Simple video/audio record application using Mediasoup and GStreamer
Recorded files are stored in the server's files directory or the directory set by the user (via process.env.RECORD_FILE_LOCATION_PATH)
File names are simply the current timestamp
This sample currently only uses VP8/opus and the output file is .webm
# For Ubuntu
sudo apt-get install libgstreamer1.0-0 gstreamer1.0-plugins-base gstreamer1.0-plugins-good gstreamer1.0-plugins-bad gstreamer1.0-plugins-ugly gstreamer1.0-libav gstreamer1.0-doc gstreamer1.0-tools gstreamer1.0-x gstreamer1.0-alsa gstreamer1.0-gl gstreamer1.0-gtk3 gstreamer1.0-qt5 gstreamer1.0-pulseaudio
cd server && npm i
cd app && npm i
# Change the listen IP in src/config.js to your local ip (config -> webRtcTransport -> listenIps)
# Create [files] directory in order for the files to be saved
# The server uses FFmpeg as default
cd server && node src/server
# To use GStreamer
PROCESS_NAME="GStreamer" node src/server
cd app
npm run build
# Copy the files from dist to a webserver etc.
# OR start the dev server
npm run dev
Argument | Type | Explanation |
---|---|---|
RECORD_FILE_LOCATION_PATH | string | Path to store the recorded files (user running node MUST have read/write permission) |
GSTREAMER_DEBUG_LEVEL | number | GStreamer Debug Level (GStreamer only) |
PROCESS_NAME | string | The command to use (GStreamer/FFmpeg) (case sensitive) default is FFmpeg |
SERVER_PORT | number | Server port number (default is 3000). Note if you change this you will also need to edit the WebSocket connection url. |
- video/audio only recording
- FFmpeg Sample (Done)
- Multiple formats (mp4/avi etc)
- Option to play the recorded file using RTP Producer after recording
- Docker support