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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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