<div dir="ltr">Hi Emrah,<br><div><br>The settings exist but they are not available from the <span tabindex="-1" id=":1sg.1" style class="">dialplan</span>, right now they can only be set globally .<br>https://<span tabindex="-1" id=":1sg.3" style class="">freeswitch</span>.org/confluence/display/<span tabindex="-1" id=":1sg.4" style class="">FREESWITCH</span>/mod_opus<br><br>You can control the bandwidth using <span tabindex="-1" id=":1sg.5" style class="">maxplaybackrate</span> and <span tabindex="-1" id=":1sg.6" style class="">maxplaybackrate</span> this will control the local encoder and also adds the corresponding <span tabindex="-1" id=":1sg.7" style class="">FMTP</span> parameters to the <span tabindex="-1" id=":1sg.8" style 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><br><a href="https://tools">https://tools</a>.<span tabindex="-1" id=":1sg.9" style class="">ietf</span>.org/html/draft-<span tabindex="-1" id=":1sg.10" style class="">ietf</span>-payload-<span tabindex="-1" id=":1sg.11" style class="">rtp</span>-opus-11<br><br>Maybe something like :<br><br><span tabindex="-1" id=":1sg.12" style class="">maxaveragebitrate</span> 24000<br><span tabindex="-1" id=":1sg.13" style class="">maxplaybackrate</span> 8000<br><br>The discussion was getting slightly more complicated when we where discussing about unnecessary <span tabindex="-1" id=":1sg.15" style class="">resampling</span> this was not a problem but it was just adding extra load on the server.<br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Sun, May 31, 2015 at 6:09 AM, 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:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi list,<br>
<br>
I re-read FS6947 and don't understand how this problematic was addressed and the issue fixed.<br>
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>
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>
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>
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>
<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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