[asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

[Philip Prindeville]

I've got the following situation.  I'm running Asterisk 1.4.18 on a 
firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones 
behind it.

I'm peering SIP with a Coppercom switch sitting behind an SBC.

On outbound calls, I get 2-way voice, no worries.

On inbound calls, I get one-way voice (I can hear the caller but they 
can't hear me).

I've looked at tcpdumps of the RTP traffic, and the addresses and port 
numbers correspond to what's in the SIP INVITE/OK messages (assuming 
that they don't somehow get munged by NAT after tcpdump looks at them -- 
there is no NAT device upstream of my Asterisk firewall).

I'll look into using Record() or Monitor() to capture the phone call, 
but if there's any conversion being done by codecs then that won't 
eliminate the possibility that the code itself is misconfigured or buggy 
and generating a bad stream on one of the legs...

Anyone have an idea about how to best go about troubleshooting this?

Thanks,

-Philip