2013/1/5 joachim zoach...@securax.org
You are pretty much limited to measuring the delay and the jitter.
The delay you can somewhat estimate prior to the call (with qualify for
example).
The jitter / packetloss you can only figure out when the call is already
up for a while. (e.g. you might
2013/1/8 Lenz Emilitri lenz.lo...@gmail.com
2013/1/5 joachim zoach...@securax.org
You are pretty much limited to measuring the delay and the jitter.
The delay you can somewhat estimate prior to the call (with qualify for
example).
The jitter / packetloss you can only figure out when the
A few years ago I spoke to a Finnish company that had a commercial
solution for automated MOS estimation. So something exists though I
have not tested it first-hand.
l.
--
You need a lot of data to calculate a MOS score, you will need the
actual call.
The only solution i can think of is
Emilitri lenz.lo...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 8, 2013 2:25 PM
Subject: Re: [asterisk-users] Detect Low Quality Calls - Realtime
2013/1/5 joachim zoach...@securax.org
You are pretty much limited
Thanks
What would you use to measure jitter / packetloss in real time?
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Sometimes just the act of collecting performance data degrades the quality
Sent from my iPhone 5
On Jan 6, 2013, at 6:00 AM, XBrian bobo...@yahoo.co.uk wrote:
Thanks
What would you use to measure jitter / packetloss in real time?
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On 5.1.2013 г. 03:37 ч., XBrian wrote:
I can only detect calls as they hit our server, do the magic and based
on latency, bandwidth and MOS (Meaning Opinion Score) - decide whether the call
should be let through. I will accept all MOS values of 4.0
You are pretty much limited to
Joachim, thanks for the reply
- delay you can somewhat estimate prior to the call (with qualify for example)
Pls be explicit. How do I use qualify to measure delay
- The jitter / packetloss you can only figure out when the call is already
up for a while.
what would you use to measure jitter
Asterisk sip show peers lists the qualify value in ms (milliseconds).
Please read up on this and the setting for it in sip.conf config file
Sent from my iPhone 5
On Jan 5, 2013, at 5:30 AM, XBrian bobo...@yahoo.co.uk wrote:
Joachim, thanks for the reply
- delay you can somewhat estimate
Hi there,
I support a large number of enterprise users who contractually must connect to
our support center via a 4G VOIP connection.
I simply want to be able to auto detect all poor quality calls in realtme (as
they are being made), play a message and drop the call - without user
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