No Audio On Supervised Call Transfer After Multicast MOH on Cube SIP

[cnorton]

Hi Guys,

I hope someone can help me in our issue with our Cube router.

We recently migrated from PRI to SIP. We just learned that MOH from the flash will not work on a SIP trunk in a cube environment. We want to enable multicast moh since unicast might saturate the bandwidth in our datacenter. So we upgraded our 2951 router ios to 15.2 version - c2951-universalk9-mz.SPA.152-4.M10.bin. We were able to make the multicast moh to work, however new issues arised. When we put the call on hold and resume, the audio dissappears. Same thing when we do a supervised transfer (click transfer on the phone > caller will hear moh > dial the number > click transfer),caller will hear nothing as soon as we transfer the call.

I was able to resolve the first  issue by creating a new sip profile and enable Early Offer support for voice and video calls (insert MTP if needed) and Require SDP Inactive Exchange for Mid-Call Media Change. So when i put the caller on hold, once i resume, audio is still fine. However, i still cant find any solution on the transfer scenario. I tried enabling other check boxes on the sip profiles but no luck. What puzzles me is when i use unicast moh,the problem gets resolved. So it looks like the issue has something to do with the multicast moh 

We are usign CUCM version 8.5 and below is a simple diagram of one of our centers.

ITSP > SIP > Cube >>vpn>> CUCM

We have 100+ sites connected to a single Datacenter where our CUCM is located so as much as possible we dont want to use unicast moh. 

I attached the running config of the router and sip debug.

I hope someone can check this one and help me.

Thanks in advance!

[Dennis Mink]

Can you add the ccsip messages debug of such a call to this post?

Please rate if helpful

[cnorton]

Hi Dennis,

Thanks for your reply. The SIP debug is already attached on my first entry (mmoh-noaudio.txt).

One more thing that i noticed, the moment the call is transferred, i still see the multicast ip 239.1.1.1 on one of the call legs. So its possible the hold session did not break when the call is transferred hence causing the audio to get lost.

***RTP when transfer button is pressed (caller will be put on hold)

VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 121668 121669 17680 6016 216.110.90.193 8.45.210.140
2 121669 121668 17682 16384 10.144.20.6 239.1.1.1

***RTP when the remote party(where the call will be transferred) answered the call

VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 121668 121669 17680 6016 216.110.90.193 8.45.210.140
2 121669 121668 17682 16384 10.144.20.6 239.1.1.1
3 121685 121686 17684 24586 10.144.20.6 10.254.253.12
4 121686 121685 17686 6018 216.110.90.193 8.45.210.140
Found 4 active RTP connections

***RTP when the call has been transferred successfully

VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 121668 121669 17680 6016 216.110.90.193 8.45.210.140
2 121669 121668 17682 17690 10.144.20.6 10.144.20.6
3 121685 121686 17684 17688 10.144.20.6 10.144.20.6
4 121686 121685 17686 6018 216.110.90.193 8.45.210.140
5 121687 121689 17688 17684 10.144.20.6 10.144.20.6
6 121688 121689 17690 17682 10.144.20.6 239.1.1.1 <<<<<<<< caller is no longer on hold yet the multicast ip is still present

Found 6 active RTP connections

[cnorton]

up

[Rajan]

Can you check whether the option "Send send-receive SDP in mid-call INVITE" is enabled in the SIP profile. 

From the logs, could see the CUCM still sending a=inactive after resume and not send-recv.

HTH

Rajan

Pls rate all useful posts

[cnorton]

Hi Rajan,

Thanks for your reply. I did not enable that option. Once "Send send-receive SDP in mid-call INVITE" is enabled, the first problem (putting caller on hold then resume = no audio) doesnt get resolved. 

Also, it does work when its unicast moh and not multicast, therefore telco is allowing a=inactive in the sdp.

[cnorton]

Any suggestions guys?