[asterisk-users] stucked calls in asterisk 1.4

[David Backeberg]

On Wed, May 27, 2009 at 10:30 AM, Stefan Schmidt <sst at sil.at> wrote:
> as i said the routing server also handles calls from an ser proxy and
> another asterisk server where iax accounts terminates and this problem
> is only on the pbx server.
>
> Maybe it is a network problem but the quality of the rtp streams is ok
> but i think that there are too much sip pakets for the system.
>
> there are around 1600 sip users registerd, 300 - 400 sip channels
> (register, options, notifys and invite) and 600 - 700 subscriptions so
> there is much sip traffic.
>
> this server also does rtp handling and have 50 to 100 calls (active
> ones) and in peek time there is around 10mbit of traffic with 5 to 6 kpps.
>
> maybe a problem with udp buffer size??

Now that I better understand your problem, I'm out of ideas.
You are correct that if a BYE sip packet gets lost,
a) it won't get retransmitted if it's UDP
b) the side that's waiting for the hangup will think the call is still active

I've seen this in my system where the network switch went down while
calls were active. The system where the call was happening caught the
hangup, but the trunk system never got the bye as the network was
down.

My only questions are:
Are you using a quality network switch, or maybe you can use a
cross-over cable to eliminate collisions with other systems?