[asterisk-bugs] [JIRA] (ASTERISK-23574) 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)]

Rusty Newton created ASTERISK-23574:
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             Summary: 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-23574
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23574
             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


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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