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.)
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://rtcmulticonnection.herokuapp.com/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
- Advance file sharing demo: https://rtcmulticonnection.herokuapp.com/demos/file-sharing.html
- P2P Screen Sharing: https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/
- Simple getDisplayMedia: https://www.webrtc-experiment.com/getDisplayMedia/
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
All WebRTC Experiments are released under MIT license . Copyright (c) Muaz Khan.