github.com/AgustinSRG/webrtc-cdn

WebRTC CDN

Live media content delivery network based on the WebRTC protocol.

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 current directory, called webrtc-cdn, or webrtc-cdn.exe if you are using Windows.

Docker Image

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

Usage

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.

Configuration

You can configure the node using environment variables

WebRTC options

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.

Redis

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.

TLS for signaling

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

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.

More options

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

Firewall configuration

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.

Documentation

Check the documentation in order to connect to the nodes:

• Signaling protocol

If you want to know about the inter-node communication protocol check:

• Inter-Node communication protocol

Client Libraries

Here is a list of available client libraries to connect to webrtc-cdn:

• Javascript client for NodeJS and web browsers

Utilities / Experiments

Here is a list of utilities and experiments based of webrtc-cdn:

• Video Publisher
• Forwarder
• Video filtering

License

This project is under the MIT License.