Live media content delivery network based on the WebRTC protocol.
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 current directory, called webrtc-cdn, or webrtc-cdn.exe if you are using Windows.
You can find the docker image for this project available in Docker Hub: https://hub.docker.com/r/asanrom/webrtc-cdn
To pull it type:
docker pull asanrom/webrtc-cdn
This project is meant to be used to create a network to deliver live media content using the WebRTC protocol.
In order to create the network, you can spawn multiple nodes connected to a Redis Pub/Sub service for inter-node communication.
Once the network is up, clients can connect to the nodes via Websocket (for signaling purposes), in order to request for publishing or receiving media streams via WebRTC.
You can configure the node using 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. |
To configure the redis connection, set the following variables:
| Variable Name | Description |
|---|---|
| STAND_ALONE | Set it to YES if you want to disable redis and just use a single node. By default, webrtc-cdn will use redis |
| REDIS_PORT | Port to connect to Redis Pub/Sub. Default is 6379 |
| REDIS_HOST | Host to connect to Redis Pub/Sub. Default is 127.0.0.1 |
| REDIS_PASSWORD | Redis authentication password, if required. |
| REDIS_TLS | Set it to YES in order to use TLS for the connection. |
If you want to use TLS for the websocket connections (recommended), you have to set 3 variables in order for it to work:
| Variable Name | Description |
|---|---|
| SSL_PORT | HTTPS listening port. Default is 443 |
| SSL_CERT | Path to SSL certificate. |
| SSL_KEY | Path to SSL private key. |
Authentication options:
| Variable Name | Description |
|---|---|
| JWT_SECRET | Secret to validate JSON web tokens used for authentication in the signaling protocol. If not set, no authentication is required. |
Here is a list with more options you can configure:
| Variable Name | Description |
|---|---|
| HTTP_PORT | HTTP listening port for insecure websocket connections, used for signaling. Default is 80 |
| BIND_ADDRESS | Bind address for signaling services. By default it binds to all network interfaces. |
| LOG_REQUESTS | Set to YES or NO. By default is YES |
| LOG_DEBUG | Set to YES or NO. By default is NO |
| MAX_IP_CONCURRENT_CONNECTIONS | Max number of concurrent connections to accept from a single IP. By default is 4. |
| CONCURRENT_LIMIT_WHITELIST | List of IP ranges not affected by the max number of concurrent connections limit. Split by commas. Example: 127.0.0.1,10.0.0.0/8 |
| MAX_REQUESTS_PER_SOCKET | Max number of active requests for a single websocket session. By default is 100 |
The ports used by the signaling websocket server must be opened, they are 80 and 443 by default.
In order for the nodes to be able to communicate via WebRTC, they need to use the port range 40000:65535/UDP
If you use a TURN server there is no need for the UDP ports to be opened, since communication can be accomplish using the TURN server as intermediate.
Check the documentation in order to connect to the nodes:
If you want to know about the inter-node communication protocol check:
Here is a list of available client libraries to connect to webrtc-cdn:
Here is a list of utilities and experiments based of webrtc-cdn:
This project is under the MIT License.