[asterisk-bugs] [JIRA] (ASTERISK-20603) Crash Asterisk 1.8.1 during SRTP

[Michael L. Young (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20603?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=198902#comment-198902 ] 

Michael L. Young commented on ASTERISK-20603:
---------------------------------------------

Thank you for your bug report. In order to move your issue forward, we require a backtrace[1] from the core file produced after the crash. Also, be sure you have DONT_OPTIMIZE enabled in menuselect within the Compiler Flags section, then:

make install

After enabling, reproduce the crash, and then execute the backtrace[1] instructions. When complete, attach that file to this issue report.

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Also, can you reproduce this on a newer version of Asterisk?  The latest version is Asterisk 1.8.17.
                
> Crash Asterisk 1.8.1 during SRTP
> --------------------------------
>
>                 Key: ASTERISK-20603
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20603
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_srtp
>    Affects Versions: 1.8.10.1
>         Environment: asterisk 1.8.1 and jitsi
>            Reporter: newbie
>
> asterisk 1.8.1 going crashed while running SRTP with jitsi.
> TLS is working fine.
> brief is below 
> sip.conf
> [general]
> context=incoming
> allowguest=no
> alwaysauthreject=yes
> allow=ulaw
> allow=alaw
> allow=gsm
> tlsenable=yes
> tlsbindaddr=0.0.0.0
> tlscertfile=/etc/asterisk/keys/newbie.pem
> tlscafile=/etc/asterisk/keys/ca.crt
> tlscipher=ALL
> tlsclientmethod=tlsv1
> [user1]
> type=peer
> defaultuser=user1
> secret=1000
> dtmfmode=rfc2833
> callerid="User one"
> host=dynamic        ; The device must always register
> canreinvite=no
> nat=yes
> encryption=yes
> transport=tls
> ; Deny registration from anywhere first
> deny=0.0.0.0/0.0.0.0
> ; Replace the IP address and mask below with the actual IP address and mask
> ; of the computer running the softphone, or the address of the hardware phone,
> ; either a host address and full mask, or a network address and correct mask,
> ; registering will be allowed from that host/network.
> permit=192.168.51.0/255.255.255.0
> context=myphones
> [user2]
> type=peer
> defaultuser=user2
> secret=1001
> dtmfmode=rfc2833
> callerid="User two"
> host=dynamic        ; The device must always register
> canreinvite=no
> nat=yes
> encryption=yes
> transport=tls
> ; Deny registration from anywhere first
> deny=0.0.0.0/0.0.0.0
> ; Replace the IP address and mask below with the actual IP address and mask
> ; of the computer running the softphone, or the address of the hardware phone,
> ; either a host address and full mask, or a network address and correct mask,
> ; registering will be allowed from that host/network.
> permit=192.168.51.0/255.255.255.0
> context=myphones
> extension.conf
> [general]
> static=yes
> writeprotect=no
> clearglobalvars=no
> [incoming]
> exten => s,1,Hangup()
> [myphones]
> exten => user1,1,Set(CHANNEL(secure_bridge_signaling)=1)
> exten => user1,n,Set(CHANNEL(secure_bridge_media)=1)
> exten => user1,n,Dial(SIP/user1)
> exten => user1,n,Hangup()
> exten => user2,1,Set(CHANNEL(secure_bridge_signaling)=1)
> exten => user2,n,Set(CHANNEL(secure_bridge_media)=1)
> exten => user2,n,Dial(SIP/user2)
> exten => user2,n,Hangup()
> exten => 201,1,Answer()
> exten => 201,n,Playback(tt-monty-knights)
> exten => 201,n,Hangup()
> exten => 202,1,Answer()
> exten => 202,n,Playback(welcome)
> exten => 202,n,Playback(demo-echotest)
> exten => 202,n,Echo()
> exten => 202,n,Playback(demo-echodone)
> exten => 202,n,Playback(vm-goodbye)
> exten => 202,n,Hangup()
> i upload srtp module also. it got loaded. But when user1 call to user2 my asterisk server getting segmentation fault and shut down.

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