LiveKit SIP ↔ SIP PBX Integration Consultant

We need an expert consultant to integrate LiveKit with a SIP PBX for a short-term project. The work includes configuring the SIP connection, ensuring reliable call routing, and verifying that the integration performs smoothly in a production environment. This is a small, focused engagement for someone with strong experience in SIP and real-time communication systems. What we need help with: Configuring LiveKit SIP trunk settings (inbound/outbound) Setting up the corresponding SIP peer/trunk on our PBX Troubleshooting NAT, codec negotiation, and signalling issues Ensuring call routing works correctly in both directions Documenting the final configuration for our team You should have: Proven experience with LiveKit and its SIP gateway Strong SIP protocol knowledge (SIP/SDP, RTP, SRTP, DTMF) Hands-on experience with at least one major PBX platform (Asterisk, FreePBX, FusionPBX, 3CX, or similar) Ability to diagnose issues via SIP traces/logs (Wireshark, sngrep, etc.) Clear communication — we'll need you to explain what you're doing as you go Nice to have: Experience with LiveKit's AI/agent pipeline alongside SIP Familiarity with cloud-hosted PBX environments To apply, please include: A brief description of a similar LiveKit or SIP integration you've completed