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

• Basic getUserMedia demo
• Use getUserMedia with canvas
• Use getUserMedia with canvas and CSS filters
• Choose camera resolution
• Audio-only getUserMedia() output to local audio element
• Audio-only getUserMedia() displaying volume
• Record stream
• Screensharing with getDisplayMedia
• Control camera pan, tilt, and zoom
• Control exposure

Devices:

Query media devices

• Choose camera, microphone and speaker
• Choose media source and audio output

Stream capture:

Stream from canvas or video elements

• Stream from a video element to a video element
• Stream from a video element to a peer connection
• Stream from a canvas element to a video element
• Stream from a canvas element to a peer connection
• Record a stream from a canvas element
• Guiding video encoding with content hints

RTCPeerConnection:

Controlling peer connectivity

• Basic peer connection demo in a single tab
• Basic peer connection demo between two tabs
• Peer connection using Perfect Negotiation
• Audio-only peer connection demo
• Change bandwidth on the fly
• Change codecs before the call
• Upgrade a call and turn video on
• Multiple peer connections at once
• Forward the output of one PC into another
• Munge SDP parameters
• Use pranswer when setting up a peer connection
• Constraints and stats
• More constraints and stats
• RTCPeerConnection and requestVideoFrameCallback()
• Display createOffer output for various scenarios
• Use RTCDTMFSender
• Display peer connection states
• ICE candidate gathering from STUN/TURN servers
• Do an ICE restart
• Web Audio output as input to peer connection
• Peer connection as input to Web Audio
• Measure how long renegotiation takes
• Choose scalablilityMode before call - Scalable Video Coding (SVC) Extension

RTCDataChannel:

Send arbitrary data over peer connections

• Transmit text
• Transfer a file
• Transfer data
• Messaging

Video chat:

Full featured WebRTC application

• AppRTC video chat client that you can run out of a Docker image

Insertable Streams:

API for processing media

• End to end encryption using WebRTC Insertable Streams
(Experimental)
• Video analyzer using WebRTC Insertable Streams
(Experimental)
• Video processing using MediaStream Insertable Streams
(Experimental)
• Audio processing using MediaStream Insertable Streams
(Experimental)
• Video cropping using MediaStream Insertable Streams in a Worker
(Experimental)
• Integrations with WebGPU for custom video rendering:
(Experimental)