3300 ISS SIP trunk audio goes away every 5 minutes

[mattyboy]

have a multi site customer with a 3300 ISS at each site rel 7.1 load 13.1.0.33
 each site has their own sip trunks( same provider at both sites) it has  been installed and working fine for years within the last week they say the calls drop after 5 minutes
what is actually happening is the audio on the call goes away  after 5 minutes if you put the call on hold and pick it back up it will work for another 5 minutes
there is no MBG   the provider has router at each  that is the gateway for the 3300  it also handles the vlan for the phones   i know it is something with the RTP stream to the phones and I know the router is the most likely cause I guess I need direction on how to prove that   the sip capture from the ISS is only the SIP portion of the call it doesn't show the rtp portion

the carrier is playing it is  the pbx card cause they see a BYE from the PBX  the only reason for that is the mitel user doesn't hear anything and hangs up  I was able to recreate the issue on a call from the customer  to my office  and I told the customer if the audio goes away hit hold and pick me back up and it worked we spoke for 15 minutes so he had to put me on hold twice

[lundah]

Sounds like a timer is mismatched between the MiVB and the carrier. Check to make sure the Registration and Subscription timers match on both sides, and try setting your Session timer to 0 in the SIP Peer Profile on the MiVB.

[mattyboy]

lundah

thanks  I will check that with the carrier

[mattyboy]

I have attached a pcap and on all the pcaps the carrier is sending an invite at around 5 minutes on a call in progress and thats when the port on the mitel phone changes  shortly after that   in the pcap the 3300 ISS is on the left   sip provider in the middle and the mitel IP phone on the right

[lundah]

Is the new Invite being sent to renegotiate the call, or is it just a session refresh? You can try changing the "Repeat SDP Answer If Duplicate Offer Is Received" option in the SIP Peer Profile.

[mattyboy]

I will have to get that answer from the carrier but on incoming calls the invite comes at around the same time  and the port changes but the audio is not disrupted so everything is updated onto the new port and there is no interuption in the rtp packets  with outgoing the invite and the port change happen but the rtp packets don't make it to the new port   here is a pcap shot of an inbound call

[lundah]

I will have to get that answer from the carrier but on incoming calls the invite comes at around the same time  and the port changes but the audio is not disrupted so everything is updated onto the new port and there is no interuption in the rtp packets  with outgoing the invite and the port change happen but the rtp packets don't make it to the new port   here is a pcap shot of an inbound call

Yeah switching the source RTP port in the middle of the call would be causing issues. You probably can work with it by messing around with SIP Peer Profile settings to allow renegotiation mid-call, but I'm not 100% of which options need to be enabled/disabled to get it to play nice. If you have access to Mitel KMS, see if there's a configuration guide for your SIP carrier there.

[mattyboy]

Thanks Lundah  i sent the pcaps to the carrier will post up the resolution

[mattyboy]

UPDATE issue resolved

after much back and forth with the carrier who stated not their issue one of their techs took a look at it and when they updated their equip. the config was not loaded correctly they restored it again and the issue went away