<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">most codecs are lossless?<div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Feb 26, 2015, at 5:29 AM, Steven Ayre <<a href="mailto:steveayre@gmail.com" class="">steveayre@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">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.<div class=""><br class=""></div><div class="">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.</div><div class=""><br class=""></div><div class=""><br class=""></div></div><div class="gmail_extra"><br class=""><div class="gmail_quote">On 25 February 2015 at 13:16, Luis Daniel Lucio Quiroz <span dir="ltr" class=""><<a href="mailto:luis.daniel.lucio@gmail.com" target="_blank" class="">luis.daniel.lucio@gmail.com</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr" class="">You shall try to void transcoding as much as possible</p><div class="HOEnZb"><div class="h5">
<div class="gmail_quote">On Feb 25, 2015 2:54 AM, "Stanislav Sinyagin" <<a href="mailto:ssinyagin@gmail.com" target="_blank" class="">ssinyagin@gmail.com</a>> wrote:<br type="attribution" class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr" class="">by the way, here are freely available test speech samples:<br class=""><br class=""><a href="http://www.voiptroubleshooter.com/open_speech/" target="_blank" class="">http://www.voiptroubleshooter.com/open_speech/</a><br class=""><a href="http://www.pscr.gov/projects/audio_quality/mrt_library/mrt_library2.php" target="_blank" class="">http://www.pscr.gov/projects/audio_quality/mrt_library/mrt_library2.php</a><br class=""><a href="http://alt-usage-english.org/audio_archive.shtml" target="_blank" class="">http://alt-usage-english.org/audio_archive.shtml</a><br class=""><a href="http://www.itu.int/net/itu-t/sigdb/menu.aspx" target="_blank" class="">http://www.itu.int/net/itu-t/sigdb/menu.aspx</a><br class=""><a href="http://www.voxforge.org/" target="_blank" class="">http://www.voxforge.org/</a><br class=""><br class=""><br class=""><br class=""></div><div class="gmail_extra"><br class=""><div class="gmail_quote">On Wed, Feb 25, 2015 at 8:33 AM, Stanislav Sinyagin <span dir="ltr" class=""><<a href="mailto:ssinyagin@gmail.com" target="_blank" class="">ssinyagin@gmail.com</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr" class=""><div class=""><div class=""><div class=""><div class="">I'd say QoS drops even more abruptly if you hit the bandwidth or performance limits.<br class=""><br class=""></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 class=""><br class=""></div>Here in scripts/ you can find a simple script that launches tshark and analyses the output:<br class=""><a href="https://github.com/voxserv/voip_qos_probe" target="_blank" class="">https://github.com/voxserv/voip_qos_probe</a><br class=""><br class=""></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 class=""><a href="http://openpreservation.org/knowledge/blogs/2012/07/09/xcorrsound-waveform-compare-new-audio-quality-assurance-tool/" target="_blank" class="">http://openpreservation.org/knowledge/blogs/2012/07/09/xcorrsound-waveform-compare-new-audio-quality-assurance-tool/</a><br class=""><a href="https://github.com/openpreserve/scape-xcorrsound" target="_blank" class="">https://github.com/openpreserve/scape-xcorrsound</a><br class=""><br class=""></div>I hope this helps :)<br class=""><div class=""><br class=""><br class=""><br class=""><div class=""><br class=""><div class=""><br class=""></div></div></div></div><div class="gmail_extra"><br class=""><div class="gmail_quote"><div class=""><div class="">On Wed, Feb 25, 2015 at 1:35 AM, Andrew V <span dir="ltr" class=""><<a href="mailto:avstarventures@gmail.com" target="_blank" class="">avstarventures@gmail.com</a>></span> wrote:<br class=""></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=""><div class=""><div dir="ltr" class=""><div class="">How do you measure quality of service across all calls?</div><div class="">What are ways to avoid bad quality of service?</div><div class="">Say the call volume is in your control.</div><div class="">Is it as easy as not letting the call volume get out of control?</div><div class="">The attached is a hypothesis of the relationship between QOS and the number of concurrent calls. Does it work that way?</div><div class="">What other factors need to be taken into account?</div><div class=""><br class=""></div><div class=""><span id="cid:ii_14bbe24a2109a3c7"><image.png></span><br class=""></div><div class=""><br class=""></div></div>
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