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[Linphone-users] Re: Linphone-users Digest, Vol 61, Issue 4


From: Joonbum Byun
Subject: [Linphone-users] Re: Linphone-users Digest, Vol 61, Issue 4
Date: Fri, 21 Dec 2007 16:49:31 -0500


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 <address@hidden>
Subject: Re: [Linphone-users] MOS reporting
To: address@hidden
Message-ID: <address@hidden>
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
address@hidden






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