[asterisk-bugs] [JIRA] (ASTERISK-27250) asymmetric_rtp_codec=no does not seem to be working anymore with Asterisk 13.17.1

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27250?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=238466#comment-238466 ] 

Asterisk Team commented on ASTERISK-27250:
------------------------------------------

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_rtp_codec=no does not seem to be working anymore with Asterisk 13.17.1
> ---------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27250
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27250
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.17.1
>            Reporter: nappsoft
>            Severity: Minor
>
> Description: We had one-way audio on some systems after upgrading to Asterisk 13.17.1 from Asterisk 13.15 or Asterisk 13.16. The traces showed that the Snom700 device was sending G711u while Asterisk was sending G711a (both codecs were offered in the SDP of both sides). Asterisk didn't switch to G711u during the whole conversation.
> Expectation: we'd expect Asterisk to switch to G711u after receiving G711u packets from the other end.
> Conclusion: It seems like Asterisk 13.17.1 would not switch the codec anymore to make the rtp codec symmetric when rtp is running through Asterisk. (Maybe this could be related to change made in ASTERISK-27013? didn't have time to look into the code, but received a SIP trace by mail that is showing the described behavior that was reported to be reproducible)



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