[asterisk-bugs] [JIRA] (ASTERISK-23572) chan_pjsip call from endpoint using WS transport to endpoint using UDP sometimes results in a crash after a call from res_rtp_asterisk into pjproject

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-23572?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-23572:
------------------------------------

    Status: Open  (was: Triage)

> chan_pjsip call from endpoint using WS transport to endpoint using UDP sometimes results in a crash after a call from res_rtp_asterisk into pjproject
> -----------------------------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-23572
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23572
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Resources/res_pjsip, Resources/res_rtp_asterisk
>    Affects Versions: SVN, 12.1.1
>            Reporter: Rusty Newton
>            Severity: Critical
>         Attachments: backtrace.txt, configs.txt, full.txt, pjsip_console_debug.txt
>
>
> While testing a WebRTC configuration, with one endpoint setup for ws transport and another setup for udp, occasionally after calling from the ws endpoint to the udp, I found this crash.
> {noformat}
> Program terminated with signal 6, Aborted.
> #0  0x00007f44e46d7425 in raise () from /lib/x86_64-linux-gnu/libc.so.6
> #0  0x00007f44e46d7425 in raise () from /lib/x86_64-linux-gnu/libc.so.6
> No symbol table info available.
> #1  0x00007f44e46dab8b in abort () from /lib/x86_64-linux-gnu/libc.so.6
> No symbol table info available.
> #2  0x00007f44e46d00ee in ?? () from /lib/x86_64-linux-gnu/libc.so.6
> No symbol table info available.
> #3  0x00007f44e46d0192 in __assert_fail () from /lib/x86_64-linux-gnu/libc.so.6
> No symbol table info available.
> #4  0x00007f44e0d54805 in grp_lock_acquire () from /usr/lib/libpj.so.2
> No symbol table info available.
> #5  0x00007f44e04e146b in pj_ice_sess_send_data () from /usr/lib/libpjnath.so.2
> No symbol table info available.
> #6  0x00007f4443ef39d0 in __rtp_sendto (instance=0x7f44840d51b8, buf=<optimized out>, size=<optimized out>, sa=0x7f4493542590, rtcp=0, ice=0x7f449354258c, use_srtp=1, flags=0) at res_rtp_asterisk.c:1605
>         len = 176
>         temp = 0x7f44840deafc
>         rtp = 0x7f44840d9700
>         srtp = <optimized out>
> #7  0x00007f4443ef6ef9 in rtp_sendto (ice=0x7f449354258c, sa=0x7f4493542590, flags=0, size=<optimized out>, buf=<optimized out>, instance=0x7f44840d51b8) at res_rtp_asterisk.c:1622
> No locals.
> {noformat}
> Will attach:
>  * full log in full.txt, 
>  * console debug in pjsip_console_debug.txt
>  * backtrace in backtrace.txt
> h3. Reproduction:
> I followed the configuration I had put on the wiki before, https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5, and configured the SIPML5 client in a browser on the same machine, to connect to localhost. 
> Call from that extension using WebSockets to another phone using a UDP transport.  Maybe try a dozen times and you should get a crash a couple seconds after one of them.



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