[Freeswitch-users] How to measure QOS?

[Stanislav Sinyagin]

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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