Calls from FXS Analog MGCP Gateway to CallManager IP Phone Failing

[Bryan Geoghan]

Hi,

I have been testing a scenario for a customer and have ran into this issue.

We have a remote site with a PBX that connects into a gateway via 4 FXS ports. The gateway is using MGCP signaling. Our intention is to use G729 between CM and the remote site. We are using CM version 6.1(3).

Both the remote site gateway and IP Phone are in the same Device Pool thus same Region. I have it set to do G729 intra and inter. However, when I place calls from either direction, the call connects but I hear no audio. The gateway is showing G729 when I do a 'show voice call status', 'show voice call summary', 'show voice port summary'.

Do I need a transcoder for this? If so I have tried that with the following configuration:

voice-card 0

dspfarm

dsp services dspfarm

sccp local FastEthernet0/0

sccp ccm 172.19.72.220 identifier 2 priority 2 version 6.0

sccp ccm 172.26.24.237 identifier 1 priority 1 version 6.0

sccp

!

sccp ccm group 10

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 10 register TRN001a6ca1f908

!

dspfarm profile 10 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 6

associate application SCCP

I added the resource in CallManager as a Cisco IOS Enhanced Media Termination Point. I'm not certain if I was to make it that or a regular 'Cisco IOS Media Termination Point'.

I added the transcoder resource to an MRG and MRGL and assigned it to the MGCP Gateway in CallManager.

The voice gateway is a 2811 with PVDM2-16 ( 16-channel (G.711) Voice/Fax PVDMII DSP SIMM PVDM daughter card).

When I change the region to be G711 intra, the call works and I hear audio. When I switch it back to G729, the cal completes but I hear no audio. It definately sounds like a codec issue.

I don't completely understand DSPs very well or how the endpoints choose their transcoder.

[Nicholas Matthews]

This is a little odd.

You only need transcoders when you have two IP devices that have a mismatch in codec capabilities. Common example of this are calls to Unity where it only supports G.711 but the regions only allow remote phones to do G.729.

In this case, both the gateway and the IP phone should be capable of doing G.729, and a transcoder isn't necessary.

I would say that by having a transcoder or MTP in this call flow you will increase the likelihood of audio issues.

Maybe you have a scenario that is requiring an annunciator, and the annunciator only supports G.711. I suppose in this case your transcoder should be able to take care of that.

I would double check your configuration to see if it's pulling in any additional resources and see if they're mucking it up.

A simple G729 call from the gateway to the IP phone should work.

Check out this doc on one-way voice and see if your packet counters are increasing on the IP phone and gateway.

hth,

nick

[MARTIN STREULE]

If the call connects you have no codec problem.

You probably have an IP connectivity problem. If there is a codec problem, the call hangs up as soon as it tries to connect (go offhook).

-> do you have a Firewall/ACLs?

-> if you have more then 1 interface: did you bind media to an interface?

Cheers,

Martin

[Bryan Geoghan]

Then why would the call work when I set the region to use G711 but not G729?

[wizardsmoke]

I'm having a similar issue. On 3 out of 5 calls I get one way audio with the recvr and sender codec showing "None" on the call statistics. Strange thing is that I can hear dtmf tones but nothing else. Even stranger, if I put the call hold and then resume...everything starts working. No firewalls or ACLs. G.711 works fine. G.729 works fine when I dial IP phone to IP phone. The problem only occurs when I try to place a call to the PSTN using g.729.

[mightyking]

What is the out put of the "show sccp" command? Do you see State "Connected" and Status "Active"?

I recommand to reset the region and the Gateway from the CUCM administration page and from the router:

router#no sccp

router#sccp

MK