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***************************************************************************************************** NOTE: THE LINKS TO THE FOLLOWING SECURITY VULNERABILITY TOOLS ARE FOR RESEARCH PURPOSES ONLY. THE INFORMATION CONTAINED IN THIS DOCUMENT WAS OBTAINED FROM SOURCES BELIEVED TO BE RELIABLE AND ACCURATE. THE AUTHOR TAKES NO GUARANTEE OR WARRANTY, EXPRESSED OR IMPLIED AS TO THE ACCURACY, TIMELINESS, OR COMPLETENESS OF THE INFORMATION WITHIN. THE AUTHOR IS NOT RESPONSIBLE OR LIABLE FOR ACCIDENTAL DESTRUCTION, MALICIOUS HARM, ERRORS, OR OMISSIONS AND SO FORTH. (VOIP RELATED TEST TOOLS)> Numerous links to VoIP testing and measurement tools http://www.sipcenter.com/sip.nsf/html/Testing+Measurement http://www.voipsa.org/Resources/tools.php > Hammer Call Analyzer http://www.empirix.com/default.asp?action=article&ID=69 The Hammer Call Analyzer enables users to visualize signaling and voice quality problems in VoIP networks. For example, the unique call list and multistage call flow display features walk engineers through the legs of a particular call. In addition, the Hammer Call Analyzer displays waveforms and the Stream Quality Signature for any call. These features allow engineers to visualize problems in the exchange of messages between the various devices and to quickly solve them. Features: · Intuitive protocol-aware searching, filtering and capture · Real-time, multi-stage call flow display · IP Stream Voice Quality Analysis · VoIP and TDM protocol decodes · Import external traces for analysis Protocols: VoIP – H.323 (H.225, H.245), Megaco (H.248), MGCP, RFC 2833, T.38, RTP, RTCP, SIP, SIP-T, Skinny (SCCP), NCS, TCP, UDP, IP, TDM – ISDN (Q.921, Q.931), SS7 (ISUP, TUP, MTP2), CCITT/ITU and JNTT variant support > TraceBuster http://www.touchstone-inc.com/tbfeatures.htm TraceBuster will save you countless hours of digging through capture files! Use the Free TraceBuster to replay/analyze call flows from LibPCap format files or step up to the Professional Editions for integrated capture and replay and an unrivaled value proposition! > IBM simulators for IMS http://www.alphaworks.ibm.com/tech/imssimulators The IP Multimedia SubSystem (IMS) provides rich multimedia services across both next-generation packet-switched and traditional circuit-switched networks for services and applications; it also enables telcos, mobile operators, and other service providers. The subsystem is standards-based and uses open interfaces and functional components that can be assembled flexibly into hardware and software systems to support real-time interactive services and applications. IBM Simulators for IP Multimedia Subsystem can be used for developing, testing, and demonstrating simple IMS applications and proofs-of-concept (POC) of specific IMS architecture components. These simulators provide an easy way for users to simulate and test the IMS components without any complex set-up of IMS servers or architecture configuration. > Asteroid http://www.infiltrated.net/asteroid/ Asteroid is a SIP Denial of Service testing tool. It consists of over 36,000 unique SIP packets and can be quickly modified to create others. Packets are grouped into their respective types (INVITES, BYE, CANCEL, etc.) and can be sent individually or called from a shell script and sent in clusters. Asteroid has effectively crashed all versions of Asterisk up until 1.2.13 and greater which were patched against the sequence which caused the crash. > SIPVicious SIPVicious tools address the need for traditional security tools to be ported to SIP. This package consists of a SIP scanner, a SIP wardialer, and a SIP PBX cracker. These tools were Written in Python. > SIP IRC BOT http://www.loria.fr/~nassar/readme.html interesting program that allows the functionality of sending spit aka spam, denial of service, scans and password cracking > SIPGREP http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/utils/sipgrep/ It is a wrapper on ngrep which: · Filters SIP messages according number in To or From header fields · Displays From tag, To tag, Call-ID and branch in different colors, thus it is possible to trace dialogs or transactions by "one look into message" · It can store received messages into a file and show them (together) > SIPSPY http://www.wesip.com/mediawiki/index.php/SipSpy Features: · Each SipSpy that connects to a spyAgent, must authenticate itself using a login and password, these are transferred using a digest method, so that passwords dont travel in clear-text, and replay attacks are avoided. Also each user is assigned a role: plain or admin. Admins can change the device on which the sipSpy is monitoring, and can change the BPF filter, whereas plain users cannot. Also, you can provide spyAgent wich a regexp for each of the users. Then, when a SipSpy provides a new regexp to match SIP packets, spyAgent will match the regexp to that regexp (that is, a regexp on a regexp), so you can limit the regexp's that users can use to monitor SIP traffic. · SIP dialogs/sessions save and load: SipSpy can save the monitored packets in an XML file, so that when someone detects a bug in the SIP network, they can save a copy of the SIP dialog and send it to the administrators to address it. · Server-based session saving: If one of your users/admins detects a bug in the SIP network, you can ask him to reproduce the bug and monitor all the SIP packets involved, and then save that SIP session to the server, so the next morning when developers go to work, they can download from the server the buggy SIP dialog. > WIST http://www.devel-it.org/index.php?modulo=projetos&id=2 This software was born as a prof concept idea to capture SIP traffic from a remote host (SIP Proxy, Gateway, etc) and show live SIP messages about an specific dialog (filtered by the From SIP user) to help debug SIP transactions in a friendly way. > SIP Proxy Tool http://sourceforge.net/projects/sipproxy/ With the SIP Proxy Tool you will have the opportunity to check and manipulate SIP messages. Furthermore you will be able to run several automated attacks and getting the results as a report. Some of these attacks will use fuzzing technology. > SIP Messenger http://www.sipcenter.com/sip.nsf/html/Compliance+Engine SIP Messenger is Java software that allows you to send SIP test messages from text files over UDP to your SIP implementation and, optionally, listen for responses. The messages can be sent using a command line utility (Messenger), suitable for invocation by automated scripting, or via a GUI (MessengerGui). Developers can use this software to construct their own SIP messages that can be pushed onto SIP servers or User Agents (possibly in conjunction with the SIP Center¹s own SIP resources – the SIP Network Server and UA). This tool is especially useful for stress testing products with scenarios that are otherwise difficult to reproduce. This software has been made available by Ubiquity Software Corporation; founder of the SIP Center > pjsip-perf http://www.pjsip.org/pjmedia/docs/html/page_pjsip_perf_c.htm pjsip-perf is a complete program to measure the performance of PJSIP or other SIP endpoints. It consists of two parts: · the server, to respond incoming requests · the client, who actively submits requests and measure the performance of the server. Both server and client part can run simultaneously, to measure the performance when both endpoints are co-located in a single program. The server accepts both INVITE and non-INVITE requests. The server exports several different types of URL, which would control how the request would be handled by the server: · URL with "0" as the user part will be handled statelessly. It should not be used with INVITE method. · URL with "1" as the user part will be handled statefully. If the request is an INVITE request, INVITE transaction will be created and 200/OK response will be sent, along with a valid SDP body. However, the SDP is just a static text body, and is not a proper SDP generated by PJMEDIA. · URL with "2" as the user part is only meaningful for INVITE requests, as it would be handled call-statefully by the server. For this URL, the server also would generate SDP dynamically and perform a proper SDP negotiation for the incoming call. Also for every call, server will limit the call duration to 10 seconds, on which the call will be terminated if the client doesn't hangup the call. > SIPp SIPp is a performance testing traffic tool for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It can also read XML scenario files describing any performance testing configuration. It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management, regular expressions and variables in scenario files, and dynamically adjustable call rates. SIPp can be used to test many real SIP platforms like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, SIP PBX…. It is also very useful to emulate thousands of user agents calling your SIP system. > rtpBreak http://xenion.antifork.org/rtpbreak/index.html rtpBreak detects, reconstructs and analyzes any RTP [rfc1889] session through heuristics over the UDP network traffic. It works well with SIP, H.323, SCCP and any other signaling protocol. In particular, it doesn't require the presence of RTCP packets (voipong needs them) that aren't always transmitted from the recent VoIP clients. > Nastysip http://phoenix.labri.fr/documentation/sip/SIP-_Tools_and_Works.html Nastysip is a simple Linux-program that generates bogus SIP-messages and sends them to any peer. The torture test messages are also part of the tarball, and the perl script nastyfront.pl can be used to send all torture messages in a loop. Nastysip is released under the General Public License. > Sip Send Fun http://www.security-scans.de/index.php?where=ssf Sip Send Fun uses netcat to send the different SIP-Payload to the tested device. The following functions are implemented: · Payload: New-Message, No-New-Message, INVITE · Test of a single device or a Class-C Scan · Source-IP Spoofing · Send Payload to a single port or portscan > SIPcrack Sipcrack is a protocol login cracker. It contains 2 programs, SIPdump to sniff SIP logins over the network and SIPcrack to bruteforce the passwords of the sniffed logins > SMAP http://www.wormulon.net/files/pub/smap-blackhat.tar.gz SMAP is a combination of nmap and sipsak. To sum up functionality in one sentence it aides in both locating and fingerprinting remote SIP devices. > SIP Analyzer http://sourceforge.net/projects/sipanalzyer or http://ant.comm.ccu.edu.tw/sip/ Distributed SIP Analyzer is a SIP protocol analyzer for Unix. It allows you to examine SIP from different local area network. You can interactively browse the capture data, viewing callflow sequence diagram and detail information for each SIP Session. > SIP Callflow Sequence Diagram Generator http://sourceforge.net/project/showfiles.php?group_id=60608 The callflow sequence diagram generator is a collection of awk and shell scripts that will take a packet capture file that can be read by ethereal and produce a time sequence diagram. This is useful to view and debug SIP callflows or other network traffic > SIPFlow Standard http://www.sipient.com/standard.html SIPFlow Standard captures data on a single host and displays SIP callflows in an intuitive graphical format. SIP messages may be viewed as ladder diagrams, or their contents may be inspected by double clicking an arrow in the ladder diagram. This allows network engineers to quickly identify the behavior of their SIP network without tracing through log files or raw captures. SIPFlow Standard currently supports: · UDP and TCP · IP Filters · SIP filters (Method, To and From) · Searching capabilities · Importing Ethereal and tcpdump captures · Reassembling fragmented packets · Mapping IP address to names · SIP message logging > Sipbomber http://www.metalinkltd.com/downloads.php Sipbomber is invaluable tool for SIP developers intended for testing SIP-protocol implementation against rfc3261. Current version can check only server implementations – (proxies, user agent servers, redirect servers, and registrars). This program is distributed under terms of GPL. > Hacking voIP Exposed Tools http://www.hackingvoip.com/sec_tools.html Tools written for the book and listed are: VLANping, SIPSCAN, TFTP Brute Forcer with TFTP Bruteforce File, IAX Flood, UDP Flooder, UDP Flooder w/VLAN support, BYE Call Teardown, RTP Flooder, Invite Flooder, Check Sync Phone Rebooter, RTP Injector, Registration Hijacker, Registration Eraser, Registration Adder > IETF SIP Torture Messages http://tools.ietf.org/wg/sipping/draft-ietf-sipping-torture-tests/ These messages were developed and refined at the SIPIt interoperability test event. During the events problematic messages were noted and released as an IETF-draft. It defines tens of valid and invalid messages, describes them and gives directions as to how the SIP application should react. > SIPSAK sipsak is a small command line tool for developers and administrators of Session Initiation Protocol (SIP) applications. It can be used for some simple tests on SIP applications and devices. Features: · sends OPTIONS request · sends text files (which should contain SIP requests) · traceroute (see section 11 in RFC3261) · user location test · flooding test · random character trashed test · interpret and react on response · authentication with qop supported (MD5 and SHA1) · short notation supported for receiving (not for sending) · unlimited string replacements in files and requests · add any header to the requests · can simulate calls in usrloc mode · uses symmetric signaling and thus should work behind NAT · can upload any given contact to a registrar · send messages to any SIP destination · Nagios compliant return codes · search for strings in reply with signaling expression · use multiple processes to create more server load · read SIP message from STDIN (e.g. from a pipe `|') · supports DNS SRV through c-ares or libruli · supports UDP and TCP transport > Protos SIP conformance test suite http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/#download The purpose of this test-suite is to evaluate implementation level security and robustness of Session Initiation Protocol (SIP) implementations. The factors behind choosing SIP included: · SIP has matured from academic interest into industrial protocol with potential for wide deployment. However, field usage appears to be in early stages. This stage of the life-cycle is both an opportunity and a challenge from software vulnerability process perspective. By applying the PROTOS approach in this context we hope to prove that the early bird catches the worm in sense that patch and penetrate cycles with respect to some trivial vulnerabilities may be avoided. · Furthermore SIP is being adopted by the Third Generation Partnership Project (3GPP) as part of the third generation mobile architecture. · The SIP family of specifications is expanding and some aspects are under development. This encourages SIP as a natural candidate for experimenting with iterative improvement of a robustness test-suite with more comprehensive releases to follow. · A HTTP-like ASCII presentation of the SIP messages may initially attract more script-kiddie level hostility (vulnerability assessment) than the rival protocols with complex encodings have attracted so far. In this test-suite, the focus was set on a specific protocol data unit (PDU), namely INVITE message. Rationale behind this selection was: · Two important SIP entity types, user agents and proxies, have to support the INVITE-method. · SIP user agents and SIP proxies are by design ready to accept incoming invitations without prior session setup. This exposes a natural attack vector that should be scrutinized with top priority. · The INVITE-method contains a wide range of header-fields and may carry Session Description Protocol (SDP) data. Thus a considerable portion of the underlying code is exposed to testing via single PDU-type. > Protos H225 Protocol Compliance http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/h2250v4/index.html The purpose of this test-suite is to evaluate implementation level security and robustness of H.225.0 implementations. H.225.0 is a protocol responsible for signaling and setting up H.323 calls. The factors behind choosing H.225.0 included: · H.323 is the de-facto standard for Voice over IP (VoIP) and conferencing and it is widely deployed. Moreover, based on lack of prior known vulnerability announcements it appears that the H.323 has not been closely scrutinized or implementations are uncommonly robust. · H.225.0 is the first and most commonly exposed interface to H.323 session establishment. · H.225.0 must be implemented by most H.323 components, namely by terminals, gateways, proxies and multi-point control units. · Due to firewall unfriendly and dynamic behavior of H.323, many firewall products contain complex H.225.0 parsing code that should be tested for robustness due to critical placement of potentially vulnerable code. The scope of the test-suite was narrowed to H.225.0 version 4 Setup-PDU. Rationale behind this selection was: · Setup is the first message sent to a target H.323 endpoint upon call signaling, it is easy to deliver test-cases and to restore the implementation back to its initial state by disconnecting. · Certain security measures can be enforced only after the Setup-PDU has been parsed and implementations are by design ready to accept incoming setup messages. · H.225.0 implements a subset of recommendation Q.931 which is used in ISDN signaling. Certain elements of Q.931 utilize BER encoded ASN.1. · Many information elements used in H.225.0 can be included in Setup-PDU. · The User-user information element in H.225.0 utilizes complex ASN.1 packet encoding rules (PER) which are also used in H.225.0 RAS (Registration, Admission, and Status) messages between H.323 endpoints and gatekeepers. > SiVuS http://www.vopsecurity.org/index.php SiVuS is the first publicly available vulnerability scanner for VoIP networks that use the SIP protocol. SiVuS is used primarily by developers, administrators, network designers, managers and consultants to verify the robustness and security of their SIP implementations by generating the attacks that are included in the SiVuS database or by crafting their own SIP messages using the SIP Message generator. > VoIPong http://www.enderunix.org/voipong/ VoIPong is a utility which detects all Voice Over IP calls on a pipeline, and for those which are G711 encoded, dumps actual conversation to separate wave files. It supports SIP, H323,Cisco's Skinny Client Protocol, RTP and RTCP. It's been written in C language for performance reasons, proved to be running on Solaris, Linux and FreeBSD; though it's thought to compile and run on other platforms as well. On a 45 Mbit/sec actual network traffic, it's been verified that VoIPong successfully detected all VoIP gateways and the VoIP calls. CPU utilization during the run has been found ranging between 66% - 80% on a 256MB RAM, Celeron 1700 Mhz Toshiba notebook. > SIP Test Tool http://voip.hcltech.com/artDisplay.asp?art_id=1226&cat_id=523 Automated test case coverage can be found at web url: http://voip.hcltech.com/pdf/Summary%20of%20Tests.pdf HCL offers a comprehensive SIP Test Tool suited for conformance, regression, integration testing and test automation needs of SIP based components such as SIP User agent and server. SIP Test Tool contains a conformance test suite for conformance testing of different SIP components such as User Agent, Proxy, Registrar, SIP B2BUA, Presence, and IM servers and STUN. SIP conformance test suite provides a number of pre-defined test cases for checking the conformance of particular network component under test. These test cases check for a specified functionality and return the test results as Pass, Fail or skip. SIP Test Tool provides the hooks for test automation and with the help of APIs, user can automate the entire test process. Features of the SIP Test Tool: · Automated test framework architecture suitable for VOIP protocols · Single platform for Protocol conformance, call flow, integration, regression testing requirements · Better protocol conformance and higher interoperability · Test automation hooks for ease of automation · High test case density with around 1000 readymade test cases available · Easy usage with GUI based execution and result analysis · Packaging flexibility Note: HCL Technologies also has a diameter tool that can be found at the below url. http://voip.hcltech.com/artdisplay.asp?cat_id=458&art_id=1306 > WinSIP http://www.touchstone-inc.com/winsip.htm You can use WinSIP to simulate user input, generate high-quality audio and video streams, and control it from the command line to automate testing. WinSIP acts as thousands of simultaneous individual endpoints or connections in any one of the following modes of operation: · Initiate Calls · Answer Calls · Unattended Answer · Registrar Test · Proxy Server > NetIQ Vivinet Diagnostics http://www.netiq.com/products/vd/default.asp The NetIQ Vivinet Diagnostics product (Vivinet Diagnostics) quickly pinpoints call quality problems in Voice over IP (VoIP) networks and explains why you are experiencing reduced call quality. Vivinet Diagnostics reduces the time needed to resolve voice quality issues and lessens the skills required for VoIP troubleshooting, in both pre- and post-deployment environments. Though simple to use, the product provides the data needed to troubleshoot complex VoIP problems in Cisco and Nortel environments. > iWar Current Features:
(IP Packet Construction Tools)> aicmpsend http://packetstormsecurity.nl/UNIX/utilities/aicmpsend.tar.gz Aicmpsend is an ICMP packet sender featuring implementation of all ICMP flags and codes, spoofing, and flooding. > Colasoft Packet Builder http://www.colasoft.com/packet_builder/ Colasoft Packet Builder enables creating custom network packets; users can use this tool to check their network protection against attacks and intruders. Colasoft Packet Builder includes a very powerful editing feature. Besides common HEX editing raw data, it features a Decoding Editor allowing users to edit specific protocol field values much easier. Users are also able to edit decoding information in two editors - Decode Editor and Hex Editor. Users can select one from the provided templates Ethernet Packet, ARP Packet, IP Packet, TCP Packet and UDP Packet, and change the parameters in the decoder editor, hexadecimal editor or ASCII editor to create a packet. Any changes will be immediately displayed in the other two windows. In addition to building packets, Colasoft Packet Builder also supports saving packets to packet files and sending packets to network. > Bit-Twist http://bittwist.sourceforge.net/index.html Bit-Twist is a simple yet powerful libpcap-based Ethernet packet generator. It is designed to compliment tcpdump, which by itself has done a great job in capturing network traffic. With Bit-Twist, you can now regenerate the captured traffic onto a live network! Packets are generated from tcpdump trace file (.pcap file). Bit-Twist also comes with a comprehensive trace file editor to allow you to change the contents of a trace file. Generally, packet generator is useful in simulating networking traffic or scenario, testing firewall, IDS, and IPS, and troubleshooting various network problems. Features: · runs on *BSD, Linux, Solaris, and Windows 2000/XP · send multiple trace files at a time · send packets at a specific speed or line rate in Mbps · comprehensive trace file editor with control over most fields in Ethernet, ARP, IP, ICMP, TCP, and UDP headers with automatic header checksum correction · append user payload to existing packets after a specific header · select a specific range of packets and save them in another trace file · If you are teaching Computer Networks classes, you may find Bit-Twist useful as a practical teaching material! It gives your students a hands-on experience to learn various networking protocols, etc. > Network Packet Generator (npg) http://www.wikistc.org/wiki/Network_packet_generator Network Packet Generator (npg) is a free GNU GPL Windows packet injector (generator) that utilizes WinPcap to send specific packets out a single or multiple network interfaces. These packets and other extended options can be defined on the command line, in a packet file, or combination of the two. A packet file can be either a Libpcap compatible capture dump or an npg formatted file that generates packets from raw byte streams providing the ability to create any packet type regardless of header, payload, or data link. > Sendip http://www.earth.li/projectpurple/progs/sendip.html SendIP makes it possible to prepare and send network packets using the NTP, BGP, RIP, RIPng, TCP, UDP and ICMP protocols, as well as raw Ipv4 and Ipv6 packets with user-supplied parameters and arbitrary data. > GASP –v1.0 http://laurent.riesterer.free.fr/gasp/ GASP stands for `Generator and Analyzer System for Protocols'. It allows you to decode and encode any protocols you specify. The main use is probably to test networks applications: you can construct packets by hand and test the behavior of your program when facing some strange packets. But you can image a lot of other application: e.g. manipulating graphical file or executable headers. Just describe the specification of the structured data. > Gspoof –v3.2 http://gspoof.sourceforge.net/ Gspoof is a tool which makes easier and accurate the building and sending of tcp-ip packets. > hping –v3 alpha 2 hping is a command-line oriented TCP/IP packet assembler/analyzer. The interface is inspired to the ping unix command, but hping isn't only able to send ICMP echo requests. It supports TCP, UDP, ICMP and RAW-IP protocols, has a traceroute mode, the ability to send files between a covered channel, and many other features. > ICMPUSH http://packetstormsecurity.org/UNIX/scanners/icmpush22.tgz ICMPUSH is a tool that builds ICMP packets fully customized from command line. It supports the following ICMP error types: Redirect, Source Quench,Time Exceeded, Destination Unreach and Parameter Problem, and the following ICMP information types: Address Mask Request, Timestamp, Information Request, Echo Request, Router Solicitation and RouterAdvertisement. > IP Sorcery - v2.0.1 http://packetstormsecurity.nl/UNIX/misc/ipsorc-1.7.0.tar.gz IP Sorcery is a TCP/IP packet generator. It has the ability to send TCP, UDP, and ICMP packets with a GTK+ interface. > Nemesis –v1.4beta3 http://sourceforge.net/project/showfiles.php?group_id=93681&release_id=273337 Nemesis is a command-line network packet injection utility for UNIX-like and Windows systems. You might think of it as an EZ-bake packet oven or a manually controlled IP stack. With Nemesis, it is possible to generate and transmit packets from the command line or from within a shell script. Nemesis is developed and maintained by Jeff Nathan. Note: A great front end gui called Jnemesis is available from the web url below. http://jnemesis.blackopscode.com/ > PacketCrafter http://www.komodia.com/tools.htm · Build custom TCP/IP/UDP packets. · Control the source address (IP spoofing) · Control IP flags (checksums, IDs and more) · Control TCP flags (state flags, sequence numbers, ack number and more) > Packet Excalibur –v1.0.2 http://www.securitybugware.org/excalibur/ A multi-platform graphical and scriptable network packet engine with extensible text based protocol descriptions > packETH –v1.1 http://packeth.sourceforge.net/ packETH is a Linux GUI packet generator tool for Ethernet. It allows you to create and send any possible packet or sequence of packets on the Ethernet. > Scapy http://www.secdev.org/projects/scapy/ Scapy is a powerful interactive packet manipulation tool, packet generator, network scanner, network discovery, packet sniffer, etc. It can for the moment replace hping, 85% of nmap, arpspoof, arp-sk, arping, tcpdump, tethereal, p0f, and etc > SING http://sourceforge.net/projects/sing/ SING stands for `Send ICMP Nasty Garbage'. It is a tool that sends ICMP packets fully customized from command line. Its main purpose is to replace the ping command but adding certain enhancements (Fragmentation, spoofing...) > TtpU http://www.poetidistrada.com/ttpu/ TTpU stands for The Dark Free Soul's TCP/IP Packets Unlimited (generator). TTpU generates TCP packets on IPv4 protocol and lets user specify: · network interface · source and destination ip · source and destination port · sequence number · acknowledgement number · tcp flags (URG, ACK, PSH, RST, SYN, FIN) · window size |
