Issues with SIP trunk calls

[shenbagarajan]

Hi all,

I am facing dtmf issues with ATA connected phones while calling a conference number through SIP trunk and then dialing the passcode for the conference. But the Analog phone is not recognising the dtmf digits dialed during the call. But when MTP required checkbox is checked on SIP trunk, the issue is resolved. But calls r dropped during when the call rate is high even though i have enough MTP resources. When mtp unchecked in sip trunk calls r working fine. Is there any other way to solve the dtmf issue without mtp checked on sip trunk. Pls suggest.

[Priyadarshini B.T · Accepted solution]

As I have understood, the issue is, that you have queries for MTP requirement on SIP trunks.
Firstly just a brief description for as to when we require the Media termination point checked on the SIP trunks.
There are two ways a call can be established using SIP trunks one is the early media and the other is delayed media.

In case of early media the session description goes out with SIP invite rather than waiting for the ringing message.

The MTP check box is only valid for the outbound calls.
1. If we use SIP trunk with early media, we are considering the configuration should be as follows. 

MRGL including MTP should be assigned to the SIP trunk.
"Media Termination Point Required" should be checked if DTMF interworking between inband RFC2833 and OOB doesn't occur.
"Media Termination Point Required" doesn't need to be checked if

DTMF interworking between inband RFC2833 and OOB occurs.

If you want the SIP trunk to send INVITE(with sdp) then all you have to do is check the "MTP Required" flag on the SIP trunk config page
and make sure MTP's are accessible. If you defined a MRGL that contains a MTP then that should be fine. This config arrangement will force the
RTP stream to go through the MTP. 

2. If we use SIP trunk with delayed media, we are considering the configuration should be as follows. 

MRGL including MTP must not be assigned to the SIP trunk.
"Media Termination Point Required" should be unchecked if DTMF interworking between inband RFC2833 and OOB doesn't occur.
"Media Termination Point Required" should be unchecked if DTMF interworking between inband RFC2833 and OOB occurs.

please find the link below for further information,

http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1123292
 

hope this helps

[shenbagarajan]

Hi Priya B.T.,

Thanks for your suggestions. Can u pls tell me how can all the SIP trunk calls be configured to use delayed media as it doesnt reuire MTP as per ur suggestion ?

[Chad Stachowicz]

Hello,

   Although Priya is correct to some extent, it ultimatly depends on your versions of IOS that you are using whether that checkbox is going to be required for DTMF to work.

I have been doing this an awfully long time and in alot of IOS trains you need to check that box to get things working.

Can you let me know your CUCM version and Gateway IOS version?

These MTP fixes I believe came in like 12.4.20

Chad

[shenbagarajan]

Hi Chad,

Thanks for ur reply. CUCM version is 7.1.3 and the SIP trunk truncates on an SBC.

[Chad Stachowicz]

what is an 'SBC'  Is this a cisco device, if so what version?

If not are you saying that the SIP trunk comes to CUCM direct from the carrier?!?

Chad

[shenbagarajan]

Its a Session Border Controller.. I will find the version and reply.

[shenbagarajan]

Its not a cisco product. Its a third party session border controller.

[Chad Stachowicz · Accepted solution]

so ultimatly your in a bind here..

The calls are gonna need MTP checked in order to use DTMF, because your not using an IOS voice gateway to terminate the SIP Trunk (the fixes for MTP lie in the IOS not CUCM)\

CUCM's built in MTP's always suck ass, thats why you have random failurs and drops..

Ultimatly your gonna need a cisco gateway somewhere with MTP resources in it, solely for your SIP trunk to act right.

Just change our that SBC with some Cisco router.

Good luck!

Chad

[shenbagarajan]

Thanks Priya & Chad for your help.

We will probably add some media resources or will change the SBC to cisco router.