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/


Hi again guys, here there is my new personal project. I think that README file is complete enough so I paste it on this post.

Next month I'll be with my colleague Antón at Kamalio World Conference showing a bit more about it. If you are there and want to talk a bit about VoIP security (or WebRTC) get in contact with us please. :)

Finally, we would like to publish the first version in one ore two months, sorry but we're developing it mostly in our free time :(. I've promised Yago to do it onSecurity by Default blog so stay tuned. 

Moreover this tool was included in Quobis personal project plan so you can always follow Quobis planet in which we publish all our experiments.

Nothing else, I hope you like it and all kind of suggestions (and coders) are welcomed :).


Bluebox-ng

Bluebox-ng is a next generation UC/VoIP security tool. It has been written in CoffeeScript using Node.js powers. This project is "our 2 cents" to help to improve information security practices in VoIP/UC environments.

Install deps

  •  cd bluebox-ng
  • npm install

Run

  • npm start

Features

  • Automatic pentesting process (VoIP, web and service vulns)
  • SIP (RFC 3261) and extensions compliant
  • TLS and IPv6 support
  • VoIP DNS SRV register support
  • SIP over websockets (and WSS) support (draft-ietf-sipcore-sip-websocket-08)
  • REGISTER, OPTIONS, INVITE, MESSAGE, SUBSCRIBE, PUBLISH, OK, ACK, CANCEL, BYE, Ringing and Busy Here requests support
  • Extension and password brute-force through different methods (REGISTER, INVITE, SUBSCRIBE, PUBLISH, etc.)
  • DNS SRV registers discovery
  • SHODAN and Google Dorks
  • SIP common vulns modules: scan, extension brute-force, Asterisk extension brute-force (CVE-2011-4597), invite attack, call all LAN endpoints, invite spoofing, registering hijacking, unregistering, bye teardown
  • SIP DoS/DDoS audit
  • SIP dumb fuzzer
  • Common VoIP servers web management panels discovery and brute-force
  • Automatic exploit searching (Exploit DB, PacketStorm, Metasploit)
  • Automatic vulnerability searching (CVE, OSVDB)
  • Geolocalization using WPS (Wifi Positioning System) or IP address (Maxmind database)
  • Colored output
  • Command completion

Roadmap

  •  Tor support
  • More SIP modules 
  • SIP Smart fuzzing (SIP Torture RFC)
  • Eavesdropping
  • CouchDB support (sessions)
  • H.323 support
  • IAX support
  • Web common panels post-explotation (Pepelux research)
  • A bit of command Kung Fu post-explotation
  • RTP fuzzing
  • Advanced SIP fuzzing with Peach
  • Reports generation
  • Graphical user interface
  • Windows support
  • Include in Debian GNU/Linux
  • Include in Kali GNU/Linux
  • Team/multi-user support
  • Documentation
  • ...
  • Any suggestion/piece of code ;) is appreciated.

Author

Jesús Pérez

Contributors

Damián Franco
Jose Luis Verdeguer

Thanks to ...

  • Quobis, some hours of work through personal projects program
  • Antón Román (@AntonRoman), he speaks SIP and I'm starting to speak it thanks to him
  • Sandro Gauci (@sandrogauci), SIPVicious was our inspiration
  • Kamailio community (@kamailioproject]), my favourite SIP Server
  • David Endler and Mark Collier (@markcollier46), authors of "Hacking VoIP Exposed" book
  • John Matherly (@achillean) for SHODAN API and GHDB
  • All VoIP, free software and security hackers that we read everyday
  • Loopsize, a music lhacker (creator of themes included in video demos)

License

This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.

This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
GNU General Public License for more details.

You should have received a copy of the GNU General Public License
along with this program.  If not, see .

2/26/2013

How to protect your WebRTC app code?

I have spent some time analyzing which could be the best way to protect a privative version of a webphone based on QoffeeSIP that we are developing now at Quobis. I have seen this same question on different sites with quite confusing responses. So I'm going to share what I learned just in case it could help to anybody.

Well, I'm not going to define what is WebRTC because Internet is full of it this year (only overtaken by cats ;). For our purposes we have to consider that our app is a Javascript library. Really there is also HTML/CSS code but what I think that is important is Javascript, but HTML/CSS can also be protected in the same way but with other tools.

First of all I want to remark that protect your code in the sense of anybody could copy/modify and redistribute it is impossible since Javascript is only text. If anybody had enough time (or money) this code could be reversed. But, as always, we can do things trying to avoid it as far as possible.

In general, I found that there is a bit confusion between minimize and obfuscate terms so we're going to speak a bit about these techniques.

Minimization

The target is to get the code as small as possible. Obviously generated code is more difficult to understand, but it could be easily reversed with tools like JSbeautifier. (really not as easy depending of the minimizing tool)

Some common possible options at this point are:

  • UglifyJS: The coolest thing right now xD. It is a Node.js package so it's easy to include. Some days ago version 2 was published. We will see that it's fast, really fast.
  • Google Closure Compiler which uses Google to its apps. It is availiable a Java command line tool but there are node modules which use the online API.
  • YUI Compressor from Yahoo, it was the facto standard but now last alternatives are beating it.
A little comparison: I can't find original link, sorry :(
  • Average time: (lower is better)
    • UglifyJS: 0.11554 seconds
    • Closure: 1.41037 seconds
  • Average reducction: (higher is better)
    •  UglifyJS: 45.6%
    • Closure: 51.5%
NOTE: Another one (more complete) with YUI included too.

In my experience Google Closure generated code is better because besides minimization tasks it includes code checking too. It provides warnings for dangerous or illegal Javascript. Moreover I like that you can use this online serviceto check your code while developing.



Obfuscation

It is defined as "the hiding of intended meaning in communication, making communication confusing, wilfully ambiguous, and harder to interpret."(Wikipedia).

We have some options here when we are working with a web app:
  • Encrypt the transport layer: needed to avoid sniffing to another users of the same LAN. So using HTTPS to serving the application is a must.
  • Encryption: Encrypt application data and decrypt it on the fly via your own javascript enccryption library.
  • Move functions to the server side, which it's not possible in the case of WebRTC because we want end to end media.
  • Use a browser plugin, it has no sense since one of the advantages of WebRTC is that the user doesn't have to install anything.
  • Implement the code in native client for Chrome browser. The advantaje is that common C code protections can be used and the app runs sandboxed. But it is not our case because we need multi-platform support.
  • To avoid legal issues you should incude a note (a Javascript comment)referencing the copyright in each copy of the .js library. Something similar to Free Software Foundation recommendations for free Javascript code. An example could be:
NOTE: Really @source tag is proposed by FSF to include a link to source code of the app. But I think that it could be a good idea to use it because browser plugins that follow the recommendations should "understand" it.

// @source: https://qoffeesip.quobis.com
// Copyright (C) Quobis
// Licensed under Quobis Commercial license
// (http://www.quobis.com/licenses/commercial-1.0.html)

I also want to point out some common obfuscation/encryption problems:
  • Performance decrement, specially speed.
  • Increase troubleshooting difficult.
  • Compatibility problems (IE!!).
  • Size increase.
  • As it was said, a skilled expert could always reverse it and get a code equivalent to ours.
All these problems are more important on the case of encryption, except the last one logically. So at this point we have some options, but I've reduced them to these ones:
  • A paid option like JsCrambler: This is the reference tool, generated code seems to be really dificult to recover and it supports an important number of encryption algorithms.
  • A free solution provided by my colleague DamiánHorrible.js. It implements obfuscation and a kind of simple (so light) optional (through "factor" parameter) encryption. Next picture shows an example using it with the three different factors.

Finally, if you don't like the ugly generated code you can always use Nice.js to get something like this example: xD



In conclusion, I like Horrible.js with factor 3. In my opinion, it has no sense to paid for mitigating a risk impossible to solve completely.

1/19/2013

SIP INVITE attack with Metasploit

Some days ago my friend @pepeluxx wrote another post about INVITE attacks. He spoke about a @sinologic project which allows to everybody passing some security tests to SIP servers. Furthermore he also published a perl script to do the same task. So I implemented it on Metasploit because I think It could be really useful during a pentesting. It’s interesting because these attacks are really dangerous, normally, attackers try to call to expensive locations. This target numbers often have special charges and they make money with this. Here there are two well known examples:


I’m not going to deep in this vector because of being a well known (and old!!) one. Basically the attacker tries to make a call using a misconfigured PBX. This is allowed because SIP RFC says that an extension has not to be registered to be able to make a call, only to receive it. Really most SIP servers implement authentication both in registering and calling process (and even to hang up a call), this is useful in eavesdropping scenarios in order to avoid SIP Teardown (BYE) attacks. But only a few systems have this configuration enabled by default, most of them use authentication only to register. In example, for Asterisk we should change “allowguest=no” in "sip.conf" file to ask for authentication in each call (INVITE). Apart from this, sysadmins should be also very carefully defining the dialplan to be secure. A common example of what not to do is the next one, in where outbound (to PSTN) calls context is included in default one:

(sip.conf file)
[general]
context=default

(extensions.conf file)
[default]
include  => outbound

I committed the module to my Github project, it only implements a SIP INVITE request where the user can provide next parameters:

Module parameters

You should try to call to a common phone number (you can see it in last picture) and with an extension because servers normally work in a different way. The code simply sends an INVITE request with provided options and then it parses the response. If it is a “Trying” you could be in a problem man. ;)

Possible insecure system

Possible insecure system

Secure system to this vector

These are the links to both UDP and TCP version of the tool. I would like to remember that Metasploit modules which support TCP also support TLS. You can change the version of the protocol and another optional parameters with command“show advanced”.
Advanced options

Finally I want to say that last days I was reviewing my SIP Metasploit modules trying to add some more features (like SIP proxy support) and I found that they are a mess. There is a lot of repeated code and they are complex to maintain. So, after speaking with some Metasploit guys on irc channel, I’m going to write a new SIP Proto ("lib/rex/proto/sip.rb") class and a Mixin ("lib/msf/core/auxiliary/sip.rb") which uses it. Once solved this I’m going to add all SIP modules I have developed to official Metasploit distribution.

Ref: http://www.sinologic.net/blog/2009-02/la-voip-mal-configurada-llama-a-cuba/

조회 수 :
38474
등록일 :
2013.04.06
22:44:35 (*.160.42.88)
엮인글 :
http://webs.co.kr/index.php?document_srl=19770&act=trackback&key=ffc
게시글 주소 :
http://webs.co.kr/index.php?document_srl=19770
List of Articles
번호 제목 글쓴이 조회 수 추천 수 날짜
162 Opensips Gateway between SIP and SMPP messages admin 78   2019-02-19
 
161 smpp sms opensips admin 75   2019-02-19
 
160 Busy Lamp Field (BLF) feature on Opensips 2.4.0 with Zoiper configuration admin 1781   2018-05-29
 
159 Documentation -> Tutorials -> WebSocket Transport using OpenSIPS admin 1639   2018-05-17
 
158 List of SIP response codes admin 3303   2017-12-20
 
157 opensips/modules/event_routing/ Push Notification Call pickup admin 2860   2017-12-20
 
156 opensips push notification How to detail file admin 2767   2017-12-20
 
155 OpenSIPS routing logic admin 2839   2017-12-12
 
154 OpenSIPS example configuration admin 2821   2017-12-12
 
153 opensips log output admin 2826   2017-12-11
 
152 opensips complete configuration example admin 2915   2017-12-10
 
151 Opensips1.6 ebook detail configuration and SIP signal and NAT etc file admin 2912   2017-12-10
 
150 dictionary.opensips radius admin 3831   2017-12-09
 
149 what is record_route() in opensips ? admin 3753   2017-12-09
 
148 what is loose_route() in opensips ? file admin 3870   2017-12-09
 
147 in opensips what is lookup(domain [, flags [, aor]]) admin 3784   2017-12-09
 
146 in opensips db_does_uri_exist() what is admin 3637   2017-12-09
 
145 in opensips what is has_totag() admin 3791   2017-12-09
 
144 opensips exec module admin 3964   2017-12-08
 
143 opensips push notification How to admin 3737   2017-12-07
 
142 OpenSIPS Module Interface admin 3873   2017-12-07
 
141 opensips configuration config explain easy basic 오픈쉽스 컨피그레이션 기본 설명 file admin 3921   2017-12-07
 
140 openssl 을 이용한 인증서 생성 절차를 정리한다. 개인키 CSR SSL 인증서 파일 생성 admin 5000   2017-09-14
 
139 Documentation -> Tutorials -> TLS opensips.cfg admin 4778   2017-09-14
 
138 Using TLS in OpenSIPS v2.2.x admin 4750   2017-09-14
 
137 opensips tls cfg admin 4888   2017-09-14
 
136 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 5393   2017-09-13
 
135 SIP to XMPP Gateway + SIP Presence Server opensips admin 4738   2017-09-13
 
134 OpenSIPS command line tricks admin 4713   2017-09-13
 
133 Fail2Ban Freeswitch How to secure admin 4995   2017-09-12
 
132 opensips.cfg. sample admin 4700   2017-09-12
 
131 Advanced SIP scenarios with Event-based-Routing admin 4850   2017-09-11
 
130 PUSH SERVER 푸시서버 안드로이드 애플 admin 5206   2017-09-11
 
129 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅(리눅스 기준) admin 12569   2017-09-09
 
128 rtpengine config basic and opensips configuration and command admin 5003   2017-09-06
 
127 WebSocket Transport using OpenSIPS configuration 웹 소켓 컨피그레이션 기본 admin 4817   2017-09-06
 
126 OpenSIPS basic configuration script 기본 컨피그 admin 4950   2017-09-05
 
125 rtpengine install and config admin 4902   2017-09-05
 
124 Installing RTPEngine on Ubuntu 14.04 admin 4995   2017-09-05
 
123 compile only the textops module make modules=modules/textops modules admin 4892   2017-09-05
 
122 opensips command /sbin/opensipsctl detail admin 4979   2017-09-04
 
121 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 4931   2017-09-04
 
120 Build-Depends debian 8.8 opensips 2.3 admin 4812   2017-09-04
 
119 What is new in 2.3.0 opensips admin 5572   2017-09-04
 
118 ubuntu 安装配置opensips,rtpproxy,mediaproxy admin 5182   2017-09-04
 
117 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 5378   2017-09-04
 
116 Using TLS in OpenSIPS v2.2.x configuration admin 5056   2017-09-04
 
115 How to 2.3 download , OpenSIPS new apt repository. DEBs for Debian / Ubuntu admin 5029   2017-09-02
 
114 You can install CDRTool in the following ways: admin 5248   2017-09-01
 
113 How to Install OpenSIPS 2.1.2 Server on Ubuntu 15.04 admin 5246   2017-09-01
 
112 Opensips 2.32 download admin 5011   2017-09-01
 
111 OpenSIPS 2.3 install admin 5293   2017-09-01
 
110 JsSIP: The JavaScript SIP Library admin 5283   2017-09-01
 
109 WebSocket Transport using OpenSIPS admin 5365   2017-09-01
 
108 A2Billing and OpenSIPS – Part 1 admin 5089   2017-08-29
 
107 A2Billing and OpenSIPS – Part 2 admin 4989   2017-08-29
 
106 A2Billing and OpenSIPS – Part 3 admin 5206   2017-08-29
 
105 OpenSIPS 2.3 philosophy admin 5696   2017-08-17
 
104 The timeline for OpenSIPS 2.3 is admin 5871   2017-08-17
 
103 OpenSIPS Control Panel and Homer integration admin 5809   2017-08-17
 
102 Opensips sip capture re designed admin 5372   2017-07-16
 
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 10345   2015-04-04
 
100 WebSocket Support in OpenSIPS 2.1 admin 11225   2015-04-04
 
99 OpenSIPS 2.1 (rc) is available, download now! admin 10238   2015-03-22
 
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 11954   2015-02-25
 
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 19561   2014-11-02
 
96 opensips.cfg for Asterisk admin 21750   2014-10-20
 
95 A2Billing and OpenSIPS config admin 21089   2014-10-20
 
94 Jitsi Videobridge meets WebRTC admin 22263   2014-10-18
 
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 20712   2014-10-18
 
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 20944   2014-10-14
 
91 Opensips TM module enables stateful processing of SIP transactions admin 18622   2014-10-04
 
90 kamailio.cfg configuration Example admin 20853   2014-10-04
 
89 opensips NAT Traversal Module admin 20169   2014-10-02
 
88 UAC Registrant Module admin 21922   2014-09-28
 
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 21068   2014-08-24
 
86 RTPPROXY Admin Guide admin 21447   2014-08-24
 
85 CANCEL MESSAGE not handled correctly admin 21211   2014-08-23
 
84 [Sipdroid] SIP data collection study tour admin 21661   2014-08-23
 
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 21671   2014-08-23
 
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 21569   2014-08-23
 
81 ICE: The ultimate way of beating NAT in SIP admin 21220   2014-08-23
 
80 Many OPENSIPS Configuration Examples This will Help you admin 20906   2014-08-23
 
79 Real-time Charging System for Telecom & ISP environments admin 21631   2014-08-23
 
78 OPENSIPS EBOOK admin 21753   2014-08-21
 
77 Opensips Documentation Function admin 21527   2014-08-21
 
76 Presence Tutorial OpenXCAP setup admin 21007   2014-08-18
 
75 Opensips Modules Documentation admin 21726   2014-08-18
 
74 A lightweight RPC library based on XML and HTTP admin 20950   2014-08-18
 
73 opensips Nat script with RTPPROXY - English Good perfect admin 19571   2014-08-15
 
72 OpenSIPS Control Panel (OCP) Installation Guide Good admin 19697   2014-08-13
 
71 Installation and configuration process record opensips opensips-cp admin 45608   2014-08-13
 
70 OpenSIPS as Homer Capture server admin 18857   2014-08-13
 
69 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 20997   2014-08-13
 
68 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 21473   2014-08-12
 
67 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 20714   2014-08-12
 
66 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 21637   2014-08-11
 
65 Kamailio Nat Traversal using RTPProxy admin 21216   2014-08-11
 
64 MediaProxy wiki page install configuration admin 21263   2014-08-11
 
63 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 38459   2014-08-11
 
62 MediaProxy Installation Guide admin 20801   2014-08-10
 
61 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 22020   2014-08-10
 
60 Opensips Installation, How to. Good guide wiki page admin 18973   2014-08-10
 
59 OpenSIPS Installation Notes admin 18510   2014-08-09
 
58 Installation and configuration process record opensips 1.9.1 admin 30796   2014-08-09
 
57 opensips 1.11.2 install Good Giide admin 21968   2014-08-09
 
56 fusionPBX install debian wheezy admin 20985   2014-08-09
 
55 opensips 1.11.2 install guide good 인스톨 가이드 admin 21247   2014-08-09
 
54 SigIMS IMS Platform admin 21582   2014-05-24
 
53 2013 2012년 분야별 최고의 오픈소스 소프트웨어 124선 admin 25639   2014-04-05
 
52 Video conference server OpenMCU-ru - Introduction admin 24157   2014-04-01
 
51 SIPSorcery admin 21974   2014-03-18
 
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 22397   2014-03-12
 
49 telepresence: Open Source SIP Telepresence/MCU admin 43964   2014-03-12
 
48 SIP PBX - OpenSIPS and Asterisk configuration admin 33455   2014-03-12
 
47 Conference Support in Kamailio (OpenSER) admin 28796   2014-03-12
 
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 20462   2014-03-12
 
45 The Impact of TLS on SIP Server Performance file admin 22058   2014-03-12
 
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 21292   2014-03-12
 
43 Where to check OpenSIPS does not start? admin 21380   2014-03-09
 
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 22400   2014-03-09
 
41 Kamailo OpenSIPs installation on Debian admin 27220   2014-03-09
 
40 Using the openSIPS Registrant Module admin 22826   2014-03-09
 
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 20802   2014-03-07
 
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 21939   2014-03-07
 
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 34547   2014-03-07
 
36 OpenSIPS Control Panel (OCP) Installation Guide admin 20514   2014-03-06
 
35 OpenSIPS Control Panel install guide admin 21700   2014-03-06
 
34 rtpproxy Module admin 21770   2014-03-06
 
33 MediaProxy Installation Guide admin 29198   2014-03-06
 
32 How to install OpenSIPS on CentOS debian module add xcap admin 22573   2014-03-06
 
31 Problem with presence_xml module Opensips 1.9 admin 22083   2014-03-06
 
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 23064   2014-03-06
 
29 Multimedia Service Platform admin 21413   2014-03-06
 
28 How to install OpenSIPS on CentOS Debian etc admin 22243   2014-03-05
 
27 Opensips Installation, How to. admin 18832   2014-03-05
 
26 100% CPU usage opensips admin 21612   2014-03-05
 
25 A2Billing and OpenSIPS admin 22842   2014-03-04
 
24 Opensips_1.9 install guide this is great I like this admin 28730   2014-03-04
 
23 Opensips install debian admin 22677   2014-03-03
 
22 Open Source VOIP applications, both clients and servers. admin 23125   2013-11-20
 
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 24280   2013-09-11
 
» My new toy: Bluebox-ng admin 38474   2013-04-06
http://nicerosniunos.blogspot.kr/ Hi again guys, here there is my new personal project. I think that README file is complete enough so I paste it on this post. Next month I'll be with my colleague ...  
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 40054   2013-04-06
 
18 Asterisk Installation Asterisk Realtime configuration admin 27102   2013-04-06
 
17 The SIP Router Project admin 26108   2013-04-06
 
16 Kamailio :: A Quick Introduction admin 23511   2013-04-06
 
15 Welcome to the Smartvox Knowledgebase admin 23873   2013-04-06
 
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 28713   2013-04-06
 
13 OpenSIPS vs Asterisk admin 69637   2013-04-06