Understanding Real time Communication with WebRTC

With WebRTC, you can add real-time communication capabilities to your application.
Web Real-Time Communications (WebRTC) is an open source project created by Google to enable peer-to-peer communication in web browsers and mobile applications through application programming interfaces.
It supports video, voice, and generic data to be sent between peers, allowing to build powerful voice and video-communication communication.

Why WebRTC:

  • It is completely free for personal or commercial use.
  • WebRTC is available in all modern browsers
  • WebRTC works in mobile application also, SDKs are available to integrate.
  • It is very powerful and versatile and can be used for audio, video, group calling, recording and many more.
  • No plugin required
  • Is it secure! WebRTC uses a secure WebSocket to establish a UDP connection between the server and the client
  • WebRTC is currently the only protocol that provides sub-500 milliseconds of real-time latency.

Who all are using WebRTC:

We will list some of the massive application those who are already using WebRTC

  • Google Meet
  • Google Hangout
  • Facebook Messenger
  • Amazon Chime
  • Discord

Brower Support:

WebRTC is compatible with Firefox, Opera, and Chrome: desktop and mobile

Community:

WebRTC have great community, here are some of the links

https://webrtc.org/support/overview
https://groups.google.com/g/discuss-webrtc?pli=1
https://dev.to/t/webrtc

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