[asterisk-bugs] [JIRA] (ASTERISK-27250) chan_pjsip: 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:all-tabpanel ]

Joshua Colp updated ASTERISK-27250:
-----------------------------------

    Assignee: nappsoft  (was: Unassigned)
      Status: Waiting for Feedback  (was: Open)

This issue should be resolved in the current release candidates. A fix was put in to resolve an issue where we would not switch our sending codec as expected. Please try and see.

> chan_pjsip: 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
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.17.0, 13.17.1
>            Reporter: nappsoft
>            Assignee: 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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