WebRTC code samples

This is a repository for the WebRTC Javascript code samples.

It is originally a fork of https://github.com/webrtc/samples, but updated to integrate the Temasys Plugin, and work on Internet Explorer and Safari.

Some of the samples use new browser features. They may only work in Chrome Canary, Firefox Beta, Microsoft Edge (available with Windows 10), and latest versions of the Temasys Plugin, and may require flags to be set.

All of the samples use AdapterJS, a shim to insulate apps from spec changes and prefix differences. In fact, the standards and protocols used for WebRTC implementations are highly stable, and there are only a few prefixed names. For full interop information, see webrtc.org/web-apis/interop.

In Chrome and Opera, all samples that use getUserMedia() must be run from a server. Calling getUserMedia() from a file:// URL will work in Firefox and the Temasys Plugin, but fail silently in Chrome and Opera.

webrtc.org/testing lists command line flags useful for development and testing with Chrome.

For more information about WebRTC, we maintain a list of WebRTC Resources. If you've never worked with WebRTC, we recommend you start with the 2013 Google I/O WebRTC presentation.

Patches and issues welcome! See CONTRIBUTING for instructions. All contributors must sign a contributor license agreement before code can be accepted. Please complete the agreement for an individual or a corporation as appropriate. The Developer's Guide for this repo has more information about code style, structure and validation. Head over to test/README.md and get started developing.

The demos

getUserMedia

Basic getUserMedia demo

getUserMedia with resolution constraints

getUserMedia with camera, mic and speaker selection

Audio-only getUserMedia output to local audio element

getUserMedia in an iFrame

[Screensharing]](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/screensharing)

Devices

Select camera, microphone and speaker

Select media source and audio output

RTCPeerConnection

Basic peer connection

Audio-only peer connection

Multiple peer connections at once

Forward output of one peer connection into another

Munge SDP parameters

Adjust constraints, view stats

Use RTCDTMFSender

Display peer connection states

ICE candidate gathering from STUN/TURN servers

Do an ICE restart

RTCDataChannel

Transmit text

Transfer a file

Transfer data

ArrayBuffer sending