Cisco Phones one way audio

[shelly mathew]

Hi,

We have a Cisco UCM 6.1 at our office, with about 300 IP Phones. We're connected to a site office over VPN, and have a few 7911 phones there that are registered to the CUCM at the main office.

Recently, we replaced the VPN gateway at the main office, with the same configuration as the previous one, and since then the phones at the site office, freeze up, or calls will fail with no audio. They work for a few calls after a reset, then fail.

Where could I begin troubleshooting? Appreciate any help.

[Nadav]

I'd put the PC port of the phone on monitoring (or use SPAN on the access switch connected to the phone). When the issue is replicated, you can see whether or not RTP arrives at the access switch or not. If so, see where it's being dropped. 

It could be that a firewall allows certain UDP high ports to pass through, and others not. 

If the RTP packets are arriving at the phone and yet you hear nothing, you should definitely open a TAC and keep the packet capture.

[shelly mathew]

hi guys,

thank you very much for your suggestions. I found an article on turning off the sccp alg on the Juniper SSG gateway, which seems to have fixed the issue. there haven't been any complaints since yesterday and the end of work today, so I'm going to assume that the problem is fixed.

thanks again.

[Manish Gogna]

Apart from the good points suggested by Hod you should try to get the ip phones to latest firmware and cucm to latest 6.1.5 at least, since the phones freezing issue could be related to a bug. Additionally TAC may not support a cucm 6.1 deployment.

Manish

- Do rate helpful posts -

[Mohammed al Baqari · Accepted solution]

Hi,

One way audio most of the time is a routing problem. So make sure that you don't have routing flapping causing intermittent RTP failure.

Another common reason (which can be your case) is a firewall device inspecting your RTP/SCCP/SIP traffic. I suggest to make sure that you look at inspection configuration and make sure that VoIP protocols are excluded.

Also, since you are having two seperate sites, make sure that your VoIP traffic isn't hitting any nat rule.