WebRTC Demos, Experiments, Libraries, Examples

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.

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

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

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

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

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

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

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

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

WebRTC Video Conferencing Demos

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

WebRTC File Sharing

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

WebRTC Screen Sharing

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

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

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

XHR-Signaling

XHR/XMLHttpRequest based WebRTC signaling implementation.

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

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

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

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

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

Disclaimer

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.

https://www.webrtc-experiment.com/disclaimer/

License

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

zideqile / webrtc-experiment Goto Github PK

webrtc-experiment's Introduction