What is WebRTC?

Fri Oct 15 2021

2 min read

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.

Reasons for Using WebRTC

There are various reasons why you should use WebRTC for real-time communication. Let’s take a look at some of them.

  • Simple to use. Since WebRTC is used via a JavaScript API, you can easily integrate it into your application.
  • WebRTC is safe and secure. All browsers involved in an interaction must agree to proceed before communication starts.
  • It is an open-source project maintained by Google. So if you get stuck somewhere, you can seek help from the community on GitHub or StackOverflow.
  • WebRTC is supported in all modern web browsers. So you can build a streaming service without the need to worry about if the user will be able to use it or not.
  • WebRTC is also available to use in mobile applications. SDKs are available for both mobile and embedded environments.
  • It provides arbitrary data exchange support.
  • WebRTC is the only protocol that provides sub-500 milliseconds of real-time latency, thus making it the fastest protocol on the market.