<p dir="ltr">You shall try to void transcoding as much as possible</p>
<div class="gmail_quote">On Feb 25, 2015 2:54 AM, "Stanislav Sinyagin" <<a href="mailto:ssinyagin@gmail.com">ssinyagin@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">by the way, here are freely available test speech samples:<br><br><a href="http://www.voiptroubleshooter.com/open_speech/" target="_blank">http://www.voiptroubleshooter.com/open_speech/</a><br><a href="http://www.pscr.gov/projects/audio_quality/mrt_library/mrt_library2.php" target="_blank">http://www.pscr.gov/projects/audio_quality/mrt_library/mrt_library2.php</a><br><a href="http://alt-usage-english.org/audio_archive.shtml" target="_blank">http://alt-usage-english.org/audio_archive.shtml</a><br><a href="http://www.itu.int/net/itu-t/sigdb/menu.aspx" target="_blank">http://www.itu.int/net/itu-t/sigdb/menu.aspx</a><br><a href="http://www.voxforge.org/" target="_blank">http://www.voxforge.org/</a><br><br><br><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Feb 25, 2015 at 8:33 AM, Stanislav Sinyagin <span dir="ltr"><<a href="mailto:ssinyagin@gmail.com" target="_blank">ssinyagin@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div>I'd say QoS drops even more abruptly if you hit the bandwidth or performance limits.<br><br></div>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.<br><br></div>Here in scripts/ you can find a simple script that launches tshark and analyses the output:<br><a href="https://github.com/voxserv/voip_qos_probe" target="_blank">https://github.com/voxserv/voip_qos_probe</a><br><br></div>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:<br><a href="http://openpreservation.org/knowledge/blogs/2012/07/09/xcorrsound-waveform-compare-new-audio-quality-assurance-tool/" target="_blank">http://openpreservation.org/knowledge/blogs/2012/07/09/xcorrsound-waveform-compare-new-audio-quality-assurance-tool/</a><br><a href="https://github.com/openpreserve/scape-xcorrsound" target="_blank">https://github.com/openpreserve/scape-xcorrsound</a><br><br></div>I hope this helps :)<br><div><br><br><br><div><br><div><br></div></div></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Wed, Feb 25, 2015 at 1:35 AM, Andrew V <span dir="ltr"><<a href="mailto:avstarventures@gmail.com" target="_blank">avstarventures@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><div>How do you measure quality of service across all calls?</div><div>What are ways to avoid bad quality of service?</div><div>Say the call volume is in your control.</div><div>Is it as easy as not letting the call volume get out of control?</div><div>The attached is a hypothesis of the relationship between QOS and the number of concurrent calls. Does it work that way?</div><div>What other factors need to be taken into account?</div><div><br></div><div><img src="cid:ii_14bbe24a2109a3c7" alt="Inline image 1" height="310" width="394"><br></div><div><br></div></div>
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