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

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27250?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=238467#comment-238467 ] 

Joshua Colp commented on ASTERISK-27250:
----------------------------------------

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> 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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