[asterisk-users] Extract a MOS value from Asterisk CDR

Thu Apr 2 04:20:06 CDT 2009

Could you share with us your Openoffice callc function?

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 2 d. 11:29
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR

Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss

What ye do not have but is needed:
- A factor, a note between 0 and 20 -> 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)

I already have an openoffice calc function to calculate the MOS regarding the 
rtt, packet loss, codec, I have to add the jitter!

Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm

Have a nice day!

-- --
Marc LEURENT
Ingénieur VoIP

DECKPOINT SA
Une société du groupe VTX Telecom
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Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
> Thank you for the interesting links on MOS values and calculations!
> It seems that many (most?) of the values that are used to construct R
> and MOS could be obtained from the data that exists within the
> dialplan, at least as far as the visible RTP path is concerned.   Or
> is there data missing in the current RTCP statistics that would be
> required to make correct R/MOS value estimates?  (If so, then that's
> on-topic for asterisk-dev, otherwise this should be moved to asterisk-
> users...)
>
> Here is the data that I think is already visible:
>
>   - codec choices
>   - round-trip delay to RTP endpoint
>   - packet loss
>   - jitter
>
> I think it is too complex to determine "Irecency", "A" or packet loss
> bursts unless there is significant additional code added to Asterisk
> to capture more granular time-slices of data on each call.  I also
> think that mid-call codec changes should not be considered due to
> complexity.  Currently, I think this is un-necessary since most people
> don't even seem to compute MOS to start with.
>
> So in your examination you may come up with a script or dialplan that
> creates a synthetic R or MOS value - could you post it to a blog, or
> if it is very short, to the asterisk-users mailing list?  I think this
> would be worthwhile.
>
> JT
>
> On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
> > Sorry for replying for the second time, but this issue is
> > interesting for me
> > also.
> >
> > I found such link: http://www.nessoft.com/kb/50
> >
> > And this:
> > http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
> >
> >
> > Regards,
> > Mindaugas Kezys
> > http://www.kolmisoft.com
> > VoIP Billing and Routing Solutions
> >
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Marc
> > Leurent
> > Sent: 2009 m. balandžio 1 d. 18:15
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
> >
> > Hello all,
> > I'm tring to retrieve a formula to calculate a MOS value from
> > Asterisk RTCP
> > stats...
> > Have you got any idea how to do it?
> > Thanks
> >
> > I'm reading all G.107 ITU docs to retrieve something...
> >
> > I'm saving the SIP RTCP stats with:
> >
> > [macro-hangupcall]
> > exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
> > exten => s,n,ResetCDR(vw)
> > exten => s,n,NoCDR()
> >
> > So I retrieve these values in my MySQL CDR table in order to
> > calculate a MOS
> >
> > value:
> > "ssrc
> > =
> > 592614191;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.00000
> > 0;txcount=20734;rlp=0;rtt=0.094000"
> > codec used: g711a
> >
> >
> > --
> > -- --
> > Marc LEURENT
> > lftsy at leurent.eu
> >
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> ---
> John Todd                       email:jtodd at digium.com
> Digium, Inc. | Asterisk Open Source Community Director
> 445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
> direct: +1-256-428-6083         http://www.digium.com/
>
>
>
>
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-- 
-- --
Marc LEURENT
Ingénieur VoIP

DECKPOINT SA
Une société du groupe VTX Telecom
================================================================
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leurent at vtx-telecom.ch
----------------------------------------------------------------
VTX, votre partenaire telecom proche de vous !
================================================================

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