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Re: [Linphone-users] Loss and Delay
From: |
Hugo Trippaers |
Subject: |
Re: [Linphone-users] Loss and Delay |
Date: |
Fri, 19 Nov 2004 16:37:21 +0100 |
Hey Eduardo,
Are you familiar with the packettracer ethereal? The most recent version
allows you to perform some basic measurements on RTP streams. It might
be what you need. I use it fairly often for determining delay and
jitter.
Regards,
Hugo
On Fri, 2004-11-19 at 11:12 -0400, Eduardo Bezerra Valentin wrote:
> Hi,
>
>
> I'm interested in QoS for VoIP. I have done some experiments on h323
> calls, using the openh323
> library. Now I want to measure the quality for SIP calls.
> I know that linphone is a good choice for SIP client. But, to measure, I
> need the information
> about the loss and the delay of each voice package transmitted and the
> linphone doesn't generate
> any kind of trace for this. I looked for how linphone creates his voice flow
> and I know that it
> is using the oRTP library, written by Saymon Morlat. I take a look at the
> documentation for the
> oRTP library and I didn't find any information about the loss and the delay.
> I want to know if anybody can help me to find out this information on the
> oRTP library.
>
>
> [ ]'s,
> ________________________________
> Eduardo Bezerra Valentin
> VoIP Laboratory
> Universidade Federal do Amazonas
>
>
>
>
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