Guide · Open source
How to use WebRTC-Experiment: demos, libraries and setup
Updated · written with AI from the project's own documentation and checked against our catalogue data
WebRTC-Experiment is a collection of separate browser-based WebRTC projects by Muaz Khan, released under the MIT licence. This guide shows you how to get the repository, find the project that matches your task, try its live demo, and know what to check before using any of it. It is aimed at developers and students who want working examples rather than a single product.
WebRTC Experimentsmuaz-khan/WebRTC-Experiment · ★ 12.1k · MIT Install options & downloads Quick start
- 1Clone the repository. Run the git clone command from the README to download the WebRTC-Experiment code to your machine.
- 2Open the live demo. Visit the demo link for the project you need and allow camera or microphone access in your browser.
- 3Choose one project. Pick RecordRTC, DetectRTC, getStats, FileBufferReader, RTCMultiConnection or another project that matches your feature.
- 4Read that project's section. Follow the GitHub link for the chosen project to see how it is used and what it depends on.
- 5Set up signaling if needed. For peer-to-peer apps like RTCMultiConnection, choose a signaling option such as the RTCMultiConnection-Server from the README.
- 6Test across browsers. Check the feature in each browser your users will run, since support varies by project and browser.
What is in the WebRTC-Experiment repository?
The repository is not one library. It is a set of many separate WebRTC projects, each with its own demo page, and each can be used on its own. WebRTC is the browser technology that lets people send audio, video and data directly to each other.
- RecordRTC: audio, video and screen recording, supporting Chrome, Firefox, Opera, Android and Microsoft Edge.
- DetectRTC: a tiny library to check for speakers, microphone, webcam, screen capture support and device counts.
- RTCMultiConnection: a library for peer-to-peer apps such as screen sharing, audio/video conferencing and file sharing.
- getStats: returns peer connection stats such as bandwidth usage, packets lost and connection type.
- FileBufferReader: reads a file into chunked array-buffers that can be sent over WebRTC data channels or socket.io.
- MultiStreamsMixer: combines multiple streams, such as screen plus camera, into a single stream.
Get a local copy of the code
The README gives one install command: cloning the repository from GitHub. The README lists no prerequisites and does not describe a package install for the whole collection, so treat the clone as a way to read and run the example code. Each project's own GitHub page is linked from the README, and the npm packages listed there (for example recordrtc, detectrtc and getstats) are the separate libraries you can add to your own project.
git clone https://github.com/muaz-khan/WebRTC-ExperimentTry a live demo before writing code
Each project in the README links to a live demo hosted on webrtc-experiment.com or another demo host. Open the demo in a browser and grant camera or microphone access when asked. This is the quickest way to confirm that a feature works on your device before you invest time in it.
- RecordRTC demo: https://www.webrtc-experiment.com/RecordRTC/
- DetectRTC demo: https://www.webrtc-experiment.com/DetectRTC/
- getStats demo: https://www.webrtc-experiment.com/getStats/
- FileBufferReader demo: https://www.webrtc-experiment.com/FileBufferReader/
- Video conferencing demo: https://rtcmulticonnection.herokuapp.com/demos/Video-Conferencing.html
Pick the right project for your task
Match your goal to one project rather than loading the whole repository. Each project has a separate README section, npm package and demo, so you only take what you need.
- Record audio, video or screen: use RecordRTC.
- Check whether a browser or device supports WebRTC features: use DetectRTC.
- Read connection statistics such as packet loss or local and remote IP and port: use getStats.
- Send a file between peers: use FileBufferReader with WebRTC data channels, or the file sharing demo.
- Build a peer-to-peer conference or screen-sharing app: use RTCMultiConnection, which needs a signaling server (the README links RTCMultiConnection-Server, based on socket.io).
Understand signaling before you build
Peers need a way to exchange connection details before they can talk directly. The README calls this signaling and lists several examples of signaling options: socket.io through RTCMultiConnection-Server, XHR-based signaling, Server Sent Events over PHP, SignalR, and WebSync. Studying these is useful for learning how WebRTC signaling and peer connections are structured, even if you later choose a different approach.
The README says that the Scalable Broadcasting module relays one broadcast to many users using socket.io, with media travelling peer-to-peer.
Check maintenance and fit before using it in production
The README shows no releases, and its last push is noted as June 2022. It also gives no release or support policy. That makes the repository a good source of examples and learning material, but a risky base for a production system that needs clear versioning and support.
If you need a supported library, the README mentions comparable projects you can evaluate: simple-peer, PeerJS, Janus WebRTC Gateway and mediasoup.
- Test each feature in the browsers and devices your users have.
- Pin and review any code you copy, since the repository itself gives no version guarantees.
- Compare alternatives before committing to a long-term dependency.
What people use it for
- Adding screen or audio recording to a browser app with RecordRTC.
- Checking whether a visitor's device has a working webcam and microphone with DetectRTC.
- Showing packet loss and connection type in a video call dashboard using getStats.
- Sending a file directly between two browsers with FileBufferReader and WebRTC data channels.
- Learning how a one-to-one WebRTC demo handles signaling, using the ASP.NET MVC example from the README.
Frequently asked questions
Is WebRTC-Experiment free to use?
Yes. The README states that all WebRTC Experiments are released under the MIT licence, with copyright held by Muaz Khan. You should still read the LICENSE file in the repository before using code in a commercial product.
How do I install WebRTC-Experiment?
The README gives one command: git clone https://github.com/muaz-khan/WebRTC-Experiment. It lists no prerequisites. Individual libraries such as recordrtc and detectrtc are also published on npm, as linked from the README.
Is WebRTC-Experiment still maintained?
The README shows no releases and says the last push was in June 2022. It gives no support policy, so check the repository activity yourself before relying on it.
What browsers does RecordRTC support?
According to the README, RecordRTC supports Chrome, Firefox, Opera, Android and Microsoft Edge, on Linux, Mac and Windows.
What are alternatives to WebRTC-Experiment?
The README lists simple-peer, PeerJS, Janus WebRTC Gateway and mediasoup as comparable projects. These may suit teams that need a more clearly versioned library.
Can I use RTCMultiConnection without a signaling server?
The README links a Socket.io signaling server, RTCMultiConnection-Server, and lists several other signaling options. It does not say whether a server is optional, so check the RTCMultiConnection project page before deciding.
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Our verdict
WebRTC-Experiment suits developers who want working, separate examples of specific WebRTC features, and students who want to see how signaling and peer connections fit together. Teams that need a single, actively maintained, versioned library should pick a comparable project such as PeerJS, simple-peer, Janus or mediasoup instead.
Install options & downloads