[asterisk-bugs] [JIRA] (ASTERISK-20499) Crash in libsrtp srtp_unprotect_rtcp when SIP channel is bridged with non-optimizing Local channel

[Jonathan Rose (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20499?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=200217#comment-200217 ] 

Jonathan Rose commented on ASTERISK-20499:
------------------------------------------

Well, from the patch...

+	ast_log(LOG_NOTICE, "srtp_diagnostic_patch: attempting srtp_create at %p\n", &temp->session);
+	res = srtp_create(&temp->session, &policy->sp);
+	ast_log(LOG_NOTICE, "srtp_diagnostic_patch: srtp_errstr = %s\n", srtp_errstr(res));

For you to have received the first log message there and not the second without something that just stops the whole process from occurring happening in srtp_create isn't possible. And we know that this didn't happen because you received the log message from the following elements of the patch:

+	if (res != err_status_ok) {
+		ast_log(LOG_NOTICE, "srtp_diagnosic_patch: result of srtp_create at %p was failure\n", &temp->session);


Also in your 1.8 log messages, it is plainly visible that there is only one line between the first log message you received in and the second (occuring on 442 and 444 respectively). In the most recent version of the patch there are 3 lines between those two log messages. Since there was only one line between the log messages mentioned in the original patch, you should pretty easily be able to see why I believe you applied the wrong patch.

While I'm working on it, the extensions you included with your file showing sip settings and such omitted some macros that seem to be essential to completing the call. Could I please get the contents of macro-filterExten, macro-setDialingExtenFromSipHeader, macro-monitor, and macro-OFFICE-Numbers?
                
> Crash in libsrtp srtp_unprotect_rtcp when SIP channel is bridged with non-optimizing Local channel
> --------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-20499
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20499
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP
>    Affects Versions: 10.8.0
>         Environment: RHEL 5.8 on IBM X3650 M4 - 12 core - Xeon E5-2640 @ 2,50 ghz
>            Reporter: tootai
>            Assignee: tootai
>            Severity: Critical
>         Attachments: asterisk-20499_20121127.log, asterisk-20499_20121127.pcap, asterisk-20499.txt, backtrace.txt, backtrace.txt, backtrace.txt, backtrace.txt, coredump20121001205609.txt, gdb.txt, gdb.txt, gdb.txt, gdb.txt, libsrtp-1.4.4-fix_crash_on_rtcp_decode.patch, srtp_diagnostic_with_sleep.diff, srtp_fixes_it_maybe.diff
>
>
> A call from snom320 in SRTP mode to echo test or to another phone *NOT* using SRTP is OK. Now we installed PhonerLite softphone with TLS/SRTP stuf and test with echo test: everything is OK too.
> Now PhonerLite calls the snom: asterisk coredump after 3~5 seconds and we are NOT able to make anymore SRTP calls after this, they all crash asterisk. We had this issue with 10.7.0 and 10.8.0
> We have logfiel from strace as well as coredump.

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