Tìm kiếm với Google


Showing posts with label PSTN. Show all posts
Showing posts with label PSTN. Show all posts

Wednesday, February 27, 2008

VoIP Security

McGrawHill Hacking Exposed VoIP Voice.Over.IP Security Secrets Solutions eBook
McGrawHill Hacking...
Hosted by eSnips

Syngress How to Cheat at VolP Security (2007)
Syngress How to Ch...
Hosted by eSnips

Asterisk Stability And Security
Asterisk Stability...
Hosted by eSnips

Voice over IP (VoIP) has finally come of age and is being rapidly embraced across most markets as an alternative to the traditional public-switched telephone network (PSTN). VoIP is a broad term, describing many different types of applications (hard phones, softphones, proxy servers, Instant Messaging clients, peer-to-peer clients, and so on), installed on a wide variety of platforms (Linux, Windows, VxWorks, mobile devices, PCs, and so on), and using a wide variety of both proprietary and open protocols (SIP, RTP, H.323, MGCP, SCCP, Unistim, SRTP, ZRTP, and so on) that depend heavily on your preexisting data network's infrastructure and services (routers, switches, DNS, TFTP, DHCP, VPNs, VLANs and so on). Correspondingly, VoIP security is just as broad a subject thanks to the heterogeneous nature of these environments found in the consumer, enterprise, carrier, and small/medium–sized business markets.


How the Book is Organized

This book is split into five completely different parts. Each part can be read without even touching the remaining four—so if the reader is interested only in the issues described in the selected part, he or she may consult only that part.

Part I. "Casing the Establishment"

The first part is introductory and describes how an attacker would first scan the whole network and then pick up specific targets and enumerate them with great precision in order to proceed with further advanced attacks through or from the hacked VoIP devices.

Chapter 1. "Footprinting a VoIP Network"

We begin the book by describing how a hacker first profiles the target organization by performing passive reconnaissance using tools such as Google, DNS, and WHOIS records, as well as the target's own website.

Chapter 2. "Scanning a VoIP Network"

A logical continuation of the previous chapter, this chapter provides a review of various remote scanning techniques in order to identify potentially active VoIP devices on the network. We cover the traditional UDP, TCP, SNMP, and ICMP scanning techniques as applied to VoIP devices.

Chapter 3. "Enumerating a VoIP Network"

Here, we show active methods of enumeration of various standalone VoIP devices, from softphones, hard phones, proxies, and other general SIP-enabled devices. Plenty of examples are provided, along with a demonstration of SIPScan, a SIP directory scanning tool we wrote.

Part II. "'Exploiting the VoIP Network"

This part of the book is focused on exploiting the supporting network infrastructure on which your VoIP applications depend. We begin with typical network denial-of-service attacks and eventually lead up to VoIP conversation eavesdropping. While many of the demonstrated techniques originate from the traditional data security world, we applied them against VoIP devices and supporting network services.

Chapter 4. "VoIP Network Infrastructure Denial of Service (DoS)"

In this chapter, we introduce quality of service and how to objectively measure the quality of a VoIP conversation on the network using various free and commercial tools. Next, we discuss various flooding and denial of service attacks on VoIP devices and supporting services such as DNS and DHCP.

Chapter 5. "VoIP Network Eavesdropping"

This section is very much focused on the types of VoIP privacy attacks an attacker can perform with the appropriate network access to sniff traffic. Techniques such as number harvesting, call pattern tracking, TFTP file snooping, and actual conversation eavesdropping are demonstrated.

Chapter 6. "VoIP Interception and Modification"

The methods described in this chapter detail how to perform man-in-the-middle attacks in order to intercept and alter an active VoIP session and conversation. We demonstrate some man-in-the-middle methods of ARP poisoning and present a new tool called sip_rogue that can sit in between two calling parties and monitor or alter their session and conversation.

Part III. "Exploiting Specific VoIP Platforms"

In this part of the book, we shift our attention to attacking specific vendor platforms where each has unique security weaknesses and countermeasures. We demonstrate some of the attacks covered in the last few chapters in order to detail the vendor-specific best practices for mitigating them.

Chapter 7. "Cisco Unified CallManager"

We installed Cisco CallManager 4.x with Cisco hard phones in a fully homogenous Cisco-switched environment in order to perform many of the attacks we've already detailed.

We also cover the various best practices to apply to the Cisco switching gear to mitigate most of the network attacks covered in Part II.

Chapter 8. "Avaya Communication Manager"

Similarly, we installed a full Avaya Communication Manager along with Avaya hard phones to detail some of the specific attacks we covered in Part I and Part II.

Chapter 9. "Asterisk"

We targeted our SIP test bed running Asterisk with the similar attacks detailed in Part I and Part II. We also performed some basic platform testing on a subset of the SIP phones in our test bed.

Chapter 10. "Emerging Softphone Technologies"

In this chapter, we discuss some security issues with the emerging softphone services, such as Skype, Gizmo, and others. While these services have not yet dominantly emerged into the enterprise space, they are poised to do so through some interesting partnerships under way.

Part IV. "VoIP Session and Application Hacking"

In this part of the book, we shift our attention from attacking the network and device to attacking the protocol. The fine art of protocol exploitation can hand intruders full control over the VoIP application traffic without any direct access and reconfiguration of the hosts or phones deployed.

Chapter 11. "Fuzzing VoIP"

The practice of fuzzing, otherwise known as robustness testing or functional protocol testing, has been around for a while in the security community. The practice has proven itself to be pretty effective at automating vulnerability discovery in applications and devices that support a target protocol. In this chapter, we demonstrate some tools and techniques for fuzzing your VoIP applications.

Chapter 12. "Flood-Based Disruption of Service"

In this chapter, we cover additional attacks that disrupt SIP proxies and phones by flooding them with various types of VoIP protocol and session-specific messages. These types of attacks partially or totally disrupt service for a SIP proxy or phone while the attack is under way. Some of the attacks actually cause the target to go out of service, requiring a restart.

Chapter 13. "Signaling and Media Manipulation"

In this chapter, we cover other attacks in which an attacker manipulates SIP signaling or RTP media to hijack, terminate, or otherwise manipulate calls. We introduce no less than ten new tools to demonstrate these attacks. As with other attacks we have covered, these attacks are simple to execute and quite lethal.

Part V. "Social Threats"

In the same way that the traditional email realm has been inundated with spam and phishing, so too are we starting to see the evolution of these social nuisances into the VoIP world. This chapter focuses on how advertisers and scam artists will likely target VoIP users and how to help counter their advance.

Chapter 14. "SPAM over Internet Telephony (SPIT)"

Voice SPAM or SPAM over Internet Telephony (SPIT) is a similar problem that will affect VoIP. SPIT, in this context, refers to bulk, automatically generated, unsolicited calls. SPIT is like telemarketing on steroids. You can expect SPIT to occur with a frequency similar to email SPAM. This chapter describes how you can use the Asterisk IP PBX and a new tool called spitter to generate your own SPIT. This chapter also details how you can detect and mitigate SPIT.

Chapter 15. "Voice Phishing"

Voice phishing relies on the effective gullibility of a victim trusting a phone number much more than an email link. Also, for a fraction of the cost, an attacker can set up an interactive voice response system through a VoIP provider that is harder to trace than a compromised web server. Also, the nature of VoIP makes this type of attack even more feasible because most VoIP services grant their customers an unlimited number of calls for a monthly fee. This chapter details how these attacks are performed and how to detect them at their various stages.

Tuesday, February 26, 2008

Fax Over IP (Part II)

FoIP Software Architecture
The facsimile interface unit (FIU) is the software module that resides within a
FoIP IWF. It demodulates voiceband signals from an analog interface and
converts them to a digital format suitable for transport over a packet network. It
also remodulates data received from the packet network and transmits it to the
analog interface. In doing so, the FIU performs protocol conversion between
Group-3 facsimile protocols and the digital facsimile protocol employed over the
packet network.
The FIU, shown in Figure 10, consists of the following three units:
fax-modem unit (FM)—This processes PCM samples based on the current
modulation mode and supports the following functions:
• V.21 Channel 2 (300 bps) binary signaling modulation and
demodulation
• high-level data link control (HDLC) framing (0 bit insertion/removal,
cyclic redundancy check (CRC) generation/checking)
• V.27 ter (2400/4800 bps) high-speed data modulation and
demodulation
• V.29 (7200/9600 bps) high-speed data modulation and demodulation
• V.17 (7200/9600/12000/14400 bps) high-speed data modulation and
demodulation
• V.33 (12000/14400 bps) high-speed modulation and demodulation
• CED detection and generation
• CNG detection and generation
• V.21 Channel-2 detection

Thursday, February 21, 2008

Installing Asterisk

Compiling Asterisk on most Linux distributions


Accomplish this by opening the folder in which the source codes were extracted (if you are using tarball files), or go to the /usr/src/asterisk folder (if using CVS server) to get the required packages.

To go to this directory, see example:

#cd /usr/src/asterisk


Once here, the final step is to compile:

Important: Follow this installation order: libpri, zaptel, asterisk

-Installing libpri

#cd /usr/src/asterisk/libpri
#make clean
#make
#make install


-Installing zaptel

#cd /usr/src/asterisk/zaptel
#make clean

Note: If you are using kernel 2.6 enter the following command '#make linux26', before doing '#make install'.

#make install


-Installing asterisk

#cd /usr/src/asterisk/asterisk
#make clean

Note: If you want to use a mp3 files for music-on-hold enter the following command '#make mpg123', before doing '#make install'.

#make install

If this is your first installation perform 'make samples' to install sample

(asteriskguru.com)

Fax Over IP (Part I)

Traditionally, there have been two approaches for sending FoIP networks:
real-time methods and store-and-forward methods. The primary difference in service between these two approaches lies in the delivery and method of receipt
confirmation. The Frame Relay Forum has defined a real-time protocol for the
transmission of fax-over–frame relay networks. Likewise, the ITU and Internet
Engineering Task Force (IETF) are working together to continue to evolve both
the real-time FoIP network standard (T.38) as well as the store-and-forward FoIP
network standard (T.37). Both T.37 and T.38 were approved by the ITU in June,
1998. Furthermore, T.38 is the fax transmission protocol selected for H.323.
There are tremendous opportunities for cost savings by transmitting fax calls
over packet networks. Fax data in its original form is digital. However, it is
modulated and converted to analog for transmission over the PSTN. This analog
form uses 64 kbps of bandwidth in both directions.
FoIP QoS
The advantages of reduced cost and bandwidth savings of carrying FoIP networks are associated with some QoS issues that are unique to packet networks and can affect the reliability of the fax transmission.
Timing
A major issue in the implementation of FoIP networks is the problem of
inaccurate timing of messages caused by delay through the network. The delay of fax packets through a packet network causes the precise timing that is required for many portions of the fax protocol to be skewed and can result in the loss of the call. The FoIP protocol in the IWF must compensate for the loss of a fixed timing of messages over the packet network so that the T.30 protocol operates without error.
There are two sources of delay in an end-to-end, FoIP call: network delay and
processing delay.
• network delay—This is caused by the physical medium and protocols
that are used to transmit the fax data and by buffers used to remove
packet jitter on the receiving end. This delay is a function of the
capacity of the links in the network and the processing that occurs as
the packets transit the network. The jitter buffers add delay when they
remove the packet-delay variation of each packet as it transits the
packet network. This delay can be a significant part of the overall delay,
as packet-delay variations can be as high as 70 to 100 milliseconds in
some frame-relay networks and even higher in IP networks.
• processing delay—This is caused by the process of demodulating and
collecting the digital fax information into a packet for transmission
over the packet network. The encoding delay is a function of both the
processor execution time and the amount of data collected before
sending a packet to the network. Low-speed data, for instance, is
usually sent out with a single byte per packet, as the time to collect a
byte of information at 300 bps is 30 milliseconds.
Jitter
Delay issues are compounded by the need to remove jitter, a variable interpacket timing caused by the network that a packet traverses. An approach to removing the jitter is to collect packets and hold them long enough so that the slowest packets to arrive are still in time to be played in the correct sequence. This approach, however, causes additional delay. In most FoIP protocols, a time stamp is incorporated in the packet to ensure that the information is played out at the proper instant.
Lost-Packet Compensation
Lost packets can be an even more severe problem, depending on the type of
packet network that is being used. In a VoIP application, the loss of packets can
be addressed by replaying last packets and other methods of interpolation. A
FoIP application, however, has more severe constraints on the loss of data, as the fax protocol can fail if information is lost. This problem varies, depending on the type of fax machine used and whether error-correction mode is enabled.
Two schemes that are used by FoIP software to address the problems of lost
frames are as follows:
• repeating information in subsequent frames so that the error can be
corrected by the receiver's playout mechanism
• using an error-correcting protocol such as TCP to transport the fax data
at the expense of added delay

Asterisk Hacking

Today,Asterisk become the most wide use solution for IP PBX.With Asterisk and one old PC ,you can building one system with many Application as:voicemail,Call forwarding,Music on hold,Call conferencing,v.vv.
So,you might be read this book,it;s truly useful for me when I'm development A NGN system called by Unified Messaging.I hope it can help you,too.Thank you.

Syngress Asterisk Hacking May 2007 eBook
Syngress Asterisk ...
Hosted by eSnips