Wireshark-dev: Re: [Wireshark-dev] Voice (RTP stream) quality - mos, delay, bandwidth, ...
From: "Newton, Don" <[email protected]>
Date: Tue, 6 Nov 2007 13:00:46 -0500
> > daniel zhang wrote:
> >
> > > 1. MOS
> > > Voice quality was traditionally reported as a Mean Opinion
> > Score (MOS)
> > > on a scale from 1-5 where 1 is the lowest and 5 the highest.
> > > I am not an expert on this but I think probably we could
> > add the MOS
> > > into Ethereal.
> >
> > At least according to the Wikipedia article on the Mean Opinion
> >
> > 	http://en.wikipedia.org/wiki/Mean_Opinion_Score
> >
> > the score comes from people listening to particular phrases
> > and giving their 1-to-5 rating of the quality of the sound,
> > not from a calculation you could perform on a raw digital
> > signal.
> > [...]
> That's true, but there have been many attempts to model MOS from raw
> directly.
> The most prominent being the ITU G.107 "E-Model".
> But still, the model includes parameters like the audio
characteristics of
> the end devices, jitter buffer implementations and so on, so MOS
cannot be
> calculated from a network trace without making specific  assumptions
> the
> end devices and audio path.
> Best regards,
> Lars Ruoff
This is a pretty interesting idea; 
Jitter buffer size is probably the biggest feature that causes MOS to
differ from what it looks like on the network.  Jitter buffer size could
easily be configurable even to point of picking the cpe and having the
defaults of that device.  

Another possibility would be to "apply" the jitter buffer settings to
both the packet list and the rendered audio file so late packets get
tossed just like they do on the phone.  This would allow operators to
pick the ideal buffer size for the network.  Generally the buffer should
be as small as possible but no smaller.

Troubleshooting a voip network would be enhanced by charting MOS slips
at various points i.e. "this stream was a solid 4.4 until it went
through this router and took a nose dive".