Fix One-Way Audio Kamailio/Asterisk

Incoming calls reach the phones, Intermittent issues such as one-way audio where caller can hear us but we cannot be heard.
The signalling path is Kamailio acting purely as the control-plane SIP proxy, then the media is anchored by a FreeSWITCH + Asterisk B2BUA cluster. Only inbound legs show the problem; outbound audio is clean.

I need a seasoned VoIP troubleshooter to:
• trace SIP and RTP on all hops (Kamailio, FreeSWITCH, Asterisk, edge SBC)
• pinpoint why RTP from the caller side never makes it to the far end (NAT, codec negotiation, rtpengine mis-pinning, firewall, wrong c= line, etc.)
• supply the minimal configuration changes or firewall rules to restore full two-way audio without disrupting live traffic

Acceptance will be:
1. SIP packet examination and diagnosing issue
2. A short report listing the root cause and the exact changes applied (diffs or CLI commands)

SSH access, sngrep, tcpdump and Wireshark captures are available the moment we start. If you prefer your own tooling, that’s fine as long as the above criteria are met.