[asterisk-users] Dropping RTP packets

[Jim Dickenson]

I have a SIP phone at home behind a NAT router registered with an * box at
my office with a routable static IP address running version
SVN-branch-1.6.0-r175638M.

If I make a call from my SIP phone out a PRI circuit to my cell phone
everything works as expected. I hear audio in both directions and all is
good.

If from the same SIP phone I make a call via our Veracity SIP account to my
cell phone I hear no audio in either direction.

In trying to find out what is wrong I used tcpdump to see if I could learn
anything. I can see the phone sending fixed length UDP packets on to my home
network heading to the IP address of the * box. If I run tcpdump on the *
box I do not see the packets being received. I do not see the * box sending
any packets to my home network either. I have not checked if the * box is
receiving packets from Veracity I only know that no audio packets are sent
to my home network.

If I use tcpdump to watch the SIP phone call via the PRI circuit I see
packets both on my home network and my * box.

If I use a SIP phone located in my office and make a call via Veracity
everything is okay. Also a co-worker has a vpn router on his home network
connected to the office vpn server and he can make calls from his SIP phone
via Veracity without problems.

I can also call his SIP phone from my SIP phone and packets pass as
expected.

It seems as if audio packets from my SIP phone disappear only if they are
involved with a call via Veracity.

Does anyone have some idea what I might look at to find what is causing this
problem?
-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/