How I Handle 1,642 WebRTC Connections at 6% CPU on a $29 Server - Pure mediasoup Benchmark
In May I stress-tested the WebRTC layer of mena-stream.com.
The result: 1,642 concurrent mediasoup SFU connections + 2,057 WebSocket signaling connections at 6% CPU, on a single Hetzner (8 vCPU, 16GB - $29/mo).
Important: This May benchmark was PURE mediasoup + uWebSockets only. No RTPEngine, no Kamailio at that time.
How I got that number with pure mediasoup:
Tuned mediasoup workers, optimized uWebSockets.js signaling, and focused on SFU forwarding efficiency. No kernel RTP forwarding was used in this specific May test just kernel tuning and proprietary optimizations.
The Grafana screenshot below is from that pure mediasoup run.
What's new after May:
After this benchmark, I added the full telephony layer on top:
Kamailio + RTPEngine (xt_RTPENGINE kernel module) SIP gateway
DTMF (RFC 2833) pass-through verified
SIP-to-Mediasoup bridge via PlainTransport, so real phone calls join WebRTC rooms
GStreamer for recording, FFmpeg for smart rendering
Automated regression suite - 10/10 passing
This new SIP layer is functionally working but its formal load test is still pending. I will publish a new benchmark with Kamailio + RTPEngine + mediasoup combined soon.
Next: Combined load test + AF_XDP kernel-bypass.
I submitted this architecture as a deep-dive to ClueCon 2026.
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