I am a VoIP engineer working with open-source projects like FreeSWITCH, Kamailio/openSIPs, RTPEngine, some other testing tools like SIPp.
My workflow includes:
My workflow includes:
- Kamailio cfg language for SIP routing logic
- FreeSWITCH XML configuration
- Lua scripting for advanced freeSWITCH dialplan apps
- Python/Go for automation and infrastructure
- With Python/Go, tools like "ChatGPT 5"/"Claude 4" are very useful.
- With VoIP-specific code, results are mixed. They often miss critical SIP/SDP nuances (e.g. Contact, Route, Record-Route header manipulation, or SDP media parameters), which makes code look right but fail in testing and practice.
- Same issue with testing tools like SIPp or handling RTP/SDP details.
- Are you using any AI or IDE tools (Cursor, Copilot, custom models, etc.) to boost your productivity in VoIP development?
- Has anyone tried training models on SIP/Kamailio/FreeSWITCH docs for better results?
- Any success stories where AI actually understands well that field and helps with configs or debugging?

