You often need to add a real-time media communication feature directly between the browser and devices in your application. For instance, you might be developing a new audio/video call application or adding it as a feature to an existing application.
Before 2011, you would have to use plugins or other means to implement this. But now, you can do this using WebRTC.
Web Real-Time Communication (WebRTC) helps achieve audio and video streaming inside the web pages by allowing direct peer-to-peer communication. It can also exchange arbitrary data between browsers without requiring an intermediary. Although internally, it is still using C/C++, WebRTC provides a JavaScript API that you can use.
The set of standards that makes up the WebRTC allows to share data and perform peer-to-peer teleconferencing without the need of installing additional plugins.
There are various reasons why you should use WebRTC for real-time communication. Let’s take a look at some of them.