Announcement plays too early over sip

[Louis-Philippe Descamps]

hi

we have a ANN that plays "Your call is being recorded" using ECC/CURRI

If called internally we hear the full message

If called externally we hear "is being recorded"

Any idea?

thanks

[R0g22]

Are you using a ITSP SIP Trunk with or without a CUBE/GW ? Try enabling PRACK on the SIP Profile.

[Louis-Philippe Descamps]

no cube. we are using voice provider over sip trunk.

PRACK is enabled as it has to be enabled in order to hear the ANN.

[R0g22]

Ok. Can you describe the complete call flow ? Call hits CUCM >> goes to ?? >> transfers ??

[Louis-Philippe Descamps]

externally

VoiceProvider->SIP Trunk CUCM->called DN (ECC profile)

Internally

calling DN -> called DN (ECC profile)

ECC profile goes to a CURRI server

[R0g22]

Ok. So in this case, your ITSP will be responsible for PRACKing not the CUCM. If there needs to be early media, ITSP will prack a 18X SIP message in order to cut through audio flowing towards them from CUCM. I believe that part is not working and the audio flows only after the final response i.e. a "200 OK" reason why there is audio clipping. Can you send a CCM SDL Trace for a test call please ? Or better a pcap from CUCM CLI using the following CLI -

utils network capture eth0 file packets count 100000 size all host ip X.X.X.X

X.X.X.X will be your ITSP IP or the SIP Trunk destination. Do a "ctrl+c" to stop the capture once done.
You can download it from RTMT >> Trace & Log Central >> Packet Capture Logs >> Select the node where you took the pcap from.

[Louis-Philippe Descamps]

see attached

[R0g22]

Who is 10.20.44.27 ?

[Louis-Philippe Descamps]

tftp

[Louis-Philippe Descamps]

tftp

[R0g22]

TFTP only ? or is it running IPVMS as well ?

[Louis-Philippe Descamps]

And IPVMS aswell
thanks

[R0g22]

Ok, so from the pcap CUCM sends a 183 w/SDP containing the IP address "10.20.44.27" and your ITSP pracks it, at that point there should be early media. Signalling looks fine. You can take a pcap from this node to check if RTP flow correctly or not for these calls every time and/or advise your ITSP to take a pcap at their end and send it across for analysis.

[Louis-Philippe Descamps]

Hi,

I setup our test sip trunk to use subscriber .23 and use ANN on .23.

the .149 is our signaling to ITSP and .13 is our rtp to ITSP.

Am I right saying that 24986 is the ANN message?

Should i go after prack?

thanks

[R0g22]

That's right. "24986" is the UDP RTP port being used.