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Sip lossy one way audio

Hi just thought I would ask and see if anyone has any ideas

I'm running ucm 8.6 I have 3 sites. HQ, 1 local site and one site over other side of the world, all on a dmvpn. The gateway is cube on a different 2921 at Hq

The local site and he all work fine no major. Issues, but the remote site has huge one way audio issues here what I have discovered.

Sccp phone works fine.

Sip phone ( only have 9971) works perfectly for a minimum of 90 seconds after reregister but that it becomes random sometimes works some times not both to pstn phones and hq or site phones after making a one way call an dither it not connecting or one way audio phone always reregistered within 10 seconds of hanging up.

When making the call it's not the audio getting miss routed when I look at the web page of the phone it's not sending any audio.

I have tried disabling the video on it and the same...

Just wondering if anyone had any ideas... Starting to really annoy me now....

Thanks again


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