한국어

소프트스위치

온누리070 플레이스토어 다운로드
    acrobits softphone
     온누리 070 카카오 프러스 친구추가온누리 070 카카오 프러스 친구추가친추
     카카오톡 채팅 상담 카카오톡 채팅 상담카톡
    
     라인상담
     라인으로 공유

     페북공유

   ◎위챗 : speedseoul


  
     PAYPAL
     
     PRICE
     

pixel.gif

    before pay call 0088 from app


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. ;)

조회 수 :
101975
등록일 :
2013.04.06
22:36:46 (*.160.42.88)
엮인글 :
http://webs.co.kr/index.php?document_srl=19768&act=trackback&key=28b
게시글 주소 :
http://webs.co.kr/index.php?document_srl=19768
List of Articles
번호 제목 글쓴이 조회 수sort 추천 수 날짜
82 Presence Tutorial OpenXCAP setup admin 41416   2014-08-18
 
81 Real-time Charging System for Telecom & ISP environments admin 40885   2014-08-23
 
80 A lightweight RPC library based on XML and HTTP admin 40840   2014-08-18
 
79 A Survey of Open Source Products for Building a SIP Communication Platform admin 40692   2014-10-18
 
78 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 40651   2014-08-24
 
77 [Sipdroid] SIP data collection study tour admin 40047   2014-08-23
 
76 UAC Registrant Module admin 40032   2014-09-28
 
75 MediaProxy Installation Guide admin 39988   2014-08-10
 
74 OpenSER_from_an_asterisk_POV file admin 39832   2013-04-06
 
73 OpenSIPS Kick Start‎: VIDEO admin 39724   2013-02-20
 
72 CANCEL MESSAGE not handled correctly admin 39716   2014-08-23
 
71 rtpproxy Module admin 39715   2014-03-06
 
70 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 39281   2014-08-12
 
69 Opensips install debian admin 39206   2014-03-03
 
68 fusionPBX install debian wheezy admin 38972   2014-08-09
 
67 OPENSIPS EBOOK admin 38902   2014-08-21
 
66 SigIMS IMS Platform admin 38896   2014-05-24
 
65 A2Billing and OpenSIPS config admin 38835   2014-10-20
 
64 opensips-1.10.0_src.tar.gz experimental source code documentation admin 38740   2014-03-09
 
63 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 38356   2014-08-11
 
62 Multimedia Service Platform admin 38140   2014-03-06
 
61 opensips Nat script with RTPPROXY - English Good perfect admin 38029   2014-08-15
 
60 OPENSIP Training VIDEO admin 38019   2013-02-20
 
59 opensips.cfg for Asterisk admin 38002   2014-10-20
 
58 opensips NAT Traversal Module admin 37811   2014-10-02
 
57 kamailio.cfg configuration Example admin 37612   2014-10-04
 
56 OpenSIPS as Homer Capture server admin 37464   2014-08-13
 
55 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 37332   2014-08-12
 
54 Kamailio Nat Traversal using RTPProxy admin 37308   2014-08-11
 
53 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 37248   2014-08-23