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-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent Sent: 2009 m. balandžio 2 d. 11:29 To: asterisk-users@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 ================================================================ Rue Eugène-Marziano 15 - 1227 Les Acacias http://www.vtx.ch - marc.leur...@vtx-telecom.ch ---------------------------------------------------------------- VTX, votre partenaire telecom proche de vous ! ================================================================ 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-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@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 > > lf...@leurent.eu > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > --- > John Todd email:jt...@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/ > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- -- -- Marc LEURENT Ingénieur VoIP DECKPOINT SA Une société du groupe VTX Telecom ================================================================ Rue Eugène-Marziano 15 - 1227 Les Acacias http://www.vtx.ch - marc.leur...@vtx-telecom.ch ---------------------------------------------------------------- VTX, votre partenaire telecom proche de vous ! ================================================================ _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users