Try "mtr", it does a continuous traceroute and shows
loss/min/max/avg/std_dev for each hop. It will also show changes in
routing as they happen
This will give you a lot of insight into the connection and where it may
be flaky, but won't show jitter.
regards,
Drew
TianLun Song wrote:
normally, what i do is Traceroute from my server to ITSP to see the number
of hops first, then run the continuous PING hop by hop to see if there is
any packet loss and round trip delay through the path. then, I do packet
capture to see the jitter value. Also, I use the phone with packet loss
concealment technology
anyway, even though the source of the packet loss or large delay is able to
be identified, you cannot do anything about it if they are in the middle
from your server to ITSP because it is out of your control
Thank you
Patrick Song
Thinking globally, Networking locally
CCVP, CCNP, M.Eng in Telecommunications
Cell:1-647-868-2950
On Mon, Jul 27, 2009 at 11:18 PM, Bruce N <[email protected]> wrote:
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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