[Freeswitch-users] SIP Re-invite

[David Knell]

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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>                 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  +19193869900 
>         
>         
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
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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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