[cisco-voip] Special considerations for IAD or PRI network-emulation on Cisco hardware...

[Jeff Anderson]

I have a couple of customers who are experiencing poor audio using an IAD to
connect their PBX to our h.323 PSTN GW.

Basic Layout.

Customer PBX PRI > IAD PRI (Network-emulate) converted to VoIP > MPLS
enabled T1 > Core > AS5850 (VoIP converted to TDM) > PSTN.

The customer reports that poor audio is experienced inbound into their PBX.
They say their outbound audio is okay. This doesnt happen on every call but
we do have more than one customer reporting the issue.

The audio can be described anywhere from echo, cutting out or distorted
audio. The problem has also been reported by users who have 38xx series
gateways so i dont think the problem is IAD specific. The AS5850 is a shared
device used by many different customers, the majority of which connect using
Cisco IP phones and this demographic of customers are not reporting the
problem.

I have done packet captures on the AS5850 and there is no packet loss
reported, the packets arrive in sequence and the jitter is very low.

In my testing, I have found double-talk to cause the far end audio (AS5850)
to cut-out. All the packets arrive but they contain a payload value of 7F.
Some improvement can be made to this condition by disabling non-linear
progression.



My questions for the community are related to experiences or lessons learned
from using PRI network emulate.

Can there be too many echo cancellers in the call flow? (i.e. Customer PBX
UC500 with Cisco PRI, IAD PRI, AS5850 PRI all with echo cancellers)

Does the cable length command have any bearing on voice quality?

Are there any special commands i should enter when connecting back to back
wth a T1 cable (echo-cancel coverage, nlp, etc)?

Is a 2 pair sheilded T1 cable necessary for the PRI back to back connection
of less than 20 ft?


Any insight or suggestions would be greatly appreciated.

Thanks,

Jeff
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