[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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161026/9182d709/attachment.html>
More information about the asterisk-users
mailing list