Re: [reSIProcate] One way audio problem

[Alan Hawrylyshen <alan@xxxxxxxxxx>]

On Jun 15, 2005, at 05.07, Eustanos wrote:

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 ?

 

Check to ensure that your SDP offer to the gateway is correct. You need to offer SDP that points at a discovered or created place in the firewall.

Tracing outside the firewall will likely show that the gateway sent RTP to the firewall, where it was dropped. Check your SDP, that is almost certainly where the problem lies.

Alan

a l a n a t j a s o m i d o t c o m