agustinsrg / webrtc-publish

Utility to publish to webrtc-cdn using a video or RTMP source.

WebRTC Publisher

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.

Compilation

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.

Usage

You can use the program from the command line:

webrtc-publisher [OPTIONS] <SOURCE> <DESTINATION>

SOURCE

The source can be a path to a video file, or an RTMP URL. Examples:

• ./path/to/video.mp4
• rtmp://localhost/test/key

DESTINATION

The destination must be a websocket URL of one of the webrtc-cdn nodes. Examples:

• ws://localhost/stream-id
• wss://www.example.com/stream-id

OPTIONS

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.

WebRTC options

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.