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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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.