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

            <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:

• https://github.com/muaz-khan/WebRTC-Experiment/blob/master/Signaling.md

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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 � [email protected]
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.