[asterisk-bugs] [JIRA] (ASTERISK-27056) codec_opus responds with incorrect RTP Payload Type

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27056?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=237427#comment-237427 ] 

Asterisk Team commented on ASTERISK-27056:
------------------------------------------

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].

> codec_opus responds with incorrect RTP Payload Type
> ---------------------------------------------------
>
>                 Key: ASTERISK-27056
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27056
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.16.0
>         Environment: x64 CentOS
>            Reporter: Luke Escude
>         Attachments: asterisk.pcap, grandstream.pcap
>
>
> A phone call to an Opus extension works just fine (Asterisk sends an invite with 107 opus). Two-way audio and all that jazz.
> However, if the Asterisk endpoint puts the call on hold, then retrieves the call (the endpoint sends an INVITE to retrieve the call with 123 opus) Asterisk agrees to use 123 opus, however it continues to send 107 opus instead.



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