VoIP Spoofing (Intigriti) 1,250€

1 year ago 79
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Hello Folks, i just want to explain a misconfiguration that affect an asset on Intigriti. So, let’s start!!

VoIP implementation allows audio calls to be made using an Internet connection instead of a conventional phone. Some VoIP gateway partners may allow you to call others who have a phone number, including local, long distance, mobile, and international numbers.

VoIP uses 5060 as a SIP signaling port by default. Used to register the phone (for example, Cisco, Polycom, etc.)

Among the most important features of VoIP are:

- Use of multiple lines

- Voicemail service

- Voice recording

- Call log

- Modular configurations

Session Initiation Protocol (SIP) allows users to establish communications, terminate, or modify voice or video calls. According to pentesting experts, voice or video traffic is transmitted via Real-Time Protocol (RTP). SIP is an application layer protocol that uses UDP or TCP for traffic. By default, SIP uses UDP/TCP port 5060.

Realize that your target is 182.x.x.x/27, so i started using nmap and i started to scan the subnet, i just found an interesting IP that have the port 5060 open:

nmap -sC -sV -A -p- -T4 182.x.x.x
Starting Nmap 7.92 ( https://nmap.org ) at 2022-11-05 06:18 CDT
Nmap scan report for 182.x.x.x (194.x.x.x)
Host is up (0.038s latency).
Not shown: 65531 filtered tcp ports (no-response)
PORT STATE SERVICE VERSION
443/tcp closed https
5060/tcp open sip?

first at all i used wireshark for take listen of this port and i found some username like:

Username Enumeration

Actually, i got so many usernames, and i can actually listen,register, delete, or spoof their call but how?

Let’s try to send an HIT request with a “Test” Username using inviteflood

Spoofing

As you can see i already sent call to the user “102” with a Test as username, let’s open wireshark and filter for SIP and see what happen:

Invite Call

As we can see i can invite to call any of their agents, using a VoIP spoofing, Currently, someone’s cell phone is ringing, and as you can see i Invite someone in their internal network IP start with 10.x.x.x . Now Let’s analyze that invite call.

Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:102@194.x.x.x SIP/2.0
Method: INVITE
Request-URI: sip:102@194.x.x.x
Request-URI User Part: 102
Request-URI Host Part: 194.x.x.x
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.1.6:9;branch=28a8d461-64d5-4636-9b0b-090000000001
Transport: UDP
Sent-by Address: 192.168.1.6
Sent-by port: 9
Branch: 28a8d461-64d5-4636-9b0b-090000000001
Max-Forwards: 70
Content-Length: 460
To: 102 <sip:102@194.x.x.x:5060>
SIP to display info: 102
SIP to address: sip:102@194.x.x.x:5060
From: Test <sip:Test@192.168.1.6:9>;tag=28a8da38-64d5-4636-b984-2a0000000001
SIP from display info: Test
SIP from address: sip:Test@192.168.1.6:9
SIP from address User Part: Test
SIP from address Host Part: 192.168.1.6
SIP from tag: 28a8da38-64d5-4636-b984-2a0000000001
Call-ID: 28a8df56-64d5-4636-b92f-5d0000000001
[Generated Call-ID: 28a8df56-64d5-4636-b92f-5d0000000001]
CSeq: 0000000001 INVITE
Sequence Number: 1
Method: INVITE
Supported: timer
Allow: NOTIFY
Allow: REFER
Allow: OPTIONS
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: BYE
Content-Type: application/sdp
Contact: <sip:Test@192.168.1.6:9>
Contact URI: sip:Test@192.168.1.6:9
Supported: replaces
User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v.3.2.6.26
Message Body

Ok as you can see i can NOTIFY, REFER, INVITE,ACK, CANCEL, BYE option. So i can CANCEL some outgoing call, i can LISTEN or Register inside call or spoof it.

This misconfiguration is marked as HIGH from the company and they rewarded me 1,250 EUR.

Rewarded

Marked High:

Resolved

Thank guys for reading, and Happy bug hunting!

0xJin
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