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 :
50509
등록일 :
2013.04.06
22:36:46 (*.160.42.88)
엮인글 :
http://webs.co.kr/index.php?document_srl=19768&act=trackback&key=8ff
게시글 주소 :
http://webs.co.kr/index.php?document_srl=19768
List of Articles
번호 제목 글쓴이 날짜 조회 수sort
162 smpp sms opensips admin 2019-02-19 1633
161 Opensips Gateway between SIP and SMPP messages admin 2019-02-19 2125
160 Documentation -> Tutorials -> WebSocket Transport using OpenSIPS admin 2018-05-17 3219
159 Busy Lamp Field (BLF) feature on Opensips 2.4.0 with Zoiper configuration admin 2018-05-29 3388
158 OpenSIPS example configuration admin 2017-12-12 4361
157 opensips log output admin 2017-12-11 4383
156 opensips push notification How to detail file admin 2017-12-20 4405
155 OpenSIPS routing logic admin 2017-12-12 4468
154 opensips/modules/event_routing/ Push Notification Call pickup admin 2017-12-20 4546
153 Opensips1.6 ebook detail configuration and SIP signal and NAT etc file admin 2017-12-10 4599
152 List of SIP response codes admin 2017-12-20 5023
151 opensips complete configuration example admin 2017-12-10 5072
150 in opensips db_does_uri_exist() what is admin 2017-12-09 5535
149 in opensips what is lookup(domain [, flags [, aor]]) admin 2017-12-09 5679
148 dictionary.opensips radius admin 2017-12-09 5733
147 OpenSIPS Module Interface admin 2017-12-07 5748
146 in opensips what is has_totag() admin 2017-12-09 5752
145 opensips push notification How to admin 2017-12-07 5769
144 what is record_route() in opensips ? admin 2017-12-09 5806
143 opensips exec module admin 2017-12-08 5946
142 opensips configuration config explain easy basic 오픈쉽스 컨피그레이션 기본 설명 file admin 2017-12-07 5988
141 what is loose_route() in opensips ? file admin 2017-12-09 6021
140 Build-Depends debian 8.8 opensips 2.3 admin 2017-09-04 6507
139 opensips.cfg. sample admin 2017-09-12 6528
138 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 6657
137 Opensips 2.32 download admin 2017-09-01 6667
136 opensips command /sbin/opensipsctl detail admin 2017-09-04 6697
135 A2Billing and OpenSIPS – Part 2 admin 2017-08-29 6721
134 Advanced SIP scenarios with Event-based-Routing admin 2017-09-11 6730
133 Installing RTPEngine on Ubuntu 14.04 admin 2017-09-05 6730