Hi, I can help with this.
I’ve worked on SIP debugging and integrations involving provider interop, SIP routing, and testing environments. I’m comfortable with Kamailio, SIP signaling analysis (INVITE flows, 4xx/5xx errors), sipp for load/testing, and voipmonitor for trace-level debugging.
Happy to start with a paid consultation to review your current setup, logs, and call flows, then help align your telephony provider and LiveKit SIP so they properly negotiate and handle signaling/media.
Feel free to DM me with a quick overview of your architecture and the specific errors you're seeing.
@victorious-architect-63325