Real-time communication (WebRTC, SIP, video calls, chat apps, apps with peer-to-peer connections) often needs more than just signaling — it needs reliable STUN/TURN servers to get traffic through NATs and firewalls.
WebRTC
I’m excited to share fastTURN — a professional, globally distributed STUN & TURN service built for speed, reliability, and simplicity.
🚦 Why fastTURN
- 🌍 Global network with 24+ locations — auto-routes users to the closest server
- ⚡ Optimized for super fast handshakes (STUN resolution avg ~49 ms)
- 🔒 Secure connections with SSL support (optional)
- 📈 Reliable uptime & bandwidth — 99.99% availability
- 💸 Affordable plans + a free tier for testing/dev
- 🔁 Works with SIP, ICE, WebRTC apps out of the box
Whether you’re building video chat, VoIP, or any peer-to-peer app, this gives you the fallback connectivity infrastructure you need without hosting your own TURN servers.
👉 Check it out: https://fastturn.net/
Would love to hear:
- What real-time stack you’re using
- How you handle TURN/STUN now
- Pain points you’ve run into in production
Happy to answer questions! 🚀
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