[asterisk-users] Audio lost on reinvite

[John A. Sullivan III]

Hello, all.  We are having a problem where audio for sip channels is
dropping upon reinvite.  Perhaps it reflects a misunderstanding of what
reinvite does.  We are running Asterisk 1.6.1.1 on CentOS 5.3.

SIP is set to canreinvite=nonat.  We have tried RTP with strictrtp set
to both yes and no.  We have also tried extending the Asterisk rtp port
range to accommodate the differing default ranges of the soft phones
(Twinkle on Linux, 3CX on Windows).

Testing revealed no problems when the soft phones we used for testing
were on the same physical and logical network.

Once we moved the soft phones to OpenVPN connections (same logical
network but different physical media), the call is setup, the receiver
hears the caller for the briefest instant (we are assuming the first
reinvite), the caller hears the receiver for some time (perhaps 20 - 30
seconds) and then the receiver's voice disappears, too.  At that very
moment, there is another redirect and RTP traffic starts on a different
set of ports from the receiver.

Packet traces revealed RTP packets flowing from the receiver to Asterisk
but no packets coming back from Asterisk except ICMP service unreachable
for port 8000 (the new port after the second reinvite).  It's as if
Asterisk does not recognize the ports after the reinvite.  We were
actually surprised to see the packets flowing between the soft phones
and Asterisk as I would have thought the reinvite would direct traffic
to flow directly between the soft phones - both of whom can ping each
other and are on the same physical network.

We've traced from the perspective of the end points, the gateway, and
Asterisk.  All show the same pattern: the caller is having a dialog with
Asterisk whereas the receiver is having a monolog - no packets back from
Asterisk.

Could someone explain why were are losing the audio, why we see a dialog
on one side but a monolog on the other, and why we are not seeing the
reinvite redirect the packet stream to be directly between the soft
phones.  Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsullivan at opensourcedevel.com

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