[asterisk-bugs] [JIRA] (ASTERISK-24639) Crash with PJSIP on SIP to SIP over WebSockets call (WebRTC, SIPML5)

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24639?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-24639:
------------------------------------

    Attachment: full.txt

> Crash with PJSIP on SIP to SIP over WebSockets call (WebRTC, SIPML5)
> --------------------------------------------------------------------
>
>                 Key: ASTERISK-24639
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24639
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>         Environment:  * Asterisk SVN-branch-13-r429983
>  * PJPROJECT 2.3 Compiled from source with (./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-speex --with-external-srtp --with-external-gsm CFLAGS='-O2 -DNDEBUG -DPJ_HAS_IPV6=1'),
>  * OpenSSL 1.0.1-4ubuntu5.20
>            Reporter: Rusty Newton
>         Attachments: backtrace.txt, full.txt
>
>
> Seemingly very similar to ASTERISK-24334, except happens when using PJSIP, newer openssl, newer PJPROJECT and Asterisk 13 as well.
> I need a developer to take a closer look and verify if this is a duplicate.
> To reproduce, I just follow the tutorial that worked in the past: https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
> The crash happens when calling from a SIP phone to the WebRTC client.



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