I'm working on a VoIP stack that includes Kamailio, Freeswitch, Asterisk, and some custom logic. Every time we make changes, we find ourselves having to do manual test calls to verify everything works as expected. We're specifically testing things like inbound call routing, IVR and DTMF interactions, voicemail, and call forwarding. While we have experimented with SIPp scripts, they've turned out to be quite challenging to maintain and don't fully test the entire call flows. I'm curious about how other teams tackle this situation. Do you rely on manual testing, use scripts, implement CI, or dare to test directly in production? I'd love to get some insights on effective practices from others in this space!
3 Answers
One approach we use is to set up the new flow with a spare number and conduct tests there. This method is particularly useful for testing call center flows, allowing us to see how everything behaves without impacting real users.
I totally agree with that! Changing one number and conducting a few inbound and outbound calls is the best way to ensure everything's working correctly. It's particularly important to test across different carriers too!
I've had similar experiences with SIPp. Could you share what issues you're facing with it? It seems like a solid option for automated testing, and I'm curious to hear more about its drawbacks.
Certainly! SIPp is indeed powerful, but we find the scenario authoring in XML can eventually get hard to manage, especially with complex flows involving multiple branches or transfers. Also, debugging failures takes time since it can be tricky to determine whether the issue is timing, NAT, or something else. We're currently working on improving the orchestration around SIPp to make it more manageable and easier to run as regression tests in CI.