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/
Live Demo: https://www.webrtc-experiment.com/DetectRTC/
Github (open sourced): https://github.com/muaz-khan/DetectRTC
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
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
Live Demo: https://www.webrtc-experiment.com/Translator/
Github (open-sourced): https://github.com/muaz-khan/Translator
Live Demo: https://www.webrtc-experiment.com/getStats/
Github (open-sourced): https://github.com/muaz-khan/getStats
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.