Hello,
I am trying to resolve a nagging problem with VOIP traffic. We use exclusively MK routers inside our entire network, and UBNT hardware for PxP and PxMP.

About 5% of the time SIP calls no not initiate properly to our clients. A caller calling in to our of our SIP lines will hear either dead air or garbled audio. If they hang up and call right back the call is generally fine. It seems to point me to think that SIP is not initiating properly 100% of the time. Generally there is no call quality issues once the call is established.

I have tried disabling ALL qos. I have tried different queue types (PFIFO, SFP, PCQ). I have tried marking all SIP traffic with a mangle rule for connection mark, then marking that connection with a packet mark, and using a queue tree to prioritize that traffic over anything else. Nothing seems to matter.

I have tested the Internet from a client's location for 1+ weeks. 0.4% packet loss, avg ping to our SIP provider is 16ms, max was 70ms. Using ping plotter (again from a client's site) I see no major drops in the connection or any other reason to explain these issues.

We have a 300Mbps pipe dedicated to these clients, and it NEVER exceeds 150Mbit even at peak times. It doesn't seem to be a QOS issue (and I don't think QOS even comes into play here because we never reach saturation).

The client I am using for testing this issue has a 20Mbit up/down pipe and is never maxing that connection out.

I have tried disabling the "SIP service" in IP -> firewall as I have read that it can cause these type of issues. Enabled or disabled it still happens.

Has anyone else ran into this? It only happens to about 5% of the calls incoming, the other 95% of calls to the same PBX route through and have no issues or call quality issues.

Do you happen to use any sort of tunnels thru those UBNT links while providing that SIP ?

Ideally we'd need packet captures to be able to diagnose this.