[asterisk-bugs] [JIRA] (ASTERISK-24832) [patch]DTLS-crashes within openssl

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24832?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-24832:
------------------------------------

    Summary: [patch]DTLS-crashes within openssl   (was: DTLS-crashes within openssl )

> [patch]DTLS-crashes within openssl 
> -----------------------------------
>
>                 Key: ASTERISK-24832
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24832
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 13.1.0
>         Environment: Fedora 20 x86_64, openssl-1.0.1e-41.fc20.x86_64, Asterisk 13.1.0, Chrome SIPML5 chan_sip peers with transport WSS
>            Reporter: Stefan Engström
>            Assignee: Rusty Newton
>         Attachments: crash1.txt, crash2.txt, crash3.txt, crash4.txt, crash5.extralog, crash5.txt, CUSTOMERRORDEBUGLOG, DTLSfailure6ErrorlogNocrashUsingNewPatch.txt, DTLSREVIEWMEcodecontribution.patch, DTLSREVIEWME.patch, SIPCONF.txt, TESTDTLS.patch
>
>
> I'm using 4 chan sip peers with transport WSS. They all use Chrome SIPml5 webrtc. 2 of them call a queue and the other 2 answer. Every 100-1000 calls or so, asterisk gets a crash due to segmentation fault or abort signal within openssl.
> Since it's load-related it's hard to provide enough information but ill try add more continuously.
> ISSUE-0
> First thing i noticed was  that dtls_perform_handshake was called too many times but that was fixed by compensating for ASTERISK-24830 
> By code inspection and tracing logs; it looks like the crashes mostly occur for dtls->ssl instances where asterisk has role: server, (SSL_set_accept_state(dtls->ssl) has been called.) 
> EDIT -- This JIRA is getting a little bigger. It seems there are many sub-problems which are all related to DTLS though... not all sub-issues below may be real issues, some are just me asking questions about code. I'd be happy if a developer took a look at it and answered questions or discussed some of the issues and possible fixes.
> ISSUE-1 - crash3 seems to prove a concurrency issue:
> thread 5 leaving asterisk code at dtls_perform_handshake is performing ssl3_clear on the same ssl struct as that which is sent to ssl_read from __rtp_recvfrom in thread 1
> ISSUE-2: Im curious about the behavior of ast_rtp_on_ice_complete() {
> ...
>         dtls_perform_handshake(instance, &rtp->dtls, 0);
>         if (rtp->rtcp) {
>                 dtls_perform_handshake(instance, &rtp->rtcp->dtls, 1);
>         }
> ...
> }
> chan_sip seems to call process_sdp which eventually calls res_asterisk_rtp::dtls_set_setup which ultimately sets SSL_set_connect_state(ssl) OR SSL_set_accept_state(ssl) on both (RTP+RTCP) ssl sessions. But this races with the firing of  dtls_perform_handshake(instance, &rtp->dtls, 0); from ast_rtp_on_ice_complete. I'm not sure if this is a problem but in my last crash crash4 the ast_ice_on_ice_complete fired before dtls_set_setup which i have never noticed during non-crash-calls,
> the big question is why is dtls_perform_handshake() called at all if we are passive? After I added a patch it doen't seem to crash anymore, but I'm pretty sure the patch is not a full fix -- note that the patches labeled test are just for the debugging process
> Possibly related to ASTERISK-24651
> Requires patch from ASTERISK-24711
> Requires patch from ASTERISK-24830  (the obvious fix of replacing USE_PJPROJECT WITH HAVE_PJPROJECT...)



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