[Asterisk-Users] SIP behind NAT: NAT'ted end has to talk first?

[Brian Capouch]

I am having problems in a couple of installations where I have SIP 
phones (both GS101 and ATA186) connecting to an asterisk box that has a 
public IP address, where the stations are behind NAT.

I'm still testing to make sure I have all the permutations looked at, 
but from what I can tell, what is happening is that in situations where 
stations behind the NAT call out, no audio is passed until after the 
party on the PUBLIC side generates some audio.

So that means if I call from the public side to one of the NAT boxes, I 
can't hear them answer.  But when (while watching the console) I can see 
that the call has been bridged, I quickly hail them with a "Hello," then 
the RTP stream starts going and everyone is happy.

I have the exact same problem using iconnecthere when I call out (to the 
PSTN) from stations behind NAT: I see the call bridge on the console; my 
party answers but I don't hear it, nor do they hear me until I say 
something, and at that point the RTP stream starts up.

This must be evidence of something wrong with the way the initial RTP 
stream is commenced when SIP stations are behind NAT.

Does anyone know what's going on, or of course better, what I can do to 
rectify this?

Thanks.

B.