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http://nicerosniunos.blogspot.kr/2012/05/flooding-asterisk-freeswitch-and.html


Flooding Asterisk, Freeswitch and Kamailio with Metasploit

Hi, it has been a long time since my last post because of my new job and my final year project ("VoIP denegation of service attacks" for curious) but there is something I found during my tests with FreeswitchKamailio and Asterisk that I want to share.
NOTE: Really, guys of Security By Default blog published us (my good friend Roi Mallo and me) two articlesabout how to develop modules for Metasploit framework, another two are coming.  ;)

During my project, among others, I developed a Metasploit module which can flood SIP protocol with common frames (INVITE, OPTIONS, REGISTER, BYE), I wrote it at Quobis (nice job ;) in order to use it for some private tests because actual software didn´t fit our needs, so we are going to probe how is the behavior of different GPL VoIP servers against this kind of attacks:
- Asterisk: I think it needs no introduction, the famous softswitch/PBX software.
- Freeswitch: It´s a newer softswitch that seems to be Asterisk replacement and I really like.
- Kamailio (former OpenSER): It is the most known GPL SIP proxy.
Virtual machines
First of all I want to be clear about two things:
- Test were made without any protection on the server side, in a real environment we shoud find (in theory xD) something like Iptables, Snort, Fail2ban, Pike or a propietary Session border controller in large arquitectures. Anyway, it should be enough for this proof of concept.
- Asterisk and Freeswitch are PBX software, they were not designed to run between the limits of the infrastructure and Internet, although they are usually placed there. In fact, one of the reason of this post is to show the importance of using a SIP Proxy because of security and performance reasons.

Next pictures show an example of the Metasploit module use and generated traffic, we will use the same attack against differents IPs, so I´m showing it once only:
Module use and config
Captured traffic
I chose INVITE packets because they are much more effective against all kind of SIP devices and TIMEOUT to 0 trying to get more traffic. Then, the results:
NOTE: With Wireshark filter "sip.Method==REGISTER or sip.Status-Code==200 and !sdp" we can see if a softphone (Jitsi in this case) could be registered , this way we can confirm if tested software losts some REGISTER packages under attack.
Metasploit vs. Asterisk

Metasploit vs. Freeswitch
 

Metasploit vs. Kamailio

Pictures show how Metasploit module can flood both Asterisk and Freeswitch, but not Kamailio. Moreover, Asterisk lost REGISTER packets under the attack and Freeswitch did "strange" things answering with a lot of "200 OK" responses. This problem would be much more important in a real environment with hundreds of phones trying to register at the same time.

As conclusion we can confirm the use of Kamailio (I think OpenSIPS or another SIP Proxy would reach the same results) as frontier with "the wild". In addition we can also use Pike module for DoS protection and we could suppose that it would respond to a high volume of traffic in a better way than other two alternatives. To sum up I would like to remark that we can see Kamailio creates different forks to manage connections, this seems to be the key of its good performance. But next times I will show how to flood Kamailio with better results and the countermeasurements to protect yourself against it. ;)

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103997
등록일 :
2013.04.06
22:36:46 (*.160.42.88)
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http://webs.co.kr/index.php?document_srl=19768&act=trackback&key=1a5
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http://webs.co.kr/index.php?document_srl=19768
List of Articles
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109 MediaProxy wiki page install configuration admin 44415   2014-08-11
 
108 Kamailio Nat Traversal using RTPProxy admin 38203   2014-08-11
 
107 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 39134   2014-08-11
 
106 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 39985   2014-08-12
 
105 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 38144   2014-08-12
 
104 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 42194   2014-08-13
 
103 OpenSIPS as Homer Capture server admin 38298   2014-08-13
 
102 Installation and configuration process record opensips opensips-cp admin 71821   2014-08-13
 
101 OpenSIPS Control Panel (OCP) Installation Guide Good admin 69971   2014-08-13
 
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99 A lightweight RPC library based on XML and HTTP admin 41903   2014-08-18
 
98 Opensips Modules Documentation admin 43886   2014-08-18
 
97 Presence Tutorial OpenXCAP setup admin 42384   2014-08-18
 
96 Opensips Documentation Function admin 42849   2014-08-21
 
95 OPENSIPS EBOOK admin 40014   2014-08-21
 
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93 Many OPENSIPS Configuration Examples This will Help you admin 50958   2014-08-23
 
92 ICE: The ultimate way of beating NAT in SIP admin 67160   2014-08-23
 
91 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 42977   2014-08-23
 
90 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 38386   2014-08-23
 
89 [Sipdroid] SIP data collection study tour admin 41195   2014-08-23
 
88 CANCEL MESSAGE not handled correctly admin 40688   2014-08-23
 
87 RTPPROXY Admin Guide admin 43222   2014-08-24
 
86 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 42198   2014-08-24
 
85 UAC Registrant Module admin 41083   2014-09-28
 
84 opensips NAT Traversal Module admin 39154   2014-10-02
 
83 kamailio.cfg configuration Example admin 39015   2014-10-04