[Freeswitch-users] How to measure QOS?

[Steven Ayre]

Whoops, obviously meant the opposite. Running on a caffeine deficit at the
time obviously...

On 26 February 2015 at 16:24, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> On Thu, Feb 26, 2015 at 5:21 PM, Michael Jerris <mike at jerris.com> wrote:
> > most codecs are lossless?
>
> 8-))
>
>
> >
> > On Feb 26, 2015, at 5:29 AM, Steven Ayre <steveayre at gmail.com> wrote:
> >
> > Two clarify this point... Most codecs are lossless, but don't necessarily
> > throw the same information away so going through multiple codecs can
> throw
> > out more information than using a single codec. Probably not an issue as
> > long as the call isn't being transcoded *many* times. Also (software)
> > transcoding uses more CPU and therefore it reduces the capacity of your
> > server, which means you may get audio issues from an overloaded CPU at
> lower
> > volumes.
> >
> > Most of the time audio issues will be network related - either jitter or
> > packet loss. Wireshark will reveal those. There are also a number of rtp
> > statistics in the XML CDR which includes information on RTP packet loss
> and
> > jitter, and a computed quality score.
> >
> >
> >
> > On 25 February 2015 at 13:16, Luis Daniel Lucio Quiroz
> > <luis.daniel.lucio at gmail.com> wrote:
> >>
> >> You shall try to void transcoding as much as possible
> >>
> >> On Feb 25, 2015 2:54 AM, "Stanislav Sinyagin" <ssinyagin at gmail.com>
> wrote:
> >>>
> >>> by the way, here are freely available test speech samples:
> >>>
> >>> http://www.voiptroubleshooter.com/open_speech/
> >>>
> http://www.pscr.gov/projects/audio_quality/mrt_library/mrt_library2.php
> >>> http://alt-usage-english.org/audio_archive.shtml
> >>> http://www.itu.int/net/itu-t/sigdb/menu.aspx
> >>> http://www.voxforge.org/
> >>>
> >>>
> >>>
> >>>
> >>> On Wed, Feb 25, 2015 at 8:33 AM, Stanislav Sinyagin <
> ssinyagin at gmail.com>
> >>> wrote:
> >>>>
> >>>> I'd say QoS drops even more abruptly if you hit the bandwidth or
> >>>> performance limits.
> >>>>
> >>>> tshark, a text terminal version of Wireshark, can analyze the quality
> of
> >>>> RTP streams and print a report that you can parse. Keep in mind that
> the
> >>>> analysis itself is quite CPU-intensive, so what I usually do is
> collect the
> >>>> RTP streams with tcpdump, and then run tshark in low-priority mode.
> >>>>
> >>>> Here in scripts/ you can find a simple script that launches tshark and
> >>>> analyses the output:
> >>>> https://github.com/voxserv/voip_qos_probe
> >>>>
> >>>> There is also a library for comparing WAV streams -- this would be the
> >>>> best for QoS measurements, and you would also be able to get the PSTN
> path
> >>>> into the test. But I didn't yet try it, as the customer was satisfied
> with
> >>>> RTP analysis:
> >>>>
> >>>>
> http://openpreservation.org/knowledge/blogs/2012/07/09/xcorrsound-waveform-compare-new-audio-quality-assurance-tool/
> >>>> https://github.com/openpreserve/scape-xcorrsound
> >>>>
> >>>> I hope this helps :)
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> On Wed, Feb 25, 2015 at 1:35 AM, Andrew V <avstarventures at gmail.com>
> >>>> wrote:
> >>>>>
> >>>>> How do you measure quality of service across all calls?
> >>>>> What are ways to avoid bad quality of service?
> >>>>> Say the call volume is in your control.
> >>>>> Is it as easy as not letting the call volume get out of control?
> >>>>> The attached is a hypothesis of the relationship between QOS and the
> >>>>> number of concurrent calls.  Does it work that way?
> >>>>> What other factors need to be taken into account?
> >>>>>
> >>>>> <image.png>
> >>>>>
> >
> >
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