Roman;

I recently found a useful document on MOS measurement published  by 
ETSI/3GPP. It contains MOS score table with four kinds of network 
parameters, bit error rate(or radio condition, 10E-2, 5x10E-4), packet 
loss rate (0%, 3%), codec bandwidth (6.7, 12.2, 12.65, 15.85kbps) and 
delay (300ms, 500ms) disturbance.  MOS value is provided for every 
combination of network paramete.  Group of hunam beings are involved in to 
score the perceptual quality of VoIP call.  Quality is categorized into 
five; Voice quality, Understanding, Interaction and perception and Global 
quality.

You may download the document  for free from 
http://portal.etsi.org/Portal_Common/home.asp --> Service Index --> ETSI 
standard --> Publication Download  and search for  "3gpp tr 26.935". You 
may need to register yourself.

Thanks and hope it helps

Joonbum




----------------------------------------------------------------------

Message: 1
Date: Sat, 08 Dec 2007 10:37:18 +0500
From: Roman Imankulov <[EMAIL PROTECTED]>
Subject: Re: [Linphone-users] MOS reporting
To: [email protected]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hi,

> 
> Does Linphone have a feature of calculating MOS (Measn Openion Score) of 
a 
> voice call? If yes, how the score is reported, RTCP, SNMP or ...?
> 

As I know there is no satisfactory way  to calculate MOS on the fly.

  * Originally proposed method (see ITU-T Rec. P.800) requires 
absolutely unsuitable conditions (such as specially-equipped studios, etc)

  * There is also PESQ (ITU-T Rec. P.862) which requires both reference 
and degraded speech samples (as I suppose, this is also unsuitable for 
the real-time quality estimation).

  * Last chance is an E-model (ITU-T Rec. G.107) which originally was 
proposed as quality estimation model for the circuit switching networks. 
   Some of parameters of this model (such as one-way delay) cannot be 
calculated without additional measurements.

I'd like to hear if somebody knows another ways to calculate speech 
quality or knows how to implement in real-time VoIP application some of 
methods proposed below.

Also I'm not sure but it seems that PESQ and E-model are patented.

-- 
WBR, Roman Imankulov
[EMAIL PROTECTED]





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