Introduction
WebRTC is a modern web application where you can easily stream audio, video and share files with people through web browsers and mobile applications.
In this blog, I will give you a short description of WebRTC. Let's start...
A short history of WebRTC
Gmail video chat become more popular in 2008, and in 2011 Google introduced Hangouts, which uses the Google Talk service. Google bought a company that name is GIPS, which had developed many components required for RTC( Real-Time Communication), such as codecs and echo cancellation techniques. Google open-sourced the technologies developed by GIPS and engaged with relevant standards bodies at the IETF and W3C to ensure industry consensus. In May 2011, Ericsson built the first implementation of WebRTC.
WebRTC is used in various apps like WhatsApp, Facebook Messenger, etc.
WebRTC APIs
WebRTC mainly works on three APIs:
- MediaStream
- RTCPeerConnection
- RTCDataChannel
All the above three APIs supported in mobile and desktop by Chrome, Safari, Firefox, Edge, and Opera.
RTCPeerConnection
RTCPeerConnection is the webRTC component that handles stable and efficient communication of streaming data between peer connection.
The main work of the RTCPeerConnection object is to set up and create a peer connection.
View the demo and source code form the below link:
https://webrtc.github.io/samples/src/content/peerconnection/pc1/
RTCDataChannel
RTCDataChannel API makes peer-to-peer exchange of arbitrary data, with low latency and high throughput.
View the demo and source code from the below link:
https://webrtc.github.io/samples/#datachannel
RTCDataChannel is available in Chrome, Safari, Firefox, Opera, and Samsung Internet
Conclusion
In this blog, I covered the Web Real-Time Communication(WebRTC) in short details. If you want to learn more about the webRTC, then go to the below link and get more information. Thanks for reading!