atcomsystems.ca/forum Forums TDM Business Telephone Systems Toshiba Telephones & Systems intermittent network problem with remote IPT2020

So I have this remote office in Mexico that has been using one of our IP phones over a VPN connection for the last couple of years. There's a Cisco 871 that maintains an IPSEC tunnel back to the home office (where the CTX is).

The remote office recently moved to a new location. Since the move we are having intermittent issues where they call me in the home office, or I call them, and I cannot hear them at all. They can hear what I'm saying but I hear dead silence. The call will stay up indefinitely so its not like the delayed call crash when a net connection goes down.

This doesn't happen all the time, and the VPN tunnel is pretty stable, far more stable than it was in the previous location where it would drop and restart several times per hour. In this new place it is up for days at a time for the most part.

When the problem happens I am still able to access resources on their network via TCP such as their file server, ICMP such as ping, etc. Likewise they can do the same, including ping the CTX IP card. As far as I can tell, the VPN tunnel remains up and functional as before when the phone problem occurs.

The remote phone is not behind traditional NAT. I have the router's vpn set up in network-extension mode which means the actual IP of all clients on that remote network are accessible from the home office.

Any suggestions on where to look for the problem? Rebooting the router doesn't seem to resolve it, yet later in the day it will just magically work.

Could be a bandwidth issue? Your TCP testing isnt gonna tell you anything really since the actual voice communication part of the voip connection is using UDP. You might want to set up QoS and see if that helps any.

Yeah I realize that the voip is using udp, but I did the other testing to verify that the tunnel was still functional. After all, if there's no tunnel there's no chance of the phone working! QoS is already configured.

Any suggestions on testing UDP when the issue crops up?

Could swings in latency cause this? I can see anywhere from 130 to 600ms on this link, pinging from the remote router to the CTX.

First, 600ms is pretty bad for VoIP, but it should not cause dead air. It usually causes bad quality.

Is the phone you are calling also an IPT? If they are at differnt locations then maybe the traffic is not routing from one site to the other. By default after the intercom call is established, the phones will creat a direct audio path between the phones. This can be turned off in emamager.

You did not say if you had access to emanager to make changings. The couple thinks I have seen the cause dead air:

Between 2 IPT phones at different locations.
QOS turned on for the phones in the Toshiba system but not supported on the network.
RTP traffic blocked or not routed properly by routers.
In a small number of cases I have fixed some problems by updationg the firmware on the phone and the IP card.

The thing that doesn't fit with your situation is that it is intermittent. Most of this issues either work or don't.

The best way to test is to run wireshark on a mirrored port. I am not sure where the nbest place to snoop, but I would guess on the side with the IP card. Since people say they can't hear you I would guess that if the RTP traffic is being blocked, it would not appear on the IP card. Again this would change if calling another IPT with default settings.

Thanks for the help so far.

While the problem extension is an IPT, the phones being called are all DKT's. I have not yet had a chance to test with an IPT on both ends. In any case, I do have emanager and the remote extension in question has the NAT peer to peer option enabled.

I'll look and see where I might be able to snoop with wireshark.

So after investigating the issue a little more and significant banging of heads against the wall, I discovered a "fix" for the one-way audio problem but its not really a great solution overall.

I believe that there is some sort of punitive traffic-shaping going on somewhere between the remote site and our HQ site. The 1-way audio is very intermittent but I have found that when it is occuring for a long period of time, if I adjust down the MTU on the remote VPN router's outside interface, then I can restore the audio to two-way. It seems like every week I have to adjust it lower by about 100 so now we're at 1000. Needless to say this isn't great for overall performance of the WAN link but it does keep things flowing.

Is there a way to make the IPT2020 discover on its own the MTU limits on its path to the PBX?