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One way Audio 9971 to PSTN (MGCP) PRI


I am having a one way audio issue with 9971's running the latest version of firmware when dialing out to

the PSTN.  IP Phone calls PSTN, and user on PSTN side can hear the caller but the IP phone user cannot

hear the called party.

This only happens when we have the "MTP Required" check box selected on the 9971.  If we uncheck the

box we get audio both ways but video internally breaks.


9971 (SIP) >> CUCM 8.6 >> MGCP/PRI >> PSTN User   

I do have the bearer-cap command configured on the PRI voice-ports. 

Any help is appreciated.

~ Rob Oliveira ~
Hall of Fame Master

Why do you enable MTP?

Why do you enable MTP?  Ideally you would not so that unless you have specific reasons to.

If you do need to use MTP, what MTP are you using CUCM based, IOS hardware or IOS software?

In either case you need to make sure that the MTP device is routable from IP phone and voice GW.


To add a few thoughts to

To add a few thoughts to Chris's very good reply;

A brief note on "MTP Required"

The reason for your one-way audio, specifically, is because when you check "MTP Required", the phone is being assigned an MTP resource from the configured MRGL (will use the default NULL MRGL if one is not specified) for media termination and the IP address of the IP phone is likely not able to fully communicate with the IP address of the MTP resource that is being selected (look for possible asymmetric routes and check all your ACL's between the IP phone and the MTP resource).

If you are using the software MTP resources of CUCM, make sure the network segment of the IP phone can communicate to ALL nodes of the cluster that has the IP Streaming Voice and Media Application running on it (not just the node the phone is registered to ... i.e a cluster with nodes in different network segments). If you are using MTP resources in the IOS configuration of a Cisco Router, again, make sure the network segment of the IP phone can communicate to that router.

While voice will work just fine with MTP, you should consider what "MTP Required" is actually doing. It is invoking an MTP resource for EVERY connected call to terminate the media stream. In cases of low volume clusters this may not be a huge concern but in every cluster I've ever worked with, there was enough volume to make "MTP Required" a BIG deal.

If you are using the MTP resources on CCM (I wouldn't recommend that), you should consider the overhead and load this will place on your cluster. If you are using IOS resources, you should again consider the overhead this places on the router.

The only place I have ever used "MTP Required" and actually had a justifiable reason for doing so was a service provider network that did a bunch of interoperability connections; outside of that -there is usually a better way than "MTP Required".

Internal Video Breakage without MTP Required

Can you define this a little more? One-way video or no-way video? Are you just doing point 2 point video (phone to phone)? Are you traversing a SIP trunk between internal endpoints?




Thanks for the thorough

Thanks for the thorough response and we do not want to use MTP and shouldn't need to.

We were only testing around a bit and found that the only way that we could get audio to work from a 9971 to the PSTN is by checking that option.

When we make an internal call (IP phone to IP phone) both audio and video work without the MTP option checked.

So really the only issue we have is internal (9971) to PSTN without MTP checked we get one way audio.  With MTP checked in this scenario audio works but breaks video on the internal calls.

Ideally we do not want to have to use the MTP but I do have a IOS device acting as a DSP farm

with plenty of resources and MTP/Conference configured and the environment is fairly small.  

No SIP trunk.  The only SIP configuration are the 9971's configured on CUCM.

Outbound Call to PSTN

9971 SIP to CUCM > MGCP Gateway >PRI > PSTN

~ Rob Oliveira ~
VIP Mentor

Can you do a test call and

Can you do a test call and send us the ff from the gateway. please include calling and called number

debug mgcp packet

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