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 :
39009
등록일 :
2013.04.06
22:44:35 (*.160.42.88)
엮인글 :
http://webs.co.kr/index.php?document_srl=19770&act=trackback&key=173
게시글 주소 :
http://webs.co.kr/index.php?document_srl=19770
List of Articles
번호 제목 글쓴이 날짜 조회 수
162 Opensips Gateway between SIP and SMPP messages admin 2019-02-19 281
161 smpp sms opensips admin 2019-02-19 264
160 Busy Lamp Field (BLF) feature on Opensips 2.4.0 with Zoiper configuration admin 2018-05-29 2006
159 Documentation -> Tutorials -> WebSocket Transport using OpenSIPS admin 2018-05-17 1888
158 List of SIP response codes admin 2017-12-20 3545
157 opensips/modules/event_routing/ Push Notification Call pickup admin 2017-12-20 3104
156 opensips push notification How to detail file admin 2017-12-20 3024
155 OpenSIPS routing logic admin 2017-12-12 3071
154 OpenSIPS example configuration admin 2017-12-12 3064
153 opensips log output admin 2017-12-11 3059
152 opensips complete configuration example admin 2017-12-10 3170
151 Opensips1.6 ebook detail configuration and SIP signal and NAT etc file admin 2017-12-10 3141
150 dictionary.opensips radius admin 2017-12-09 4093
149 what is record_route() in opensips ? admin 2017-12-09 4023
148 what is loose_route() in opensips ? file admin 2017-12-09 4178
147 in opensips what is lookup(domain [, flags [, aor]]) admin 2017-12-09 4046
146 in opensips db_does_uri_exist() what is admin 2017-12-09 3887
145 in opensips what is has_totag() admin 2017-12-09 4056
144 opensips exec module admin 2017-12-08 4229
143 opensips push notification How to admin 2017-12-07 4031
142 OpenSIPS Module Interface admin 2017-12-07 4139
141 opensips configuration config explain easy basic 오픈쉽스 컨피그레이션 기본 설명 file admin 2017-12-07 4212
140 openssl 을 이용한 인증서 생성 절차를 정리한다. 개인키 CSR SSL 인증서 파일 생성 admin 2017-09-14 5306
139 Documentation -> Tutorials -> TLS opensips.cfg admin 2017-09-14 5070
138 Using TLS in OpenSIPS v2.2.x admin 2017-09-14 5064
137 opensips tls cfg admin 2017-09-14 5151
136 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 2017-09-13 5700
135 SIP to XMPP Gateway + SIP Presence Server opensips admin 2017-09-13 5024
134 OpenSIPS command line tricks admin 2017-09-13 4993
133 Fail2Ban Freeswitch How to secure admin 2017-09-12 5275
132 opensips.cfg. sample admin 2017-09-12 4958
131 Advanced SIP scenarios with Event-based-Routing admin 2017-09-11 5130
130 PUSH SERVER 푸시서버 안드로이드 애플 admin 2017-09-11 5589
129 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅(리눅스 기준) admin 2017-09-09 13890
128 rtpengine config basic and opensips configuration and command admin 2017-09-06 5302
127 WebSocket Transport using OpenSIPS configuration 웹 소켓 컨피그레이션 기본 admin 2017-09-06 5122
126 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 5223
125 rtpengine install and config admin 2017-09-05 5199
124 Installing RTPEngine on Ubuntu 14.04 admin 2017-09-05 5308
123 compile only the textops module make modules=modules/textops modules admin 2017-09-05 5177
122 opensips command /sbin/opensipsctl detail admin 2017-09-04 5276
121 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 2017-09-04 5203
120 Build-Depends debian 8.8 opensips 2.3 admin 2017-09-04 5097
119 What is new in 2.3.0 opensips admin 2017-09-04 5929
118 ubuntu 安装配置opensips,rtpproxy,mediaproxy admin 2017-09-04 5464
117 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 2017-09-04 5690
116 Using TLS in OpenSIPS v2.2.x configuration admin 2017-09-04 5357
115 How to 2.3 download , OpenSIPS new apt repository. DEBs for Debian / Ubuntu admin 2017-09-02 5317
114 You can install CDRTool in the following ways: admin 2017-09-01 5623
113 How to Install OpenSIPS 2.1.2 Server on Ubuntu 15.04 admin 2017-09-01 5527
112 Opensips 2.32 download admin 2017-09-01 5306
111 OpenSIPS 2.3 install admin 2017-09-01 5628
110 JsSIP: The JavaScript SIP Library admin 2017-09-01 5575
109 WebSocket Transport using OpenSIPS admin 2017-09-01 5665
108 A2Billing and OpenSIPS – Part 1 admin 2017-08-29 5367
107 A2Billing and OpenSIPS – Part 2 admin 2017-08-29 5274
106 A2Billing and OpenSIPS – Part 3 admin 2017-08-29 5488
105 OpenSIPS 2.3 philosophy admin 2017-08-17 6024
104 The timeline for OpenSIPS 2.3 is admin 2017-08-17 6161
103 OpenSIPS Control Panel and Homer integration admin 2017-08-17 6202
102 Opensips sip capture re designed admin 2017-07-16 5650
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 2015-04-04 10673
100 WebSocket Support in OpenSIPS 2.1 admin 2015-04-04 11994
99 OpenSIPS 2.1 (rc) is available, download now! admin 2015-03-22 10558
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 2015-02-25 12367
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 2014-11-02 19895
96 opensips.cfg for Asterisk admin 2014-10-20 22138
95 A2Billing and OpenSIPS config admin 2014-10-20 21457
94 Jitsi Videobridge meets WebRTC admin 2014-10-18 23116
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 2014-10-18 21078
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 2014-10-14 21289
91 Opensips TM module enables stateful processing of SIP transactions admin 2014-10-04 18975
90 kamailio.cfg configuration Example admin 2014-10-04 21246
89 opensips NAT Traversal Module admin 2014-10-02 20535
88 UAC Registrant Module admin 2014-09-28 22300
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 2014-08-24 21472
86 RTPPROXY Admin Guide admin 2014-08-24 21823
85 CANCEL MESSAGE not handled correctly admin 2014-08-23 21631
84 [Sipdroid] SIP data collection study tour admin 2014-08-23 22042
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 2014-08-23 22036
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 2014-08-23 21938
81 ICE: The ultimate way of beating NAT in SIP admin 2014-08-23 21581
80 Many OPENSIPS Configuration Examples This will Help you admin 2014-08-23 21238
79 Real-time Charging System for Telecom & ISP environments admin 2014-08-23 21991
78 OPENSIPS EBOOK admin 2014-08-21 22138
77 Opensips Documentation Function admin 2014-08-21 21829
76 Presence Tutorial OpenXCAP setup admin 2014-08-18 21426
75 Opensips Modules Documentation admin 2014-08-18 22101
74 A lightweight RPC library based on XML and HTTP admin 2014-08-18 21274
73 opensips Nat script with RTPPROXY - English Good perfect admin 2014-08-15 20078
72 OpenSIPS Control Panel (OCP) Installation Guide Good admin 2014-08-13 20285
71 Installation and configuration process record opensips opensips-cp admin 2014-08-13 46507
70 OpenSIPS as Homer Capture server admin 2014-08-13 19188
69 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 2014-08-13 21364
68 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 2014-08-12 21845
67 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 2014-08-12 21135
66 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 2014-08-11 22006
65 Kamailio Nat Traversal using RTPProxy admin 2014-08-11 21563
64 MediaProxy wiki page install configuration admin 2014-08-11 21620
63 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 2014-08-11 40413
62 MediaProxy Installation Guide admin 2014-08-10 21134
61 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 2014-08-10 22371
60 Opensips Installation, How to. Good guide wiki page admin 2014-08-10 19384
59 OpenSIPS Installation Notes admin 2014-08-09 18898
58 Installation and configuration process record opensips 1.9.1 admin 2014-08-09 32450
57 opensips 1.11.2 install Good Giide admin 2014-08-09 22455
56 fusionPBX install debian wheezy admin 2014-08-09 21328
55 opensips 1.11.2 install guide good 인스톨 가이드 admin 2014-08-09 21626
54 SigIMS IMS Platform admin 2014-05-24 21878
53 2013 2012년 분야별 최고의 오픈소스 소프트웨어 124선 admin 2014-04-05 26333
52 Video conference server OpenMCU-ru - Introduction admin 2014-04-01 24561
51 SIPSorcery admin 2014-03-18 22306
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 2014-03-12 22727
49 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 47065
48 SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 35547
47 Conference Support in Kamailio (OpenSER) admin 2014-03-12 29968
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 2014-03-12 20902
45 The Impact of TLS on SIP Server Performance file admin 2014-03-12 22341
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 2014-03-12 21579
43 Where to check OpenSIPS does not start? admin 2014-03-09 21673
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 2014-03-09 22713
41 Kamailo OpenSIPs installation on Debian admin 2014-03-09 28416
40 Using the openSIPS Registrant Module admin 2014-03-09 23177
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 21136
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 2014-03-07 22220
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 35997
36 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 20885
35 OpenSIPS Control Panel install guide admin 2014-03-06 22121
34 rtpproxy Module admin 2014-03-06 22007
33 MediaProxy Installation Guide admin 2014-03-06 30366
32 How to install OpenSIPS on CentOS debian module add xcap admin 2014-03-06 22891
31 Problem with presence_xml module Opensips 1.9 admin 2014-03-06 22391
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 2014-03-06 23398
29 Multimedia Service Platform admin 2014-03-06 21709
28 How to install OpenSIPS on CentOS Debian etc admin 2014-03-05 22489
27 Opensips Installation, How to. admin 2014-03-05 19182
26 100% CPU usage opensips admin 2014-03-05 21897
25 A2Billing and OpenSIPS admin 2014-03-04 23715
24 Opensips_1.9 install guide this is great I like this admin 2014-03-04 29214
23 Opensips install debian admin 2014-03-03 23025
22 Open Source VOIP applications, both clients and servers. admin 2013-11-20 23413
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 2013-09-11 24619
» My new toy: Bluebox-ng admin 2013-04-06 39009
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 41498
18 Asterisk Installation Asterisk Realtime configuration admin 2013-04-06 27505
17 The SIP Router Project admin 2013-04-06 26455
16 Kamailio :: A Quick Introduction admin 2013-04-06 23978
15 Welcome to the Smartvox Knowledgebase admin 2013-04-06 24288
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 2013-04-06 29200
13 OpenSIPS vs Asterisk admin 2013-04-06 72780