<div dir="ltr">> As far as an Opus manual, I also love the idea and can't wait to read it.<div><br></div><div>This is the document trying to address the need for an "Opus manual":</div><div><a href="https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec">https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec</a><br></div><div><br></div><div>As mentioned there, questions and other feedback are welcome.</div><div><br></div><div>It can be easily exported from Confluence as PDF, should it be desired.</div><div><br></div><div>Giacomo</div><div><br></div><div><br></div><div><div class="gmail_extra"><br><div class="gmail_quote">On 14 October 2016 at 18:48, Emrah <span dir="ltr"><<a href="mailto:lists@kavun.ch" target="_blank">lists@kavun.ch</a>></span> wrote:<br><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">Hey there,<div>Thanks for this extensive response. I am not sure I got everything, but I'll try to answer what I can.</div><div><br></div><div><span class="gmail-"><blockquote type="cite"><span style="float:none;display:inline">Why would you say opus@8000@20i does not sound very good ? What do you</span><br><span style="float:none;display:inline">mean exactly ? Your call quality is bad or you mean you don't like the</span><br><span style="float:none;display:inline">fact that the sampling rate is low ?</span><br></blockquote></span><div>Opus at 8khz doesn't give you the same fidelity as PCM at 8khz. You definitely hear the compression. However, Opus at 48khz transcoded into PCMU / PCMA is hardly noticeable. Therefore, I wanted to find a decent compromise, one that would save bandwidth and CPU for calls that don't require a full band audio.</div><div><br></div><div><span class="gmail-"><blockquote type="cite"><span style="float:none;display:inline">You can play with the sampling rates by setting maxplaybackrate and</span><br><span style="float:none;display:inline">sprop_maxcapurerate and enabling asymmetric_samplerates (which is an</span><br><span style="float:none;display:inline">experimental feature so far) in your opus.conf.xml .</span><br></blockquote></span>That sounds very interesting, however I do not want to impose global limitations as many of my Opus calls are full band calls. Would these settings help me work on a per call basis?</div><div><br></div><div><span class="gmail-"><blockquote type="cite"><span style="float:none;display:inline">FS does not have 12 khz or 24 khz because they are not much used for</span><br><span style="float:none;display:inline">Voip. As for opus @ 16 khz perhaps it will be added.</span></blockquote></span>I guess this reconnects with my observation above. The idea would be to use something slightly higher than 8khz, yet not as intensive as 48khz, just to get a clearer call for "PSTN like" connections.</div><div><br></div><div><span class="gmail-"><blockquote type="cite"><span style="float:none;display:inline">But if you use opus @ 48 khz and you just change maxaveragebitrate and</span><br><span style="float:none;display:inline">maxplaybackrate accordingly you should get only WIDEBAND from the</span><br><span style="float:none;display:inline">encoder anyway .</span><br></blockquote></span>Not sure I got that right. Could you elaborate more? Can this work on a per call basis?</div><div><br></div><div><span class="gmail-"><blockquote type="cite"><span style="float:none;display:inline">As for the issue why codec settings cannot be set from FS's dialplan , I</span><br><span style="float:none;display:inline">think its a missing feature that affects other audio codecs too and I</span><br><span style="float:none;display:inline">think it would be very useful.</span><br></blockquote></span>Yes, absolutely. This would definitely help a lot. Not only on the FS side, but also on how to involve the Opus stack on the client side. </div><div><br></div><div>As far as an Opus manual, I also love the idea and can't wait to read it.</div><div><br></div><div>Thanks again for this response,</div><div>Emrah</div><div><div class="gmail-h5"><div><blockquote type="cite"><div>On Oct 14, 2016, at 2:27 PM, Dragos Oancea <<a href="mailto:dragos.oancea@athonet.com" target="_blank">dragos.oancea@athonet.com</a>> wrote:</div><br class="gmail-m_3250631716145889568Apple-interchange-newline"><div><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">Hi</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">Why would you say opus@8000@20i does not sound very good ? What do you</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">mean exactly ? Your call quality is bad or you mean you don't like the</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">fact that the sampling rate is low ?</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">You can play with the sampling rates by setting maxplaybackrate and</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">sprop_maxcapurerate and enabling asymmetric_samplerates (which is an</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">experimental feature so far) in your opus.conf.xml .</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">FS does not have 12 khz or 24 khz because they are not much used for</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">Voip. As for opus @ 16 khz perhaps it will be added.</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">But if you use opus @ 48 khz and you just change maxaveragebitrate and</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">maxplaybackrate accordingly you should get only WIDEBAND from the</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">encoder anyway .</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">The decoder should decode at any sample rate.</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">We're working on a document (sort of manual) for the Opus module and</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">hopefully it will be released soon.</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">As for the issue why codec settings cannot be set from FS's dialplan , I</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">think its a missing feature that affects other audio codecs too and I</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">think it would be very useful.</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">We needed opus@8000hz for transcoding and I tried to explain some things</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">here (see my comment at the bottom of the page ):</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><a href="https://freeswitch.org/confluence/display/FREESWITCH/mod_opus" style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px" target="_blank">https://freeswitch.org/<wbr>confluence/display/FREESWITCH/<wbr>mod_opus</a><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">Basically if you do heavy transcoding to PCMA / PCMU which is 8000 hz</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">you'll want to avoid resampling 48 khz <-> 8 khz - we did tests and by</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">avoiding resampling we were saving 20-30 % CPU .</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">Regards,</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">Dragos</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;float:none;display:inline">On 14/10/2016 14:02, Emrah wrote:</span><br style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><blockquote type="cite" style="font-family:helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px">Hi there,<br>Revisiting this issue. I see that I can set my Opus codec with<br>Opus@8000@20i, but 8khz seems to be the only alternative profile I can<br>use. I see only 2 extreme options when the module is loaded or unloaded<br>that it's either 48khz, mono or stereo and packet size, or 8khz, mono or<br>stereo and packet size. Can someone clarify why there is nothing in<br>between? And what exactly this setting does? Opus@8000h@20i definitely<br>doesn't sound very good. I'd rather have a compromise for tough network<br>conditions.<br>Since these are parameters I can dynamically set on my dialplan, the<br>question then becomes why can't I fully manipulate my Opus stack from<br>the dialplan?<br><br>Thanks!<br><blockquote type="cite">On Jun 2, 2015, at 6:23 AM, Emrah <<a href="mailto:lists@kavun.ch" target="_blank">lists@kavun.ch</a><br><<a href="mailto:lists@kavun.ch" target="_blank">mailto:lists@kavun.ch</a>>> wrote:<br><br>Hi there,<br>@Mike: yes, but in a commonsensical approach the Opus library on the<br>client's side would resample and therefore optimize the codec and the<br>bandwidth accordingly up to FS.<br>@Julien, I saw the setting for Opus globally, but it defeats the<br>purpose. I don't want to limit the bandwidth of Opus for all<br>instances. I'd like to optimize Opus on a per call basis.<br><br>Thanks for the replies<br><blockquote type="cite">On May 31, 2015, at 2:13 PM, Michael Jerris <<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a><br><<a href="mailto:mike@jerris.com" target="_blank">mailto:mike@jerris.com</a>>> wrote:<br><br>Side note, opening at the different rate I believe just makes the<br>opus library do the re sampling instead of FreeSWITCH.<br><br>On Sunday, May 31, 2015, Julien Chavanton <<a href="mailto:jchavanton@gmail.com" target="_blank">jchavanton@gmail.com</a><br><<a href="mailto:jchavanton@gmail.com" target="_blank">mailto:jchavanton@gmail.com</a>>> wrote:<br><br> Hi Emrah,<br><br> The settings exist but they are not available from the dialplan,<br> right now they can only be set globally .<br> <a href="https://freeswitch.org/confluence/display/FREESWITCH/mod_opus" target="_blank">https://freeswitch.org/<wbr>confluence/display/FREESWITCH/<wbr>mod_opus</a><br><br> You can control the bandwidth using maxplaybackrate and<br> maxplaybackrate this will control the local encoder and also adds<br> the corresponding FMTP parameters to the SDP to be used by the<br> remote encoder (if it does implement the following draft, the<br> draft is evolving but I think it as not changed)<br><br> <a href="https://tools/" target="_blank">https://tools</a><br> <<a href="https://tools/" target="_blank">https://tools/</a>>.<a href="http://ietf.org/html/draft-ietf-payload-rtp-opus-11" target="_blank">ietf.org/<wbr>html/draft-ietf-payload-rtp-<wbr>opus-11</a><br><br> Maybe something like :<br><br> maxaveragebitrate 24000<br> maxplaybackrate 8000<br><br> The discussion was getting slightly more complicated when we<br> where discussing about unnecessary resampling this was not a<br> problem but it was just adding extra load on the server.<br><br> On Sun, May 31, 2015 at 6:09 AM, Emrah <<a href="mailto:lists@kavun.ch" target="_blank">lists@kavun.ch</a><br> <javascript:_e(%7B%7D,'<wbr>cvml','<a href="mailto:lists@kavun.ch" target="_blank">lists@kavun.ch</a>');>> wrote:<br><br> Hi list,<br><br> I re-read FS6947 and don't understand how this problematic<br> was addressed and the issue fixed.<br> The scope is simple. There should be a setting in the<br> dialplan that allows downsampling of Opus for applications<br> that do not require the 48khz / 2 channels framework. I.e.:<br> terminating to the PSTN with Opus to take advantage of low<br> bandwidth and great PLC.<br> There seems to be a lot of confusion around bandwidth in<br> general there. It doesn't matter if the internal clock of the<br> device is always sampling at 48khz / 2ch. There are settings<br> that can facilitate a lower bandwidth consumption for<br> particular use cases, and it seems the reason it is not being<br> implemented in FS is just a matter of being confused about<br> the intent of the 48khz 2ch base.<br> Please revisit this issue. FS should allow tuning of Opus<br> audio / network bandwidth in the dialplan. 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