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 🤝

Hello devs, Is there any pay as you go plan in livekit for agent deployment in there cloud.