[Linphone-developers] Loss and Delay

[Eduardo Bezerra Valentin]

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.
   Anyone knows how to produce trace about the RTP flow in oRTP library?? How 
can I
do this in linphone???
   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