[Freeswitch-users] SIP Re-invite
anthony.minessale at gmail.com
Thu Dec 17 07:54:21 PST 2009
Is the packet capture running on the FS box itself?
On Thu, Dec 17, 2009 at 9:36 AM, Michael Jerris <mike at jerris.com> wrote:
> if you don't see it in sofia siptrace but do see it in tcpdump capture then
> something very ugly is going on. Either sofia has hung up completely and is
> not listening on that port anymore (can other calls go through?) or the
> packet you see in tcpdump is not really going to the right port. Can you
> confirm which one?
> On Dec 16, 2009, at 6:29 PM, DJB wrote:
> We have a customer that we are sending calls to off the FS and here is the
> Call is initially setup fine and they send a first re-invite with media
> 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first
> re-invite fine
> They then send a second re-invite with their media IP to cut through media
> and the FS sends a 200 OK to this fine. At this point the call is fine
> 30 minutes later they send a third re-invite because according to them it
> is strictly for the purpose of “keep alive” per RFC 4028. This third
> re-invite has the exact same media IP and UDP pot information as the second
> re-invite does. The problem is FS does not respond to this third re-invite
> AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the
> call to be dropped as the other end does not recieve a response from FS.
> One more thing, we did not see the third re-invite in sofia siptrace, but
> we do see it in ethereal, which is kind of odds.
> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
Anthony Minessale II
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