10-24-2012 03:43 PM - edited 03-16-2019 01:52 PM
Hi all.
I want to know if it's possible to force rtp traffic between remote sites to pass through cucm at head quarter.
All remote sites have connectivity to hq but not between each other. So the is no audio between ip phones. Only when i make a conference call from hq we can hear both parties, but when the the conference originator (hq) hangs up the audio packets are dropped.
thank you in advance
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10-24-2012 11:20 PM
We have a feature exactly for this and it's Called "Trusted Relay Points". It's just a fancy term for an MTP.
Basically you force the phone to require a Trusted Relay point. The phone will allocate a trusted relay point otherwise the call will fail.
You then enable the Trusted Relay Point check, on the XCODE, MTP you want the call to allocate. This way you can have some of your phones have their audio going through an MTP like the CUCM.
This feature was designed for CIPC over VPNS where crazy routing would create all sort of audio issues.
MTP Required would not work, since that feature would only be available for Trunks and Gateways, not internal IP phone to IP Phone calls.
10-24-2012 03:59 PM
I can think of two options off the top of my head:
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10-24-2012 11:20 PM
We have a feature exactly for this and it's Called "Trusted Relay Points". It's just a fancy term for an MTP.
Basically you force the phone to require a Trusted Relay point. The phone will allocate a trusted relay point otherwise the call will fail.
You then enable the Trusted Relay Point check, on the XCODE, MTP you want the call to allocate. This way you can have some of your phones have their audio going through an MTP like the CUCM.
This feature was designed for CIPC over VPNS where crazy routing would create all sort of audio issues.
MTP Required would not work, since that feature would only be available for Trunks and Gateways, not internal IP phone to IP Phone calls.
10-25-2012 04:12 AM
thanks Robert and Jonathan for your answers.
I tried to configure the trusted point relay option with no luck.
What i did is:
1-check the trusted relay point on the software MTP's (3 CUCM , 3 MTPs).
2- add this MTPs to the MRG
3-Change the trusted relay point on the devices to on.
after that, i check the ip addresses involved on site to site call and there is no change, direct rtp audio between phones.
Best regards
10-25-2012 04:53 AM
hi there,
resetting the MTPs did the trick.
Thank you very much for your help.
Regards
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