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

[Sean Bright (JIRA)]

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

Sean Bright updated ASTERISK-24639:
-----------------------------------

    Assignee: Rusty Newton
      Status: Waiting for Feedback  (was: Open)

Is this reproducible with current Asterisk 13?

> 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
>            Assignee: Rusty Newton
>         Attachments: backtrace.txt, extensions.txt, full.txt, http.txt, jssip_full.txt, pjsip.txt, rtp.txt
>
>
> Seemingly very similar to ASTERISK-24334, except happens when using PJSIP, newer openssl, newer PJPROJECT and Asterisk 13 as well.
> h1. Reproduction
> 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. In this case, a Digium D40 to SIPML5 (live demo).
> h1. Notes
> backtrace.txt is the trace from the crash occurring when calling from a Digium D40 to SIPML5.  The full.txt is the full log trace with pjsip logger output.
> The jssip_full.txt contains a full log from the same call scenario, but swapping out the SIPML5 client with JsSIP. Calling from the D40 to JsSIP results in a failed call, but no crash. JsSIP responds to our INVITE with 488 Not Acceptable Here.



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