11-21-2014 03:57 AM - edited 03-17-2019 01:02 AM
Hi Folks,
I have an issue on a SIP Trunk, running CUBE 15.2(3) and CUCM 9.1. When I call in from a mobile and perform either a called-party release or a calling-party release the call disconnects both sides normally.
If I call out from an IP Phone to a mobile and again perform a called or calling-party release the opposite side of the call takes 10 seconds to clear.
The two attached debugs start from the moment the call is ended. On the Inbound call you can see clearly the BYE and ACK messages from both call legs tearing down the call. However, on the Outbound call there are a few 200 OK messages sent from CUCM to CUBE before the BYE messages start. Once the BYE messages are sent the call clears, but it takes 8-9 seconds for them to start from when the call is cancelled on either end.
Does anyone have any idea where to start looking for this delay?
Thanks in advance for any help gratefully received.
Rob
Solved! Go to Solution.
11-25-2014 12:57 AM
Hi Robert,
There are many sip messages missing like second INVITE from outside interface to Service provider , Trying , Proxy authentication acknowledgement with new INVITE.
When i look at your CUBE config , the intended dial-peer for outbound call is 9000 but the logs shows matching with DP 9011 (due to answer address preference over destination pattern).
Nov 24 17:01:24.482: //-1/842E0F800002/SIP/Info/sipSPIGetCallConfig: Peer tag 9011 matched for incoming call
I think due to wrong DP match many SIP messages are missing , can you separate inbound and outbound dial-peer from CUCM and ITSP.
And one more change i would ask you do to uncheck the "Run on all active CM nodes" from sip trunk because it do create complication when its configured incorrectly. Check this link.
Thanks
Manish
11-21-2014 07:16 AM
Hi Rob.
Can you please post your CUBE config?
Thanks
Regards
Carlo
11-21-2014 07:57 AM
11-21-2014 08:56 AM
Hi Rob.
From your debug I can see that CUCM does not send BYE to CUBE infact the BYE sent to remote host is with cause code 86 (network timeout)
Can you check trunk config on CUCM?
Let me know
Regards
Carlo
11-24-2014 03:43 AM
Thanks for your reply Carlo. Can you suggest the settings I should be looking at? Would it be the Trunk itself or the SIP Profile that may be at fault. I have double checked it with other SIP Trunks I have configured and the settings seem very similar, however this is the only CUBE Trunk I have worked on.
11-24-2014 05:53 AM
Hi Robert.
Can you please post a screenshot of your trunk configuration?
Thanks
Regards
Carlo
11-24-2014 06:03 AM
11-24-2014 07:07 AM
Did you tried to trace it why this error is coming ?
Nov 21 10:54:36.710: %IP-4-DUPADDR: Duplicate address ext.ext.ext.ext on GigabitEthernet0/1, sourced by 00a0.8e7d.a74b
11-24-2014 07:52 AM
Hi Manish,
Yes we have traced it and it appears to be a false alarm. The external range is assigned to our Checkpoint Firewalls, that is where the message is being generated from despite the fact that nothing is assigned to this particular IP Address on the Checkpoint side. Despite the message I have no problems with external connectivity.
Thanks
Rob
11-24-2014 08:15 AM
Do you observe delay in call setup for an outbound call from an IP phone ?
When the call is setup (connected) , the contact header from SIP message is used to route further sip messages whether its BYE, UPDATE,NOTIFY or anything.
I need to look at SIP transport messages , can you please collect "debug ccsip all" for an outbound call and attach it here. You can sanitize important ip addresses.
11-24-2014 08:31 AM
Hi Rob.
In your Trunk configuration, I see that you defined a new sip trunk security profile.
What is the difference with the default one?
Please let us know.
Regards
Carlo
11-24-2014 09:15 AM
Hi Carlo,
The only difference between my Customised SIP Profile and Default is "Early offer support for voice and video calls (insert MTP if needed)" is selected. I was advised to do this by my SIP Provider.
Thanks
Rob
11-25-2014 04:12 AM
Hi Robert,
CUCM send remote-id by default and a lot of ISP don't like this attribute, try to remove it in the CUBE, under sip-ua
no remote-party-id
Outbound:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.2.52:5060;branch=z9hG4bK123e49758524e6
From: "OrbTalk 0100" <sip:441216550100@10.0.2.52>;tag=2664251~558b08dc-be7e-476e-8dc0-eeef8522c7f1-50169393
To: <sip:447795826135@10.0.0.2>;tag=334E2454-12DC
Date: Fri, 21 Nov 2014 10:54:17 GMT
Call-ID: bbd35800-46f119d9-30a29-3402000a@10.0.2.52
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:447795826135@ext.ext.ext.ext>;party=called;screen=no;privacy=off
Contact: <sip:447795826135@ext.ext.ext.ext:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.3.T1
Session-Expires: 7200;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 191
Inbound :
Nov 21 10:57:10.910: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.20.39.101:5060;branch=z9hG4bKa069.ae50e615.0
From: "447795826135" <sip:447795826135@sip.orbtalk.co.uk>;tag=KmsqOjIwOC43Ni4xNi43ODo0MDgx
To: <sip:441216550100@sip.orbtalk.co.uk>;tag=33508C10-151C
Date: Fri, 21 Nov 2014 10:57:10 GMT
Call-ID: hs216_b6a256b74-75baded8-2c60967c-d207@10.16.171.4
Server: Cisco-SIPGateway/IOS-15.2.3.T1
CSeq: 2 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=456,OS=72960,PR=446,OR=71360,PL=0,JI=0,LA=0,DU=9
Content-Length: 0
11-24-2014 09:10 AM
Hi Manish,
Please find full "ccsip all" call flow output attached. I don't see any delays in call setup, however, the strange thing is that I don't get ringback while the call is being setup. I get a brief ringback tone, but then silence. When I answer I have full stream, though.
Thanks
Rob
11-25-2014 12:57 AM
Hi Robert,
There are many sip messages missing like second INVITE from outside interface to Service provider , Trying , Proxy authentication acknowledgement with new INVITE.
When i look at your CUBE config , the intended dial-peer for outbound call is 9000 but the logs shows matching with DP 9011 (due to answer address preference over destination pattern).
Nov 24 17:01:24.482: //-1/842E0F800002/SIP/Info/sipSPIGetCallConfig: Peer tag 9011 matched for incoming call
I think due to wrong DP match many SIP messages are missing , can you separate inbound and outbound dial-peer from CUCM and ITSP.
And one more change i would ask you do to uncheck the "Run on all active CM nodes" from sip trunk because it do create complication when its configured incorrectly. Check this link.
Thanks
Manish
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