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 > >>> _________________________________________________________________ > >>> Send and receive email from all of your webmail accounts. > >>> http://go.microsoft.com/?linkid=9671356 > >>> --------------------------------------------------------------------- > >>> To unsubscribe, e-mail: [email protected] > >>> For additional commands, e-mail: [email protected] > >>> > >> > >> --------------------------------------------------------------------- > >> To unsubscribe, e-mail: [email protected] > >> For additional commands, e-mail: [email protected] > >> > >> > >> > > > > > > --------------------------------------------------------------------- > > To unsubscribe, e-mail: [email protected] > > For additional commands, e-mail: [email protected] > > > _________________________________________________________________ > Attention all humans. We are your photos. Free us. > http://go.microsoft.com/?linkid=9666047 > --------------------------------------------------------------------- > To unsubscribe, e-mail: [email protected] > For additional commands, e-mail: [email protected] > > -- Thank you Patrick Song Thinking globally, Networking locally CCVP, CCNP, M.Eng in Telecommunications Cell:1-647-868-2950
