[asterisk-bugs] [JIRA] (ASTERISK-28346) useless transcoding
Joshua C. Colp (JIRA)
noreply at issues.asterisk.org
Thu Mar 28 08:16:47 CDT 2019
[ https://issues.asterisk.org/jira/browse/ASTERISK-28346?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Joshua C. Colp updated ASTERISK-28346:
--------------------------------------
Assignee: Thomas Sevestre (was: Unassigned)
Status: Waiting for Feedback (was: Triage)
> 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
> Assignee: 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?
--
This message was sent by Atlassian JIRA
(v6.2#6252)
More information about the asterisk-bugs
mailing list