MRGL selection questions

[marco_81]

Hi all,

i have some questions regarding MRGL selection when SIP trunk is involved in the call. Here is the call flow:

PSTN --> MGCP Voice Gateways (two 3845) --> Call Manager 10.x --> SIP Trunk (dtmf no preference) --> Third party IVR 

These are the constraints of live environment:

- inbound calls from PSTN are delivered to MGCP voice gateways in a round robin fashion

- MGCP voice gateways are configured to work with OOB DTMF method  

- Third party IVR only support RFC2833 inband method

When a call is made from PSTN to IVR, Call Manager claims MTP resource to accomplish DTMF mismatch (OOB to RFC2833).

Now let's have a look at two possible configuration scenario:

First configuration case: PRI on MGCP gateway is configured with MRGL that contains software MTP provided by CallManager. The call setup is successful , but IVR does not recognize DTMF, so i suppose DTMF translation does not take place.

Second configuration case: same as previous, plus the same MRGL used by PRI applied to SIP trunk. The call setup is successful , and IVR recognize DTMF, i suppose DTMF translation takes place.

I don't understand why the calling device cannot allocate MTP resource. Is it an expected behaviour?

Is it possible to set MGCP gateway to exchange both RFC2833 and OOB to overcome the needed DTMF translation?

regards 

[Jaime Valencia · Accepted solution]

Because resources are invoked by the device that requires them, at the point in time at which the call arrives at the MGCP GW, there is no need for the MTP, once the call is sent to the IVR, that's when the DTMF mismatch is observed, and that SIP trunk is the one that will try to invoke the MTP to translate from OOB to 2833.

[marco_81]

Thanks for reply, it's clear. Another question related to this call flow:

supposing MGCP gateways receive inbound calls in a round robin fashion (gw1 receives first call, gw2 receives second call, end so on), do you see any trouble if a configure SIP trunk to use MTP software residing on both mgcp gws?

I'm a little bit confused about what would happen if a call arrives at gw1 (from PSTN) and the sip trunk select mtp on gw2, for example.

Does it make sense this question?

[Jaime Valencia]

Nothing will happen, the voice will simply go from VG1 to MTP on VG2 to SIP trunk.