Summary:ASTERISK-23017: Crash on inbound calls using WebRTC config with ICE Servers -signal 6 abort, while in ice_worker_thread
Reporter:Agustí Ubalde (aubalde)Labels:
Date Opened:2013-12-17 10:20:38.000-0600Date Closed:2014-01-27 12:40:12.000-0600
Status:Closed/CompleteComponents:Resources/res_rtp_asterisk Resources/res_srtp
Versions:11.6.1 Frequency of
is related toASTERISK-22889 Segmentation fault when RTP going via ICE
is related toASTERISK-20762 Asterisk Crash, assertion failed, in res_rtp_asterisk thread (ice_worker_thread)
is related toASTERISK-21696 Assertion error results in crash in pjproject's ICE worker thread
Environment:CentOS 6.4 64bits Google Chrome Version 31.0.1650.63m SIPML5 API version = 1.2.185Attachments:( 0) backtrace.txt
Description:Asterisk crash occurs when incoming calls are received on the registered web phone registered. This error occurs randomly, but frequently.

I add traces captured with the "gdb" tool from a core file generated by the application.
Comments:By: Rusty Newton (rnewton) 2013-12-17 16:09:41.252-0600

This is likely a duplicate of one of the several ICE-related crashes already reported, however your backtrace is optimized out. Can you get a backtrace from Asterisk compiled without optimizations so we can double-check things?

If you follow the instructions here: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace you'll be sure to get what we need. That is being sure you compile with both the DONT_OPTIMIZE and BETTER_BACKTRACES options.

By: Agustí Ubalde (aubalde) 2013-12-19 03:24:08.232-0600

Hi Rusty,

Thanks for your response.

Now, I have a problem with the environment in DEBUG MODE. If I compile with the flag "DON'T OPTIMIZE" checked, the audio does not work. I've seen that RTP packets have been sent only one way and no audio is heard in the browser.

If you want I can upload asterisk traces.

Thanks and regards,

By: Agustí Ubalde (aubalde) 2013-12-19 03:45:26.706-0600

Hi Rusty,

Before the crash, these error messages are displayed on the console:

[Dec 18 12:53:37] ERROR[1470][C-0000000a]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
[Dec 18 12:53:37] WARNING[1470][C-0000000a]: chan_sip.c:15853 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
[Dec 18 12:53:37] ERROR[1470][C-0000000a]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported

By: Agustí Ubalde (aubalde) 2013-12-19 03:51:48.520-0600

Hi Rusty,

This warning message is displayed on the console when audio is being heard correctly on both sides:

[Dec 18 12:59:36] WARNING[1625][C-00000000]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

Thanks and regards,

By: Rusty Newton (rnewton) 2014-01-07 13:01:56.655-0600

Agusti. Does the crash still occur where you can get a new trace, despite the audio not working after you have compiled with DONT_OPTIMIZE?

If you can't get the crash with DONT_OPTIMIZE compiled, can you get provide an Asterisk log leading up to the crash? Be sure VERBOSE and DEBUG are enabled: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

By: Rusty Newton (rnewton) 2014-01-27 12:40:06.172-0600

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines