Skip to content

WebRTC Demos & WebRTC Experiments. Audio/Video/Screen Hangout & Conferencing; Audio/Video Recording; Screen/Data/and FileSharing; Media Servers & Signaling e.g. XMPP, SIP, WebSockets, Socket.io, WebSycn, SignalR etc. Translator.js; ffmpeg.js; RTCMultiConnection.js; RecordRTC.js; MediaStreamRecorder.js; getMediaElement.js; File.js; DataChannel.js…

License

Notifications You must be signed in to change notification settings

rayj00/WebRTC-Experiment

 
 

Folders and files

NameName
Last commit message
Last commit date

Latest commit

 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 

Repository files navigation

Realtime/Working WebRTC Experiments

  1. It is a repository of uniquely experimented WebRTC demos; written by Muaz Khan!
  2. No special requirement! Just WebRTC supported browser (e.g. chrome/firefox/opera on desktop/android)
  3. These demos/experiments are entirely client-side; i.e. no server installation needed!

=

Libraries

Library Name Short Description Documentation Demos
RecordRTC.js A library for audio/video recording Documentation Demos
RTCMultiConnection.js An ultimate wrapper library for RTCWeb APIs Documentation Demos
DataChannel.js An ultimate wrapper library for RTCDataChannel APIs Documentation Demos
SdpSerializer.js An easiest way to modify SDP Documentation Demos
RTCall.js A library for voice (i.e. audio-only) calls Documentation Demos
Meeting.js A library for audio/video conferencing Documentation Demos
File.js A standalone library for file sharing functionalities Documentation Demos
getMediaElement.js A library for audio/video media elements' layout Documentation Demos
Translator.js Voice & Text Translator Documentation Demos

=

Important Experiments
Experiment Name Short Description Source Code Demo
Pre-recorded Media Streaming Stream video files in realtime; same like webcam streaming! Source Demo
Part of Screen Sharing Share a region of the screen; not the entire screen! Source Demo
Plugin-free Screen Sharing Share the entire screen Source Demo
One-Way Broadcasting Same like radio stations; transmit audio/video/screen streams in one-way direction. Though, it is browser-to-browser streaming! Source Demo
Audio-only Calls Realtime, plugin-free, voice-only calls Source Demo

=

Useful Experiments
Experiment Name Previous Demos New Demos
video-conferencing / multi-user (group) video sharing Demo / Source Demo / Source Code
file sharing / multi-user (group) files hangout Demo / Source Demo / Source Code
file sharing using SCTP data channels Demo / -- -- / Source Code
text chat / multi-user (group) text chat Demo / Source Demo / Source Code
MultiRTC Demo / -- -- / Source Code

=

One-to-Many style of WebRTC Experiments
Experiment Name Previous Demos New Demos
video-broadcasting Demo / Source Demo / Source Code
audio-broadcasting Demo / Source Demo / Source Code

=

One-to-One style of WebRTC Experiments
Experiment Name Demo Source Code
One-to-one WebRTC video chat using WebSocket Demo Source
One-to-one WebRTC video chat using socket.io Demo Source

=

Single-Page / One-Page / Client Side
Experiment Name Demo Source Code
Switch streams from screen-sharing to audio+video. (Renegotiation) Demo Source
Share screen and audio/video from single peer connection! Demo Source
Text chat using RTCDataChannel APIs Demo Source
Simple video chat Demo Source
Sharing video - using socket.io for signaling Demo Source
Sharing video - using WebSockets for signaling Demo Source
Audio Only Streaming Demo Source
MediaStreamTrack.getSources Demo Source

=

Experiments to share tab/screen/desktop
Experiment Name Previous Demos New Demos
Plugin-free screen sharing / share the entire screen Demo / Source Demo / Source Code
Desktop sharing / using desktopCapture APIs Demo / Source --
Tab sharing / using tabCapture APIs Demo / Source --

=

Experiments to share region/part of the screen
Experiment Name Demo Source Code
Share part-of-screen using RTCDataChannel APIs Demo Source
Share part-of-screen using Firebase Demo Source
A realtime chat using RTCDataChannel Demo Source
A realtime chat using Firebase Demo Source

=

Demos using MediaStreamRecorder.js library
Experiment Name Demo Source Code
Audio Recording Demo Source
Video/Gif Recording Demo Source

=

Demos using DataChannel.js library
Experiment Name Demo Source Code
DataChannel basic demo Demo Source
Auto Session Establishment Demo Source
Share part-of-screen using DataChannel.js Demo Source
Private Chat Demo ----

=

Experimental (Non-Functional)
Experiment Name Demo Source Code
Attaching Remote Media Streams Demo Source
mozCaptureStreamUntilEnded for pre-recorded media streaming Demo Source
Remote audio stream recording Demo Source

=

Demos using RTCMultiConnection
Experiment Name Demo Source Code
All-in-One test Demo Source
Renegotiation & Mute/UnMute/Stop Demo Source
Multi-streams attachment Demo Source
Admin/Guest audio/video calling Demo Source
Session-Reinitiation Demo Source
Audio/Video Recording Demo Source
Mute/UnMute Demo Source
Password Protected Rooms Demo Source
WebRTC remote media stream forwarding Demo Source
Video Conferencing Demo Source
Multi-Session Establishment Demo Source
RTCMultiConnection-v1.3 testing demo Demo Source
Video Broadcasting Demo Source
File Sharing + Text Chat Demo Source
Audio Conferencing Demo Source
Join with/without camera Demo Source
Screen Sharing Demo Source
One-to-One file sharing Demo Source
Manual session establishment + extra data transmission + video conferencing Demo Source
Customizing Bandwidth Demo Source
Users ejection and presence detection Demo Source
RTCMultiConnection-v1.3 and socket.io ---- Source

=

A few documents for newbies and beginners
RTCMultiConnection Documentation
DataChannel Documentation
RTCPeerConnection Documentation
How to use RTCPeerConnection.js?
RTCDataChannel for Beginners
How to use RTCDataChannel? - single code for both canary and nightly
WebRTC for Beginners: A getting stared guide!
WebRTC for Newbies
How to switch streams?
How to echo cancellation? / Noise management?
STUN or TURN? Which one to prefer; and why?
WebRTC RTP Usage
webrtcpedia!
Are you want to learn WebRTC?

=

  1. Transcoding WAV into Ogg
  2. Transcoding WebM into mp4
  3. Transcoding WebM into mp4; then merging WAV+mp4 into single mp4

=

Custom Signaling

  1. Socket.io over Node.js
  2. WebSocket over Node.js
  3. WebSync / ASP.NET MVC

=

How to record audio using RecordRTC?
<script src="//www.webrtc-experiment.com/RecordRTC.js"></script>
var recordRTC = RecordRTC(mediaStream);

recordRTC.startRecording();
recordRTC.stopRecording();

var blob = recordRTC.getBlob();
var blobURL = recordRTC.toURL();

recordRTC.getDataURL(function(dataURL) {});
  1. RecordRTC to Node.js
  2. RecordRTC to PHP
  3. RecordRTC to ASP.NET MVC
  4. RecordRTC & HTML-2-Canvas i.e. Canvas/HTML Recording!
  5. MRecordRTC i.e. Multi-RecordRTC!
  6. RecordRTC on Ruby!
  7. RecordRTC over Socket.io
  8. ffmpeg-asm.js and RecordRTC! Audio/Video Merging & Transcoding!
  9. Recording Audio+Video in single WebM on Firefox

=

You can write entire skype-like web-app using RTCMultiConnection! It supports all complex renegotiation scenarios!

<button id="openNewSessionButton">open New Session Button</button><br />

<script src="http://www.RTCMultiConnection.org/latest.js"> </script>
<script>
var connection = new RTCMultiConnection().connect();
document.getElementById('openNewSessionButton').onclick = function() {
    connection.open();
};
</script>

RTCMultiConnection Documentation

=

DataChannel.js / A library for RTCDataChannel APIs
<script src="//www.webrtc-experiment.com/DataChannel.js"> </script>
<script>
    var channel = new DataChannel();
    channel.onopen = function(userid) {};
    channel.onmessage = function(message) {};
	
    // search for existing channels
    channel.connect();

    document.getElementById('new-channel').onclick = function() {
        channel.open(); // setup new channel
    };
</script>

DataChannel Documentation

=

Translator.js is a JavaScript library built top on Google Speech-Recognition & Translation API to transcript and translate voice and text. It supports many locales and brings globalization in WebRTC!

<script src="//www.webrtc-experiment.com/Translator.js"> </script>
var translator = new Translator();

translator.voiceToText(function (text) {
    console.log('Your voice as text!', text);
}, 'your-language');

translator.translateLanguage(textToConvert, {
    from: 'language-of-the-text',
    to: 'convert-into',
    callback: function (translatedText) {
        console.log('translated text', translatedText);
    }
});

translator.speakTextUsingRobot(textToPlay);

translator.speakTextUsingGoogleSpeaker({
    textToSpeak: 'text-to-convert',
    targetLanguage: 'your-language'
});

=

openSignalingChannel for RTCMultiConnection.js and DataChanel.js (Client-Side Code)
var channels = {};
var currentUserUUID = Math.round(Math.random() * 60535) + 5000;
var socketio = io.connect('http://localhost:8888/');

socketio.on('message', function(data) {
    if(data.sender == currentUserUUID) return;
    
    if (channels[data.channel] && channels[data.channel].onmessage) {
        channels[data.channel].onmessage(data.message);
    };
});

connection.openSignalingChannel = function (config) {
    var channel = config.channel || this.channel;
    channels[channel] = config;

    if (config.onopen) setTimeout(config.onopen, 1000);
    return {
        send: function (message) {
            socketio.emit('message', {
                sender: currentUserUUID,
                channel: channel,
                message: message
            });
        },
        channel: channel
    };
};

=

Nodejs/Socketio Server-Side Code
io.sockets.on('connection', function (socket) {
    socket.on('message', function (data) {
        socket.broadcast.emit('message', data);
    });
});

Read more here.

=

Browser Support

WebRTC Experiments works fine on following web-browsers:

Browser Support
Firefox Stable / Aurora / Nightly
Google Chrome Stable / Canary / Beta / Dev
Opera Stable / NEXT
Android Chrome / Firefox

=

=

Want to Donate?

https://www.webrtc-experiment.com/donate/

=

License

All WebRTC Experiments are released under MIT licence . Copyright (c) Muaz Khan.

About

WebRTC Demos & WebRTC Experiments. Audio/Video/Screen Hangout & Conferencing; Audio/Video Recording; Screen/Data/and FileSharing; Media Servers & Signaling e.g. XMPP, SIP, WebSockets, Socket.io, WebSycn, SignalR etc. Translator.js; ffmpeg.js; RTCMultiConnection.js; RecordRTC.js; MediaStreamRecorder.js; getMediaElement.js; File.js; DataChannel.js…

Resources

License

Stars

Watchers

Forks

Releases

No releases published

Packages

No packages published