/WebRTC-Experiment

WebRTC libraries, WebRTC demos, WebRTC experiments, audio, video, screen, conferencing, file sharing, screen sharing, recording, MCU, media stacks, media servers, signaling, SIP, XMPP, XHR, websockets, socket.io, websync, signalR, Translator.js, RecordRTC.js, ffmpeg.js, RTCMultiConnection.js, DataChannel.js, DetectRTC, Meeting.js, MediaRecorder, MediaStreamRecorder, SDP-Serializer, captureStreamUntilEnded, pre-recorded, media streaming, live streaming, mp3 live, desktop sharing, part of screen sharing and everything regarding webrtc i.e. realtime web!

Primary LanguageJavaScriptOtherNOASSERTION

Realtime/Working WebRTC Experiments

  1. It is a repository of uniquely experimented WebRTC demos; written by Muaz Khan!
  2. No special requirement! Just WebRTC compatible web-browser (e.g. chrome/firefox/opera on desktop/android)
  3. These demos/experiments are entirely client-side; i.e. no server installation needed!
  4. You can use all these demos in PHP/Python/Ruby/ASP.NET/etc. everywhere!

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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
getScreenId.js Single chrome extension for all domains! Documentation Demos
Conversation.js Enjoy Skype-like Conversations! 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
DetectRTC.js A library for detecting WebRTC features Documentation Demos
navigator.customGetUserMediaBar.js Keep your users Privacy! Documentation Demos

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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

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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

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Google Chrome Extensions for WebRTC!
  1. desktopCapture API / Install App Store Extension
  2. tabCapture API / Install App Store Extension
  3. Desktop Sharing / Install App Store Extension

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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

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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
WebRTC 1-1 Audio/Video/Screen Sharing Realtime, pluginfree! Source

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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

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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 --

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Experiments to share region/part of the screen
Experiment Name Demo Source Code
Share part-of-screen RTCMultiConnection Demo Source
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

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

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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 ----

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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

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Demos using RTCMultiConnection
  1. MultiRTC! RTCMultiConnection all-in-one demo!
  2. All-in-One test
  3. Multi-Broadcasters and Many Viewers
  4. OneWay Screen & Two-Way Audio
  5. Stream Mp3 Live
    6.             <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/remote-stream-forwarding.html">Remote Stream Forwarding</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/getMediaDevices.html">navigator.getMediaDevices AKA navigator.enumerateDevices</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/changeBandwidth.html">connection.changeBandwidth({ audio: 128, video: 256 })</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/RTCMultiConnection.sharePartOfScreen.html">connection.sharePartOfScreen({ element: 'body'})</a></li>
            
			<li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/Renegotiation.html">Renegotiation & Mute/UnMute/Stop</a></li>
			<li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/videoconferencing.html">Video-Conferencing</a></li>
			<li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/video-broadcasting.html">Video Broadcasting</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/audioconferencing.html">Audio Conferencing</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/multi-streams-attachment.html">Multi-streams attachment</a></li>
			<li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/admin-guest.html">Admin/Guest audio/video calling</a></li>
			<li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/session-reinitiation.html">Session Re-initiation Test</a></li>
			<li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/rooms-screenshots.html">Preview Screenshot of the room</a></li>
			<li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/RecordRTC-and-RTCMultiConnection.html">RecordRTC & RTCMultiConnection</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/features.html">Explains how to customize ice servers; and resolutions</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/mute-unmute.html">Mute/Unmute and onmute/onunmute</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/one-page-demo.html">One-page demo: Explains how to skip external signalling gateways</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/password-protect-rooms.html">Password Protect Rooms: Explains how to authenticate users</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/session-management.html">Session Management: Explains difference between "leave" and "close" methods</a></li>
            
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/multi-sessions-management.html">Multi-Sessions Management</a></li>
			
			<li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/RTCMultiConnection-v1.3-demo.html">RTCMultiConnection-v1.3 test</a></li>
			<li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/bandwidth.html">Customizing Bandwidth</a></li>
			<li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/users-ejection.html">Users ejection and presence detection</a></li>				
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/multi-session-establishment.html">Multi-Session Establishment</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/group-file-sharing-plus-text-chat.html">File Sharing + Text Chat</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/audio-conferencing-data-sharing.html">Audio Conferencing + File Sharing + Text Chat</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/join-with-or-without-camera.html">Join with/without camera</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/screen-sharing.html">Screen Sharing</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/one-to-one-filesharing.html">One-to-One file sharing</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission.html">Manual session establishment + extra data transmission</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html">Manual session establishment + extra data transmission + video conferencing</a></li>
            <li><a href="https://www.webrtc-experiment.com/RTCMultiConnection/chrome-to-firefox-screen-sharing.html">Chrome-to-Firefox Screen Sharing</a></li>

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Documents for newcomers/newbies/beginners
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?
WebRTC Tips & Tricks
  1. http://muaz-khan.blogspot.com/search/label/WebRTC
  2. https://www.webrtc-experiment.com/#documentations
  3. https://www.facebook.com/WebRTC
  4. https://plus.google.com/+WebRTC-Experiment/posts

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ffmpeg-asm.js Demos

  1. Transcoding WAV into Ogg / Source Code
  2. Transcoding WebM into mp4 / Source Code
  3. Transcoding WebM into mp4; then merging WAV+mp4 into single mp4 / Source Code
  4. Recording Audio+Canvas and merging in single mp4 / Source Code

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Custom Signaling

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

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

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

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

recordRTC.getDataURL(callback_function);
  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
  10. RecordRTC / PHP / FFmpeg

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RTCMultiConnection.js

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="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
<script>
var connection = new RTCMultiConnection().connect();
document.getElementById('openNewSessionButton').onclick = function() {
    connection.open();
};
</script>

RTCMultiConnection Documentation

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DataChannel.js / A library for RTCDataChannel APIs
<script src="//cdn.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

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Translator.js / Demo

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="//cdn.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'
});

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getScreenId.js / Demo

Simply use getScreenId.js and enjoy screen capturing from any domain. You don't need to deploy chrome extension yourself. You can refer your users to install this chrome extension instead. Also, getScreenId.js auto-fallbacks to command-line based screen capturing if chrome extension isn't installed or disabled. getScreenId.js throws clear exceptions which is helpful for end-user experiences.

Demo: https://www.webrtc-experiment.com/getScreenId/

<script src="//cdn.WebRTC-Experiment.com/getScreenId.js"></script>

<script>
getScreenId(function (error, sourceId, screen_constraints) {
    // error    == null || 'permission-denied' || 'not-installed' || 'installed-disabled' || 'not-chrome'
    // sourceId == null || 'string'

    navigator.webkitGetUserMedia(screen_constraints, function (stream) {
        document.querySelector('video').src = URL.createObjectURL(stream);
    }, function (error) {
        console.error(error);
    });
});
</script>

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openSignalingChannel for RTCMultiConnection.js and DataChanel.js (Client-Side Code)
var onMessageCallbacks = {};
var socketio = io.connect('http://localhost:8888/');

socketio.on('message', function(data) {
    if(data.sender == connection.userid) return;
    
    if (onMessageCallbacks[data.channel]) {
        onMessageCallbacks[data.channel](data.message);
    };
});

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

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

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

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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 / Opera

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Muaz Khan

  1. Personal Webpage � http://www.muazkhan.com
  2. Email � muazkh@gmail.com
  3. Twitter � https://twitter.com/muazkh and https://twitter.com/WebRTCWeb

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License

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