During the last couple of years there has been a shift in technology within the Swedish elderly care where the analog personal security alarm is being replaced with personal security alarm that uses internet to communicate. This transition happens due to the ever increasing access to the internet among the elderly and the decreasing availability of analog personal security alarms. This paper describes the work whose purpose is to develop a system that will act as a prototype of a personal security alarm that uses Voice Over IP and the protocol Session Initiation Protocol. The final system is to be comprised by a Raspberry Pi that uses the SIP protocol, a keypad and a soundcard that is built around PCM3060 chip. / De senaste åren pågår det ett teknikskifte inom den svenska äldreomsorgen där de analoga trygghetslarmen ersätts av larm som använder internet för att kommunicera. Denna övergång sker på grund av att tillgång till internet ökar hos äldre personer samt att hushåll med analoga anslutningarna blir allt färre. Denna rapport beskriver arbetet med att ta fram ett system som ska fungera som en prototyp för ett trygghetslarm. Detta system använder sig av Voice Over IP och protokollet Session Initiation Protocol. Det slutliga systemet består av en Raspberry Pi som använder sig av SIP protokollet, en knappsats samt ett ljudkort som är byggt runt ett PCM3060 chip.

Nowadays, VoIP is getting more and more popular. It helps company to reduce cost, extends service to remote area, produce more service opportunities, etc. Besides these advantages, VoIP also put forward security problems. In this paper, we introduce the popular protocols in VoIP and their security mechanisms, by introducing threats to VoIP, we point out the vulnerabilities with the security mechanisms of each VoIP protocol, and give recommendation for each VoIP protocol. In the conclusion part, we evaluate the vulnerabilities of each protocol, and point out in the future, with better protocol architecture, enhanced security policies, VoIP will has a brighter future.

This thesis focuses on enterprise VoIP telephony network security. Introduces brief comparison of old analog and digital voice networks and IP telephone networks with special focus on VoIP system security. The goal of the thesis is to identify the risks of implementation and operation of VoIP technologies in enterprise environment and so thesis brings some conclusion how to minimalize or avoid these risks. First two chapters briefly introduce the development of telephony technologies with differentiation of enterprise telephone network from public telephone networks. Further it describes individual technologies, digitalization of voice, processing the signal and VoIP protocols and components. Third chapter focuses on infrastructure of telephony networks with special interest for architecture of IP telephony and ways of establishing call processing. It describes data flows for further security risk analysis, which this technology came with. Fifth chapter is about enterprise security standards in common and is trying to describe information security management system (ISMS) adopting VoIP technology. Individual security threats and risks are described in sixth chapter, along with known methods how to avoid them. Final parts of thesis concludes of two real situation studies of threats and risks of VoIP technologies implemented in environment of small commercial enterprise and medium size enterprise, in this example represented by University of economics. These chapters conclude theoretical problems shown on practical examples.

À travers les années, l'industrie de la téléphonie a su déployer une multitude de moyens de communication fiables. Son infrastructure garantit une transmission efficace de données, incluant la voix et d'autres contenus, par des voies filaires et sans-fil avec une fiabilité visant les 99,999 %. Cette fiabilité a toutefois un prix; le déploiement des infrastructures nécessaires doit être planifié soigneusement. Ce modèle rigide est bien mal adapté aux situations nécessitant le déploiement rapide d'un réseau, comme une situation d'urgence ou un déploiement militaire. Afin de mettre en place efficacement un système temporaire de communications, un réseau "ad hoc" mobile peut être utilisé. Ce type de réseau dynamique utilise tous les noeuds qui le composent afin de transmettre l'information entre une source et une destination. Toutefois, les protocoles de routage utilisés dans ce type de réseau ne sont présentement pas bien adaptés pour les contenus multimédia nécessitant un flux constant de données, comme la téléphonie sur IP (VoIP). Ce projet de recherche a été divisé en trois volets. Tout d'abord, une étude des solutions existantes a été effectuée. Cette dernière a identifié quatre métriques d'intérêt pour mesurer la performance des réseaux "ad hoc" dans un contexte de VoIP: le délai, la gigue, la fréquence d'interruptions et leur durée moyenne. Il a été montré que sous sa forme actuelle, le protocole AODV est présentement mal adapté pour les réseaux transportant des flux de type VoIP. De plus, le standard IEEE 802.11 a été identifié comme l'un des principaux responsables des piètres performances des réseaux "ad hoc" mobiles. Le deuxième volet du projet a porté sur la conception d'un algorithme, basé sur un filtre de Kalman, devant prévenir la perte de route afin de réduire le nombre d'interruptions subies par un utilisateur de VoIP sur un réseau "ad hoc" mobile. L'algorithme consiste à estimer la puissance de signal reçu de chacun des noeuds voisins ainsi que la vitesse de variation de cette puissance. En connaissant le niveau minimum de puissance nécessaire à la réception d'un message, il est ainsi possible d'estimer la durée de vie restante de chacun des liens vers les noeuds voisins. Une recherche de route aléatoire est déclenchée lorsque la durée de vie prévue d'un lien devient trop courte. L'algorithme a été ajouté aux protocoles AODV et OLSR. Les résultats de simulations ont été mitigés avec AODV. Par contre, pour OLSR, les résultats ont été fort prometteurs. Globalement, la fréquence d'interruptions à survenir sur le réseau a diminué, tout comme la durée totale de ces dernières. Finalement, le troisième volet du projet a porté sur la construction d'un banc d'essais physiques permettant de tester l'algorithme de prévention de perte de route. Ce banc d'essais s'est avéré une preuve de concept pour la construction d'environnements de plus grande envergure. Les résultats obtenus à l'aide de ce dernier sont venus appuyer les résultats obtenus lors des simulations.

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Approved for public release, distribution is unlimited / In this thesis, Voice over Internet Protocol (VoIP) technology will be explored and a recommendation of the operational benefit of VoIP will be provided. A network model will be used to demonstrate improvement of voice End-to-End delay by implementing quality of service (QoS) controls. An overview of VoIP requirements will be covered and recommended standards will be reviewed. A clear definition of a Battle Group will be presented and an overview of current analog RF voice technology will be explained. A comparison of RF voice technology and VoIP will modeled using OPNET Modeler 9.0. / Lieutenant, United States Navy

The appearance of Voice over IP (VoIP) revolutionized the telecommunications word, this technology delivers voice communications over the internet protocol (IP) networks instead of the public switched telephone network (PSTN), calls can be made between two VoIP phones as well as between a VoIP phone and an analog phone connected to a VoIP adapter [1]. The use of this technology gives access to more communication options compared to the conventional telephony but the users face different problems, mostly connectivity and performance issues related to different factors such as latency and jitter [2], these factors affect directly the call quality and can result in choppy voice, echoes, or even in a call failure. The main objective of this work was to create a tool for automatic analysis and evaluation from packet traces, identify connectivity and performance issues, reconstruct the audio streams and estimate the call quality. The results of this work showed that the objectives sated above are met, where a tool that automatically analyzes VoIP calls is created, this tool takes non encrypted pcap files as input and returns a list of calls with different parameters related to connectivity and performance such as delay and jitter, it does as well reconstruct the audio of every VoIP stream and plots the waveform and spectrum of the reconstructed audio for evaluation purposes.

4G LTE is a new cellular phone network standard to provide both the capacity and Quality of Service (QoS) needed to support multimedia applications. Recent research in LTE has explored modifications to the current QoS setup, creating MAC layer schedulers and modifying the current QoS architecture. However, what has not been fully explored are the effects of LTE retransmission choices and capabilities on QoS. This thesis examines the impact of using acknowledgments to recover lost data over the wireless interface on VoIP, FTP and MPEG video applications. Issues explored include interaction between application performance, network transport protocols, LTE acknowledgment mode, and wireless conditions. Simulations show that LTE retransmissions improve FTP throughput by 0.1 to 0.8 Mb/s. With delay sensitive applications, like VoIP and video, the benefits of retransmissions are dependent on the loss rate. When the wireless loss rate is less than 20%, VoIP has similar performance with and without LTE retransmissions. At higher loss rates the use of LTE retransmissions adds degrading the VoIP quality by 71%. With UDP video, the choice of retransmissions or not makes little change when the wireless loss rates are less than 10%. With higher wireless loss rates, the frame arrival delay increases by up to 539% with LTE retransmissions, but the frame rate of the video decreases by up to 34% without those retransmissions. LTE providers should configure their networks to use retransmission policies appro- priate for the type of application traffic. This thesis shows that VoIP, FTP and video require different configurations in the LTE network layers.

Orientador: João Henrique Kleinschmidt / Dissertação (mestrado) - Universidade Federal do ABC. Programa de Pós-Graduação em Engenharia da Informação, 2013