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WebRTC for Absolute beginners, an overview.

With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. It supports video, voice, and generic data to be sent between peers, allowing developers to build powerful voice- and video-communication solutions. The technology is available on all modern browsers as well as on native clients for all major platforms. The technologies behind WebRTC are implemented as an open web standard and available as regular JavaScript APIs in all major browsers. For native clients, like Android and iOS applications, a library is available that provides the same functionality. The WebRTC project is an open source project and supported by Apple, Google, Microsoft and Mozilla, amongst others.

What can WebRTC really do?

There are many different use-cases for WebRTC, from basic web apps that uses the camera or microphone, to more advanced video-calling applications and screen sharing. WebRTC can be used to build anything and everything starting from weekend side projects to build a simple one to one video chat app or build a complex enterprise grade video conferencing apps with security and other necessary features.

What exactly is WebRTC?

A WebRTC application usually goes through a common application flow. To make it simple, it can be understood in 4 steps, accessing the media devices, opening peer connections, discovering peers, and start streaming. It is a collection of a set of APIs which facilitate these above mentioned steps. Creating a new application based on the WebRTC technologies can be overwhelming if one is unfamiliar with these APIs.


The WebRTC standard covers, on a high level, two different technologies: media capture devices and peer-to-peer connectivity.

Media capture devices includes video cameras and microphones, but also screen capturing "devices". For cameras and microphones, we use navigator.mediaDevices.getUserMedia() to capture MediaStreams. For screen recording, we use navigator.mediaDevices.getDisplayMedia() instead.

The peer-to-peer connectivity is handled by the RTCPeerConnection interface. This is the central point for establishing and controlling the connection between two peers in WebRTC.

In the upcoming posts, the APIs will be elaborated with easy understand and follow examples. Stay tuned to learn WebRTC from scratch.



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