calling to Trunk SIP, I not hear the first 10 seconds of audio

[gsilva]

Hi guys,

I have problem with the calls to a trunk SIP. In some calls, i not hear the first 10 seconds of audio.

Scenario

PSTN--->GW MGCP---->CCM--->Trunk SIP---PBX SIP with IVR

The trunk SIP is using MTP (MTP checked).

The capture show a parameter a:inactive

Please your help

Thanks a lot

[George Thomas]

What version of CUCM are you running? There is asetting in the SIP Profile: SIP Rel1XX Options. I would modify this setting for the SIP profile that is assigned to the SIP trunk. If CUCM is 7.x, then this option is under the Service parameters.

[gsilva]

Hi Geroge,

CUCM 8.5, I tested with the diferents options in the sip profile SIP Rel1XX Options, but the problem continue.

I attach, the capture from cli cucm.

Here the call fail, I not hear after 12 seconds

[Rajkumar Yadav]

Hi,

Could you please use the Early offer or the Early media for the call.

Also please share the log for affected call.

Thanks.

Raaj.

[gsilva]

Hi,

I try use Early offer (unchecked the MTP) and the error is solved but, with this way the DTMF is problem.

PSTN- GW MGCP- CCM- SIP Trunk- Third device SIP.

I Called and Hear a IVR,  option 1 and next request a ID( 7 number) but this digits not are received.

How set Early media ?

I attach the capture, is a file .cap (please change the name).

Thanks a lot

[Rajkumar Yadav]

Hi, 

MTP should be checked since the third party SIP phone support only RFC 2833 DTMF method and DTMF RFC method should be RFC 2833 OR OOB & RFC 2833.

(SCCP Phone support both inband and out of band)

MGCP would be configured to support OOB.

From the SIP profile configuration you can select the early offer mode for all calls.

For Early media it requires a PRACK message (SDP answer) as response to  183 session in progress (containing SDP offer).

I think as from the setup as you presented it should work fine with MTP checked, DTMF method as RFC 2833 and from SIP profile it should have Early offer.

I would like to see the SDI or SDL traces for that particular call.

Even I have the same setup for one of my client (CUCM with Genesys Contact center).

I can provide you the SIP Trunk working configuration.

One query from your attached text file, if you say MTP unchecked then the content length should had been 0 (i.e no SDP parameter).

Please clarify.

Kind Regards,

Raaj.

[gsilva]

Raa,

Please could you provide you SIP Trunk working configuration.

My email: guille.sb99@gmail.com

Today I will obtain the SDI SDL Traces with new call.

Remember the problem is, some calls I Hear the audio record(greetings of the contact Center, after 10 seconds) and the other calls I hear all the record audio.

So, I analize the capture and see that in the call with the fail, the invite include this parameter

Media Attribute (a) : inactive

and the 200 Ok from the third device SIP side,  a

connection information (c) : IN IP4 0.0.0.0

Do you think the problem could be in the CCM Side ?

About you query, The attached file, have MTP checked.

When I uncheck the MTP the problem is with the DTMF, but I don´t have this capture.

Thank a lot for your help

[Rajkumar Yadav]

Hi,

When i was facing somewhat similar issue at that time the Genesys was the culprit.

So it could be Genesys side issue (guess).

for your fail call issue 

Media Attribute (a) : inactive  // means the audio disconnect, usually when you put some one on hold then this parameter happens

and the 200 Ok from the third device SIP side,  // other side accepting the same

connection information (c) : IN IP4 0.0.0.0 // this is part of the media attribute (a) .

Could please check show call active voice brief  in router and check the rtp for send and receive for that particular call increases or stopped.

Kind Regards,

Raaj.