[cisco-voip] Anyone ever seen this message before?

[Andreas Sikkema]

> Site is a CUBE running SIP to the Telco and SIP to a CUCM 8.5.1 Calls from a
> specific number out in the PSTN come in and when you answer there is no
> audio, it only happens with one site’s CLID the customer has that has not
> yet been moved to the IP Network(voice or Data) and call the central site
> via PSTN. Any other call from any other number in the world works just fine.
>
>
>
> I see this error on the CUBE whenever that number calls in
>
> %SIP-3-UNSUPPORTED: Unsupported ptime value VCC, mtp1 index, mtp2 index,
> stream1, stream2 = 0 0 35 36 VCC, mtp1 index, mtp2 index, stream1, stream2 =
> 0 0 36 35

The ptime value in the SDP body is for the packetization time. Usually
it is 20 (for 20milliseconds), but some SIP UAs also send ptime=30.
Most hardware supports either value, but when it doesn't you could get
no audio. Removing ptime=30 with some kind of SIP message manipulation
might work, because most SIP hardware uses ptime=20 as default.

I could be wrong, but it might be worth checking this out.

-- 
Andreas Sikkema