Utility to publish to webrtc-cdn using a video or RTMP source.
It uses FFMpeg to convert the source to RTP, and then the pion/webrtc library to convert it to WebRTC.
In order to install dependencies, type:
go get github.com/AgustinSRG/webrtc-cdn
To compile the code type:
go build
The build command will create a binary in the currenct directory, called webrtc-publish, or webrtc-publish.exe if you are using Windows.
You can use the program from the command line:
webrtc-publisher [OPTIONS] <SOURCE> <DESTINATION>
The source can be a path to a video file, or an RTMP URL. Examples:
./path/to/video.mp4rtmp://localhost/test/key
The destination must be a websocket URL of one of the webrtc-cdn nodes. Examples:
ws://localhost/stream-idwss://www.example.com/stream-id
Here is a list of all the options:
| Option | Description |
|---|---|
--help, -h |
Shows the command line options |
--version, -v |
Shows the version |
--debug |
Enables debug mode (prints more messages) |
--ffmpeg-path <path> |
Sets the FFMpeg path. By default is /usr/bin/ffmpeg. You can also change it with the environment variable FFMPEG_PATH |
--loop, -l |
Enables loop (for video files). |
--auth, -a <auth-token> |
Sets auth token. |
--secret, -s <secret> |
Provides secret to generate authentication tokens. |
You can configure WebRTC configuration options with environment variables:
| Variable Name | Description |
|---|---|
| STUN_SERVER | STUN server URL. Example: stun:stun.l.google.com:19302 |
| TURN_SERVER | TURN server URL. Set if the server is behind NAT. Example: turn:turn.example.com:3478 |
| TURN_USERNAME | Username for the TURN server. |
| TURN_PASSWORD | Credential for the TURN server. |