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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
> 
> 
> 
> 
> _______________________________________________
> Linphone-users mailing list
> address@hidden
> http://lists.nongnu.org/mailman/listinfo/linphone-users





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