[Freeswitch-users] Help needed with audio test setup

[Stanislav Sinyagin]

should be quite easy to do, probably a day of scripting.
The only issue, how to measure the RTP quality in real time.






________________________________
 From: Victor Chukalovskiy <victor.chukalovskiy at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Friday, September 27, 2013 6:57 PM
Subject: [Freeswitch-users] Help needed with audio test setup
 

Hello Community,

The problem is that a 3-rd party SIP hardware causes audio isssues in 
less then 1% of the cases. Need to do 100's of manual test calls to 
identify it and bring to the vendor.

Is there a quick recepie to make the following FS test setup? Should be 
as simple as possible. If someone has something ready-made to offer, 
please contact me off-list.

Here is the idea:

-FS generates test SIP call and keeps it up for N minutes.
-One side initates the call and sends test tone at X Hz
-another side (also FS) answers the call and sends test tone at Y Hz
-both sides listen to the tone coming from the other side.
-If tone in either direction degrades into siginificant noise or 
silence, FS leaves the call up indifentely long AND sends email 
notification "got the sucker!"
-If in N minutes tone in both directions did not degrade, FS hangs up 
the test call and initates a new one. Infinite testing cycle until we 
turn it off.

Thank you,
Victor

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