From SIP to AI: A Real Call Finally Worked
Sharing a small but exciting milestone from my learning journey 🚀
Over the last few weeks, I’ve been deep into voice infrastructure and SIP, and I finally have a full working setup:
📞 Local Phone → SIP (from Signalwire) → FreeSWITCH → Voice Agent → Response back to the caller
The FreeSWITCH server is running on a Debian server on DigitalOcean, and everything is now talking to each other smoothly — SIP, RTP, real-time audio, and AI responses.
I’m currently working with a client, and initially we’re aiming to scale this setup to handle ~100 concurrent calls, which is pushing me to really understand:
  • SIP call flows
  • Audio streaming
  • Server performance & scaling
  • Latency trade-offs vs managed platforms
Honestly, this stuff is challenging but insanely exciting. Every time a real phone call hits the server and the agent responds correctly, it feels like magic — but the earned kind 😄
Just wanted to share this win and keep building.
If anyone here is working on voice agents, SIP, or FreeSWITCH — would love to connect and exchange notes 🤝
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5 comments
Khizar Hayat
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From SIP to AI: A Real Call Finally Worked
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