<div dir="ltr">Hello fellow VoIPers and RTCers,<div><br></div><div>on GitHub there is an early release of sipnagios, opensource.</div><div><br></div><div>check it out: <a href="https://github.com/gmaruzz/sipnagios">https://github.com/gmaruzz/sipnagios</a></div><div><br></div><div>sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)<br></div><div>
<p>sipnagios implements the Nagios plugin API for monitoring and performance data.</p>
<p>sipnagios.c is a modification of the original siprtp.c sample in
pjproject distribution. Supposedly, it works on Linux, Windows, and
anywhere you can compile pjproject on.</p>
<p>It makes a call, checks all the various resulting values (mos, rtt,
pdd, tta, jitter, packet loss, bytes and packets transferred, and so
on). It verifies these values are included into acceptable, warning, or
critical ranges.</p>
<p>If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0.</p>
<p>If the call fails, or if its measured values are not inside
acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL
values.<br><br>mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand :) (merci Julien!)</p><div>Enjoy!<br>-giovanni<br><br><br></div>-- <br><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature">Sincerely,<br><br>Giovanni Maruzzelli<br>OpenTelecom.IT<br>cell: +39 347 266 56 18<br><br></div></div></div>