P2P WebRTC and Own Server Options
Source page: https://webrtchost.com/hosting-plans/
P2P WebRTC services focus on low-latency peer connections using the VideoWhisper WebRTC signaling server plus STUN/TURN. This path is useful for private calls, high-interactivity sessions, developer tests, and custom deployments.
Free Developer Account Context
The free developer account is designed for limited testing by 1-3 developers with constrained quality settings. The public plan page lists:
| Feature | Free Developers Account |
|---|---|
| Streams around 532kbps | 5 |
| 1-way viewers | 3 |
| 2-way private calls | 1 |
| Max resolution | 640x360p |
| Max framerate | 15 fps |
| Max video bitrate | 750 kbps |
| Max audio bitrate | 32 kbps |
| Total account bitrate | 2.7 Mbps |
| Max connections | 10 |
Own Server Deployment
VideoWhisper can install the P2P WebRTC signaling server on a client VPS or dedicated server. The open-source server is available at:
https://github.com/videowhisper/videowhisper-webrtc
Typical requirements include:
- Linux server, preferably CentOS, CloudLinux, or similar.
- SSL certificate for a pointed domain.
- Node.js signaling server over WSS/HTTPS.
- Coturn STUN/TURN server configuration.
- Open ports for signaling, STUN/TURN, and WebRTC media relay.
- Starting hardware around 4+ CPU cores, 8 GB RAM, and 250 Mbps connection, adjusted for project scale.
Limitations
P2P WebRTC is excellent for private calls and low-latency interactivity, but it is not optimized for 1-to-many scaling. For large broadcast audiences, a relay streaming server is usually recommended so the broadcaster sends one stream to the server and the server distributes it to viewers.
Optional RTMP/HLS Module
Some projects may require additional RTMP/HLS stream management around the Node.js server and Nginx. This adds integration capabilities for broadcast/playback control, PINs, live stream listings, and web callbacks.