Thanks for the input guys.
 
That won't do for a server that doesn't have GNOME installed. I am trying to 
test a hosted server. If there is any console command that keep some sort of a 
test live for a day or so and check QOS then it might do for me.
 
-Bruce


----------------------------------------
> Date: Mon, 27 Jul 2009 20:37:53 -0400
> From: [email protected]
> To: [email protected]
> Subject: Re: [on-asterisk] Source of jitter and delay
>
> If you need a definitive measure of the jitter and other SIP QOS try this link
> 
>
> Henry L.Coleman [VoIP-PBX.ca]
> -------------------------------------------------
>
>
>
>> Andre Courchesne - Consultant<
>> Hi Bruce, I have a similar issue and too do not know how to address it.
>> Who is your SIP carrier?
>>
>> ----
>> Andre Courchesne - Consultant
>> http://www.net-forces.com
>> Phone: (514) 667-8448
>> MSN: [email protected]
>> Skype: VoipForces
>>
>> L'information contenue dans le présent document est la propriété de
>> Andre Courchesne. Et est divulguée en toute confidentialité. Cette
>> information ne doit pas être utilisée, divulguée à d'autres personnes ou
>> reproduite sans le consentement écrit explicite de Andre Courchesne.
>>
>> The information contained in this document is confidential and property
>> of Andre Courchesne. It shall not be used, disclosed to others or
>> reproduced without the express written consent of Andre Courchesne.
>>
>> Bruce N wrote:
>>> Hello Everyone,
>>>
>>> What are some of the ways to find out what causes jitter and delay in a SIP 
>>> call using G711. I am testing a hosted
>>> server on (amazon Ec2) and I am under the impression that the link is at 
>>> least a 100mbps (an overkill for a single
>>> g711 call). The provider is supposed to be a really quality provider 
>>> (direct termination) and using G711 all the
>>> way. Still, at times there is a lot of jitter and delay that is introduced 
>>> right in the middle of the call. After a
>>> while it gets better or it doesn't.
>>>
>>> What should I do to find the source of the problem? what are some of the 
>>> tools that I can use. I am thinking it is
>>> the SIP provider issue but I need some solid evidence.
>>>
>>> Thanks,
>>> Bruce
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