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Rtp traffic between remote sites

Youssef Aoufi
Level 3
Level 3

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

Sent from Cisco Technical Support Android App

1 Accepted Solution

Accepted Solutions

Robert Thomas
Level 7
Level 7

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.

View solution in original post

4 Replies 4

Jonathan Schulenberg
Hall of Fame
Hall of Fame

I can think of two options off the top of my head:

  1. Fix your routing. I'm guessing you're using DMVPN or EasyVPN. You could implement spoke-to-spoke tunnels with DMVPN but in either case just advertise routes for the other sites from HQ. Voice relys on IP routing like any other app.
  2. Check the "MTP Required" checkbox and ensure that the remote office phones MRGL only contains an MTP located in HQ. The downside to this is that even intra-office calls would hairpin through HQ.

Please remember to rate helpful responses and identify helpful or correct answers.

Robert Thomas
Level 7
Level 7

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.

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

hi there,

resetting the MTPs did the trick.

Thank you very much for your help.

Regards