[asterisk-bugs] [JIRA] (ASTERISK-30071) rtp_timeout on websocket

[nappsoft (JIRA)]

nappsoft created ASTERISK-30071:
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             Summary: rtp_timeout on websocket
                 Key: ASTERISK-30071
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30071
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_pjsip
    Affects Versions: 18.12.0
            Reporter: nappsoft


We recently migrated from asterisk 16.19.0 (with security patches) to asterisk 18.12.0 and PJSIP 2.12.

We now have a problem with connections to ConfBridges over websockets as soon as the the user connects with audio and video. If the user only connects with audio or if we set rtp_timeout to 0, everything works as expected.

We didn't have this behavior with asterisk 16.19.0.



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