[asterisk-bugs] [JIRA] (ASTERISK-22381) Inbound SIP call to a valid extension results in segfault in multicast_rtp_new at res_rtp_multicast.c

[Rusty Newton (JIRA)]

Rusty Newton created ASTERISK-22381:
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             Summary: Inbound SIP call to a valid extension results in segfault in multicast_rtp_new at res_rtp_multicast.c
                 Key: ASTERISK-22381
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22381
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_pjsip, Core/RTP, Resources/res_pjsip
    Affects Versions: 12
         Environment: SVN-branch-12-r397614
            Reporter: Rusty Newton
            Severity: Critical


To reproduce:

* See attached pjsip.conf
* Dial from a SIP endpoint to an extension calling either Playback or Dial applications. Playback(demo-congrats) works just fine. Dialing another pjsip endpoint also reproduces the same crash.

Note:

Unloading res_rtp_multicast.so allows calls to Playback to function normally. Calls from pjsip to pjsip endpoints (on same LAN) have no audio, RTP debug does not indicate RTP flowing in either direction.

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