[asterisk-users] Opus codec in codecs.conf

Igor Goncharovsky igor.goncharovsky at gmail.com
Wed Oct 26 00:39:40 CDT 2016


Hello,

George, thank you for pointing this, but there is other question. It is not
clear for some parameters names, what is possible values?

For example this parameters:
packet_loss
max_bandwidth
signal
application

Is there example of configured opus with full set of parameters?


2016-10-25 18:42 GMT+06:00 George Joseph <gjoseph at digium.com>:

>
>
> On Mon, Oct 24, 2016 at 6:54 PM, Igor Goncharovsky <
> igor.goncharovsky at gmail.com> wrote:
>
>> Hello,
>>
>> I am trying to configure new opus codec in asterisk 14, but unable to
>> find any examples of codecs.conf settings for this codec.
>>
>> All I am trying to do - setup peer with using opus in narrow band mode
>> (8kHz sampling rate). Does anybody know how to configure chan_opus?
>>
>>
> If you run "config show help condec_opus opus" from teh Asterisk command
> line, you'll get a list of the configuration options....
>
> pbx1*CLI> config show help codec_opus opus
> opus: [category !~ /.?/]
>
> Codec opus module for Asterisk options
>
> type                      -- Must be of type 'opus'
> sample_rate               -- Codec's sample rate.
> packet_loss               -- Encoder's packet loss percentage.
>
> complexity                -- Encoder's computational complexity.
>
> max_bandwidth             -- Encoder's maximum bandwidth allowed.
>
> signal                    -- Encoder's signal type.
> application               -- Encoder's application type.
> max_playback_rate         -- Encoder's maximum playback rate.
>
> max_ptime                 -- Encoder's maximum packetization rate.
>
> ptime                     -- Encoder's packetization rate.
> bitrate                   -- Encoder's bit rate.
> cbr                       -- Encoder's constant bit rate value.
> fec                       -- Encoder's forward error correction value.
> dtx                       -- Encoder's discontinuous transmission value.
>
>
>
>> --
>> Regards, Igor Goncharovsky
>> Unistim Dev: http://unistim.igorg.ru
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>       http://www.asterisk.org/community/astricon-user-conference
>>
>> New to Asterisk? Start here:
>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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>
>
>
> --
> George Joseph
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>       http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Regards, Igor Goncharovsky
Unistim Dev: http://unistim.igorg.ru
Blog: http://igorg.ru
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