[cisco-voip] SIP trunk and is MTP Required does not need to be checked

Tue Nov 23 18:43:39 EST 2010

So I am here to attest that for the following topology you don't need Device > Trunk > MTP Required checked. It's usually a bad thing to require all calls use a MTP and a workaround for those too lazy to debug and find out why/what is broke or mis-configured, gather/analyze ccmtrace files.

I have this working in production.  You have to make sure your voice-class codec or codec doesn't require a MTP/Transcoder, that your phone/conference bridge resource are in same DP (here I have the router as a registered sccp conf bridge).  I'm advertising G722 and using that phone-phone internal and G711 to gateway. Since it's a PRI  on Gateway the dtmf on phones and gateway is rtp-nte (RFC2833) thus no resource need.

SIP 7900 Phone Load 9.0.3S  > CallManager 8.03a-------SIP Trunk--------IOS 151-2.T2-----PRI----PSTN

The root cause of why my calls initially were not going out without a MTP was found in the configuration error on the gateway's running-config, the preference level was set for 722 which invoked the MTP resource (which would cause SIP SRST call preservation to fail)
voice class codec 1
codec preference 1 g722-64
codec preference 2 g711ulaw
codec preference 3 g711alaw
codec preference 4 g729r8
codec preference 5 g729br8

From: Mark Holloway [mailto:mh at markholloway.com]
Sent: Monday, November 15, 2010 11:21 AM
To: Bill Riley
Cc: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP trunk one way audio

That would make sense.  My experience is based on 7.x and it sounds like Jason had a similar experience with 8.x. Sometime in the next couple of weeks I will try to isolate the differences and get to the bottom of it.

On Nov 15, 2010, at 9:00 AM, Bill Riley wrote:

I am not saying you don't need an MTP. I am saying you don't need to use the MTP required check box. If an MTP is required it should allocate one from the MRGL.

From: Mark Holloway [mailto:mh at markholloway.com]
Sent: Monday, November 15, 2010 9:59 AM
To: Jason Aarons (US)
Cc: Bill Riley; cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] SIP trunk one way audio

Assuming you aren't using a SIP phone load, then when generating a DTMF tone from the SIP trunk towards a SIP provider and using an out of band method, the trunk needs the resource to generate the out of band tone. In CUCM 7 if you don't allocated the MTP to the SIP trunk DTMF does not work. I have not tried to do this by strictly relying on CUBE, which may supplement it, but it may also depend if you are using SDP transparency.

On Nov 15, 2010, at 8:49 AM, Jason Aarons (US) wrote:

If using RFC2833 (rtp-nte) which both 7900 SIP LOAD and IOS and Unity support, why is MTP Required checkmark required technically? What's happening that needs MTP?

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mark Holloway
Sent: Monday, November 15, 2010 10:34 AM
To: Bill Riley
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] SIP trunk one way audio

Unless you are using inband DTMF it will be required.

On Nov 15, 2010, at 7:07 AM, Bill Riley wrote:

I don't think that's correct. It needs access to an MTP but you shouldn't need to use the MTP required checkbox on the trunk.

From: Mark Holloway [mailto:mh at markholloway.com]
Sent: Friday, November 12, 2010 12:24 PM
To: Bill Riley
Cc: 'Ryan Ratliff'; cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] SIP trunk one way audio

CUCM needs it assigned to the trunk for DTMF to work properly for calls egressing the SIP Trunk.

On Nov 12, 2010, at 10:50 AM, Bill Riley wrote:

I don't think that is correct. It should only need to have one available in the MRGL, not one allocated every time a call comes in.

voice service voip
ip address trusted list
  ipv4 x.x.x.x
  ipv4 x.x.x.x
  ipv4 x.x.x.x
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
modem passthrough nse codec g711ulaw
sip
  bind control source-interface Serial0/0/0:1
  bind media source-interface Serial0/0/0:1
  early-offer forced
  midcall-signaling passthru
!

voice class codec 100
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
dial-peer voice 201 voip
preference 1
destination-pattern 91[2-9]..[2-9]......
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp cs3 signaling
!
dial-peer voice 202 voip
preference 1
destination-pattern 9[2-9]......T
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp ef signaling
!
dial-peer voice 203 voip
preference 1
destination-pattern ^9556[2-9]......
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp ef signaling
!
dial-peer voice 204 voip
preference 1
destination-pattern 9555[2-9]......
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
fax rate 14400
ip qos dscp ef signaling
!
dial-peer voice 411 voip
preference 1
destination-pattern 5555555..
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp cs3 signaling
!
dial-peer voice 412 voip
preference 2
destination-pattern 5555555..
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp cs3 signaling
!
dial-peer voice 413 voip
preference 3
destination-pattern 5555555..
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4: x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
fax rate 14400
ip qos dscp cs3 signaling

From: Mark Holloway [mailto:mh at markholloway.com]
Sent: Friday, November 12, 2010 11:32 AM
To: Bill Riley
Cc: 'Ryan Ratliff'; cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] SIP trunk one way audio

You should have it on the trunk.

On Nov 12, 2010, at 10:29 AM, Bill Riley wrote:

Your right, but I don't need to have the check box to require one on the trunk. It should allocate one from the MRGL on an as needed basis.

From: Mark Holloway [mailto:mh at markholloway.com]
Sent: Friday, November 12, 2010 11:28 AM
To: Bill Riley
Cc: 'Ryan Ratliff'; cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] SIP trunk one way audio

You should be using an MTP for your SIP trunk to support DTMF. It does not need to be a hardware MTP resource.

On Nov 12, 2010, at 10:05 AM, Bill Riley wrote:

When I change it to MTP required on the sip trunk everything works as expected. The point is that I shouldn't need to have this configuration.

From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Friday, November 12, 2010 9:00 AM
To: Bill Riley
Cc: 'Cheng, Karen'; cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] SIP trunk one way audio

I see lots of discussion around config and call flow but have you actually done any troubleshooting?  One way audio most of the time comes down to the simple fact that one party is not receiving RTP from the other.   For these one-way audio calls you need to determine what IP addresses are involved.  Next verify this in the signaling via the SDPs in the SIP messages.   You can then use show commands on the router to confirm where it thinks it should be sending and receiving RTP to/from and if in fact packet counters are incrementing.

If no MTP is being used for the call, try forcing it to use one and see if that fixes the issue.
If you are using media flow-through (default) does changing it to flow-around fix the issue?

-Ryan

On Nov 12, 2010, at 8:09 AM, Bill Riley wrote:

I shouldn't need an MTP for this connection. All SIP traffic is sourced from one interface.  I do have SCCP traffic sourced from a different interface but it is only used for a conference bridge, not MTP.

From: Cheng, Karen [mailto:Karen.Cheng at racq.com.au]
Sent: Thursday, November 11, 2010 9:14 PM
To: 'Bill Riley'
Subject: RE: [cisco-voip] SIP trunk one way audio

Not sure if you have checked already but is your SIP trunk using one interface and your MTP/SCCP interface a different interface?

I had one-way audio/no audio problems ages back due to this because our integrator had configured the SIP trunk to point to int gi0/0's IP and then configured the SCCP interface to loopback0.

Regards

Karen Cheng
Voice Network Engineer

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley
Sent: Friday, 12 November 2010 2:15 AM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] SIP trunk one way audio

I have a new SIP trunk terminating in a 2921 CUBE bundle. When I call in from the Trunk directly to an IP phone it works correctly. If I call from the trunk to IP phone and the IP phone transfers the call without waiting for the remote party to answer I get one way audio. From reading this looks like and MTP issue but I have an MTP set in the MRGL for the trunk.

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