# WebRTC-Experiment **Repository Path**: eaco_hu/WebRTC-Experiment ## Basic Information - **Project Name**: WebRTC-Experiment - **Description**: 很好的WebRTC功能范例，体验WebRTC - **Primary Language**: Unknown - **License**: MIT - **Default Branch**: master - **Homepage**: None - **GVP Project**: No ## Statistics - **Stars**: 0 - **Forks**: 0 - **Created**: 2016-07-17 - **Last Updated**: 2024-05-31 ## Categories & Tags **Categories**: Uncategorized **Tags**: None ## README ## Realtime/Working [WebRTC Experiments](https://www.webrtc-experiment.com/) 1. It is a repository of uniquely experimented WebRTC demos; written by Muaz Khan! 2. No special requirement! Just WebRTC compatible web-browser (e.g. chrome/firefox/opera on desktop/android) 3. These demos/experiments are entirely client-side; i.e. no server installation needed! 4. You can use all these demos in PHP/Python/Ruby/ASP.NET/etc. everywhere! ## How to use? **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 :) ## Libraries | Library Name | Short Description | Doc | Demos | | ------------- |-------------|-------------|-------------| | `RecordRTC.js` | Supports cross-browser audio/video recordings! | [Doc](https://github.com/muaz-khan/RecordRTC) | [Demos](https://www.webrtc-experiment.com/RecordRTC/) | | `Translator.js` | Voice & Text Translator | [Doc](https://github.com/muaz-khan/Translator) | [Demos](https://www.webrtc-experiment.com/Translator/) | | `RTCMultiConnection.js` | Single Library for Everything! Just imagine :) | [Doc](https://github.com/muaz-khan/RTCMultiConnection) | [Demos](https://www.webrtc-experiment.com/RTCMultiConnection/) | | `FileBufferReader.js` | File buffers reader & chunkifier | [Doc](https://github.com/muaz-khan/FileBufferReader) | [Demos](https://www.WebRTC-Experiment.com/FileBufferReader/) | | `getScreenId.js` | Single chrome extension for all domains! Again, imagine :) | [Doc](https://github.com/muaz-khan/getScreenId) | [Demos](https://www.webrtc-experiment.com/getScreenId/) | | `Conversation.js` | Enjoy Skype-like Conversations! Oops :) | [Doc](https://github.com/muaz-khan/Conversation.js) | [Demos](https://www.webrtc-experiment.com/Conversationjs/) | | `DataChannel.js` | Supports data-streaming among multiple peers | [Doc](https://github.com/muaz-khan/DataChannel) | [Demos](https://www.webrtc-experiment.com/#DataChannel) | | `getMediaElement.js` | A library for audio/video media elements' layout | [Doc](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/getMediaElement) | [Demos](https://www.webrtc-experiment.com/getMediaElement/) | | `DetectRTC.js` | A library for detecting WebRTC features | [Doc](https://github.com/muaz-khan/DetectRTC) | [Demos](https://www.webrtc-experiment.com/DetectRTC/) | | `ConcatenateBlobs.js` | Concatenate Array of Blobs | [Doc](https://github.com/muaz-khan/ConcatenateBlobs) | [Demos](https://www.webrtc-experiment.com/ConcatenateBlobs/) | | `getStats.js` | Get peers statistics | [Doc](https://github.com/muaz-khan/getStats) | None | Other libraries: * [navigator.customGetUserMediaBar.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/navigator.customGetUserMediaBar) / [Demo](https://www.webrtc-experiment.com/navigator.customGetUserMediaBar/) * [File.js](https://www.webrtc-experiment.com/docs/Share-Files-using-Filejs.html) / [Demo](https://www.webrtc-experiment.com/WebRTC-File-Sharing/) * [Meeting.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/meeting) / [Demo](https://www.webrtc-experiment.com/meeting/) * [RTCall.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCall) / [Demo](https://www.webrtc-experiment.com/RTCall/) * [SdpSerializer.js](https://github.com/muaz-khan/SdpSerializer) / [Demo](https://www.webrtc-experiment.com/SdpSerializer/demo.html) ## Other Repositories 13. [WebRTC Scalable Broadcast](https://github.com/muaz-khan/WebRTC-Scalable-Broadcast) 14. [Reliable Signaling](https://github.com/muaz-khan/Reliable-Signaler) 2. [RTCMultiConnection.js](https://github.com/muaz-khan/RTCMultiConnection) 1. [RecordRTC.js](https://github.com/muaz-khan/RecordRTC) 4. [Collaborate Canvas Designer](https://github.com/muaz-khan/Canvas-Designer) 1. [Translator.js](https://github.com/muaz-khan/Translator) 10. [FileBufferReader.js](https://github.com/muaz-khan/FileBufferReader) 11. [Chrome-Extensions](https://github.com/muaz-khan/Chrome-Extensions) 12. [Firefox-Extensions](https://github.com/muaz-khan/Firefox-Extensions) 3. [DetectRTC.js](https://github.com/muaz-khan/DetectRTC) 3. [getStats.js](https://github.com/muaz-khan/getStats) 3. [Conversation.js](https://github.com/muaz-khan/Conversation.js) 3. [Ffmpeg.js](https://github.com/muaz-khan/Ffmpeg.js) 3. [DataChannel.js](https://github.com/muaz-khan/DataChannel) 3. [MultiRTC Demos](https://github.com/muaz-khan/MultiRTC) 5. [XHR-Signaling](https://github.com/muaz-khan/XHR-Signaling) 6. [PluginRTC: IE/Safari Plugins compatible WebRTC-Experiments](https://github.com/muaz-khan/PluginRTC) 7. [ASP.NET MVC based WebRTC 1:1 Demo](https://github.com/muaz-khan/WebRTC-ASPNET-MVC) 8. [WebSync Signaling](https://github.com/muaz-khan/WebSync-Signaling) 9. [SdpSerializer.js](https://github.com/muaz-khan/SdpSerializer) ## Important [Experiments](https://github.com/muaz-khan/WebRTC-Experiment) | Experiment Name | Short Description | Source | Demo | | ------------- |-------------|-------------|-------------| | `Pre-recorded Media Streaming` | Stream video files in realtime; same like webcam streaming! | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pre-recorded-Media-Streaming) | [Demo](https://www.webrtc-experiment.com/Pre-recorded-Media-Streaming/) | | `Part of Screen Sharing` | Share a region of the screen; not the entire screen! | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing) | [Demo](https://www.webrtc-experiment.com/part-of-screen-sharing/) | | `Plugin-free Screen Sharing` | Share the entire screen | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharing) | [Demo](https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/) | | `One-Way Broadcasting` | Same like radio stations; transmit audio/video/screen streams in one-way direction. Though, it is browser-to-browser streaming! | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/webrtc-broadcasting) | [Demo](https://www.webrtc-experiment.com/webrtc-broadcasting/) | ## Useful [Experiments](https://github.com/muaz-khan/WebRTC-Experiment) | Experiment Name | Previous Demos | New Demos | | ------------- |-------------|-------------| | **video-conferencing** / multi-user (group) video sharing | [Demo](https://www.webrtc-experiment.com/video-conferencing/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing) | [Demo](https://www.webrtc-experiment.com/meeting/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/meeting) | | **file sharing** / multi-user (group) files hangout | [Demo](https://www.webrtc-experiment.com/file-hangout/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/file-hangout) | [Demo](https://www.webrtc-experiment.com/file-sharing/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/file-sharing) | | **file sharing using SCTP data channels** | [Demo](https://www.webrtc-experiment.com/WebRTC-File-Sharing/) / -- | -- / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/WebRTC-File-Sharing) | | **text chat** / multi-user (group) text chat | [Demo](https://www.webrtc-experiment.com/chat-hangout/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/chat-hangout) | [Demo](https://www.webrtc-experiment.com/text-chat/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/text-chat) | | **MultiRTC** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/MultiRTC/) / -- | -- / [Source Code](https://github.com/muaz-khan/MultiRTC) | ## [Google Chrome Extensions](https://github.com/muaz-khan/Chrome-Extensions) for [WebRTC](https://www.webrtc-experiment.com/)! 1. [desktopCapture API](https://github.com/muaz-khan/Chrome-Extensions/tree/master/desktopCapture) / [Install App Store Extension](https://chrome.google.com/webstore/detail/screen-capturing/ajhifddimkapgcifgcodmmfdlknahffk) 2. [tabCapture API](https://github.com/muaz-khan/Chrome-Extensions/tree/master/tabCapture) / [Install App Store Extension](https://chrome.google.com/webstore/detail/tab-capturing-sharing/pcnepejfgcmidedoimegcafiabjnodhk) 3. [Desktop Sharing](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/desktop-sharing) / [Install App Store Extension](https://chrome.google.com/webstore/detail/webrtc-desktop-sharing/nkemblooioekjnpfekmjhpgkackcajhg) 4. [File Sharing](https://github.com/muaz-khan/Chrome-Extensions/tree/master/file-sharing) / [Install App Store Extension](https://chrome.google.com/webstore/detail/webrtc-file-sharing/nbnncbdkhpmbnkfngmkdbepoemljbnfo) ## [Firefox Extensions](https://github.com/muaz-khan/Firefox-Extensions) for [WebRTC](https://www.webrtc-experiment.com/)! 1. [enable-screen-capturing](https://github.com/muaz-khan/Firefox-Extensions/tree/master/enable-screen-capturing) ## One-to-Many style of WebRTC [Experiments](https://github.com/muaz-khan/WebRTC-Experiment) | Experiment Name | Previous Demos | New Demos | | ------------- |-------------|-------------| | **video-broadcasting** | [Demo](https://www.webrtc-experiment.com/broadcast/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/broadcast) | [Demo](https://www.webrtc-experiment.com/one-to-many-video-broadcasting/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/one-to-many-video-broadcasting) | | **audio-broadcasting** | [Demo](https://www.webrtc-experiment.com/audio-broadcast/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/audio-broadcast) | [Demo](https://www.webrtc-experiment.com/one-to-many-audio-broadcasting/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/one-to-many-audio-broadcasting) | ## One-to-One Calls | Experiment Name | Demo | Source Code | | ------------- |-------------|-------------| | One-to-one WebRTC video chat using WebSocket | [Demo](https://www.webrtc-experiment.com/websocket/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/websocket) | | One-to-one WebRTC video chat using socket.io | [Demo](https://www.webrtc-experiment.com/socket.io/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/socket.io) | | WebRTC 1-1 Audio/Video/Screen Sharing | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/realtime-pluginfree-calls) | [Demo](https://www.webrtc-experiment.com/realtime-pluginfree-calls/) | | WebRTC 1-1 Calls | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/realtime-pluginfree-calls) | [Demo](https://www.webrtc-experiment.com/realtime-pluginfree-calls/) | ## Single-Page / One-Page / Client Side | Experiment Name | Demo | Source Code | | ------------- |-------------|-------------| | **Switch streams from screen-sharing to audio+video. (Renegotiation)** | [Demo](https://www.webrtc-experiment.com/demos/switch-streams.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/switch-streams.html) | | **Share screen and audio/video from single peer connection!** | [Demo](https://www.webrtc-experiment.com/demos/screen-and-video-from-single-peer.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/screen-and-video-from-single-peer.html) | | **Text chat using RTCDataChannel APIs** | [Demo](https://www.webrtc-experiment.com/demos/client-side-datachannel.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/) |client-side-datachannel.html | **Simple video chat** | [Demo](https://www.webrtc-experiment.com/demos/client-side.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/client-side.html) | | **Sharing video - using socket.io for signaling** |[Demo](https://www.webrtc-experiment.com/demos/client-side-socket-io.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/client-side-socket-io.html) | | **Sharing video - using WebSockets for signaling** | [Demo](https://www.webrtc-experiment.com/demos/client-side-websocket.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/client-side-websocket.html) | | **Audio Only Streaming** | [Demo](https://www.webrtc-experiment.com/demos/audio-only-streaming.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/audio-only-streaming.html) | | **MediaStreamTrack.getSources** | [Demo](https://www.webrtc-experiment.com/demos/MediaStreamTrack.getSources.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/MediaStreamTrack.getSources.html) | ## Experiments to share tab/screen/desktop | Experiment Name | Previous Demos | New Demos | | ------------- |-------------|-------------| | **Plugin-free screen sharing** / share the entire screen | [Demo](https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharing) | [Demo](https://www.webrtc-experiment.com/screen-sharing/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/screen-sharing) | | **Desktop sharing** / using `desktopCapture` APIs | [Demo](https://www.webrtc-experiment.com/desktop-sharing/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/desktop-sharing) | -- | | **Tab sharing** / using `tabCapture` APIs | [Demo](https://www.webrtc-experiment.com/screen-broadcast/) / [Source](https://github.com/muaz-khan/Chrome-Extensions/tree/master/tabCapture) | -- | ## Experiments to [share region/part of the screen](https://www.webrtc-experiment.com/part-of-screen-sharing/) | Experiment Name | Demo | Source Code | | ------------- |-------------|-------------| | Share part-of-screen **RTCMultiConnection** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/RTCMultiConnection.sharePartOfScreen.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/demos/RTCMultiConnection.sharePartOfScreen.html) | | Share part-of-screen **using RTCDataChannel APIs** | [Demo](https://www.webrtc-experiment.com/part-of-screen-sharing/webrtc-data-channel/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing/webrtc-and-part-of-screen-sharing) | | Share part-of-screen **using Firebase** | [Demo](https://www.webrtc-experiment.com/part-of-screen-sharing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing) | | **A realtime chat using RTCDataChannel** | [Demo](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/realtime-chat/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing/realtime-chat/RTCDataChannel) | | **A realtime chat using Firebase** | [Demo](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/realtime-chat/No-WebRTC-Chat.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing/realtime-chat) | ## [Demos](https://www.webrtc-experiment.com/msr/) using [MediaStreamRecorder.js](https://github.com/streamproc/MediaStreamRecorder) library | Experiment Name | Demo | Source Code | | ------------- |-------------|-------------| | **Audio Recording** | [Demo](https://www.webrtc-experiment.com/msr/audio-recorder.html) | [Source](https://github.com/streamproc/MediaStreamRecorder/tree/master/demos/audio-recorder.html) | | **Video Recording** | [Demo](https://www.webrtc-experiment.com/msr/video-recorder.html) | [Source](https://github.com/streamproc/MediaStreamRecorder/tree/master/demos/video-recorder.html) | | **Gif Recording** | [Demo](https://www.webrtc-experiment.com/msr/gif-recorder.html) | [Source](https://github.com/streamproc/MediaStreamRecorder/tree/master/demos/gif-recorder.html) | Or [RecordRTC entire Meeting using MediaStreamRecorder.js](https://github.com/streamproc/Record-Entire-Meeting) ## Demos using [DataChannel.js](https://github.com/muaz-khan/DataChannel) library | Experiment Name | Demo | Source Code | | ------------- |-------------|-------------| | **DataChannel basic demo** | [Demo](https://www.webrtc-experiment.com/DataChannel/) | [Source](https://github.com/muaz-khan/DataChannel/blob/master/index.html) | | **Auto Session Establishment** | [Demo](https://www.webrtc-experiment.com/DataChannel/auto-session-establishment.html) | [Source](https://github.com/muaz-khan/DataChannel/blob/master/auto-session-establishment.html) | | **Share part-of-screen using DataChannel.js** | [Demo](https://www.webrtc-experiment.com/part-of-screen-sharing/webrtc-data-channel/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing/webrtc-and-part-of-screen-sharing) | | **Private Chat** | [Demo](https://muazkh.appspot.com/privatechat/) | ---- | | **Text Chat using Pusher and DataChannel.js** | [Demo](http://webrtc-chat-demo.pusher.io/) | [Source](https://github.com/pusher/pusher-webrtc-chat-demo) | ## Experimental (Non-Functional) | Experiment Name | Demo | Source Code | | ------------- |-------------|-------------| | **Attaching Remote Audio Streams** | [Demo](https://www.webrtc-experiment.com/experimental/remote-media-stream-attachment/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/experimental/remote-media-stream-attachment) | | **mozCaptureStreamUntilEnded for pre-recorded media streaming** | [Demo](https://www.webrtc-experiment.com/experimental/mozCaptureStreamUntilEnded/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/experimental/mozCaptureStreamUntilEnded) | | **Remote audio stream recording** | [Demo](https://www.webrtc-experiment.com/demos/remote-stream-recording.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/experimental/remote-stream-recording.html) | ## RTCMultiConnection v2.2.2 Demos | Experiment Name | Demo | Source Code | | ------------- |-------------|-------------| | AppRTC like RTCMultiConnection demo! | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/AppRTC-Look.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/AppRTC-Look.html) | | MultiRTC! RTCMultiConnection all-in-one demo! | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/MultiRTC/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/MultiRTC-simple) | | Collaborative Canvas Designer | [Demo](https://www.webrtc-experiment.com/Canvas-Designer/) | [Source](https://github.com/muaz-khan/Canvas-Designer) | | Conversation.js - Skype like library | [Demo](https://www.webrtc-experiment.com/Conversationjs/) | [Source](https://github.com/muaz-khan/Conversation.js) | | All-in-One test | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/all-in-one.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/all-in-one.html) | | Multi-Broadcasters and Many Viewers | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/Multi-Broadcasters-and-Many-Viewers.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/Multi-Broadcasters-and-Many-Viewers.html) | | Select Broadcaster at runtime | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/select-broadcaster-at-runtime.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/select-broadcaster-at-runtime.html) | | OneWay Screen & Two-Way Audio | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/OneWay-Screen-TwoWay-Audio.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/OneWay-Screen-TwoWay-Audio.html) | | Stream Mp3 Live | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/stream-mp3-live.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/stream-mp3-live.html) | | Socket.io auto Open/Join rooms | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/socketio-auto-open-join-room.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/socketio-auto-open-join-room.html) | | Screen Sharing & Cropping | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/cropped-screen-sharing.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/cropped-screen-sharing.html) | | Share Part of Screen without cropping it | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/RTCMultiConnection.sharePartOfScreen.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/RTCMultiConnection.sharePartOfScreen.html) | | getMediaDevices/enumerateDevices | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/getMediaDevices.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/getMediaDevices.html) | | Renegotiation & Mute/UnMute/Stop | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/Renegotiation.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/Renegotiation.html) | | Video-Conferencing | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/videoconferencing.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/videoconferencing.html) | | Video Broadcasting | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/video-broadcasting.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/video-broadcasting.html) | | Many-to-One Broadcast | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/many-to-one-broadcast.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/many-to-one-broadcast.html) | | Audio Conferencing | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/audioconferencing.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/audioconferencing.html) | | Multi-streams attachment | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/multi-streams-attachment.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/multi-streams-attachment.html) | | Admin/Guest audio/video calling | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/admin-guest.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/admin-guest.html) | | Session Re-initiation Test | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/session-reinitiation.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/session-reinitiation.html) | | Preview Screenshot of the room | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/rooms-screenshots.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/rooms-screenshots.html) | | RecordRTC & RTCMultiConnection | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/RecordRTC-and-RTCMultiConnection.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/RecordRTC-and-RTCMultiConnection.html) | | Explains how to customize ice servers; and resolutions | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/features.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/features.html) | | Mute/Unmute and onmute/onunmute | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/mute-unmute.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/mute-unmute.html) | | One-page demo: Explains how to skip external signalling gateways | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/one-page-demo.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/one-page-demo.html) | | Password Protect Rooms: Explains how to authenticate users | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/password-protect-rooms.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/password-protect-rooms.html) | | Session Management: Explains difference between "leave" and "close" methods | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/session-management.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/session-management.html) | | Multi-Sessions Management | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/multi-sessions-management.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/multi-sessions-management.html) | | Customizing Bandwidth | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/bandwidth.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/bandwidth.html) | | Users ejection and presence detection | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/users-ejection.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/users-ejection.html) | | Multi-Session Establishment | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/multi-session-establishment.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/multi-session-establishment.html) | | Group File Sharing + Text Chat | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/group-file-sharing-plus-text-chat.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/group-file-sharing-plus-text-chat.html) | | Audio Conferencing + File Sharing + Text Chat | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/audio-conferencing-data-sharing.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/audio-conferencing-data-sharing.html) | | Join with/without camera | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/join-with-or-without-camera.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/join-with-or-without-camera.html) | | Screen Sharing | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/screen-sharing.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/screen-sharing.html) | | One-to-One file sharing | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/one-to-one-filesharing.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/one-to-one-filesharing.html) | | Manual session establishment + extra data transmission | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/manual-session-establishment-plus-extra-data-transmission.html) | | Manual session establishment + extra data transmission + video conferencing | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) | | takeSnapshot i.e. Take Snapshot of Local/Remote streams | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/takeSnapshot.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/takeSnapshot.html) | | Audio/Video/Screen sharing and recording | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/audio-video-screen-sharing-recording.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/audio-video-screen-sharing-recording.html) | | Broadcast Multiple-Cameras | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/Broadcast-Multiple-Cameras.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/Broadcast-Multiple-Cameras.html) | | Remote Stream Forwarding | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/remote-stream-forwarding.html) | [Source](https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2/demos/remote-stream-forwarding.html) | | WebRTC Scalable Broadcast | Socketio/Nodejs | [Source](https://github.com/muaz-khan/WebRTC-Scalable-Broadcast) | v2.2.2 is available here: * https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2 ## [Demos](https://www.webrtc-experiment.com/Conversationjs/) using [Conversation.js](https://github.com/muaz-khan/Conversation.js)
  1. AndroidRTC
  2. Search Users
  3. Cross-Language (Multi-Lingual) Text Chat
  4. Old Conversation.js demos
## [Documents for newcomers/newbies/beginners](https://www.webrtc-experiment.com/docs/) | A few documents for newbies and beginners | | ------------- | | [How to use RTCPeerConnection.js?](https://www.webrtc-experiment.com/docs/how-to-use-rtcpeerconnection-js-v1.1.html) | | [RTCDataChannel for Beginners](https://www.webrtc-experiment.com/docs/rtc-datachannel-for-beginners.html) | | [How to use RTCDataChannel?](https://www.webrtc-experiment.com/docs/how-to-use-rtcdatachannel.html) - single code for both canary and nightly | | [WebRTC for Beginners: A getting stared guide!](https://www.webrtc-experiment.com/docs/webrtc-for-beginners.html) | | [WebRTC for Newbies ](https://www.webrtc-experiment.com/docs/webrtc-for-newbies.html) | | [How to switch streams?](https://www.webrtc-experiment.com/docs/how-to-switch-streams.html) | | [How to echo cancellation? / Noise management?](https://www.webrtc-experiment.com/docs/echo-cancellation.html) | | [STUN or TURN? Which one to prefer; and why?](https://www.webrtc-experiment.com/docs/STUN-or-TURN.html) | | [WebRTC RTP Usage](https://www.webrtc-experiment.com/docs/RTP-usage.html) | | [webrtcpedia!](https://www.webrtc-experiment.com/webrtcpedia/) | | [Are you want to learn WebRTC?](http://muaz-khan.blogspot.com/2013/12/i-want-to-learn-webrtc.html) | | [WebRTC Tips & Tricks](http://muaz-khan.blogspot.com/2014/05/webrtc-tips-tricks.html) | 1. http://muaz-khan.blogspot.com/search/label/WebRTC 2. https://www.webrtc-experiment.com/#documentations 3. https://www.facebook.com/WebRTC 4. https://plus.google.com/+WebRTC-Experiment/posts = ## [ffmpeg-asm.js && Ffmpeg Demos](https://github.com/muaz-khan/Ffmpeg.js) | Demo Name | Live Demo | Source Code | | ------------- |-------------|-------------| | Transcoding WAV into Ogg | [Live Demo](https://www.webrtc-experiment.com/ffmpeg/wav-to-ogg.html) | [Source Code](https://github.com/muaz-khan/Ffmpeg.js/blob/master/wav-to-ogg.html) | | Transcoding WebM into mp4 | [Live Demo](https://www.webrtc-experiment.com/ffmpeg/webm-to-mp4.html) | [Source Code](https://github.com/muaz-khan/Ffmpeg.js/blob/master/webm-to-mp4.html) | | Transcoding WebM into mp4; then merging WAV+mp4 into single mp4 | [Live Demo](https://www.webrtc-experiment.com/ffmpeg/merging-wav-and-webm-into-mp4.html) | [Source Code](https://github.com/muaz-khan/Ffmpeg.js/blob/master/merging-wav-and-webm-into-mp4.html) | | Recording Audio+Canvas and merging in single mp4 | [Live Demo](https://www.webrtc-experiment.com/ffmpeg/audio-plus-canvas-recording.html) | [Source Code](https://github.com/muaz-khan/Ffmpeg.js/blob/master/audio-plus-canvas-recording.html) | = ## Custom Signaling 1. [Socket.io over Node.js](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/socketio-over-nodejs) 2. [WebSocket over Node.js](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/websocket-over-nodejs) 3. [WebSync / ASP.NET MVC](https://github.com/muaz-khan/WebSync-Signaling) 4. [XHR Signaling](https://github.com/muaz-khan/XHR-Signaling) 5. [openSignalingChannel](http://www.rtcmulticonnection.org/docs/openSignalingChannel/) ## How to record audio using [RecordRTC](http://recordrtc.org/)? ```html ``` Documentation page: http://recordrtc.org/RecordRTC.html ```javascript var recordRTC = RecordRTC(mediaStream, { type: 'video' // audio or video or gif or canvas }); recordRTC.startRecording(); recordRTC.stopRecording(function(videoURL) { video.src = videoURL; var blob = recordRTC.blob; var arrayBuffer = recordRTC.buffer; recordRTC.getDataURL(callback_function); }); ``` 1. [RecordRTC to Node.js](https://github.com/muaz-khan/RecordRTC/tree/master/RecordRTC-to-Nodejs) 2. [RecordRTC to PHP](https://github.com/muaz-khan/RecordRTC/tree/master/RecordRTC-to-PHP) 3. [RecordRTC to ASP.NET MVC](https://github.com/muaz-khan/RecordRTC/tree/master/RecordRTC-to-ASPNETMVC) 4. [RecordRTC & HTML-2-Canvas i.e. Canvas/HTML Recording!](https://github.com/muaz-khan/RecordRTC/tree/master/Canvas-Recording) 5. [MRecordRTC i.e. Multi-RecordRTC!](https://github.com/muaz-khan/RecordRTC/tree/master/MRecordRTC) 6. [RecordRTC on Ruby!](https://github.com/cbetta/record-rtc-experiment) 7. [RecordRTC over Socket.io](https://github.com/muaz-khan/RecordRTC/tree/master/RecordRTC-over-Socketio) 8. [ffmpeg-asm.js and RecordRTC! Audio/Video Merging & Transcoding!](https://github.com/muaz-khan/Ffmpeg.js) 9. [Recording Audio+Video in single WebM on Firefox](https://www.webrtc-experiment.com/RecordRTC/AudioVideo-on-Firefox.html) 10. [RecordRTC / PHP / FFmpeg](https://github.com/muaz-khan/RecordRTC/tree/master/PHP-and-FFmpeg) ## [RTCMultiConnection.js](http://www.RTCMultiConnection.org/docs/) You can write entire skype-like web-app using RTCMultiConnection! It supports all complex renegotiation scenarios! ```html
``` [RTCMultiConnection Documentation](http://www.RTCMultiConnection.org/docs/) ## [DataChannel.js](https://github.com/muaz-khan/DataChannel) / A library for RTCDataChannel APIs ```html ``` [DataChannel Documentation](https://github.com/muaz-khan/DataChannel) ## [Translator.js](https://github.com/muaz-khan/Translator) / [Demo](https://www.webrtc-experiment.com/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! ```html ``` ```javascript var translator = new Translator(); translator.voiceToText(function (text) { console.log('Your voice as text!', text); }, 'your-language'); translator.translateLanguage(textToConvert, { from: 'language-of-the-text', to: 'convert-into', callback: function (translatedText) { console.log('translated text', translatedText); } }); translator.speakTextUsingRobot(textToPlay); translator.speakTextUsingGoogleSpeaker({ textToSpeak: 'text-to-convert', targetLanguage: 'your-language' }); ``` ## [FileBufferReader.js](https://github.com/muaz-khan/FileBufferReader) / [Demo](https://www.WebRTC-Experiment.com/FileBufferReader/) 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. ```javascript var fileBufferReader = new FileBufferReader(); fileBufferReader.readAsArrayBuffer(file, function(uuid) { // var file = fileBufferReader.chunks[uuid]; // var listOfChunks = file.listOfChunks; // get first chunk, and send using WebRTC data channels // NEVER send chunks in loop; otherwise you'll face issues in slow networks // remote peer should notify if it is ready for next chunk fileBufferReader.getNextChunk(uuid, function(nextChunk, isLastChunk) { if(isLastChunk) { alert('File Successfully sent.'); } // sending using WebRTC data channels datachannel.send(nextChunk); }); }); datachannel.onmessage = function(event) { var chunk = event.data; if (chunk instanceof ArrayBuffer || chunk instanceof DataView) { // array buffers are passed using WebRTC data channels // need to convert data back into JavaScript objects fileBufferReader.convertToObject(chunk, function(object) { datachannel.onmessage({ data: object }); }); return; } // if you passed "extra-data", you can access it here: // chunk.extra.senderUserName or whatever else // if target peer requested next chunk if(chunk.readyForNextChunk) { fileBufferReader.getNextChunk(chunk.uuid, function(nextChunk, isLastChunk) { if(isLastChunk) { alert('File Successfully sent.'); } // sending using WebRTC data channels datachannel.send(nextChunk); }); return; } // if chunk is received fileBufferReader.addChunk(chunk, function(promptNextChunk) { // request next chunk datachannel.send(promptNextChunk); }); }; ``` ## [getScreenId.js](https://github.com/muaz-khan/getScreenId) / [Demo](https://www.webrtc-experiment.com/getScreenId/) Simply use getScreenId.js and enjoy screen capturing from any domain. You don't need to deploy chrome extension yourself. You can refer your users to install this chrome extension instead. Also, getScreenId.js auto-fallbacks to command-line based screen capturing if chrome extension isn't installed or disabled. getScreenId.js throws clear exceptions which is helpful for end-user experiences. ```html ``` ## Signaling? * https://github.com/muaz-khan/WebRTC-Experiment/blob/master/Signaling.md ## Browser Support [WebRTC Experiments](https://www.webrtc-experiment.com/) works fine on following web-browsers: | Browser | Support | | ------------- |-------------| | Firefox | [Stable](http://www.mozilla.org/en-US/firefox/new/) / [Aurora](http://www.mozilla.org/en-US/firefox/aurora/) / [Nightly](http://nightly.mozilla.org/) | | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) | | Opera | [Stable](http://www.opera.com/) / [NEXT](http://www.opera.com/computer/next) | | Android | [Chrome](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) / [Firefox](https://play.google.com/store/apps/details?id=org.mozilla.firefox) / [Opera](https://play.google.com/store/apps/details?id=com.opera.browser) | ## Credits * Muaz Khan - https://github.com/muaz-khan # License All [WebRTC Experiments](https://github.com/muaz-khan/WebRTC-Experiment) are released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) [Muaz Khan](http://www.muazkhan.com/).