Performance evaluation of a technology independent security gateway for Next Generation Networks

With the all IP based Next Generation Networks being deployed around the world, the use of real-time multimedia service applications is being extended from normal daily communications to emergency situations. However, currently different emergency providers utilise differing networks and different technologies. As such, conversations could be terminated at the setup phase or data could be transmitted in plaintext should incompatibility issues exit between terminals. To this end, a novel security gateway that can provide the necessary security support for incompatible terminals was proposed, developed and implemented to ensure the successful establishment of secure real-time multimedia conversations. A series of experiments were conducted to evaluate the security gateway through the use 40 Boghe softphone acting as the terminals. The experimental results demonstrate that the best performance of the prototype was achieved by utilising a multithreading and multi-buffering technique, with an average of 582 microseconds processing overhead. Based upon the ITU-Ts 150 milliseconds one way delay recommendation for voice communications, it is envisaged that such a marginal overhead will not be noticed by users in practice

A methodology for obtaining More Realistic Cross-Layer QoS Measurements in mobile networks: A VoIP over LTE Use Case

Los servicios de voz han sido durante mucho tiempo la primera fuente de ingresos para los operadores móviles. Incluso con el protagonismo creciente del tráfico de datos, los servicios de voz seguirán jugando un papel importante y no desaparecerán con la transición a redes basadas en el protocolo IP. Por otra parte, hace años que los principales actores en la industria móvil detectaron claramente que los usuarios no aceptarían una degradación en la calidad de los servicios de voz. Es por esto que resulta crítico garantizar la experiencia de usuario (QoE) en la transición a redes de nueva generación basadas en conmutación de paquetes. El trabajo realizado durante esta tesis ha buscado analizar el comportamiento y las dependencias de los diferentes servicios de Voz sobre IP (VoIP), así como identificar configuraciones óptimas, mejoras potenciales y metodologías que permitan asegurar niveles de calidad aceptables al mismo tiempo que se trate de minimizar los costes. La caracterización del rendimiento del tráfico de datos en redes móviles desde el punto de vista de los usuarios finales es un proceso costoso que implica la monitorización y análisis de un amplio rango de protocolos y parámetros con complejas dependencias. Para abordar desde la raíz este problema, se requiere realizar medidas que relacionen y correlen el comportamiento de las diferentes capas. La metodología de caracterización propuesta en esta tesis proporciona la posibilidad de recoger información clave para la resolución de problemas en las comunicaciones IP, relaciolándola con efectos asociados a la propagación radio, como cambios de celda o pérdida de enlaces, o con carga de la red y limitaciones de recursos en zonas geográficas específicas. Dicha metodología se sustenta en la utilización de herramientas nativas de monitorización y registro de información en smartphones, y la aplicación de cadenas de herramientas para la experimentación extensiva tanto en redes reales y como en entornos de prueba controlados. Con los resultados proporcionados por esta serie de herramientas, tanto operadores móviles y proveedores de servicio como desarrolladores móviles podrían ganar acceso a información sobre la experiencia real del usuario y sobre cómo mejorar la cobertura, optimizar los servicios y adaptar el funcionamiento de las aplicaciones y el uso de protocolos móviles basados en IP en este contexto. Las principales contribuciones de las herramientas y métodos introducidos en esta tesis son los siguientes: - Una herramienta de monitorización multicapa para smartphones Android, llamada TestelDroid, que permite la captura de indicadores clave de rendimiento desde el propio equipo de usuario. Asimismo proporciona la capacidad de generar tráfico de forma activa y de verificar el estado de alcanzabilidad del terminal, realizando pruebas de conectividad. - Una metodología de post-procesado para correlar la información presente en las diferentes capas de las medidas realizadas. De igual forma, se proporciona la opción a los usuarios de acceder directamente a la información sobre el tráfico IP y las medidas radio y de aplicar metodologías propias para la obtención de métricas. - Se ha realizado la aplicación de la metodología y de las herramientas usando como caso de uso el estudio y evaluación del rendimiento de las comunicaciones basadas en IP a bordo de trenes de alta velocidad. - Se ha contribuido a la creación de un entorno de prueba realista y altamente configurable para la realización de experimentos avanzados sobre LTE. - Se han detectado posibles sinergias en la utilización de instrumentación avanzada de I+D en el campo de las comunicaciones móviles, tanto para la enseñanza como para la investigación en un entorno universitario

Interworking Architectures in Heterogeneous Wireless Networks: An Algorithmic Overview

The scarce availability of spectrum and the proliferation of smartphones, social networking applications, online gaming etc., mobile network operators (MNOs) are faced with an exponential growth in packet switched data requirements on their networks. Haven invested in legacy systems (such as HSPA, WCDMA, WiMAX, Cdma2000, LTE, etc.) that have hitherto withstood the current and imminent data usage demand, future and projected usage surpass the capabilities of the evolution of these individual technologies. Hence, a more critical, cost-effective and flexible approach to provide ubiquitous coverage for the user using available spectrum is of high demand. Heterogeneous Networks make use of these legacy systems by allowing users to connect to the best network available and most importantly seamlessly handover active sessions amidst them. This paper presents a survey of interworking architectures between IMT 2000 candidate networks that employ the use of IEFT protocols such as MIP, mSCTP, HIP, MOBIKE, IKEV2 and SIP etc. to bring about this much needed capacity

TechNews digests: Jan - Mar 2010

TechNews is a technology, news and analysis service aimed at anyone in the education sector keen to stay informed about technology developments, trends and issues. TechNews focuses on emerging technologies and other technology news. TechNews service : digests september 2004 till May 2010 Analysis pieces and News combined publish every 2 to 3 month

Prospects of peer-to-peer SIP for mobile operators

Tämän diplomityön tarkoituksena on esitellä kehitteillä oleva Peer-to-Peer Session Initiation Protocol (P2PSIP), jonka avulla käyttäjät voivat itsenäisesti ja helposti luoda keskenään puhe- ja muita multimediayhteyksiä vertaisverkko-tekniikan avulla. Lisäksi tarkoituksena on arvioida P2PSIP protokollan vaikutuksia ja mahdollisuuksia mobiilioperaattoreille, joille sitä voidaan pitää uhkana. Tästä huolimatta, P2PSIP:n ei ole kuitenkaan tarkoitus korvata nykyisiä puhelinverkkoja. Työn alussa esittelemme SIP:n ja vertaisverkkojen (Peer-to-Peer) periaatteet, joihin P2PSIP-protokollan on suunniteltu perustuvan. SIP mahdollistaa multimedia-istuntojen luomisen, sulkemisen ja muokkaamisen verkossa, mutta sen monipuolinen käyttö vaatii keskitettyjen palvelimien käyttöä. Vertaisverkon avulla käyttäjät voivat suorittaa keskitettyjen palvelimien tehtävät keskenään hajautetusti. Tällöin voidaan ylläpitää laajojakin verkkoja tehokkaasti ilman palvelimista aiheutuvia ylläpito-kustannuksia. Mobiilioperaattorit ovat haasteellisen tilanteen edessä, koska teleliikennemaailma on muuttumassa yhä avoimemmaksi. Tällöin operaattoreiden asiakkaille aukeaa mahdollisuuksia käyttää kilpailevia Internet-palveluja (kuten Skype) helpommin ja tulevaisuudessa myös itse muodostamaan kommunikointiverkkoja P2PSIP:n avulla. Tutkimukset osoittavat, että näistä uhista huolimatta myös operaattorit pystyvät näkemään P2PSIP:n mahdollisuutena mukautumisessa nopeasti muuttuvan teleliikennemaailman haasteisiin. Nämä mahdollisuudet sisältävät operaattorin oman verkon optimoinnin lisäksi vaihtoehtoisten ja monipuolisempien palveluiden tarjoamisen asiakkailleen edullisesti. Täytyy kuitenkin muistaa, että näiden mahdollisuuksien toteuttamisten vaikutusten ei tulisi olla ristiriidassa operaattorin muiden palveluiden kanssa. Lisäksi tulisi muistaa, että tällä hetkellä keskeneräisen P2PSIP-standardin lopullinen luonne ja ominaisuudet voivat muuttaa sen vaikutuksia.The purpose of this thesis is to present the Peer-to-Peer Session Initiation Protocol (P2PSIP) being developed. In addition, the purpose of this thesis is to evaluate the impacts and prospects of P2PSIP to mobile operators, to whom it can be regarded as a threat. In P2PSIP, users can independently and easily establish voice and other multimedia connections using peer-to-peer (P2P) networking. However, P2PSIP is not meant to replace the existing telephony networks of the operators. We start by introducing the principles of SIP and P2P networking that the P2PSIP is intended to use. SIP enables to establish, terminate and modify multimedia sessions, but its versatile exploitation requires using centralized servers. By using P2P networking, users can decentralize the functions of centralized servers by performing them among themselves. This enables to maintain large and robust networks without maintenance costs resulted of running such centralized servers. Telecommunications market is transforming to a more open environment, where mobile operators and other service providers are challenged to adapt to the upcoming changes. Subscribers have easier access to rivalling Internet-services (such as Skype) and in future they can form their own communication communities by using P2PSIP. The results show that despite of these threats, telecom operators can find potential from P2PSIP in concurrence in adaptation to the challenges of the rapidly changing telecom environment. These potential roles include optimization of the network of the operator, but as well roles to provide alternative and more versatile services to their subscribers at low cost. However, the usage of P2PSIP should not conflict with the other services of the operator. Also, as P2PSIP is still under development, its final nature and features may change its impacts and prospects

Security Enhancements in Voice Over Ip Networks

Voice delivery over IP networks including VoIP (Voice over IP) and VoLTE (Voice over LTE) are emerging as the alternatives to the conventional public telephony networks. With the growing number of subscribers and the global integration of 4/5G by operations, VoIP/VoLTE as the only option for voice delivery becomes an attractive target to be abused and exploited by malicious attackers. This dissertation aims to address some of the security challenges in VoIP/VoLTE. When we examine the past events to identify trends and changes in attacking strategies, we find that spam calls, caller-ID spoofing, and DoS attacks are the most imminent threats to VoIP deployments. Compared to email spam, voice spam will be much more obnoxious and time consuming nuisance for human subscribers to filter out. Since the threat of voice spam could become as serious as email spam, we first focus on spam detection and propose a content-based approach to protect telephone subscribers\u27 voice mailboxes from voice spam. Caller-ID has long been used to enable the callee parties know who is calling, verify his identity for authentication and his physical location for emergency services. VoIP and other packet switched networks such as all-IP Long Term Evolution (LTE) network provide flexibility that helps subscribers to use arbitrary caller-ID. Moreover, interconnecting between IP telephony and other Circuit-Switched (CS) legacy telephone networks has also weakened the security of caller-ID systems. We observe that the determination of true identity of a calling device helps us in preventing many VoIP attacks, such as caller-ID spoofing, spamming and call flooding attacks. This motivates us to take a very different approach to the VoIP problems and attempt to answer a fundamental question: is it possible to know the type of a device a subscriber uses to originate a call? By exploiting the impreciseness of the codec sampling rate in the caller\u27s RTP streams, we propose a fuzzy rule-based system to remotely identify calling devices. Finally, we propose a caller-ID based public key infrastructure for VoIP and VoLTE that provides signature generation at the calling party side as well as signature verification at the callee party side. The proposed signature can be used as caller-ID trust to prevent caller-ID spoofing and unsolicited calls. Our approach is based on the identity-based cryptography, and it also leverages the Domain Name System (DNS) and proxy servers in the VoIP architecture, as well as the Home Subscriber Server (HSS) and Call Session Control Function (CSCF) in the IP Multimedia Subsystem (IMS) architecture. Using OPNET, we then develop a comprehensive simulation testbed for the evaluation of our proposed infrastructure. Our simulation results show that the average call setup delays induced by our infrastructure are hardly noticeable by telephony subscribers and the extra signaling overhead is negligible. Therefore, our proposed infrastructure can be adopted to widely verify caller-ID in telephony networks