[asterisk-bugs] [JIRA] (ASTERISK-28436) Transcoding happening even though it is not necessary

Asterisk Team (JIRA) noreply at issues.asterisk.org
Mon Jun 3 17:29:47 CDT 2019


    [ https://issues.asterisk.org/jira/browse/ASTERISK-28436?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=247291#comment-247291 ] 

Asterisk Team commented on ASTERISK-28436:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

> Transcoding happening even though it is not necessary
> -----------------------------------------------------
>
>                 Key: ASTERISK-28436
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28436
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 15.3.0
>            Reporter: Flole Systems
>
> A phone has G722,ulaw,alaw set as allowed codecs. The SIP Provider only allows G722 when the number dialed does support it. For the provider I set allow to G722,ulaw,alaw aswell.
> When setting up a call asterisk first negotiates a codec with the phone (G722) and then with the provider (ulaw in the worst case). Instead of switching the phone to ulaw aswell (it does support that), asterisk starts to transcode using 5 times the CPU power a normal call would take. This is not only a waste of resources but also reduces audio quality while having absolutely no benefit.
> I would expect asterisk to start re-negotiating the codec once it knows which codec each party supports selecting the most prefered one (aka the first one in my allow list) that both parties support. Only if there is none, use transcoding. If transcoding is not possible, fail.



--
This message was sent by Atlassian JIRA
(v6.2#6252)



More information about the asterisk-bugs mailing list