webrtc-samples

    1.0.0 • Public • Published

    Build Status

    WebRTC code samples

    This is a repository for the WebRTC Javascript code samples.

    Some of the samples use new browser features. They may only work in Chrome Canary and/or Firefox Beta, and may require flags to be set.

    All of the samples use adapter.js, 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, 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 + canvas

    getUserMedia + canvas + CSS Filters

    getUserMedia with resolution constraints

    getUserMedia with camera, mic and speaker selection

    Audio-only getUserMedia output to local audio element

    Audio-only getUserMedia displaying volume

    Face tracking

    Record stream

    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

    Use pranswer when setting up a peer connection

    Adjust constraints, view stats

    Display createOffer output

    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

    RTCDataChannel

    Transmit text

    Transfer a file

    Transfer data

    Video chat

    AppRTC video chat client powered by Google App Engine

    AppRTC URL parameters

    Install

    npm i webrtc-samples

    DownloadsWeekly Downloads

    0

    Version

    1.0.0

    License

    BSD-3-Clause

    Last publish

    Collaborators

    • kaptenjansson