Call Failed -- Error : ACK, dest SDP null

[Akash Sharma]

Hi All

Can someone please explain what does it mean by ACK, dest SDP null  .

The explanation I found is :

 An acknowledgement request was recieved but the destination SDP Body was unavailable for the delayed media call.

The actual error I am getting is :

Result: Call Failed. Ringing, not able to answer. Multiple Invite Message

Error: %VOICE_IEC-3-GW: SIP: Internal Error (ACK, dest sdp null): IEC=1.1.96.7.25.0

The environment for the call is like :

Site A ----> Cluster A ----------------> SIP Trunk ---------------> Cluster B

Does it mean that The destination body is not reachable or there is any call routing thing missing ?

Please help me to make it understand more clear.

Regards

[Vivek Batra]

Seems SDP answer should be there in ACK for SIP delayed offer, but it didn't.

Can you please share the SIP traces between the clusters to check further?

- Vivek

[Akash Sharma]

Hi Vivek

Thanks again for the reply !!

But sorry for the inconvinience as I cant have the SIP Traces as I am working as L1.

But what I can understand from Internal Error (ACK, dest sdp null) is that may be the call routing is not properly defined between the clusters or may be there is something missing w.r.t. the call routing. Can it be the reason ?

[Vivek Batra]

I doubt the issue is because of call routing configuration since the error seems at SIP layer. If you've access to RTMT, you can simply get the SIP traces from either cluster and we can quickly identify the issue.

Please check the following document and error code 25 under SIP section which clearly stated that SDP was expected in delayed call but it was unavailable.

http://www.cisco.com/c/en/us/td/docs/ios/voice/monitor/configuration/guide/12_4/vt_12_4_book/vt_voip_err_cds.pdf

- Vivek

[Akash Sharma]

Hi Vivek

Please find attached the Traces for your reference.

Regards

[Vivek Batra]

Akash - I think you've not enabled SIP traces in unified serviceability hence full SIP debugs are not there. Please enable SIP level traces and pull the debug again.

- Vivek

[Akash Sharma]

HI Vivek

The SIP Traces is enabled. PFA the screenshot for your reference.

[Vivek Batra]

Well in that case, you can go to RTMT -> Call Manager -> Real Time Data, double click on desired call and it will show you the SIP flow. Copy the SIP traces and share the same.

- Vivek

[Akash Sharma]

Hi Vivek

I am getting no record for the same call in RTMT.

Is there any other way to troubleshoot the same ?

[Vivek Batra]

There is no reason not to have this record in RTMT if you're following the correct procedure.

First make a test call. Login to respective call manager node and under real time data, set the date/time couple of minutes before of test call. Add 30 minutes capture. Please note the current date/time from CUCM CLI using 'utils ntp status'.

Also take traces from calling cluster instead of called cluster.

- Vivek

[Akash Sharma]

Hi Vivek

Please find attached the Traces for your reference.

Regards