[asterisk-bugs] [JIRA] (ASTERISK-26880) Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26880?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235956#comment-235956 ] 

Rusty Newton edited comment on ASTERISK-26880 at 3/20/17 9:12 AM:
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[~ktyerman] , I had to delete your patch as it wasn't submitted under license agreement.

https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process

Please verify you have signed a license agreement and then re-submit your patch and select "mark as contribution".

You can then submit it to Gerrit and push it through the review process.

Also, thanks for all the extra debug!


was (Author: rnewton):
[~ktyerman] , I had to delete your patch as it wasn't submitted under license agreement.

https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process

Please verify you have signed a license agreement and then re-submit your patch and select "mark as contribution".

You can then submit it to Gerrit and push it through the review process.

> Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
> --------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-26880
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26880
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/Bridging
>    Affects Versions: 13.9.1, 14.3.0
>         Environment: Fedora 25
>            Reporter: Kirsty Tyerman
>            Assignee: Unassigned
>         Attachments: backtrace.txt, codecs.conf, debug_log_26880.txt, extensions.conf, Screenshot from 2017-03-16 16-43-50.png, sip.conf
>
>
> I have compiled Asterisk 14.3.0 on a Fedora 25 Workstation. 
> I have configured speex in asterisk to enable preprocessing and preprocessing voice activity detection. I have two sip clients (linphones) using speex. When the two sip clients join the same Confbridge asterisk crashes with a segmentation fault with the following output (after running asterisk -vvvg -c):
> {noformat}
> warning: The VAD has been replaced by a hack pending a complete rewrite
> warning: The VAD has been replaced by a hack pending a complete rewrite
> FRACK!, Failed assertion user_data is NULL (0) at line 5727 in ast_set_write_format of channel.c
> [Mar 16 18:30:32] ERROR[31673][C-00000001]: channel.c:5727 ast_set_write_format: FRACK!, Failed assertion user_data is NULL (0)
>     -- <CBAnn/3000-00000000;1> Playing 'confbridge-join.gsm' (language 'en')
> Got 22 backtrace records
> #0: [0x62966c] asterisk(__ast_assert_failed+0x8d) [0x62966c]
> #1: [0x45ef3f] asterisk() [0x45ef3f]
> #2: [0x45f816] asterisk(__ao2_ref+0x89) [0x45f816]
> #3: [0x533d2c] asterisk(__ast_format_cap_append+0xa7) [0x533d2c]
> #4: [0x4c3e96] asterisk(ast_set_write_format+0x70) [0x4c3e96]
> #5: [0x4c29ae] asterisk(ast_write+0x10aa) [0x4c29ae]
> #6: [0x48a0f5] asterisk() [0x48a0f5]
> #7: [0x48a7ee] asterisk() [0x48a7ee]
> #8: [0x48af54] asterisk(bridge_channel_internal_join+0x558) [0x48af54]
> #9: [0x47039c] asterisk(ast_bridge_join+0x2c1) [0x47039c]
> #10: [0x7f53456fa9b5] /usr/lib/asterisk/modules/app_confbridge.so(+0xb9b5) [0x7f53456fa9b5]
> #11: [0x5a2952] asterisk(pbx_exec+0x119) [0x5a2952]
> #12: [0x58ed5b] asterisk() [0x58ed5b]
> #13: [0x592947] asterisk(ast_spawn_extension+0x50) [0x592947]
> #14: [0x593580] asterisk() [0x593580]
> #15: [0x594ca3] asterisk() [0x594ca3]
> #16: [0x62663e] asterisk() [0x62663e]
> Segmentation fault (core dumped)
> {noformat}
> This error is also caused when dtx is enabled in codecs.conf. 
> When pp_vad and dtx is disabled in codecs.conf, asterisk will not crash.
> Attached are the asterisk config files that were configured to produce the error and a screen grab of the asterisk console after crash.
> *STEPS TO REPRODUCE*
> 1. dnf install asterisk-13.9.1
> 2. use configuration files supplied in attatchments
> 3. configure two sip phones using speex and register to the asterisk server using sip accounts in sip.conf (may have to change sip bindaddr)
> 4. dial each sip phone into confbridge 3000
> *UPDATE*
> Please see further attached items, including a backtrace and an asterisk log. I have compiled asterisk from the git repository (commit b05d2fda0c8b3473c3d6d7bd1cc0473e2728b744) with the debugging flags on to obtain the backtrace. 
> bridge_channel.c:2348, is the source of the problem. It is not handling AST_FRAM_VOICE in switch statement correctly. Asterisk is caused to crash due to format being NULL.
> Applying patch.txt to main/bridge_channel.c does not cause asterisk to crash when multiple speex users join a confbridge.



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