technologies, like HTML, TCP/IP protocol, and Hypertext Transfer Protocol. It is a full package that turns the browser into a communication machine in the cheapest manner possible.
It supports different kinds of proxies, and has an abstract key like NAT. It also uses the latest firewall technology via ICE, TURN, STUN and RTP-over-TCP.
It is integrated with the best engines for processing voice and video quality, which are deployed on many different end points.
The unique signalling process of this framework is due to a special and unique signalling machine. This machine is a state machine, which maps directly to a peer-to-peer connection. It builds the strength of the browser. The developer can choose any protocol, depending upon the situation.

Codecs

There are few codecs that are very important for this technology:

Opus Audio Codec: This is a royalty-free codec. It supports both constant as well as variable types of bit rate encoding techniques. It also supports sampling rates from 8 kHz to 48 kHz.
iSAC Audio Codec: This is an adaptive and robust technique that is used in many different kinds applications using voice over IP and audio streaming features.
iLBC Audio Codec: This is also an audio codec used in different platforms. It uses a narrowband technique and the latest version of this codec has a profile draft feature in it.
VP8: This is a very efficient video codec used by different kinds of platforms. It uses a unique compression technique, which is known to reduce the size but the not the quality of the images. It’s developed by On2 technologies, but since On2 is a part of Google, this framework uses the codec without any cost.

Components of the WebRTC Package

There are many different components that comprise the total WebRTC package. The major components are given below with explanations:

Audio: This framework provides a complete set of capabilities to provide flawless sound in the communication process. It has many different codecs and audio components, which increase the rich experience of the sound. It also has a software-based functionality, which reduces any echo, using acoustic echo cancellation techniques. This framework also works on suppression of noise and reduces it, automatic gain control and controls the hardware access in different types of platforms.
Video: It uses VP8 for its video as it is the latest video codec to be introduced. Using this codec for the video component, this framework can fix all kinds of packet loss. Additionally, the framework can clean up all types of blurry, unfocused and noisy images and also has the capability of capturing and performing playback in many different types of platforms.
Network: As previously mentioned, this framework works on a secure peer-to-peer connection using different technology. This also has a dynamic jitter buffer and error-correcting techniques which work on any unreliable network and stabilize it. These techniques together help the framework to utilize every different kind of platform with the same quality and also conceal the loss of packets in the network to increase the quality of video and audio.

WebRTC Facts

You can access the code for this framework from the site here. You can also implement your own renderer file and different types of hook in the platform of WebRTC. If you have good enough programming skills to render the file of WebRTC, then you can create your own software application and also can contribute your code to contribute to the future of this technology. You only have to know JavaScript API and a few Web development skills. This framework is also supported by Opera and Mozilla. But few of its components, like NetEQ, AEC, voice and a video engine are from Google’s acquisition of GIPS (Global IP Solutions).

Conclusion

The WebRTC framework is free of cost and highly efficient. It uses different kinds of codecs and a robust technology to make real-time communication smooth, simple and cheap. In the days ahead it is sure to have a big impact in the world of real-time communication.