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.
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@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.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Call-ID are the same for 1st, 2nd, and 3rd INVITE. The only thing I saw difference was the Via Branch value. Would that be a problem, since 1st and 2nd INVITE was also different and was okay. Is there any other values that I should look at? Thank you. ________________________________ From: Anthony Minessale <anthony.minessale@gmail.com> To: freeswitch-users@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@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.
_______________________________________________
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
FWIW, we’ve seen the same thing intermittently, haven’t had time/been able to get a solid repro to capture debug information. Call ID and tags are all matching. After the re-invite fails and the remote end sends a BYE, FS does indeed respond to the re-invite. -Michael From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of DJB Sent: Wednesday, December 16, 2009 6:00 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP Re-invite Call-ID are the same for 1st, 2nd, and 3rd INVITE. The only thing I saw difference was the Via Branch value. Would that be a problem, since 1st and 2nd INVITE was also different and was okay. Is there any other values that I should look at? Thank you. ________________________________ From: Anthony Minessale <anthony.minessale@gmail.com> To: freeswitch-users@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@yahoo.com<mailto:djbinter@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. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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@gmail.com> To: freeswitch-users@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@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.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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@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@gmail.com> *To:* freeswitch-users@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@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.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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@gmail.com> To: freeswitch-users@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@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@gmail.com>
To: freeswitch-users@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@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.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn: +19193869900 +19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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@gmail.com> To: freeswitch-users@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@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@gmail.com> To: freeswitch-users@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@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.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn: +19193869900 +19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yes, I have a complete trace here: http://pastebin.freeswitch.org/11541 ________________________________ From: David Knell <dave@3c.co.uk> To: freeswitch-users@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@gmail.com> To: freeswitch-users@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@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@gmail.com> To: freeswitch-users@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@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.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn: +19193869900 +19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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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@gmail.com> To: freeswitch-users@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@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@gmail.com>
To: freeswitch-users@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@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.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900
_______________________________________________
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
I am sorry; here is the complete one: http://pastebin.freeswitch.org/11540 Thank you. ________________________________ From: DJB <djbinter@yahoo.com> To: freeswitch-users@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@gmail.com> To: freeswitch-users@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@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@gmail.com>
To: freeswitch-users@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@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.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900
_______________________________________________
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Please advise whether I should put a request in JIRA. http://pastebin.freeswitch.org/11541 Thank you. ________________________________ From: DJB <djbinter@yahoo.com> To: freeswitch-users@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@gmail.com> To: freeswitch-users@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@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@gmail.com>
To: freeswitch-users@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@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.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900
_______________________________________________
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote:
Anthony,
I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info.
Thank you.
Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. ________________________________ From: Michael Jerris <mike@jerris.com> To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony,
I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info.
Thank you.
I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote:
Mike,
My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541
Thank you, Dorn B. From: Michael Jerris <mike@jerris.com> To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite
are you doing this trace from the freeswitch box itself?
Mike
On Dec 17, 2009, at 10:48 AM, DJB wrote:
Anthony,
I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info.
Thank you.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Mike, There are 2 traces in there. 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 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. ________________________________ From: Michael Jerris <mike@jerris.com> To: freeswitch-users@lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike,
My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541
Thank you, Dorn B.
________________________________ From: Michael Jerris <mike@jerris.com>
To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite
are you doing this trace from the freeswitch box itself?
Mike
On Dec 17, 2009, at 10:48 AM, DJB wrote:
Anthony,
I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info.
Thank you.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Mike, There are 2 traces in there. 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 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. ________________________________ From: Michael Jerris <mike@jerris.com> To: freeswitch-users@lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike,
My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541
Thank you, Dorn B.
________________________________ From: Michael Jerris <mike@jerris.com>
To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite
are you doing this trace from the freeswitch box itself?
Mike
On Dec 17, 2009, at 10:48 AM, DJB wrote:
Anthony,
I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info.
Thank you.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thank you Mike for your suggestion on IRC. We did what you recommend and found out it's the iptables issue that we thought it was not there at the beginning since we saw the first 2 invites from the far end fine, but somehow it has something to do with the 3rd invite. I did close the Jira that I thought it was a bug. Thank you again for the community and your support. Dorn B. ________________________________ From: Michael Jerris <mike@jerris.com> To: freeswitch-users@lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike,
My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541
Thank you, Dorn B.
________________________________ From: Michael Jerris <mike@jerris.com>
To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite
are you doing this trace from the freeswitch box itself?
Mike
On Dec 17, 2009, at 10:48 AM, DJB wrote:
Anthony,
I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info.
Thank you.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I'd be suspicious of: (a) the CSeq going backwards from 4 to 3 between re-invites 2 and 3; (b) the branch on the Via tag changing (c) (not sure about this one) the SDP session ID and version changing for what's the same session. --Dave
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@gmail.com> To: freeswitch-users@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@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.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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? Mike 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 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.
Is the packet capture running on the FS box itself? On Thu, Dec 17, 2009 at 9:36 AM, Michael Jerris <mike@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?
Mike
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 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.
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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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
It only happened to the calls from this customer that keeps sending re-invite every 30 minutes, since their switch is expecting a reply back from those re-invite and FS did not respond back to those re-invite. Thank you. ________________________________ From: Michael Jerris <mike@jerris.com> To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:36:44 AM Subject: Re: [Freeswitch-users] SIP Re-invite 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? Mike 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 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.
participants (5)
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Anthony Minessale
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David Knell
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DJB
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Michael Giagnocavo
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Michael Jerris