github.com/muaz-khan/webrtc-experiment

WebRTC Demos, Experiments, Libraries, Examples

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RecordRTC | WebRTC Audio+Video+Screen Recording

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.

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MultiStreamsMixer

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

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DetectRTC | Is WebRTC Supported In Your Browser?

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

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RTCMultiConnection

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

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WebRTC Scalable Broadcasting

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

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WebRTC Dashboard | Canvas2D Drawing Tool

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

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WebRTC Voice & Text Translator

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

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getStats | Get WebRTC Peer Connection Stats

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

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FileBufferReader | File Sharing

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

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WebRTC Video Conferencing Demos

• Simple Demo: https://rtcmulticonnection.herokuapp.com/demos/Video-Conferencing.html

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WebRTC File Sharing

• Advance file sharing demo: https://rtcmulticonnection.herokuapp.com/demos/file-sharing.html

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WebRTC Screen Sharing

• P2P Screen Sharing: https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/
• Simple getDisplayMedia: https://www.webrtc-experiment.com/getDisplayMedia/

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Ffmpeg.js demos, both for browsers and node.js

• https://github.com/muaz-khan/Ffmpeg.js

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XHR-Signaling

XHR/XMLHttpRequest based WebRTC signaling implementation.

Github (open-sourced): https://github.com/muaz-khan/XHR-Signaling

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ASP.NET MVC based WebRTC Demo

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

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WebSync-Signaling

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

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Server Sent Events (SSE) over PHP

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

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SignalR

SignalR project for RTCMultiConnection: https://github.com/muaz-khan/RTCMultiConnection-SignalR

• https://github.com/muaz-khan/WebRTC-Experiment/blob/master/Signaling.md#how-to-use-signalr-for-signaling

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License

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