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

[Thomas Sevestre (JIRA)]

Thomas Sevestre created ASTERISK-28346:
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             Summary: opus 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


When I force opus codec with SIP_CODEC and SIP_CODEC_OUTBOUND, asterisk needlessly transcode both channels to opus at 48000 to slin at 48000 to opus at 48000

Here is the revelant informations :

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

I force opus codec with SIP_CODEC and SIP_CODEC_OUTBOUND variables and establish the call 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 sip.conf to allow only opus, it works as expected without transcoding.

Is it a known issue?



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