[asterisk-bugs] [Asterisk 0018674]: [patch] Unable to choose which SRTP suite to offer

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18674 
====================================================================== 
Reported By:                bbeers
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18674
Category:                   Channels/chan_sip/SRTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-3142 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 303637 
Request Review:              
====================================================================== 
Date Submitted:             2011-01-25 09:56 CST
Last Modified:              2011-02-24 10:28 CST
====================================================================== 
Summary:                    [patch] Unable to choose which SRTP suite to offer
Description: 
Setting encryption=yes in sip.conf will cause asterisk to
 generate a line in SIP INVITE SDP:

 a=crypto: AES_CM_128_HMAC_SHA1_80 ...

There is no way to specify that asterisk should offer
 AES_CM_128_HMAC_SHA1_32 instead of
 AES_CM_128_HMAC_SHA1_80.

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0018187 Indicate SRTP + Feature reqest
====================================================================== 

---------------------------------------------------------------------- 
 (0132359) Irontec (reporter) - 2011-02-24 10:28
 https://issues.asterisk.org/view.php?id=18674#c132359 
---------------------------------------------------------------------- 
Hi,

I've tested latest patch (*22.patch) with the same result. Asterisk
Crash.


[Feb 24 17:14:36] WARNING[20316]: chan_sip.c:10446 get_crypto_attrib: No
SRTP key management enabled
    -- Called 200
    -- SIP/200-00000001 is ringing
[Feb 24 17:14:52] WARNING[20293]: chan_sip.c:8421 process_sdp: We are
requesting SRTP, but they responded without it!
    -- SIP/200-00000001 answered SIP/100-00000000
  == Using 0x1 (AES_CM_128_HMAC_SHA1_32) for srtp crypto offer.
[Feb 24 17:14:52] WARNING[20316]: chan_sip.c:10446 get_crypto_attrib: No
SRTP key management enabled
  == Spawn extension (remoteParty, 200, 2) exited non-zero on
'SIP/100-00000000'
*** glibc detected *** asterisk: double free or corruption (!prev):
0x000000000291db30 ***
======= Backtrace: =========
/lib/libc.so.6[0x7f4ac0f359a8]
/lib/libc.so.6(cfree+0x76)[0x7f4ac0f37ab6]
/usr/lib/asterisk/modules/chan_sip.so[0x7f4abdb194ca]
/usr/lib/asterisk/modules/chan_sip.so[0x7f4abdb1a385]
/usr/lib/asterisk/modules/chan_sip.so[0x7f4abda94bbb]
/usr/lib/asterisk/modules/chan_sip.so[0x7f4abda95b0e]
/usr/lib/asterisk/modules/chan_sip.so[0x7f4abda95a75]
asterisk[0x442783]
asterisk(__ao2_ref+0x38)[0x4426b2]
/usr/lib/asterisk/modules/chan_sip.so[0x7f4abda85b45]
/usr/lib/asterisk/modules/chan_sip.so[0x7f4abda87b6e]
/usr/lib/asterisk/modules/chan_sip.so[0x7f4abdacb4ea]
asterisk[0x443297]
asterisk(__ao2_callback+0x59)[0x4436ce]
/usr/lib/asterisk/modules/chan_sip.so[0x7f4abdaf4026]
asterisk[0x55d68b]
/lib/libpthread.so.0[0x7f4ac074cfc7]
/lib/libc.so.6(clone+0x6d)[0x7f4ac0f9164d]


With encryption=no there is also a crash but is a different one. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-24 10:28 Irontec        Note Added: 0132359                          
======================================================================