Hi there,
Revisiting this issue. I see that I can set my Opus codec with Opus@8000@20i, but 8khz seems to be the only alternative profile I can use. I see only 2 extreme options when the module is loaded or unloaded that it's either 48khz, mono or stereo and packet size, or 8khz, mono or stereo and packet size. Can someone clarify why there is nothing in between? And what exactly this setting does? Opus@8000h@20i definitely doesn't sound very good. I'd rather have a compromise for tough network conditions.
Since these are parameters I can dynamically set on my dialplan, the question then becomes why can't I fully manipulate my Opus stack from the dialplan?

Thanks!
On Jun 2, 2015, at 6:23 AM, Emrah <lists@kavun.ch> wrote:

Hi there,
@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.
@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.

Thanks for the replies
On May 31, 2015, at 2:13 PM, Michael Jerris <mike@jerris.com> wrote:

Side note, opening at the different rate I believe just makes the opus library do the re sampling instead of FreeSWITCH.

On Sunday, May 31, 2015, Julien Chavanton <jchavanton@gmail.com> wrote:
Hi Emrah,

The settings exist but they are not available from the dialplan, right now they can only be set globally .
https://freeswitch.org/confluence/display/FREESWITCH/mod_opus

You can control the bandwidth using maxplaybackrate and maxplaybackrate this will control the local encoder and also adds the corresponding FMTP parameters to the SDP 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)

https://tools.ietf.org/html/draft-ietf-payload-rtp-opus-11

Maybe something like :

maxaveragebitrate 24000
maxplaybackrate 8000

The discussion was getting slightly more complicated when we where discussing about unnecessary resampling this was not a problem but it was just adding extra load on the server.

On Sun, May 31, 2015 at 6:09 AM, Emrah <lists@kavun.ch> wrote:
Hi list,

I re-read FS6947 and don't understand how this problematic was addressed and the issue fixed.
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.
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.
Please revisit this issue. FS should allow tuning of Opus audio / network bandwidth in the dialplan. It would optimize greatly lots of use cases.
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.

Jira here: https://freeswitch.org/jira/browse/FS-6947

Emrah
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