[Asterisk-Users] MOS Scores and LCR

[trixter aka Bret McDanel]

On Sat, 2006-06-17 at 10:16 +0200, Florian Overkamp wrote:
> There are ways to guesstimate MOS scores on a call by continuously 
> getting some decent statistics from the jitterbuffer. We've had an 
> intern do some work on this using IAXclient.
> 
> http://www.speakup.nl/en/opensource/jitterbuffer/

yes and I suggested that however, MOS is an opinion, so its totally
subjective and not based on anything 'real'.  That was kinda my point
earlier.  Personally I think that its better to isolate the network/cpu
issues and correct them to get what a given implementation of a codec is
supposed to be rated at (ideally the two would be intertwined).  

By looking at the jitterbuffer (assuming you have one, if not you may
have to get some code that will inspect packets on the network which can
be done but isnt as easy, its still not that difficult), latency,
dropped packets, etc and generating stats based off that you can
probably make a better guestimation of call quality without actually
listening and then scoring the calls.  

The work that you have done so far is a great step towards a product
that many people might find useful.  In a nutshell the concept I am
thinking about is a tool that you drop onto your network and it will
monitor the data (presumably not just iax but sip, h.323, whatever) and
generate live stats of the call and possibly even have an alarm system
that would send off a page or something if conditions get too far from
'normal'.  

The approach to wait for customer complaints before reacting to voice
degradation is likely not going to work for many.  A tool like I
described, which it seems you are well on your way towards even if that
wasnt your intention, could add a more proactive approach to it and
increase reliability and quality for many.  

I do not mean to trivialize network management and do understand that
sometimes problems are outside your control (ie packets on the
uncontrolled internet may experience problems because of a tertiary
provider between you and the remote end) and as such this is just one
peice of the puzzle as I see it.


-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
Belfast IE +44 28 9099 6461    DE +49 801 777 555 3402
Utrecht NL +31 306 553058      US WA +1 360 207 0479
US NY +1 516 687 5200          FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/e4de4d8c/attachment.pgp