Ecosyste.ms: Awesome
An open API service indexing awesome lists of open source software.
https://github.com/agustinsrg/webrtc-cdn
Live media content delivery network based on the WebRTC protocol.
https://github.com/agustinsrg/webrtc-cdn
audio cdn go golang video webrtc webrtc-signaling websocket
Last synced: about 1 month ago
JSON representation
Live media content delivery network based on the WebRTC protocol.
- Host: GitHub
- URL: https://github.com/agustinsrg/webrtc-cdn
- Owner: AgustinSRG
- License: mit
- Created: 2022-03-15T08:16:51.000Z (almost 3 years ago)
- Default Branch: master
- Last Pushed: 2024-06-29T09:37:51.000Z (6 months ago)
- Last Synced: 2024-11-20T07:14:58.202Z (about 1 month ago)
- Topics: audio, cdn, go, golang, video, webrtc, webrtc-signaling, websocket
- Language: Go
- Homepage:
- Size: 146 KB
- Stars: 3
- Watchers: 2
- Forks: 0
- Open Issues: 0
-
Metadata Files:
- Readme: README.md
- License: LICENSE
Awesome Lists containing this project
README
# 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](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](./doc/signaling.md)
If you want to know about the inter-node communication protocol check:
- [Inter-Node communication protocol](./doc/redis.md)
## Client Libraries
Here is a list of available client libraries to connect to webrtc-cdn:
- [Javascript client for NodeJS and web browsers](https://github.com/AgustinSRG/webrtc-cdn-client)
## Utilities / Experiments
Here is a list of utilities and experiments based of webrtc-cdn:
- [Video Publisher](https://github.com/AgustinSRG/webrtc-publish)
- [Forwarder](https://github.com/AgustinSRG/webrtc-forwarder)
- [Video filtering](https://github.com/AgustinSRG/webrtc-video-filter)
## License
This project is under the [MIT License](./LICENSE).