WebRTC samples

This is a collection of small samples demonstrating various parts of the WebRTC APIs. The code for all samples are available in the GitHub repository.

Most of the samples use adapter.js, a shim to insulate apps from spec changes and prefix differences.

https://webrtc.org/getting-started/testing lists command line flags useful for development and testing with Chrome.

Patches and issues welcome! See CONTRIBUTING.md for instructions.

Warning: It is highly recommended to use headphones when testing these samples, as it will otherwise risk loud audio feedback on your system.

getUserMedia():

Access media devices

Devices:

Query media devices

Stream capture:

Stream from canvas or video elements

RTCPeerConnection:

Controlling peer connectivity

RTCDataChannel:

Send arbitrary data over peer connections

Video chat:

Full featured WebRTC application

Insertable Streams:

API for processing media