WebRTC Experiments
Realtime/Working- It is a repository of uniquely experimented WebRTC demos; written by Muaz Khan!
- No special requirement! Just WebRTC compatible web-browser (e.g. chrome/firefox/opera on desktop/android)
- These demos/experiments are entirely client-side; i.e. no server installation needed!
- You can use all these demos in PHP/Python/Ruby/ASP.NET/etc. everywhere!
How to use?
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 :)
Libraries
Library Name | Short Description | Documentation | Demos |
---|---|---|---|
RecordRTC.js |
Supports cross-browser audio/video recordings! | Documentation | Demos |
RTCMultiConnection.js |
Single Library for Everything! Just imagine :) | Documentation | Demos |
getScreenId.js |
Single chrome extension for all domains! Again, imagine :) | Documentation | Demos |
Conversation.js |
Enjoy Skype-like Conversations! Oops :) | Documentation | Demos |
DataChannel.js |
Supports data-streaming among multiple peers | 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 |
Other Repositories
- PluginRTC: IE/Safari Plugins compatible WebRTC-Experiments
- RecordRTC.js
- RTCMultiConnection.js
- Conversation.js
- Collaborate Canvas Designer
- XHR-Signaling
- ASP.NET MVC based WebRTC 1:1 Demo
- WebSync Signaling
- SdpSerializer.js
Experiments
ImportantExperiment 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 |
Experiments
UsefulExperiment 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 |
Google Chrome Extensions for WebRTC!
- desktopCapture API / Install App Store Extension
- tabCapture API / Install App Store Extension
- Desktop Sharing / Install App Store Extension
Experiments
One-to-Many style of WebRTCExperiment Name | Previous Demos | New Demos |
---|---|---|
video-broadcasting | Demo / Source | Demo / Source Code |
audio-broadcasting | Demo / Source | Demo / Source Code |
One-to-One Calls
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 | Source | Demo |
WebRTC 1-1 Calls | Source | Demo |
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 | -- |
share region/part of the screen
Experiments toExperiment 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 |
Demos using MediaStreamRecorder.js library
Experiment Name | Demo | Source Code |
---|---|---|
Audio Recording | Demo | Source |
Video Recording | Demo | Source |
Gif Recording | Demo | Source |
DataChannel.js library
Demos usingExperiment 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 | ---- |
Text Chat using Pusher and DataChannel.js | Demo | Source |
Experimental (Non-Functional)
Experiment Name | Demo | Source Code |
---|---|---|
Attaching Remote Audio Streams | Demo | Source |
mozCaptureStreamUntilEnded for pre-recorded media streaming | Demo | Source |
Remote audio stream recording | Demo | Source |
Demos using RTCMultiConnection
- AppRTC like RTCMultiConnection demo!
- MultiRTC! RTCMultiConnection all-in-one demo!
- Collaborative Canvas Designer
- All-in-One test
- Multi-Broadcasters and Many Viewers
- Select Broadcaster at runtime
- OneWay Screen & Two-Way Audio
- Stream Mp3 Live
- Socket.io auto Open/Join rooms
- navigator.getMediaDevices / navigator.enumerateDevices / MediaStreamTrack.getSources
- Renegotiation & Mute/UnMute/Stop
- Video-Conferencing
- Video Broadcasting
- Audio Conferencing
- Multi-streams attachment
- Admin/Guest audio/video calling
- Session Re-initiation Test
- Preview Screenshot of the room
- RecordRTC & RTCMultiConnection
- Explains how to customize ice servers; and resolutions
- Mute/Unmute and onmute/onunmute
- One-page demo: Explains how to skip external signalling gateways
- Password Protect Rooms: Explains how to authenticate users
- Session Management: Explains difference between "leave" and "close" methods
- Multi-Sessions Management
- RTCMultiConnection-v1.3 test
- Customizing Bandwidth
- Users ejection and presence detection
- Multi-Session Establishment
- File Sharing + Text Chat
- Audio Conferencing + File Sharing + Text Chat
- Join with/without camera
- Screen Sharing
- One-to-One file sharing
- Manual session establishment + extra data transmission
- Manual session establishment + extra data transmission + video conferencing
Demos using Conversation.js
- 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>
Documents for newcomers/newbies/beginners
A few documents for newbies and beginners |
---|
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 |
- http://muaz-khan.blogspot.com/search/label/WebRTC
- https://www.webrtc-experiment.com/#documentations
- https://www.facebook.com/WebRTC
- https://plus.google.com/+WebRTC-Experiment/posts
=
ffmpeg-asm.js Demos
- Transcoding WAV into Ogg / Source Code
- Transcoding WebM into mp4 / Source Code
- Transcoding WebM into mp4; then merging WAV+mp4 into single mp4 / Source Code
- Recording Audio+Canvas and merging in single mp4 / Source Code
=
Custom Signaling
- Socket.io over Node.js
- WebSocket over Node.js
- WebSync / ASP.NET MVC
- XHR Signaling
- openSignalingChannel
RecordRTC?
How to record audio using<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);
- RecordRTC to Node.js
- RecordRTC to PHP
- RecordRTC to ASP.NET MVC
- RecordRTC & HTML-2-Canvas i.e. Canvas/HTML Recording!
- MRecordRTC i.e. Multi-RecordRTC!
- RecordRTC on Ruby!
- RecordRTC over Socket.io
- ffmpeg-asm.js and RecordRTC! Audio/Video Merging & Transcoding!
- Recording Audio+Video in single WebM on Firefox
- RecordRTC / PHP / FFmpeg
RTCMultiConnection.js
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
DataChannel.js / A library for RTCDataChannel APIs
<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>
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'
});
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.getUserMedia = navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
navigator.getUserMedia(screen_constraints, function (stream) {
document.querySelector('video').src = URL.createObjectURL(stream);
}, function (error) {
console.error(error);
});
});
</script>
Signaling?
=
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 |
=
Muaz Khan
- Personal Webpage � http://www.muazkhan.com
- Email � muazkh@gmail.com
- Twitter � https://twitter.com/muazkh and https://twitter.com/WebRTCWeb
=
License
All WebRTC Experiments are released under MIT licence . Copyright (c) Muaz Khan.