<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="">Hi there,<div class="">@Mike: yes, but in a commonsensical approach the Opus library on the client's side would resample and therefore optimize the codec and the bandwidth accordingly up to FS.</div><div class="">@Julien, I saw the setting for Opus globally, but it defeats the purpose. I don't want to limit the bandwidth of Opus for all instances. I'd like to optimize Opus on a per call basis.</div><div class=""><br class=""></div><div class="">Thanks for the replies<br class=""><div><blockquote type="cite" class=""><div class="">On May 31, 2015, at 2:13 PM, Michael Jerris <<a href="mailto:mike@jerris.com" class="">mike@jerris.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class="">Side note, opening at the different rate I believe just makes the opus library do the re sampling instead of FreeSWITCH.<span class=""></span><br class=""><br class="">On Sunday, May 31, 2015, Julien Chavanton <<a href="mailto:jchavanton@gmail.com" class="">jchavanton@gmail.com</a>> 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="">Hi Emrah,<br class=""><div class=""><br class="">The settings exist but they are not available from the <span class="">dialplan</span>, right now they can only be set globally .<br class="">https://<span class="">freeswitch</span>.org/confluence/display/<span class="">FREESWITCH</span>/mod_opus<br class=""><br class="">You can control the bandwidth using <span class="">maxplaybackrate</span> and <span class="">maxplaybackrate</span> this will control the local encoder and also adds the corresponding <span class="">FMTP</span> parameters to the <span class="">SDP</span> to be used by the remote encoder (if it does implement the following draft, the draft is evolving but I think it as not changed)<br class=""><br class=""><a href="https://tools/" target="_blank" class="">https://tools</a>.<span class="">ietf</span>.org/html/draft-<span class="">ietf</span>-payload-<span class="">rtp</span>-opus-11<br class=""><br class="">Maybe something like :<br class=""><br class=""><span class="">maxaveragebitrate</span> 24000<br class=""><span class="">maxplaybackrate</span> 8000<br class=""><br class="">The discussion was getting slightly more complicated when we where discussing about unnecessary <span class="">resampling</span> this was not a problem but it was just adding extra load on the server.<br class=""></div></div><div class="gmail_extra"><br class=""><div class="gmail_quote">On Sun, May 31, 2015 at 6:09 AM, Emrah <span dir="ltr" class=""><<a href="javascript:_e(%7B%7D,'cvml','lists@kavun.ch');" target="_blank" class="">lists@kavun.ch</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi list,<br class="">
<br class="">
I re-read FS6947 and don't understand how this problematic was addressed and the issue fixed.<br class="">
The scope is simple. There should be a setting in the dialplan that allows downsampling of Opus for applications that do not require the 48khz / 2 channels framework. I.e.: terminating to the PSTN with Opus to take advantage of low bandwidth and great PLC.<br class="">
There seems to be a lot of confusion around bandwidth in general there. It doesn't matter if the internal clock of the device is always sampling at 48khz / 2ch. There are settings that can facilitate a lower bandwidth consumption for particular use cases, and it seems the reason it is not being implemented in FS is just a matter of being confused about the intent of the 48khz 2ch base.<br class="">
Please revisit this issue. FS should allow tuning of Opus audio / network bandwidth in the dialplan. It would optimize greatly lots of use cases.<br class="">
If I'm calling the PSTN, I'd rather have my client downsample and stream at a lower bandwidth, even if my audio capture would still be at 48khz / 2ch as per the RFC, and save on bandwidth, than transcode the full 48khz spectrum into PCM on my FS and minimize processing power on the client's side.<br class="">
<br class="">
Jira here: <a href="https://freeswitch.org/jira/browse/FS-6947" target="_blank" class="">https://freeswitch.org/jira/browse/FS-6947</a><br class="">
<br class="">
Emrah<br class="">
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