[Freeswitch-users] How to measure QOS?

[Michael Jerris]

most codecs are lossless?

> 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 <mailto: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 <mailto:ssinyagin at gmail.com>> wrote:
> by the way, here are freely available test speech samples:
> 
> http://www.voiptroubleshooter.com/open_speech/ <http://www.voiptroubleshooter.com/open_speech/>
> http://www.pscr.gov/projects/audio_quality/mrt_library/mrt_library2.php <http://www.pscr.gov/projects/audio_quality/mrt_library/mrt_library2.php>
> http://alt-usage-english.org/audio_archive.shtml <http://alt-usage-english.org/audio_archive.shtml>
> http://www.itu.int/net/itu-t/sigdb/menu.aspx <http://www.itu.int/net/itu-t/sigdb/menu.aspx>
> http://www.voxforge.org/ <http://www.voxforge.org/>
> 
> 
> 
> 
> On Wed, Feb 25, 2015 at 8:33 AM, Stanislav Sinyagin <ssinyagin at gmail.com <mailto: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 <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/ <http://openpreservation.org/knowledge/blogs/2012/07/09/xcorrsound-waveform-compare-new-audio-quality-assurance-tool/>
> https://github.com/openpreserve/scape-xcorrsound <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 <mailto: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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