[asterisk-users] Dropping RTP packets

[Brent Davidson]

You need "canreinvite=no" in the config for your sip phone and the 
veracity connection, otherwise Asterisk will just mediate the call setup 
then try to allow the sip phone and veracity to talk directly to one 
another.

Jim Dickenson wrote:
> 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?
>