[Freeswitch-users] SIP Re-invite

[DJB]

Please advise whether I should put a request in JIRA. 
http://pastebin.freeswitch.org/11541

Thank you.



________________________________
From: DJB <djbinter at yahoo.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, December 17, 2009 9:35:27 AM
Subject: Re: [Freeswitch-users] SIP Re-invite


Anthony/Michael,

I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539

There are 2 traces in there from within the same box.  One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison.

The missing re-invite in FS is at 2009/12/17 17:25:55.207747 


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
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>>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
>
>
>_______________________________________________
>>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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