[Freeswitch-users] FS6947 - Tuning Opus bandwidth

[Michael Jerris]

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 at 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 at kavun.ch
> <javascript:_e(%7B%7D,'cvml','lists at 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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