SIP BYE Delay CUBE and CUCM 9.1

[Robert Shaw]

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

[Robert Shaw]

Hi Manish,

I think you're onto something.  To test the theory I temporarily removed DP 9011 and it works as expected.  Although I still don't get ringback until about the third ring of my mobile phone.  However, when I hang up both sides clear down at the same time.  So it is just the question now of separating the inbound and outbound calls.  This will be tricky as we both use the e.164 numbering scheme.

Thanks a lot for spending the time to go through the Outputs and you're assistance in coming to the right conclusion.

Kind Regards

Rob

[Manish Prasad]

To find out where exactly delay is happening we need to do some packet capturing.

Please enable packet capture on internal and external interfaces of CUBE and also on CUCM where phone is registered to.

We can check the RTP stream (ringback) after the exchange of 180/183 message from ITSP , if the RTP delay is coming from Service provider then you can contact provider for this or if the delay is happening at CUBE internal interface or CUCM then we need to look at CUCM traces.

Just for testing can you enable MTP required on sip trunk and check the ringback.