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Projects Archive - Before 2008-2009 |
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This project continues the work begun in Spring, 2006. (See below IMS-based services through an IMS-compliant firewall) The current stage of this project uses OpenIMS code to create the P-, I- and S-CSCFs and HSS, and includes the Reef Point IMS-enabled, carrier grade router as a SEG. The goals this semester include installation of the four open-source elements on different physical devices, characterization of the behavior of the test environment by means of call traces taken under different scenarios, and verification of compliance with 3GPP requirements. |
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In the coming semesters the performance and security characteristics of the test environment will be recorded using bulk call generators based on SIP-P, the Spirent Abacus and other tools as they become available. Security aspects of the test environment will be characterized using several attack mechanisms developed in the lab, including the “Klingon Kluster”, a collection of PCs scripted to perform various types of DoS attacks. |
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This project continues the work begun in Spring, 2007. During that period we installed the Open Source P2PSIP implementation from Telecom Italia found at: sourceforge . Following recent decisions at the IETF p2p working group, our current P2P test environment is based on the Chord DHash code. The team is currently characterizing the behavior of this implementation, collecting traces of protocol messages and observing the population of the finger tables and logs. Future work will include a study of the behaviors of PASTRY and CAN-based overlay networks. The experience gained with this early research will be used by the team to write code that allows VoIP applications to run on these overlay networks and to study the performance and security characteristics of these implementations. |
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The Asterisk IP PBX is widely deployed both in products such as the Epygi IP PBX (available in the VoIP Lab) and in custom installations developed for individual enterprises. The goal of this project is to develop a simple conference service for the IEEE Communications Society to use in its planning meetings. The experience gained by this first Asterisk project will then be applied to other projects, including the development of a remote classroom application and also an application to provide authentication services for entry into a flat-rate VoIP long distance service. |
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Signaling protocols need to identify the port to which media streams are delivered. The ports are often behind Network Address Translators and thus have private IP Addresses that are not routable on the public Internet. Many techniques have been developed for literally “getting around” this obstacle. Projects in this area will explore specific examples of these techniques and characterize their effectiveness. |
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Last semester students downloaded an open source STUN server, configured and installed it and characterized its behaviors. This semester a new team was challenged to download the NATA Check software at sorceforge , to lear how this tool works and to build NAT Check capability into the VoIP Lab web site so that visitors can learn about the type of NAT behind which they are situated. |
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The RTP stream created by SIP and other signaling protocols presents several types of network security vulnerabilities. It can be hijacked using Man In The Middle (MITM) techniques. It is easy to eavesdrop on this stream as well. Eavesdropping can be prevented by encryption of the RTP stream using SRTP. But this function requires an exchange of encryption keys and this exchange is usually provided by a separate key service. ZRTP provides keys in the RTP stream, and thus makes secure RTP easier to achieve. ZRTP also provides defense against MITM attacks. Zfone is an application that incorporates the ZRTP protocol. |
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The goal of this project is to learn how Zfone operates and to test its functions as they develop. The students will use the Gizmo soft phone and create a set of test scenarios to demonstrate how Gizmo and the Zfone work together to create a secure audio call. |
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SIPConnect is a SIP Trunking standard first developed by Cbeyond, Broadsoft and Cisco. It enables the direct connection of an IP PBX with a SIP Proxy on a host site. SIP Trunking represents one of many initiatives that port telecommunications services onto an IP Infrastructure. Operations and scalability as well as the performance and security characteristics of the services that use the SIPConnect standard will be studied in this project. |
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This study will enable the student to work with the service provider to develop new features, functions and procedures that will improve the various aspects of the service. Cbeyond has donated accounts on its SIP Trunking service for this project and has provided training to the student team. |
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This project continues the work begun in Fall, 2005. (See below) Using the wireless mesh network built in the first semester, the students used the Abacus5000 to generate a SIP-based bulk call load. Results are included in the presentation and project reports below.   |
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Suggested and supported by the Center for Neighborhood Technology (CNT), this project tests and characterizes the ability of a wireless mesh network using the CUWIN routing protocol to support SIP-based VoIP applications. |
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The students built a test bed in the lab and created an initial set of tests. They demonstrated that both a hosted SIP-based service and lab-centered SIP UA/Proxy were able to complete calls on the test bed. |
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In the second semester, the students will
create a variety of stresses and impairments, repeat the
same tests under these new conditions, and characterize the
system's behavior under stress. They also plan to test the
operation of special features and the behavior of a variety
of freely-downloadable UA/Proxy combinations. You may click
below to view the slide presentation that describes the
current status and future direction of this project, and you
can also contact the authors for more detail. |
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This project continues the work begun in
fall, 2005. (See below.) The team continued to study
the performance characteristics of the Reef Point Systems
firewall. They made use of the Abacus5000 load
generator provided by Spirent Communications and thus were
able to send a high capacity load through the firewall and
collect data relative to dropped calls, delayed calls and
voice quality. |
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The call load was further increased using an enhanced version of the freely-available SIP-P software. This enhanced version is applicable to our testing since it carries SDP information and so causes the firewall to open the necessary pinholes. |
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A new student team picked up the work over the summer and is continuing to add tests using freely available software tools to begin various forms of torture testing. The next project report will be available at the end of the fall semester in December, 2006. |
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Proposed and supported by Reef Point Systems, this project
tests and characterizes the behavior of SIP applications operating through
the Reef Point Firewall. The students set-up and configured the firewall,
then integrated it into a simple test bed. Tests of basic SIP call functions
were conducted using a selection of SIP UA/Proxy free applications
downloaded from the Internet. An important by-product of this phase of the
project is a library of traces of the registrations, call set-ups and
disconnects as interpreted by various SIP implementations. This library will
be valuable to many different studies and will also contribute to improving
interoperability between implementations. |
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In the second semester the students will create a more
extensive test plan and examine the behavior of various call features as
well as behavior under various stresses. You may click below to view the
slide presentation that describes the current status and future direction of
this project, and you can also contact the authors for more detail.  |
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Suggested and supported by IP3, Inc, this project explores
the possibility that a thief internal to an enterprise could send
proprietary information cloaked in the RTP stream of a SIP application
through the company’s firewall. |
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The student selected free software packages and created
prototype software that takes a text file and encapsulates it into RTP
packets. These packets then flow through a firewall that would otherwise
reject the export of the file. The recipient of these phony RTP packets then
runs a program to reconstitute the original file. |
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Patent Pending |
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Develop an application to prevent the data
theft in the RTP stream that was demonstrated in the Data
Theft Project. The presentation slides have been
removed for the present in order to support the patent
application which is in progress. |
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SA new student team picked up the work begun in the Fall, 2005 semester. This created the architecture for a proof of concept implementation of the distributed classroom, illustrated in the presentation slides below. The team chose to use Push To Talk model since it seemed adequate to the requirements that only person have the floor at any given time. |
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The project goal is to develop a SIP-based application that will allow students and Instructor to have a classroom experience while students are remote both from the physical classroom and from each other. This scenario supports the situation in which there are several geographical distant university campuses as well as the situation in which students must travel and my need to attend class while in a hotel room or other remote site. |
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The design requirements include floor control by the Instructor, including the
ability to recognize requests to speak, relinquish the floor to a recognized
speaker, and take the ‘microphone’ back as needed. The number of students, their
ability to provide video as well as audio streams, and many other aspects are
considered. |
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Engineers at Lucent Technologies suggested that the students select a project and design that supports and anticipates special needs and future applications. The students chose to create a SIP phone that would the support the special needs of hearing impaired people. This phone employs speech-to-text and text-to-speech technologies, and can have applicability as well in high noise environments and in battle-field applications. |
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The students selected free software packages and created
prototype software that allows a hearing impaired person to place a SIP phone
call by typing a text message. The typed messages are translated into speech at
the receiving side if the person receiving the message is a hearing-enabled
individual. To communicate back to the hearing-impaired individual, the
hearing-enabled participant speaks and their speech is in turn translated into
text at the hearing-impaired participant’s end of the conversation. (Spring - 2005) , (Spring - 2006), |
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The goal of the project is to create an IP PBX using freely
available software packages. The application can be extended to
create a wide variety of features. In the current phase the user
indicates which people are required for a conference call and
the application finds the prospective participants and launches
the call. |
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