[asterisk-bugs] [JIRA] (ASTERISK-28346) useless transcoding

[Thomas Sevestre (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28346?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Thomas Sevestre updated ASTERISK-28346:
---------------------------------------

    Description: 
Here is the revelant informations :

/etc/asterisk/sip.conf :
[100]
disallow=all
allow=alaw
allow=opus
...
[200]
disallow=all
allow=alaw
allow=opus
...

/etc/asterisk/extensions.conf :
I force opus codec with SIP_CODEC and SIP_CODEC_OUTBOUND variables
The call is established with Dial application.

In asterisk console, both channels have the following transcoding information :
  NativeFormats: (opus)
    WriteFormat: slin48
     ReadFormat: slin48
 WriteTranscode: Yes (slin at 48000)->(opus at 48000)
  ReadTranscode: Yes (opus at 48000)->(slin at 48000)

If I force alaw on both channels, it works as expected without transcoding.

If I change codec order in sip.conf :
[100]
disallow=all
allow=opus
allow=alaw
...
[200]
disallow=all
allow=opus
allow=alaw
...
Then I can force opus passthrough but when I force alaw there is a useless transcoding :
  NativeFormats: (alaw)
    WriteFormat: slin48
     ReadFormat: slin48
 WriteTranscode: Yes (slin at 48000)->(slin at 8000)->(alaw at 8000)
  ReadTranscode: Yes (alaw at 8000)->(slin at 8000)->(slin at 48000)

Is it a known issue?

  was:
Here is the revelant informations :

/etc/asterisk/sip.conf :
[100]
disallow=all
allow=alaw
allow=opus
...
[200]
disallow=all
allow=alaw
allow=opus
...

/etc/asterisk/extensions.conf :
I force opus codec with SIP_CODEC and SIP_CODEC_OUTBOUND variables
The call is established with Dial application.

In asterisk console, both channels have the following transcoding information :

          State: Up (6)
  NativeFormats: (opus)
    WriteFormat: slin48
     ReadFormat: slin48
 WriteTranscode: Yes (slin at 48000)->(opus at 48000)
  ReadTranscode: Yes (opus at 48000)->(slin at 48000)

If I force alaw on both channels, it works as expected without transcoding.

If I change codec order in sip.conf :
[100]
disallow=all
allow=opus
allow=alaw
...
[200]
disallow=all
allow=opus
allow=alaw
...
Then I can force opus passthrough but when I force alaw there is a useless transcoding :
  NativeFormats: (alaw)
    WriteFormat: slin48
     ReadFormat: slin48
 WriteTranscode: Yes (slin at 48000)->(slin at 8000)->(alaw at 8000)
  ReadTranscode: Yes (alaw at 8000)->(slin at 8000)->(slin at 48000)

Is it a known issue?


> useless transcoding
> -------------------
>
>                 Key: ASTERISK-28346
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28346
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/codec_opus
>    Affects Versions: 13.25.0
>            Reporter: Thomas Sevestre
>
> Here is the revelant informations :
> /etc/asterisk/sip.conf :
> [100]
> disallow=all
> allow=alaw
> allow=opus
> ...
> [200]
> disallow=all
> allow=alaw
> allow=opus
> ...
> /etc/asterisk/extensions.conf :
> I force opus codec with SIP_CODEC and SIP_CODEC_OUTBOUND variables
> The call is established with Dial application.
> In asterisk console, both channels have the following transcoding information :
>   NativeFormats: (opus)
>     WriteFormat: slin48
>      ReadFormat: slin48
>  WriteTranscode: Yes (slin at 48000)->(opus at 48000)
>   ReadTranscode: Yes (opus at 48000)->(slin at 48000)
> If I force alaw on both channels, it works as expected without transcoding.
> If I change codec order in sip.conf :
> [100]
> disallow=all
> allow=opus
> allow=alaw
> ...
> [200]
> disallow=all
> allow=opus
> allow=alaw
> ...
> Then I can force opus passthrough but when I force alaw there is a useless transcoding :
>   NativeFormats: (alaw)
>     WriteFormat: slin48
>      ReadFormat: slin48
>  WriteTranscode: Yes (slin at 48000)->(slin at 8000)->(alaw at 8000)
>   ReadTranscode: Yes (alaw at 8000)->(slin at 8000)->(slin at 48000)
> Is it a known issue?



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