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

[newbie (JIRA)]

newbie created ASTERISK-20603:
---------------------------------

             Summary: 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: . I did not set the category correctly.
    Affects Versions: 1.8.10.1
         Environment: asterisk 1.8.1 and jitsi
            Reporter: newbie
            Severity: Blocker


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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