[Freeswitch-users] SIP Re-invite
DJB
djbinter at yahoo.com
Thu Dec 17 10:31:56 PST 2009
Yes, I have a complete trace here: http://pastebin.freeswitch.org/11541
________________________________
From: David Knell <dave at 3c.co.uk>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, December 17, 2009 9:50:22 AM
Subject: Re: [Freeswitch-users] SIP Re-invite
Can you post the full packets with Ethernet, IP, UDP headers as well, or
upload a pcap file?
I'll add the change in 'Max-Forwards' from 70 to 69 between the two
packets to my things to be suspicious of list.
--Dave
> The trace that I pasted on the pastebin was from our
> analyzer,Tektronix spectra2 that was sitting between FS and customer.
> I also had the FS sip trace on and compare with the trace from Spectra
> when I found out about the 3rd re-invite was missing from FS.
>
> Thank you.
>
>
>
> ______________________________________________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, December 17, 2009 7:57:42 AM
> Subject: Re: [Freeswitch-users] SIP Re-invite
>
> The question was:
>
> Are you doing the packet capture on the actual FS box using tshark or
> tcpdump?
>
>
> On Thu, Dec 17, 2009 at 9:48 AM, DJB <djbinter at yahoo.com> wrote:
> Anthony,
>
> I have pasted the invite sip trace here:
> http://pastebin.freeswitch.org/11536
> Please advise if you need further info.
>
> Thank you.
>
>
>
> ______________________________________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wed, December 16, 2009 3:42:48 PM
> Subject: Re: [Freeswitch-users] SIP Re-invite
>
>
>
> that means the invite is not matching the call dialog
> compare the via tags and call-id etc
>
>
> On Wed, Dec 16, 2009 at 5:29 PM, DJB <djbinter at yahoo.com>
> wrote:
> We have a customer that we are sending calls to off
> the FS and here is the issue:
>
>
>
> 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.
>
>
> Thank you very much.
>
>
>
>
>
>
> _______________________________________________
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>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn: +19193869900 +19193869900
>
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> _______________________________________________
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