Hi,
Currently, I face an issue that I could not hear the audio sound at all when I try to call to FreeSWITCH (9386) via its public domain name from the SIP client of my mobile phone with mobile internet.
In this case, SIP registration with TLS and media encryption with SRTP is working perfectly because I could see the secure symbol in Linphone but I could not hear any audio at all. After 30 seconds of the silent call, it hanged up automatically.

SIP Client on the 3G Mobile Internet > Freeswitch (tls:voip.test.com, SRTP) 
SIP Registration Sucess, Call No Media

My question is that is this the problem of NAT? If so, it would be great if you could help me solve this problem. 
Any help would be really appreciated.

Best regards,
Chhorm Chhatra