Summary:	ASTERISK-20603: Crash Asterisk 1.8.1 during SRTP
Reporter:	newbie (newbie1234)	Labels:
Date Opened:	2012-10-25 01:43:35	Date Closed:	2012-11-08 15:56:55.000-0600
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Resources/res_srtp
Versions:	1.8.10.1	Frequency of Occurrence	Constant
Related Issues:
Environment:	asterisk 1.8.1 and jitsi	Attachments:
Description:	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.
Comments:	By: Michael L. Young (elguero) 2012-10-25 08:28:17.213-0500 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. By: Jonathan Rose (jrose) 2012-10-25 15:46:01.636-0500 I'd like to echo that sentiment. If at all possible, can you please see if this is reproducible in 1.8.17.  1.8.1 was released quite a good while ago (nearly two years ago) and there have been numerous patches involving SRTP since then. *also worth checking if you are using 1.8.1 or 1.8.10.1... the issue tags suggest this is against 1.8.10.1, but your description says 1.8.1. By: Rusty Newton (rnewton) 2012-11-08 15:56:36.103-0600 Suspended due to lack of activity. Should you have the additional information requested, then please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue .Further information can be found at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines