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!

Each demo has a unique directory. Simply download that directory, upload in your webserver and use it; and it'll work!

You don't need to modify any single line to use it. No single installation or modification is needed :)

1. PluginRTC: IE/Safari Plugins compatible WebRTC-Experiments
2. RecordRTC.js
3. RTCMultiConnection.js
4. Conversation.js
5. Collaborate Canvas Designer
6. XHR-Signaling
7. ASP.NET MVC based WebRTC 1:1 Demo
8. WebSync Signaling
9. SdpSerializer.js

1. desktopCapture API / Install App Store Extension
2. tabCapture API / Install App Store Extension
3. Desktop Sharing / Install App Store Extension

1. AppRTC like RTCMultiConnection demo!
2. MultiRTC! RTCMultiConnection all-in-one demo!
3. Collaborative Canvas Designer
4. All-in-One test
5. Multi-Broadcasters and Many Viewers
6. Select Broadcaster at runtime
7. OneWay Screen & Two-Way Audio
8. Stream Mp3 Live
9. Renegotiation & Mute/UnMute/Stop
10. Video-Conferencing
11. Video Broadcasting
12. Audio Conferencing
13. Multi-streams attachment
14. Admin/Guest audio/video calling
15. Session Re-initiation Test
16. Preview Screenshot of the room
17. RecordRTC & RTCMultiConnection
18. Explains how to customize ice servers; and resolutions
19. Mute/Unmute and onmute/onunmute
20. One-page demo: Explains how to skip external signalling gateways
21. Password Protect Rooms: Explains how to authenticate users
22. Session Management: Explains difference between "leave" and "close" methods

            <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>

1. AndroidRTC

            <li>
				<a href="https://www.webrtc-experiment.com/Conversationjs/search-user.html">Search Users</a>
            </li>
            
            <li>
				<a href="https://www.webrtc-experiment.com/Conversationjs/cross-language-chat.html">Cross-Language (Multi-Lingual) Text Chat</a>
            </li>
            
            <li>
                <a href="https://www.rtcmulticonnection.org/conversationjs/demos/">Old Conversation.js demos</a>
            </li>

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

<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

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

<button id="openRoom">Open Room</button>
<button id="joinRoom">Join Room</button><br />

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

RTCMultiConnection Documentation

<script src="//cdn.webrtc-experiment.com/DataChannel.js"> </script>
<script>
    var datachannel = new DataChannel();
    datachannel.onopen = function(remoteUserid) {};
    datachannel.onmessage = function(message, remoteUserid) {};
	
    // search for existing channels
    datachannel.connect();

    document.getElementById('new-channel').onclick = function() {
        datachannel.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="//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'
});

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

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

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WebRTC Experiments works fine on following web-browsers:

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1. Personal Webpage � https://www.muazkhan.com
2. Email � [email protected]
3. Twitter � https://twitter.com/muazkh and https://twitter.com/WebRTCWeb

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