WebRTC – Introduction

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WebRTC stands for Web Real Time Communication. It is an open source project to enable real-time communication of audio, video and data in Web and Native applications.

WebRTC API includes media capture, audio/video encoding/decoding, transportation layer and session management.

Media capture

  • Detect the type of devices available
  • Get user permission to access the devices
  • Manage the stream


  • Choosing the codec between the two users

Transportation layer

  • Manager order of packets
  • Deal with packet loss
  • Connecting with other users

Session management

  • Managing, opening and organising connections (Signalling)

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