WebRTC Video/Audio Broadcast

WebRTC PeerToPeer broadcast application that allows the broadcaster to send video and audio stream to all connected users (watchers).

You can find the tutorial that explains the code and functionality on my website.

Getting started

Starting the application

Start the application using Node:

# Install dependencies for server
npm install

# Run the server
node server

Start the application using Docker:

# Building the image
docker build --tag webrtcvideobroadcast .

# Run the image in a container
docker run -d -p 4000:4000 webrtcvideobroadcast

Testing the application

The application should now be running on your localhost:4000 and you test it by connecting to localhost:4000/broadcast.html to add a new broadcaster.

After that, you just need to visit localhost:4000 to connect to the server as a client and you should get the video that is streamed from the broadcaster.

Adding a TURN server

A TURN server is used to relay traffic if a direct peer to peer connection fails and is required for most WebRTC application since a direct socket is often not possible between two clients that aren't on the same network. This repository doesn't include the usage of a TURN server by default, but you can add one by commenting in the turn configuration in the broadcast.js and watch.js file and filling in your TURN credentials.

There are several options on how you can create your own TURN server. Here are just two common ones:

• Coturn
• Golang WebRTC pion library TURN examples

You can also use TURN servers from cloud providers or other companies.

Author

Gabriel Tanner

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License

This project is licensed under the MIT License - see the LICENSE.md file for details