[reSIProcate] One way audio problem

[Eustanos]

Hi, I know this is SIP Forum, but maybe there will be somebody who will find
some time to help me.
I want to applogise everyone who feels disturbed by this mail.
 
I'm behind Port Restricted Router. I write softphone using reSIPprocate
stack.
My software also uses STUN. The problem I fight is "one way audio problem".
When somedoby calls me from PSTN to my computer behind NAT, everything
works fine (two way audio). But when I make a call from my softphone to the
PSTN gateway
and the other person answers, his voice is cut off, I can't hear anything.
But he can hear
me.
 
I know this is NAT problem, when I use computer not behind NAT everything
works just fine.
It's difficult for me to explain why this happens. RTP packets are not send
to my computer from
PSTN gateway. Why ?
 
Best regards,
Mariusz.
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