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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I'm currently handling about 6k calls per day between a few different enterprises. We're likely going to implement and build this all out in LiveKit but I also need a better eval + simulation solution and I don't really want to build it out as well. I have seen Coval but I don't know the cost yet -- what else it out there that you guys use that can give me actionable feedback of missed tool calls, hallucinations, etc.

I’ve been building voice agents primarily with Retell and VAPI so far. I’ve explored platforms like Bland and Bolna as well, but I’ve mostly stuck with Retell since it’s been solid and reliable. Recently, I wanted to branch out into custom-built voice agents using frameworks like Pipecat, Cartesia, and others. I’m not really a coder, but I started experimenting with the Cartesia Line SDK through Antigravity. I managed to put together a decent agent but running into a lot of errors as I'm trying to match the responsiveness I get with Retell. No luck yet. I came across this community through a friend and thought I’d ask: has anyone here tried building with the Cartesia Line SDK? I’d love to hear your experiences.

Hey Guys! I'm fairly experienced with VAPI, but 100% new to open source voice AI. I'm currently looking to get deeper into that, and wondering what the state of the art/commonly used providers/models for STT, TTS, LLM, turntaking?, ... are, that are open source/hosted/usable in the EU. I had an advisory call today with a start-up that wants to integrate voice AI, and since they're in GER they need to be compliant and I want to recommend them a few good and solid options for their infra. Thank you already for letting me know your thoughts! Happy to learn! Cheers!