Ecosyste.ms: Awesome
An open API service indexing awesome lists of open source software.
https://github.com/muaz-khan/WebRTC-Experiment
WebRTC, WebRTC and WebRTC. Everything here is all about WebRTC!!
https://github.com/muaz-khan/WebRTC-Experiment
webrtc webrtc-demos webrtc-examples webrtc-experiments webrtc-libraries webrtc-samples webrtc-tools
Last synced: about 2 months ago
JSON representation
WebRTC, WebRTC and WebRTC. Everything here is all about WebRTC!!
- Host: GitHub
- URL: https://github.com/muaz-khan/WebRTC-Experiment
- Owner: muaz-khan
- License: mit
- Created: 2012-11-14T05:53:13.000Z (about 12 years ago)
- Default Branch: master
- Last Pushed: 2022-06-15T09:37:48.000Z (over 2 years ago)
- Last Synced: 2024-05-14T01:42:26.968Z (7 months ago)
- Topics: webrtc, webrtc-demos, webrtc-examples, webrtc-experiments, webrtc-libraries, webrtc-samples, webrtc-tools
- Language: JavaScript
- Homepage: https://www.webrtc-experiment.com/
- Size: 32 MB
- Stars: 11,606
- Watchers: 663
- Forks: 3,943
- Open Issues: 533
-
Metadata Files:
- Readme: README.md
- License: LICENSE
Awesome Lists containing this project
- awesome-starts - muaz-khan/WebRTC-Experiment - WebRTC, WebRTC and WebRTC. Everything here is all about WebRTC!! (JavaScript)
- awesome-webrtc - WebRTC Experiment - WebRTC, WebRTC and WebRTC. Everything here is all about WebRTC. (Samples)
- awesome-starred - muaz-khan/WebRTC-Experiment - WebRTC, WebRTC and WebRTC. Everything here is all about WebRTC!! (others)
README
# 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/
[![npm](https://img.shields.io/npm/v/recordrtc.svg)](https://npmjs.org/package/recordrtc) [![downloads](https://img.shields.io/npm/dm/recordrtc.svg)](https://npmjs.org/package/recordrtc) [![Build Status: Linux](https://travis-ci.org/muaz-khan/RecordRTC.png?branch=master)](https://travis-ci.org/muaz-khan/RecordRTC)
Github (open sourced): https://github.com/muaz-khan/RecordRTC
RecordRTC extension is [available in the Chrome Web Store](https://chrome.google.com/webstore/detail/recordrtc/ndcljioonkecdnaaihodjgiliohngojp).
----
# 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/
[![npm](https://img.shields.io/npm/v/detectrtc.svg)](https://npmjs.org/package/detectrtc) [![downloads](https://img.shields.io/npm/dm/detectrtc.svg)](https://npmjs.org/package/detectrtc) [![Build Status: Linux](https://travis-ci.org/muaz-khan/DetectRTC.png?branch=master)](https://travis-ci.org/muaz-khan/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.)
[![npm](https://img.shields.io/npm/v/rtcmulticonnection.svg)](https://npmjs.org/package/rtcmulticonnection) [![downloads](https://img.shields.io/npm/dm/rtcmulticonnection.svg)](https://npmjs.org/package/rtcmulticonnection) [![Build Status: Linux](https://travis-ci.org/muaz-khan/RTCMultiConnection.png?branch=master)](https://travis-ci.org/muaz-khan/RTCMultiConnection)
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
[![npm](https://img.shields.io/npm/v/webrtc-scalable-broadcast.svg)](https://npmjs.org/package/webrtc-scalable-broadcast) [![downloads](https://img.shields.io/npm/dm/webrtc-scalable-broadcast.svg)](https://npmjs.org/package/webrtc-scalable-broadcast)
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
[![npm](https://img.shields.io/npm/v/canvas-designer.svg)](https://npmjs.org/package/canvas-designer) [![downloads](https://img.shields.io/npm/dm/canvas-designer.svg)](https://npmjs.org/package/canvas-designer) [![Build Status: Linux](https://travis-ci.org/muaz-khan/Canvas-Designer.png?branch=master)](https://travis-ci.org/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/
[![npm](https://img.shields.io/npm/v/getstats.svg)](https://npmjs.org/package/getstats) [![downloads](https://img.shields.io/npm/dm/getstats.svg)](https://npmjs.org/package/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
[![npm](https://img.shields.io/npm/v/fbr.svg)](https://npmjs.org/package/fbr) [![downloads](https://img.shields.io/npm/dm/fbr.svg)](https://npmjs.org/package/fbr) [![Build Status: Linux](https://travis-ci.org/muaz-khan/FileBufferReader.png?branch=master)](https://travis-ci.org/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
----
# License
All [WebRTC Experiments](https://github.com/muaz-khan/WebRTC-Experiment) are released under [MIT license](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/LICENSE) . Copyright (c) [Muaz Khan](https://muazkhan.com/).