[asterisk-bugs] [JIRA] (ASTERISK-23310) bridged channel crashes in bridge_p2p_rtp_write

Kinsey Moore (JIRA) noreply at issues.asterisk.org
Tue Mar 4 09:46:49 CST 2014


    [ https://issues.asterisk.org/jira/browse/ASTERISK-23310?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=216045#comment-216045 ] 

Kinsey Moore commented on ASTERISK-23310:
-----------------------------------------

The test for this is up at https://reviewboard.asterisk.org/r/3297/

> bridged channel crashes in bridge_p2p_rtp_write
> -----------------------------------------------
>
>                 Key: ASTERISK-23310
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23310
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 11.7.0
>         Environment: Debian wheezy
>            Reporter: Jeremy Lainé
>            Assignee: Kinsey Moore
>         Attachments: ASTERISK-23310.diff, broken-peer, crash-backtrace.txt, extensions.conf, refcount-and-log.tar.gz, sip.conf
>
>
> I have encountered numerous crashes using Bridge, which always involve bridge_p2p_rtp_write. To sumarize the setup:
> - two calls are fired up via Originate (one to an operator, one to a contact)
> - the operator gets put into an extension which basically does Answer + Wait(1800)
> - when the contact picks up, he gets bridged to the operator
> - the contact hangs up => the operator returns to the extension and waits some more
> .. and at some point things fall apart, and I start getting "bad magic number" warnings and eventually asterisk crashes, but always in bridge_p2p_rtp_write.
> An example:
> [Feb 14 15:47:15] DEBUG[25480][C-00000196] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped
> [Feb 14 15:47:15] DEBUG[25480][C-00000196] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped
> [Feb 14 15:47:15] DEBUG[25480][C-00000196] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped
> [Feb 14 15:47:15] DEBUG[24287] chan_sip.c: Auto destroying SIP dialog '13e862723108b2aa2dbf50f14b7327ed at MY.IP.ADDRESS.HERE:5060'
> [Feb 14 15:47:15] DEBUG[24287] chan_sip.c: Destroying SIP dialog 13e862723108b2aa2dbf50f14b7327ed at MY.IP.ADDRESS.HERE:5060
> [Feb 14 15:47:15] VERBOSE[24287] chan_sip.c: Really destroying SIP dialog '13e862723108b2aa2dbf50f14b7327ed at MY.IP.ADDRESS.HERE:5060' Method: BYE
> [Feb 14 15:47:15] DEBUG[24287] rtp_engine.c: Destroyed RTP instance '0x7f5b980d17b8'
> [Feb 14 15:47:15] ERROR[25480][C-00000196] astobj2.c: bad magic number for object 0x7f5b98146488. Object is likely destroyed.
> [Feb 14 15:47:15] ERROR[25480][C-00000196] astobj2.c: bad magic number for object 0x7f5b98146488. Object is likely destroyed.
> [Feb 14 15:47:15] DEBUG[25480][C-00000196] res_rtp_asterisk.c: Unsupported payload type received



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