[asterisk-bugs] [JIRA] (ASTERISK-24666) RTP not closed after sip call using unsupported codec

[Y Ateya (JIRA)]

Y Ateya created ASTERISK-24666:
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             Summary: RTP not closed after sip call using unsupported codec
                 Key: ASTERISK-24666
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24666
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_pjsip
    Affects Versions: 13.1.0
         Environment: ubuntu 12.04; pjproject build from asterisk git repo.
            Reporter: Y Ateya


This is similar to ASTERISK-23721; but on asterisk 13.1.0.
Attached pjsip.conf
To reproduce the bug:
   - Run watch -n1 "netstat -lp | grep aster"
   - Make a call using sip client (which don't support g729)
   - You will get messasge "No joint capabilities for 'audio' media stream between our configuration((g729)) and incoming SDP((ulaw|gsm|alaw))"
   - Check netstat result; you will find 2 RTP ports opened and not closed.
   - Allow ulaw; make same call from same sip client
  - ports will be opened for the call duration and then removed after hangup.





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