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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US">Hi Guys<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US">I install FS on a CentOS7 but seems the OPUS libraries that came with the Centos repository doesn’t allow the OPUS to be compiled (I
compiled from source).<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US">What libraries should be install?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US">BR<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US">Jaoquin<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif">From:</span></b><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif"> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
<b>On Behalf Of </b>Giacomo Vacca<br>
<b>Sent:</b> 26 October 2016 10:02<br>
<b>To:</b> FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br>
<b>Subject:</b> Re: [Freeswitch-users] FS6947 - Tuning Opus bandwidth<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
<div>
<p class="MsoNormal">> As far as an Opus manual, I also love the idea and can't wait to read it.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<div>
<p class="MsoNormal">This is the document trying to address the need for an "Opus manual":<o:p></o:p></p>
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<p class="MsoNormal"><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><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<div>
<p class="MsoNormal">As mentioned there, questions and other feedback are welcome.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">It can be easily exported from Confluence as PDF, should it be desired.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">Giacomo<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
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<div>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On 14 October 2016 at 18:48, Emrah <<a href="mailto:lists@kavun.ch" target="_blank">lists@kavun.ch</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal">Hey there,<o:p></o:p></p>
<div>
<p class="MsoNormal">Thanks for this extensive response. I am not sure I got everything, but I'll try to answer what I can.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
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<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal">Why would you say opus@8000@20i does not sound very good ? What do you<br>
mean exactly ? Your call quality is bad or you mean you don't like the<br>
fact that the sampling rate is low ?<o:p></o:p></p>
</blockquote>
<div>
<p class="MsoNormal">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.<o:p></o:p></p>
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<div>
<p class="MsoNormal"><o:p> </o:p></p>
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<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal">You can play with the sampling rates by setting maxplaybackrate and<br>
sprop_maxcapurerate and enabling asymmetric_samplerates (which is an<br>
experimental feature so far) in your opus.conf.xml .<o:p></o:p></p>
</blockquote>
<p class="MsoNormal">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?<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal">FS does not have 12 khz or 24 khz because they are not much used for<br>
Voip. As for opus @ 16 khz perhaps it will be added.<o:p></o:p></p>
</blockquote>
<p class="MsoNormal">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.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal">But if you use opus @ 48 khz and you just change maxaveragebitrate and<br>
maxplaybackrate accordingly you should get only WIDEBAND from the<br>
encoder anyway .<o:p></o:p></p>
</blockquote>
<p class="MsoNormal">Not sure I got that right. Could you elaborate more? Can this work on a per call basis?<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
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<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal">As for the issue why codec settings cannot be set from FS's dialplan , I<br>
think its a missing feature that affects other audio codecs too and I<br>
think it would be very useful.<o:p></o:p></p>
</blockquote>
<p class="MsoNormal">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. <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">As far as an Opus manual, I also love the idea and can't wait to read it.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">Thanks again for this response,<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">Emrah<o:p></o:p></p>
</div>
<div>
<div>
<div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class="MsoNormal">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:<o:p></o:p></p>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<div>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">Hi<br>
<br>
Why would you say opus@8000@20i does not sound very good ? What do you<br>
mean exactly ? Your call quality is bad or you mean you don't like the<br>
fact that the sampling rate is low ?<br>
<br>
You can play with the sampling rates by setting maxplaybackrate and<br>
sprop_maxcapurerate and enabling asymmetric_samplerates (which is an<br>
experimental feature so far) in your opus.conf.xml .<br>
<br>
FS does not have 12 khz or 24 khz because they are not much used for<br>
Voip. As for opus @ 16 khz perhaps it will be added.<br>
<br>
But if you use opus @ 48 khz and you just change maxaveragebitrate and<br>
maxplaybackrate accordingly you should get only WIDEBAND from the<br>
encoder anyway .<br>
The decoder should decode at any sample rate.<br>
We're working on a document (sort of manual) for the Opus module and<br>
hopefully it will be released soon.<br>
<br>
As for the issue why codec settings cannot be set from FS's dialplan , I<br>
think its a missing feature that affects other audio codecs too and I<br>
think it would be very useful.<br>
<br>
We needed opus@8000hz for transcoding and I tried to explain some things<br>
here (see my comment at the bottom of the page ):<br>
</span><a href="https://freeswitch.org/confluence/display/FREESWITCH/mod_opus" target="_blank"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">https://freeswitch.org/confluence/display/FREESWITCH/mod_opus</span></a><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif"><br>
<br>
<br>
Basically if you do heavy transcoding to PCMA / PCMU which is 8000 hz<br>
you'll want to avoid resampling 48 khz <-> 8 khz - we did tests and by<br>
avoiding resampling we were saving 20-30 % CPU .<br>
<br>
Regards,<br>
Dragos<br>
<br>
<br>
<br>
On 14/10/2016 14:02, Emrah wrote:<br style="font-variant-caps:normal;text-align:start;word-spacing:0px">
<br>
</span><o:p></o:p></p>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">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>
<br>
<o:p></o:p></span></p>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">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>
<br>
<o:p></o:p></span></p>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">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/confluence/display/FREESWITCH/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/html/draft-ietf-payload-rtp-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,'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. It would optimize<br>
greatly lots of use cases.<br>
If I'm calling the PSTN, I'd rather have my client downsample<br>
and stream at a lower bandwidth, even if my audio capture<br>
would still be at 48khz / 2ch as per the RFC, and save on<br>
bandwidth, than transcode the full 48khz spectrum into PCM on<br>
my FS and minimize processing power on the client's side.<br>
<br>
Jira here: <a href="https://freeswitch.org/jira/browse/FS-6947" target="_blank">
https://freeswitch.org/jira/browse/FS-6947</a><br>
<br>
Emrah<br>
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<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
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_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<span class="gmail-m3250631716145889568apple-converted-space"> </span><br>
</span><a href="mailto:consulting@freeswitch.org" target="_blank"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">consulting@freeswitch.org</span></a><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif"><br>
</span><a href="http://www.freeswitchsolutions.com/" target="_blank"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">http://www.freeswitchsolutions.com</span></a><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif"><br>
<br>
Official FreeSWITCH Sites<br>
</span><a href="http://www.freeswitch.org/" target="_blank"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">http://www.freeswitch.org</span></a><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif"><br>
</span><a href="http://confluence.freeswitch.org/" target="_blank"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">http://confluence.freeswitch.org</span></a><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif"><br>
</span><a href="http://www.cluecon.com/" target="_blank"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">http://www.cluecon.com</span></a><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif"><br>
<br>
FreeSWITCH-users mailing list<br>
</span><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">FreeSWITCH-users@lists.freeswitch.org</span></a><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif"><br>
</span><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</span></a><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif"><br>
UNSUBSCRIBE:</span><a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">http://lists.freeswitch.org/mailman/options/freeswitch-users</span></a><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif"><br>
</span><a href="http://www.freeswitch.org/" target="_blank"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif">http://www.freeswitch.org</span></a><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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</div>
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<p class="MsoNormal"><br>
_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><o:p></o:p></p>
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