@conference {237, title = {Socio-technical aspects of video phones}, booktitle = {2010 Second International Conference on COMmunication Systems and NETworks (COMSNETS 2010)}, year = {2010}, month = {Jan}, abstract = {

The widespread use of voice over Internet protocol has paved the way for video over Internet protocol. In the past, certain technical shortcomings have prevented the popularity of videophones in the market. With present-day technology, videophones have just about everything required for day-to-day functions. Under such circumstances, certain socio-technical aspects require attention so that videophones can become as widespread and as technically streamlined as a plain old telephone system (POTS) with its additional benefits. A frequently brushed-upon topic is optimum features in the video phone for day-to-day social interactions. We carried out several experiments on different kinds of codecs and video formats to address two issues: i) the size of a video screen and ii) perception of motion and distance. From the measurements, we observed that a small frame rate with low bandwidth is adequate and can result in satisfactory video quality. We also observed that H263 performs well for all the day-to-day social networking activities. Standing 4 feet from the camera can still give reasonably good video quality in the currently available codecs. We believe that socio-technical issues will emerge more clearly over the next several years and they are germane to deployment of PC-based soft phones as well as hard phones.

}, keywords = {Bandwidth, Cameras, Codecs, Computer science, Internet telephony, plain old telephone system, Protocols, social aspects of automation, Social factors, social interactions, social issues, Social network services, social networking activities, sociotechnical aspects, Streaming media, technical issues, video codecs, video coding, video phones, video quality, Videoconference, videophone quality, videophones, videotelephony, voice over Internet protocol}, doi = {10.1109/COMSNETS.2010.5431971}, author = {Chandrasekaran, Vikram and Dantu, Siva and Kadiyala, Priyanka and Ram Dantu and Phithakkitnukoon, Santi} } @conference {76, title = {A society-integrated testbed architecture for peer-to-peer telecommunications}, booktitle = {Testbeds and Research Infrastructures for the Development of Networks Communities and Workshops, 2009. TridentCom 2009. 5th International Conference on}, year = {2009}, month = {April}, abstract = {

Exploiting and deploying telecommunication services on Internet Telephony are imperative for a large market share. However, the distributed nature of peer-to-peer (P2P) network makes services implementation very difficult. In order to meet the research goals of achieving P2P telecommunication services through the union of social and computer networks, we present a society-integrated testbed architecture for P2P voice over IP (VoIP). By introducing an architecture that exploits and entailing social structure based connection establishment, the testbed presents an organic integration of social functionalities with good portability. The testbed also serves as a flexible tool suite that can be used to run experiments and obtain timely feedback based on the performance measurements.

}, keywords = {Computer architecture, computer network, computer networks, Computer science, Feedback, file organisation, Internet telephony, network traffic, P2P voice-over-IP, Peer to peer computing, peer-to-peer computing, peer-to-peer telecommunication service, portable social DHT function, Routing protocols, Social network services, society-integrated testbed architecture, software architecture, Telecommunication services, telecommunication traffic, Testing}, doi = {10.1109/TRIDENTCOM.2009.4976215}, author = {Yang, Xiaohui and Ram Dantu and Wijesekera, Duminda} } @conference {99, title = {Call Algebra}, booktitle = {2008 5th IEEE Consumer Communications and Networking Conference}, year = {2008}, month = {Jan}, abstract = {

In every day life, people communicate through a voice(e.g., VoIP) network with different social groups that range from known people such as family members, friends, and distant relatives to unknown people such as spammers, telemarketers, and phishers. We believe that there exists a human/social dynamics between individuals by the way calls are generated, handled and received. In this paper we present how this dynamics can be used for detecting and filtering unwanted calls. In this paper, we first enumerate the communication patterns between the called party (callee) and the calling parties (callers). Next, we discuss operations on caller-callee matrices constructed based on their communication patterns, and derive call-constructs that can be used for determining the legitimacy of the calls and the callers. Finally, we discuss how these communication patterns and operations can be grouped for solutions to few of the existing IP telephony problems. These solutions can complement the existing no-call-lists in voice networks.

}, keywords = {Algebra, call algebra, caller-callee matrices, communication pattern, Computer science, Context modeling, Filtering, Humans, Internet telephony, IP networks, IP telephony problem, matrix algebra, Mood, Peer to peer computing, social aspects of automation, social group, Social network services, voice-over-IP, VoIP network}, doi = {10.1109/ccnc08.2007.199}, author = {Kolan, Prakash and Ram Dantu} } @conference {90, title = {Experiences in building a multi-university testbed for research in multimedia communications}, booktitle = {Parallel and Distributed Processing, 2008. IPDPS 2008. IEEE International Symposium on}, year = {2008}, month = {April}, abstract = {

The next generation Internet needs to support multimedia services such as voice/video over IP (VoIP) and IP-based television (IPTV) and media distribution. The goal of the project is to develop a testbed for understanding and analysis of vulnerabilities of IP-based multimedia networks. This is a collaborative project between University of North Texas, Columbia University, Purdue University, and University of California at Davis. This project was awarded in 2006 and in one year, we have developed a multi-university testbed with variety of PBX solutions including Asterisk, a multitude of voice and video IP phones, and multiple universities were connected through a secure VPN and Internet!. As with any endeavor, experiences and we have many unique issues, which sometimes cause setbacks. We have dealt with issues concerning interoperability between phones and servers, network address translation (NAT) connectivity issues, and many other collaboration issues concerning people and technology. An interworking of students from multiple universities, faculty, system administrators and support personnel has brought the testbed resources together for a working environment. In this paper we described our progress and experiences in more detail and how to fulfill our mission statement while supporting the various collaborative efforts involved with this project. In addition, we described the current research activities based on the testbed. Finally we described the next steps in the testbed development.

}, keywords = {Asterisk, Collaboration, Collaborative work, digital television, Internet telephony, IP-based television, IPTV, media distribution, Multimedia communication, Multimedia communications, multiuniversity testbed, Network address translation, network address translation connectivity issues, Network servers, next generation Internet, open systems, PBX solutions, phones-servers interoperability, private telephone exchanges, Testing, TV, video IP phones, video over IP, virtual private network, virtual private networks, voice over IP, Web and internet services}, doi = {10.1109/IPDPS.2008.4536392}, author = {Ram Dantu and Schulzrinne, Henning and Sroufe, Paul} } @conference {58, title = {Automatic Calibration Using Receiver Operating Characteristics Curves}, booktitle = {2007 2nd International Conference on Communication Systems Software and Middleware}, year = {2007}, month = {Jan}, abstract = {

Application-level filters, such as e-mail and VoIP spam filters, that analyze dynamic behavior changes are replacing static signature-recognition filters. These application-level filters learn behavior and use that knowledge to filter unwanted requests. Because behavior of a service request{\textquoteright}s participating entities changes rapidly, filters must adapt quickly by using end user{\textquoteright}s preferences about receiving that service request message. Many adaptive filters learn from the participating entities{\textquoteright} behavior; however, none configure themselves automatically to an end user{\textquoteright}s changing tolerance levels. Also, filter administrators cannot manually change the threshold for each service request in real time. Traditional adaptive filters fail when administrators must optimize multiple filter thresholds manually and often. Thus, to improve a filter{\textquoteright}s learning, we must automate its threshold-update process. We propose an automatic threshold-calibration mechanism using Receiver Operating Characteristics (ROC) curves that updates the threshold based on an end user{\textquoteright}s feedback. To demonstrate the mechanism{\textquoteright}s real-time applicability, we integrated it in a Voice over IP (VoIP) spam filter that analyzes incoming Spam over IP Telephony (SPIT) calls. Using this mechanism, we observed good improvement in the VoIP spam filter{\textquoteright}s accuracy. Further, computing and updating the optimum threshold in realtime does not impede the filter{\textquoteright}s temporal performance because we update thresholds after each call{\textquoteright}s completion. Because we reach an optimum threshold for any initial setting, this mechanism works efficiently when we cannot predict end-user behavior. Furthermore, automatic calibration proves efficient when using multiple threshold values.

}, keywords = {adaptive filters, application-level filter learning, automatic threshold-calibration mechanism, behavior, calibration, Computer science, Computer worms, Electronic mail, end user feedback, Filtering, information filtering, information filters, Internet telephony, learning (artificial intelligence), Protocols, receiver operating characteristics curves, sensitivity analysis, service request message, telecommunication traffic, Threshold, tolerance, Viruses (medical)}, doi = {10.1109/COMSWA.2007.382484}, author = {Kolan, Prakash and Vaithilingam, Ram and Ram Dantu} } @article {180, title = {Securing voice over IP}, journal = {IEEE Network}, volume = {20}, year = {2006}, month = {Sept}, pages = {4-5}, keywords = {Authentication, Fingerprint recognition, information filtering, information filters, Information security, Internet telephony, Markup languages, Protection, Protocols, Standardization}, issn = {0890-8044}, doi = {10.1109/MNET.2006.1705876}, author = {Ram Dantu and Ghosal, Dipak and Schulzrinne, Henning} } @conference {33, title = {Securing VoIP and PSTN from integrated signaling network vulnerabilities}, booktitle = {1st IEEE Workshop on VoIP Management and Security, 2006.}, year = {2006}, abstract = {

The liberalization of public switched telephone network (PSTN) and growing acceptance of SIGTRAN protocol suite have introduced new and yet to be trusted signaling entities. Thus security threats emerging from one network not only affects itself but other network also. We show how this integrated signaling environment can become a security threat to emerging VoIP and PSTN networks. We propose a security solution as a fix. Our proposal goes beyond "Gateway Screening" and "SS7 Gatekeeper" proposed by Telcordia and Verizon respectively to defend vulnerable SS7 network.

}, keywords = {Information security, Information systems, Internet telephony, IP networks, ISDN, National security, Protocols, Routing, Signal processing, Switching circuits}, doi = {10.1109/VOIPMS.2006.1638116}, author = {Sengar, Hemant and Ram Dantu and Wijesekera, Duminda} } @article {39, title = {SS7 over IP: signaling interworking vulnerabilities}, journal = {IEEE Network}, volume = {20}, year = {2006}, month = {Nov}, pages = {32-41}, abstract = {

Public telephony - the preferred choice for two-way voice communication over a long time - has enjoyed remarkable popularity for providing acceptable voice quality with negligible connection delays, perhaps due to its circuit-switched heritage. Recently, IP telephony, a packet-based telephone service that runs as an application over the IP protocol, has been gaining popularity. To provide seamless interconnectivity between these two competing services, the Internet Engineering Task Force (IETF) has designed a signaling interface commonly referred to as SIGTRAN. This seamless intersignaling provided by SIGTRAN facilitates any subscriber in one network to reach any other subscriber in the other network, passing through any heterogeneous maze of networks consisting of either of these. Unfortunately, the same intersignaling potentially can be exploited from either side to disrupt the services provided on the other side. We show how this can be done and propose a solution based on access control, signal screening, and detecting anomalous signaling. We argue that to be effective, the latter two should consider syntactic correctness, semantic validity of the signal content, and the appropriateness of a particular signal in the context of earlier exchanged messages

}, keywords = {anomalous signaling detection, circuit-switched heritage, Delay effects, Design engineering, IETF, Integrated circuit interconnections, Internet Engineering Task Force, Internet telephony, intersignaling, IP protocol, ISDN, packet-based telephone service, Protocols, public telephony, Routing, Signal detection, signal screening, signaling interworking, SIGTRAN, SS7 over IP, Switches, telecommunication signalling, two-way voice communication, Web and internet services}, issn = {0890-8044}, doi = {10.1109/MNET.2006.273119}, author = {Sengar, Hemant and Ram Dantu and Wijesekera, Duminda and Jajodia, Sushil} }