[Freeswitch-users] FS6947 - Tuning Opus bandwidth
Emrah
lists at kavun.ch
Fri Oct 14 20:48:00 MSD 2016
Hey there,
Thanks for this extensive response. I am not sure I got everything, but I'll try to answer what I can.
> Why would you say opus at 8000@20i does not sound very good ? What do you
> mean exactly ? Your call quality is bad or you mean you don't like the
> fact that the sampling rate is low ?
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.
> You can play with the sampling rates by setting maxplaybackrate and
> sprop_maxcapurerate and enabling asymmetric_samplerates (which is an
> experimental feature so far) in your opus.conf.xml .
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?
> FS does not have 12 khz or 24 khz because they are not much used for
> Voip. As for opus @ 16 khz perhaps it will be added.
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.
> But if you use opus @ 48 khz and you just change maxaveragebitrate and
> maxplaybackrate accordingly you should get only WIDEBAND from the
> encoder anyway .
Not sure I got that right. Could you elaborate more? Can this work on a per call basis?
> As for the issue why codec settings cannot be set from FS's dialplan , I
> think its a missing feature that affects other audio codecs too and I
> think it would be very useful.
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.
As far as an Opus manual, I also love the idea and can't wait to read it.
Thanks again for this response,
Emrah
> On Oct 14, 2016, at 2:27 PM, Dragos Oancea <dragos.oancea at athonet.com> wrote:
>
> Hi
>
> Why would you say opus at 8000@20i does not sound very good ? What do you
> mean exactly ? Your call quality is bad or you mean you don't like the
> fact that the sampling rate is low ?
>
> You can play with the sampling rates by setting maxplaybackrate and
> sprop_maxcapurerate and enabling asymmetric_samplerates (which is an
> experimental feature so far) in your opus.conf.xml .
>
> FS does not have 12 khz or 24 khz because they are not much used for
> Voip. As for opus @ 16 khz perhaps it will be added.
>
> But if you use opus @ 48 khz and you just change maxaveragebitrate and
> maxplaybackrate accordingly you should get only WIDEBAND from the
> encoder anyway .
> The decoder should decode at any sample rate.
> We're working on a document (sort of manual) for the Opus module and
> hopefully it will be released soon.
>
> As for the issue why codec settings cannot be set from FS's dialplan , I
> think its a missing feature that affects other audio codecs too and I
> think it would be very useful.
>
> We needed opus at 8000hz for transcoding and I tried to explain some things
> here (see my comment at the bottom of the page ):
> https://freeswitch.org/confluence/display/FREESWITCH/mod_opus <https://freeswitch.org/confluence/display/FREESWITCH/mod_opus>
>
>
> Basically if you do heavy transcoding to PCMA / PCMU which is 8000 hz
> you'll want to avoid resampling 48 khz <-> 8 khz - we did tests and by
> avoiding resampling we were saving 20-30 % CPU .
>
> Regards,
> Dragos
>
>
>
> On 14/10/2016 14:02, Emrah wrote:
>> Hi there,
>> Revisiting this issue. I see that I can set my Opus codec with
>> Opus at 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 at 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 at kavun.ch
>>> <mailto:lists at kavun.ch <mailto:lists at 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 at jerris.com <mailto:mike at jerris.com>
>>>> <mailto:mike at jerris.com <mailto:mike at 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 at gmail.com <mailto:jchavanton at gmail.com>
>>>> <mailto:jchavanton at gmail.com <mailto: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 <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 <https://tools/>
>>>> <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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>>
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