<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=""><meta http-equiv="Content-Type" content="text/html charset=us-ascii" class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Even forced enabled jitter doesn't make asterisk to ignore late arrived packets. <div class="">During my tests jb was always enabled (forced).<br class=""><div class="">I made also tests without delay, only with drops - quality of Digium Opus not acceptable for voice conversation anyway.</div><div class="">The fact is open source opus handles better dropped packets.</div><div class="">You can't disagree with quality after all,</div><div class="">I can send you recorded samples, you can compare.</div><div class=""><br class=""></div><div class="">Yury</div><div class=""><br class=""></div><div class=""><br class=""></div><div class=""><div class=""><blockquote type="cite" class=""><div class="">On 3 Apr 2017, at 21:22, Kevin Harwell <<a href="mailto:kharwell@digium.com" class="">kharwell@digium.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class=""><br class=""><div class="gmail_extra"><br class=""><div class="gmail_quote">On Mon, Apr 3, 2017 at 1:28 PM, Yury Tsaregorodtsev <span dir="ltr" class=""><<a href="mailto:aero.1080@icloud.com" target="_blank" class="">aero.1080@icloud.com</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style="word-wrap:break-word" class=""><div style="word-wrap:break-word" class=""><div class=""><div class=""><snip> </div></div></div></div></blockquote><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style="word-wrap:break-word" class=""><div style="word-wrap:break-word" class=""><div class="">MOS on calls using open source opus higher almost twice.</div><div class="">Subjective opinion regarding audio quality: using open source codec quality almost same as in example on <a href="http://opus-codec.org/examples/" target="_blank" class="">http://opus-codec.org/examp<wbr class="">les/</a> with 30% loss and FEC, acceptable for ears, but</div><div class="">using digium opus quality is not acceptable, a lot of spikes, interruptions.</div><div class=""><br class=""></div><div class="">I also double checked the fact before applying ASTERISK-25629 patch asterisk don't drop lately arrived RTP.</div></div></div></blockquote><div class=""><br class=""></div><div class="">Dropped packets and late arriving packets are two separate issue and are handled as such in Asterisk. The Digium Opus codec can handle dropped packets by enabling FEC. If the problem is late arriving packets than applying a jitter buffer to the audio stream exhibiting the problem should help alleviate that.</div></div>
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