Is the packet capture running on the FS box itself?<br><br><br><div class="gmail_quote">On Thu, Dec 17, 2009 at 9:36 AM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com">mike@jerris.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div style="word-wrap: break-word;">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?<div>
<br></div><div>Mike</div><div><div></div><div class="h5"><div><br><div><div>On Dec 16, 2009, at 6:29 PM, DJB wrote:</div><br><blockquote type="cite"><span style="border-collapse: separate; font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;"><div>
<div style="margin: 0px; font-family: verdana,helvetica,sans-serif; font-size: 10pt;"><div style="margin: 0px;"><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial,sans-serif;">We have a customer that we are sending calls to off the FS and here is the issue:</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial,sans-serif;"> </span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial,sans-serif;">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</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial,sans-serif;"> </span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial,sans-serif;">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</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial,sans-serif;"> </span></p><p class="MsoNormal"><span style="font-size: 10pt; font-family: Arial,sans-serif;">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. </span></p>
<p class="MsoNormal"><font face="Arial, sans-serif"><br></font></p><p class="MsoNormal"><font face="Arial, sans-serif">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.</font></p>
<p class="MsoNormal"><font face="Arial, sans-serif"><br></font></p><p class="MsoNormal"><font face="Arial, sans-serif">We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.</font></p></div></div></div>
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