Audio gone after a random call duration

I have a strange behavior with some call FreeSWITCH process. I receive the calls from provider over a lan and internal FreeSWITCH sip profile, then send the call to client over public IP using external sip profile. The problem is after a random time the media flow stop from internal sip profile to the provider. None SIP signaling problem.
This freeSWITCH process inbound calls from customer over public IP, external sip profiles (about 1000/15000 concurrent calls) and send to a OpenSIPs. If I stop inbound calls from customers, the problem disappear.
Any hint, please?

Would start with finding how to replicate the problem on demand and inspect FreeSWITCH debug logs and capture a pcap of it happening.

It’s very hard to inspect FreeSWITCH debug with this calls number. In the call where audio gone nothing interessing in the pcap capture. Maybe a ulimit configuration?
Thank you and regards

More hint, please?
thank you in davance