Call Quality troubleshooting 1

[Sparticle500]

Hi

We have an issue with our SIP call quality. If a call is initiated externally (ie someone dials in) the call quality is perfect, the caller can hear us and we can hear them perfectly.

If we make the call, the other party can hear us OK, but sometimes the call quality that we hear is poor, with a sort of 'warble' in the sound. It only happens occuasionally and it only affects what we hear.

The network Jitter showing in the phone (we can see on the 9650 phones) is about 20ms, it is the same on good calls and poor calls.

On the poor quality occuasions a Quality of service alarm is recorded in the Alarms area of IP500. Jitter when this occurs is often higher than 20ms.

We have a dedicated internet connection over a PSTN line for the SIP service, system is IPoffice 500 v1.

This is driving me nuts, as there is no consistancy to the problem. Any suggestions as to the problem and what troubleshooting logs/tests/traces could we run to try and isolate where the problem is?
Appreciate any guidance.

 

[Westi]

I had issues not too long ago and my STUN server I used was no good, it sometimes worked and sometimes didn't and also had weird quality issues, so you cuold try that one and find a new STUN server to test. If you have a static IP then this is of course no good. 20ms is not that bad

Joe W.

FHandw, ACSS

 

[montyzummer]

also have you got a managed connection for voice it may be youre upload speed is ok but somebody is down loading the entire wikipedia website hense killing download .?

APSS (SME)
ACSS (SME)

 

[Shortcode]

How many IP Phones are on the network? Are they on their own VLAN? Is there a router between the dedicated internet hit and the IP Office?

"When you do things right, people won't be sure you have done anything at all.

 

[Sparticle500]

We're looking into the STUN server, we have one entered, but now looking to see if there is a problem there. thanks for that suggestion, we'll see if it helps.

The phones run on their own network and have their own switch. So not mixed with other network traffic.

We have 2 internet connections, with one used only for the SIP service, all other traffic goes out though a seperate connection, so there is no issue with bandwidth being chewed up by something else, ie downloading etc

 

[Sparticle500]

So far no joy. The service provider has said they don't recommend a STUN server and that they use HNT (Hosted NAT traversal).

We've tried it with and without a STUN server. We still get the poor quality on some outgoing calls. It is also from the instance the phone rings, so the poor quality is instantly when we hear the other phone ringing.

Any other suggestions on what traces (using Monitor or other)we could run while on the call?

 

[FFrey]

Try turning off direct IP to IP media if it is enabled. Also make sure the IP phones are using the same codec as the trunk. When it comes to network issues I like to use wireshark.

 

[Sparticle500]

Codecs are the same. Sorry FFrey, I'm not sure what you mean by IP to IP media, is there a setting you are refering to in the Avaya? Thanks for the suggestion of wireshark, I'll have a look.