<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><a href="https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_wait_for_silence">https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_wait_for_silence</a><div><br><div><div>On Sep 27, 2013, at 2:34 PM, Victor Chukalovskiy <<a href="mailto:victor.chukalovskiy@gmail.com">victor.chukalovskiy@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">
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<div class="moz-cite-prefix">Clarification:<br>
<br>
No need to measure RTP quality. Just need to detect when a test
tone changes into white noise or dead air.<br>
<br>
I see mod_spandsp can fire events on tone detection. Is there a
way to do inverse? That is, fire event on tone loss?<br>
<br>
On 13-09-27 01:22 PM, Stanislav Sinyagin wrote:<br>
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<blockquote cite="mid:1380302556.1196.YahooMailNeo@web126201.mail.ne1.yahoo.com" type="cite">
<div style="background-color: rgb(255, 255, 255); font-family: arial, helvetica, sans-serif; font-size: 10pt; ">should be quite easy to
do, probably a day of scripting.<br>
The only issue, how to measure the RTP quality in real time.<br>
<br>
<br>
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<hr size="1"> <font face="Arial" size="2"> <b><span style="font-weight:bold;">From:</span></b> Victor
Chukalovskiy <a class="moz-txt-link-rfc2396E" href="mailto:victor.chukalovskiy@gmail.com"><victor.chukalovskiy@gmail.com></a><br>
<b><span style="font-weight: bold;">To:</span></b>
FreeSWITCH Users Help
<a class="moz-txt-link-rfc2396E" href="mailto:freeswitch-users@lists.freeswitch.org"><freeswitch-users@lists.freeswitch.org></a> <br>
<b><span style="font-weight: bold;">Sent:</span></b>
Friday, September 27, 2013 6:57 PM<br>
<b><span style="font-weight: bold;">Subject:</span></b>
[Freeswitch-users] Help needed with audio test setup<br>
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Hello Community,<br>
<br>
The problem is that a 3-rd party SIP hardware causes audio
isssues in <br>
less then 1% of the cases. Need to do 100's of manual test
calls to <br>
identify it and bring to the vendor.<br>
<br>
Is there a quick recepie to make the following FS test
setup? Should be <br>
as simple as possible. If someone has something ready-made
to offer, <br>
please contact me off-list.<br>
<br>
Here is the idea:<br>
<br>
-FS generates test SIP call and keeps it up for N minutes.<br>
-One side initates the call and sends test tone at X Hz<br>
-another side (also FS) answers the call and sends test
tone at Y Hz<br>
-both sides listen to the tone coming from the other side.<br>
-If tone in either direction degrades into siginificant
noise or <br>
silence, FS leaves the call up indifentely long AND sends
email <br>
notification "got the sucker!"<br>
-If in N minutes tone in both directions did not degrade,
FS hangs up <br>
the test call and initates a new one. Infinite testing
cycle until we <br>
turn it off.<br>
<br>
Thank you,<br>
Victor<br>
<br>
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