[asterisk-bugs] [JIRA] (ASTERISK-22380) Inbound SIP call to a valid extension results in segfault in multicast_rtp_new at res_rtp_multicast.c

[Digium Subversion (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-22380?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Digium Subversion closed ASTERISK-22380.
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    Resolution: Fixed
    
> Inbound SIP call to a valid extension results in segfault in multicast_rtp_new at res_rtp_multicast.c
> -----------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-22380
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22380
>             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
>         Attachments: ASTERISK-22380-12.diff, backtrace5.txt, full5.txt, pjsip.txt
>
>
> 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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