sip problem

[stsguy]

here goes nothing! have a customer with a 3300 mxe using a multitec voip gatewayon both ends. sip station going into a bogen commander quantum and calling classrooms. the problem is when the class room makes a call outside the mitel system thier is jitter.any sugestions?

 

[IrwinMFletcher]

Define 'jitter'?

Any chance the default for the SIP is 30ms packets?

 

[stsguy]

whre would i find that info?

 

[stsguy]

jitter: like talking into a fan but worse.

 

[IrwinMFletcher]

Maintenace logs might tell you, something like this:

1 WARNING Media Negotiation 2008/02/02 12:00:00 MaintenanceLog(0)
Incompatible RTP packetization capabilities detected on call between SIP port xxxxxxxx and port xxxxxxxx


Do both endpoints report the same problem?

 

[stsguy]

far end does not here jitter.

 

[IrwinMFletcher]

What software version are you running?

The best way to find out (if there are no logs) is to get a capture of the packets heading to the phone.

 

[stsguy]

9.0.2.31_1

 

[IrwinMFletcher]

And there are no logs?

Search for "packet rate' in the help files and make sure that you have everything programmed correctly. The Bogen system should also have some configuration panels.

 

[slapin]

Do you have any network monitoring? Do you have end-to-end QoS deployed? What are the chances that you network is congested? Is the problem consistent or intermittent?

 

[Canuckvoip]

Download the 30 day trial version of Pingplotter Pro onto a PC or laptop.
Put that machine on the voice subnet of the 3300 and target the Bogen.
Set it up so that you are pinging the target every 1 second.
You will see all the hops in-between and you will see any packet loss/delay/jitter/MOS.

Dave

You can't believe anything you read... unless of course it's this sentence.

 

[yeahbutno]

Sounds like a QoS issue,have you any IP phones running and if so, you can monitor some by using the Mitel IP Phone Analyzer to see if there are any loss of packets and Jitter on your network which would probably suggest again your QoS.

 

[stsguy]

theproblem only occurs when the bogen is in the mix. the jitter is intermit.

 

[slapin]

Canuckvoip, echo reply is the last task which any TCP/IP stack will perform when it has really nothing else to do. So ping response is the not the metric you should use when evaluating network performance.

I will recommend to capture all voice traffic using Wireshark close to the point where you experience problems. Encryption should be turned off. Then Wireshark can parse RTP/RTCP information and show real situation with packet loss and jitter.

 

[IrwinMFletcher]

slapin has it right, the only true way to determine what is causing the audio issues is to capture as close to the endpoint that is experiencing the problem as possible.

stsguy, can you be overly specific, do you mean intermittent during a call, or intermittent from call to call (but it last for the duration of the call). They point to different possible problems.

 

[stsguy]

the jitter is thru out the call

 

[IrwinMFletcher]

If it's for the duration of the call, then I would suggest that it's a negotiation problem with the sip endpoint(s).

Are you able to get a trace?