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| Name | Name | Last commit date | ||
|---|---|---|---|---|
Each demo has a unique directory. Simply download that directory, upload in your webserver and use it; and it'll work!
You don't need to modify any single line to use it. No single installation or modification is needed :)
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 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
Cross browser audio/video/screen recording. It supports Chrome, Firefox, Opera and Microsoft Edge. It even works on Android browsers. It follows latest MediaRecorder API standards and provides similar APIs.
Live Demos: https://www.webrtc-experiment.com/msr/
Github (open-sourced): https://github.com/streamproc/MediaStreamRecorder
Source codes and demos are available here: http://webrtcweb.com/cordova-apps/
A few demo apps published to Google Play Store:
WebRTC JavaScript library for peer-to-peer applications (screen sharing, audio/video conferencing, file sharing, media streaming etc.)
Documentation: http://www.rtcmulticonnection.org/
Version 3 demos: https://rtcmulticonnection.herokuapp.com/
Version 2 and older demos: https://www.webrtc-experiment.com/RTCMultiConnection/
Github (open-sourced): https://github.com/muaz-khan/RTCMultiConnection
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 chrome extensions for screen sharing, screen recording, file sharing, youtube+audio sharing, etc.
Github (open sourced): https://github.com/muaz-khan/Chrome-Extensions
Firefox extension API are used to enable screen capturing flag for specific domains, etc.
Install from Firefox addons store: https://addons.mozilla.org/en-US/firefox/addon/enable-screen-capturing/
Github (open-sourced): https://github.com/muaz-khan/Firefox-Extensions
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
getScreenId | Capture Screen on Any Domain! This script is a hack used to support single chrome extension usage on any HTTPs domain.
Live Demo: https://www.webrtc-experiment.com/getScreenId/
Youtube video presentation: https://www.youtube.com/watch?v=UHrsfe9RYAQ
Install this chrome extension (required): https://chrome.google.com/webstore/detail/screen-capturing/ajhifddimkapgcifgcodmmfdlknahffk
Github (open-sourced): https://github.com/muaz-khan/getScreenId
| Experiment Name | Demo | Source Code |
|---|---|---|
| Audio Recording | Demo | Source |
| Video Recording | Demo | Source |
| Gif Recording | Demo | Source |
| MultiStreamRecorder Demo | Demo | Source |
A reliable socketlio/node.js based signaling implementation for DataChannel.js, RTCMultiConnection.js and WebRTC Experiments.
Github (open-sourced): https://github.com/muaz-khan/Reliable-Signaler
Github (open-sourced): https://github.com/muaz-khan/Ffmpeg.js
| Demo Name | Demo | Code |
|---|---|---|
| Transcoding WAV into AAC | Demo | Source |
| Transcoding WAV into Ogg | Demo | Source |
| Transcoding WebM into mp4 | Demo | Source |
| Transcoding WebM into mp4; then merging WAV+mp4 into single mp4 | Demo | Source |
| Recording Audio+Canvas and merging in single mp4 | Demo | Source |
| Recording Audio+Screen and merging in single mp4 | Demo | Source |
DataChannel.js is a JavaScript library useful to write many-to-many i.e. group file/data sharing or text chat applications. Its syntax is easier to use and understand. It highly simplifies complex tasks like any or all user rejection/ejection; direct messages delivery; and more.
Github (open-sourced): https://github.com/muaz-khan/DataChannel
Github (open-sorced): https://github.com/muaz-khan/MultiRTC
XHR/XMLHttpRequest based WebRTC signaling implementation.
Github (open-sourced): https://github.com/muaz-khan/XHR-Signaling
Github (open-sourced): https://github.com/muaz-khan/PluginRTC
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
SdpSerializer.js — An easiest way to modify SDP. It is an object-oriented way of sdp declaration, manipulation and serialization.
Github (open-sourced): https://github.com/muaz-khan/SdpSerializer
| Experiment Name | Short Description | Source | Demo |
|---|---|---|---|
| Pre-recorded Media Streaming | Stream video files in realtime; same like webcam streaming! | Source | Demo |
| Part of Screen Sharing | Share a region of the screen; not the entire screen! | Source | Demo |
| Plugin-free Screen Sharing | Share the entire screen | Source | Demo |
| One-Way Broadcasting | Same like radio stations; transmit audio/video/screen streams in one-way direction. Though, it is browser-to-browser streaming! | Source | Demo |
| Experiment Name | Previous Demos | New Demos |
|---|---|---|
| video-conferencing / multi-user (group) video sharing | Demo / Source | Demo / Source Code |
| file sharing / multi-user (group) files hangout | Demo / Source | Demo / Source Code |
| file sharing using SCTP data channels | Demo / -- | -- / Source Code |
| text chat / multi-user (group) text chat | Demo / Source | Demo / Source Code |
| MultiRTC | Demo / -- | -- / Source Code |
| Experiment Name | Previous Demos | New Demos |
|---|---|---|
| video-broadcasting | Demo / Source | Demo / Source Code |
| audio-broadcasting | Demo / Source | Demo / Source Code |
| Experiment Name | Demo | Source Code |
|---|---|---|
| One-to-one WebRTC video chat using WebSocket | Demo | Source |
| One-to-one WebRTC video chat using socket.io | Demo | Source |
| WebRTC 1-1 Audio/Video/Screen Sharing | Source | Demo |
| WebRTC 1-1 Calls | Source | Demo |
| Experiment Name | Demo | Source Code |
|---|---|---|
| Switch streams from screen-sharing to audio+video. (Renegotiation) | Demo | Source |
| Share screen and audio/video from single peer connection! | Demo | Source |
| Text chat using RTCDataChannel APIs | Demo | Source |
| Simple video chat | Demo | Source |
| Sharing video - using socket.io for signaling | Demo | Source |
| Sharing video - using WebSockets for signaling | Demo | Source |
| Audio Only Streaming | Demo | Source |
| MediaStreamTrack.getSources | Demo | Source |
| A few documents for newbies and beginners |
|---|
| How to use RTCPeerConnection.js? |
| RTCDataChannel for Beginners |
| How to use RTCDataChannel? - single code for both canary and nightly |
| WebRTC for Beginners: A getting stared guide! |
| WebRTC for Newbies |
| How to switch streams? |
| How to echo cancellation? / Noise management? |
| STUN or TURN? Which one to prefer; and why? |
| WebRTC RTP Usage |
| webrtcpedia! |
| Are you want to learn WebRTC? |
| WebRTC Tips & Tricks |
WebRTC Experiments works fine on following web-browsers:
| Browser | Support |
|---|---|
| Firefox | Stable / Aurora / Nightly |
| Google Chrome | Stable / Canary / Beta / Dev |
| Opera | Stable / NEXT |
| Android | Chrome / Firefox / Opera |
| DemoTitle | TestLive | ViewSource |
|---|---|---|
| Audio+Video+File+TextChat | Demo | Source |
| Pre-recorded media streaming (webm/mp3 live streaming) | Demo | Source |
| FileSharing | Demo | Source |
| Scalable Audio/Video Broadcast | Demo | Source |
| Scalable Screen Broadcast | Demo | Source |
| Scalable Video Broadcast | Demo | Source |
| Scalable File Sharing | Demo | Source |
| Video Conferencing | Demo | Source |
| SSEConnection (Server Sent Events) | Demo | Source |
| Audio+Video+Screen Sharing | Demo | Source |
| One-to-One Video Chat | Demo | Source |
| Audio Conferencing | Demo | Source |
| Video Broadcasting | Demo | Source |
| TextChat+FileSharing | Demo | Source |
| addStream in a Chat room | Demo | Source |
| Part-of-Screen Sharing | Demo | Source |
| Share Audio+Screen | Demo | Source |
| Screen Sharing | Demo | Source |
| Disconnect/Rejoin rooms | Demo | Source |
| Password Protected Rooms | Demo | Source |
| replaceTrack in Firefox | Demo | Source |
| applyConstraints in Firefox | Demo | Source |
| Firebase-Demo | Demo | Source |
| PubNub Demo | Demo | Source |
| Socket.io Custom-Messaging | Demo | Source |
| Check Rooms Presence | Demo | Source |
| getPublicModerators | Demo | Source |
| Change Cameras/Microphone | Demo | Source |
| MultiRTC: Skype-like app | Demo | Source |
| Change Video Resolutions in your Live Sessions | Demo | Source |
| Admin/Guest demo | Demo | Source |
| Check if StreamHasData | Demo | Source |
| Capture & Share Screen from any domain! | Demo | Source |
| Experiment Name | Demo | Source Code |
|---|---|---|
| AppRTC like RTCMultiConnection demo! | Demo | Source |
| MultiRTC! RTCMultiConnection all-in-one demo! | Demo | Source |
| Collaborative Canvas Designer | Demo | Source |
| Conversation.js - Skype like library | Demo | Source |
| All-in-One test | Demo | Source |
| Multi-Broadcasters and Many Viewers | Demo | Source |
| Select Broadcaster at runtime | Demo | Source |
| OneWay Screen & Two-Way Audio | Demo | Source |
| Stream Mp3 Live | Demo | Source |
| Socket.io auto Open/Join rooms | Demo | Source |
| Screen Sharing & Cropping | Demo | Source |
| Share Part of Screen without cropping it | Demo | Source |
| getMediaDevices/enumerateDevices | Demo | Source |
| Renegotiation & Mute/UnMute/Stop | Demo | Source |
| Video-Conferencing | Demo | Source |
| Video Broadcasting | Demo | Source |
| Many-to-One Broadcast | Demo | Source |
| Audio Conferencing | Demo | Source |
| Multi-streams attachment | Demo | Source |
| Admin/Guest audio/video calling | Demo | Source |
| Session Re-initiation Test | Demo | Source |
| Preview Screenshot of the room | Demo | Source |
| RecordRTC & RTCMultiConnection | Demo | Source |
| Explains how to customize ice servers; and resolutions | Demo | Source |
| Mute/Unmute and onmute/onunmute | Demo | Source |
| One-page demo: Explains how to skip external signalling gateways | Demo | Source |
| Password Protect Rooms: Explains how to authenticate users | Demo | Source |
| Session Management: Explains difference between "leave" and "close" methods | Demo | Source |
| Multi-Sessions Management | Demo | Source |
| Customizing Bandwidth | Demo | Source |
| Users ejection and presence detection | Demo | Source |
| Multi-Session Establishment | Demo | Source |
| Group File Sharing + Text Chat | Demo | Source |
| Audio Conferencing + File Sharing + Text Chat | Demo | Source |
| Join with/without camera | Demo | Source |
| Screen Sharing | Demo | Source |
| One-to-One file sharing | Demo | Source |
| Manual session establishment + extra data transmission | Demo | Source |
| Manual session establishment + extra data transmission + video conferencing | Demo | Source |
| takeSnapshot i.e. Take Snapshot of Local/Remote streams | Demo | Source |
| Audio/Video/Screen sharing and recording | Demo | Source |
| Broadcast Multiple-Cameras | Demo | Source |
| Remote Stream Forwarding | Demo | Source |
| WebRTC Scalable Broadcast | Socketio/Nodejs | Source |
RecordRTC wiki pages:
Other wiki pages:
All WebRTC Experiments are released under MIT licence . Copyright (c) Muaz Khan.
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