[asterisk-bugs] [JIRA] (ASTERISK-26745) Asymmetric codecs when when asymmetric_rtp_codec=no

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26745?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=234821#comment-234821 ] 

Asterisk Team commented on ASTERISK-26745:
------------------------------------------

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].

> Asymmetric codecs when when asymmetric_rtp_codec=no
> ---------------------------------------------------
>
>                 Key: ASTERISK-26745
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26745
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.13.1, 14.2.1, GIT
>            Reporter: Jesse Ross
>
> Calling from a Grandstream HT802 (opus,ulaw) to a trunk (ulaw), causes one way audio (no audio heard on HT802)
> Wireshark shows HT802 send opus and receives ulaw packets.
> I pulled the 13 and 14 branches from the git repo and had the same issue.
> In my case, it was not solved by this:
> https://issues.asterisk.org/jira/browse/ASTERISK-26603
> Attached are 
> debug.txt (verbose=10, debug=10, rtp and pjsip debugging on)
> endpoint1.txt (HT802 endpoint calling from)
> endpoint2.txt (sip trunk endpoint)
> ht802.txt (snippet of syslog from GS ATA)
> I originally made this post: https://community.asterisk.org/t/problems-with-opus-grandstream-ht802-directmedia-native-rtp/69459/5



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