[asterisk-bugs] [JIRA] (ASTERISK-22920) Crash while Forwarding from TLS extension

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22920?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214615#comment-214615 ] 

Rusty Newton commented on ASTERISK-22920:
-----------------------------------------

Well.. Apparently most of the freely available soft phones that support TLS and SRTP don't also support call forwarding or call transfers with a 302, so it wasn't quick to reproduce.

Fortunately a colleague had a Snom 370 setup and was able to reproduce your issue. We reproduced it with optimizations, so the backtrace is pretty much the same as yours. Tomorrow he'll try to get one without optimizations. I'll attach the very minimal sip.conf required to help any developer that reproduces this.

The key to reproduction is the dialplan:

{noformat}
same => n,Set(CHANNEL(secure_bridge_signaling)=1)
same => n,Set(CHANNEL(secure_bridge_media)=1)
{noformat}
Without this, the forward works fine.

With the options set, the crash occurs. We didn't narrow it down to one option or the other.

                
> Crash while Forwarding from TLS extension
> -----------------------------------------
>
>                 Key: ASTERISK-22920
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22920
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_srtp
>    Affects Versions: 1.8.14.0, 1.8.24.0, 11.2.2, 11.5.0, 11.6.0, 11.7.0
>         Environment: CentOS release 5.8 (Final)  kernel 2.6.18-308.24.1.el5 64bit, libsrtp 1.4.2(compiled manually)  with 1.8.14 with and without patch (https://issues.asterisk.org/jira/browse/ASTERISK-18345)
> Debian GNU/Linux 7 (wheezy) kenrel 3.2.0-4-amd64 (3.2.51-1 64bit), with above patch on 11.5.0 and without patch on 1.8.24.0 11.7.0-rc1 11.6.0
> with libsrtp 1.4.4 (from debian repo), self compiled 1.4.2, as well as 1.4.4 self compiled and self compiled with patch ( http://srtp.cvs.sourceforge.net/viewvc/srtp/srtp/crypto/replay/rdb.c?r1=1.4&r2=1.5) as mentioned on https://issues.asterisk.org/jira/browse/ASTERISK-16665
> 2 phones were tested snom 710 and fanvil C62 
>            Reporter: Shlomi Gutman
>            Assignee: Shlomi Gutman
>         Attachments: backtrace_ldd.log, debug.log, exten_incoming.conf, extension_realtime.info, gdb.log, ldd.log, sip.conf
>
>
> Steps to reproduce:
> 1)Asterisk with self signed certificates or GoDaddy certificates
> 2)Extension connected with TLS transport (behind NAT in our case)
> 3)Route incoming call to that extension, while forward call from it without answering (302 - FORWARD)
> 4)Crash
> I know that this bug may be related to srtp, but as we see it was not developed and maintained for a long time and as asterisk srtp based on itץ
> I think at least it should crash the call only, but not whole asterisk.

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