Github: https://github.com/muaz-khan/WebRTC-Experiment
WebRTC JavaScript library for audio/video as well as screen activity recording. It supports Chrome, Firefox, Opera, Android, and Microsoft Edge. Platforms: Linux, Mac and Windows.
Live Demo: https://www.webrtc-experiment.com/RecordRTC/
Github (open sourced): https://github.com/muaz-khan/RecordRTC
RecordRTC extension is available in the Chrome Web Store.
Pass multiple streams (e.g. screen+camera or multiple-cameras) and get single stream.
Live Demo: https://www.webrtc-experiment.com/MultiStreamsMixer/
Github: https://github.com/muaz-khan/MultiStreamsMixer
A tiny JavaScript library that can be used to detect WebRTC features e.g. system having speakers, microphone or webcam, screen capturing is supported, number of audio/video devices etc.
Live Demo: https://www.webrtc-experiment.com/DetectRTC/
Github (open sourced): https://github.com/muaz-khan/DetectRTC
WebRTC JavaScript library for peer-to-peer applications (screen sharing, audio/video conferencing, file sharing, media streaming etc.)
Demos: https://muazkhan.com:9001/demos/
Github: https://github.com/muaz-khan/RTCMultiConnection
Socket.io signaling server: https://github.com/muaz-khan/RTCMultiConnection-Server
This module simply initializes socket.io and configures it in a way that single broadcast can be relayed over unlimited users without any bandwidth/CPU usage issues. Everything happens peer-to-peer!
Live Demo: https://muazkhan.com:9001/demos/Scalable-Broadcast.html
Github (open sourced): https://github.com/muaz-khan/WebRTC-Scalable-Broadcast
Collaborative, extendable, JavaScript Canvas2D drawing tool, supports dozens of builtin tools, as well as generates JavaScript code for 2D animations.
Live Demo: https://www.webrtc-experiment.com/Canvas-Designer/
Github (open-sourced): https://github.com/muaz-khan/Canvas-Designer
You video presentation: https://www.youtube.com/watch?v=pvAj5l_v3cM
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!
Live Demo: https://www.webrtc-experiment.com/Translator/
Github (open-sourced): https://github.com/muaz-khan/Translator
A tiny JavaScript library using WebRTC getStats API to return peer connection stats i.e. bandwidth usage, packets lost, local/remote ip addresses and ports, type of connection etc.
Live Demo: https://www.webrtc-experiment.com/getStats/
Github (open-sourced): https://github.com/muaz-khan/getStats
FileBufferReader is a JavaScript library reads file and returns chunkified array-buffers. The resulting buffers can be shared using WebRTC data channels or socket.io.
Live Demo: https://www.webrtc-experiment.com/FileBufferReader/
Github (open-sourced): https://github.com/muaz-khan/FileBufferReader
Youtube video presentation: https://www.youtube.com/watch?v=gv8xpdGdS4o
XHR/XMLHttpRequest based WebRTC signaling implementation.
Github (open-sourced): https://github.com/muaz-khan/XHR-Signaling
A simple WebRTC one-to-one demo written in September, 2012! It supports public rooms as well as password-protected private rooms! MS-SQL database is used as signaling gateway!
Github (open-sourced): https://github.com/muaz-khan/WebRTC-ASPNET-MVC
WebSync is used as signaling gateway with/for WebRTC-Experiments e.g. RTCMultiConnection.js, DataChannel.js, Plugin-free screen sharing, and video conferencing.
Github (open-sourced): https://github.com/muaz-khan/WebSync-Signaling
Server Sent Events (SSE) are used to setup WebRTC peer-to-peer connections.
Github (open-sourced): https://github.com/muaz-khan/RTCMultiConnection/tree/master/demos/SSEConnection
SignalR project for RTCMultiConnection: https://github.com/muaz-khan/RTCMultiConnection-SignalR
There is no warranty, expressed or implied, associated with thse experiments. Use at your own risk. I'm not responsible to maintain any of these experiments. So please fix bugs yourselves or simply do not use them.
All WebRTC Experiments are released under MIT license . Copyright (c) Muaz Khan.