Land mobile radio (LMR) systems are used for communication by public safety and other government and commercial organizations. LMR systems offer mission-critical or even life-critical service in the day-to-day activities of such organizations. Traditionally, a variety of different LMR systems have been deployed by different organizations, leading to a lack of radio interoperability. A voice application that connects LMR systems via a packet-switched network is called an LMR Voice over IP (LMRVoIP) system and is a potential solution to the interoperability problem. LMRVoIP systems are time critical, i.e., are delay and jitter sensitive. Transmission of LMRVoIP traffic in a congested packet-switched network with no quality of service (QoS) or priority mechanisms in place could lead to high delays and extreme variations in delay, i.e., high jitter, thus resulting in poor application performance. LMRVoIP systems may also have performance issues with the use of virtual private networks (VPNs). To the best of our knowledge, there has been no prior thorough investigation of the performance of an LMRVoIP system with different queuing schemes for QoS and with the use of VPN. In this thesis, we investigate the performance of an LMRVoIP system with different queuing schemes and with the use of VPN. An experimental test bed was created to evaluate four QoS queuing schemes: first-in first-out queuing (FIFO), priority queuing (PQ), weighted fair queuing (WFQ), and class-based weighted fair queuing (CBWFQ). Quantitative results were obtained for voice application throughput, delay, jitter, and signaling overhead. Results show that, compared to a baseline case with no background traffic, LMRVoIP traffic suffers when carried over links with heavy contention from other traffic sources when FIFO queuing is used. There is significant packet loss for voice and control traffic and jitter increases. FIFO queuing provides no QoS and, therefore, should not be used for critical applications where the network may be congested. The situation can be greatly improved by using one of the other queuing schemes, PQ, WFQ, or CBWFQ, which perform almost equally well with one voice flow. Although PQ has the best overall performance, it tends to starve the background traffic. CBWFQ was found to have some performance benefits over WFQ in most cases and, thus, is a good candidate for deployment. The LMRVoIP application was also tested using a VPN, which led to a modest increase in latency and bandwidth utilization, but was found to perform well. / Master of Science

Background. SIP and RTP are two protocols that are widely used, and they play an important role in VoIP services. VoIP is an integral part of many communication services, e.g., Microsoft Teams, Skype, Discord, and communications over cellular networks (VoLTE and VoWiFi). Since these technologies are so widely used, a high level of security is paramount. Objectives. The aim of this study is threefold: (1) To investigate if it is possible to create a penetration testing methodology for SIP and RTP, where the target group is penetration testers with no previous knowledge of these protocols. (2) To identify previously discovered vulnerabilities and attacks. (3) Due to the lack of domain experts, a methodology of this kind will hopefully help penetration testers without prior knowledge, easing them into a new work area. Further, the aim is to increase awareness of potential vulnerabilities in such systems. Methods. Through a literature review, threat modeling, and exploratory penetration testing on three different testbeds, several vulnerabilities and attacks were identified and validated. From the results, a methodology was compiled. For evaluation purposes, it was evaluated by a third party, who tested it on a testbed and gave feedback. Results. The results from our research show that SIP and RTP are susceptible to a wide array of different attacks even to this day. From our literature study, it was determined that most of these attacks have been known for a long time. Using exploratory penetration testing, we managed to verify most of these attacks on three different systems. Additionally, we discovered a few novel attacks that we did not find in previous research. Conclusions. Our literature study suggests that SIP and RTP based systems are relatively susceptible to multiple attacks. Something we also validated during the exploratory testing phase. We successfully executed multiple existing attacks and some new attacks on three different testbeds. The methodology received mostly positive feedback. The results show that many of the participants appreciated the simplicity and concrete model of the methodology. Due to the low number of participants in the evaluation, an improvement to the study and results would be to increase the population and also have multiple novice penetration testers test several different systems. An increase in the number of testbeds would also further support the results and help generalize the methodology. / Bakgrund. SIP och RTP  är två protokoll som är vitt använda och spelar en väldigt viktig roll i VoIP-tjänster. VoIP utgör en viktig del i många kommunikationstjänster, t.ex. Microsoft Teams, Skype och Discord, men även i kommunikation över mobilnätet (VoLTE och VoWiFi). Eftersom dessa teknologier används i så stor utsträckning, är säkerhet av största vikt. Syfte. Syftet med denna studie är trefaldig: (1) Undersöka om det är möjligt att utforma en penetration testningsmetod för SIP och RTP, för en målgrupp av penetrationstestare utan förkunskaper kring dessa protokoll. (2) Att identifiera sårbarheter och attacker från tidigare studier. (3) På grund av brist på kompentens inom området penetrationstestning och telekommunikation kan en sådan här metod förhoppningsvis hjälpa till att introducera penetrationstestare utan tidigare erfarenhet till det här specifika området. Ytterligare är också målet att att öka medvetheten när det kommer till sårbarheter i sådana system. Metod. Genom en literaturstudie, hotmodellering och utforskande penetrationstestning på tre olika testmiljöer har ett flertal sårbarheter och attacker identifieras och utförts. Från resultatet utformades en metod för penetrationstesning, som sedan evaluerades genom att en tredje part testade metoden och gav återkoppling som rör metodens format och struktur. Resultat. Resultaten från vår studie visar att SIP och RTP är sårbara för en rad olika attacker än idag. Resultaten från vår litteraturstudie visar att många av dessa attacker har varit kända under en lång tid. Vi lyckades verifiera de flesta av dessa attacker genom utforskande penetationstestning på tre olika system. Dessutom lyckades vi identifiera ett antal nya attacker som inte tidigare nämnts i forskning inom området. Slutsatser. Resultaten från vår litteraturstudie visar att system som använder sig av SIP och RTP är relativt sårbara för en mängd olika attacker. Detta bekräftades i den utforskande testningen, där ett flertal kända samt nya attacker utfördes framgångsrikt. Den interna evalueringen i studien visar på att metoden kan appliceras framgångsrikt på ett flertal olika system, med begränsningen att endast tre system testats. Resultaten från den externa evalueringen, där penetrationstestare blev tillfrågade att utvärdera och testa metoden visar att de hade en relativt positiv inställning till metoden. För att ytterligare underbygga detta påstående krävs en större population, både för testningen och utvärderingen. Det krävs också att en större mängd testmiljöer används för att kunna generalisera metoden.

Voice Over IP (VoIP) telephony is becoming widespread, and is often integrated into computer networks. Because of his, it is likely that malicious software will threaten VoIP systems the same way traditional computer systems have been attacked by viruses, worms, and other automated agents. While most users have become familiar with email spam and viruses in email attachments, spam and malicious traffic over telephony currently is a relatively unknown threat. VoIP networks are a challenge to secure against such malware as much of the network intelligence is focused on the edge devices and access environment. A novel security architecture is being developed which improves the security of a large VoIP network with many inexperienced users, such as non-IT office workers or telecommunication service customers. The new architecture establishes interaction between the VoIP backend and the end users, thus providing information about ongoing and unknown attacks to all users. An evaluation of the effectiveness and performance of different implementations of this architecture is done using virtual machines and network simulation software to emulate vulnerable clients and servers through providing apparent attack vectors.

Approved for public release, distribution is unlimited / Converging voice and data networks has the potential to save money and is the main reason Voice over Internet Protocol (VoIP) is quickly becoming mainstream in corporate America. The potential VoIP offers to more efficiently utilize the limited connectivity available to ships at sea makes it an attractive option for the Navy. This thesis investigates the usefulness of VoIP for the communications needs of a unit level ship. This investigation begins with a review of what VoIP is and then examines the ship to shore connectivity for a typical unit level ship. An OMNeT++ model was developed and used to examine the issues that affect implementing VoIP over this type of link and the results are presented. / Lieutenant Commander, United States Navy

O projeto VoIPFix surgiu da necessidade de uma ferramenta que complementasse as demais existentes no ramo de análise de redes de computadores para telefonia IP. Ele foi construído para ser uma ferramenta de gerenciamento eficiente e exclusiva para VoIP, com funcionalidades necessárias para dar suporte ao profissional de rede de computadores e telefonia IP a observar e diagnosticar problemas de VoIP. / The VoIPFix project arose from the need for a tool to complement similar tools in the analysis of computer networks for IP telephony. It was built to be an efficient and unique management tool for VoIP, with advanced features required to support the computer network and IP telephony professionals to observe and diagnose problems related to VoIP.

A utilização das redes sem fio nos mais diversos segmentos vem crescendo vertiginosamente nos últimos anos. Aliado ao crescimento da utilização das redes, está a diversificação de aplicações utilizadas por todos os usuários, sejam eles corporativos ou domésticos. Isto tem contribuído significativamente para o desenvolvimento de tecnologias que ofereçam mecanismos de qualidade de serviço, de forma a permitir o uso satisfatório de aplicações como voz e vídeo em tempo real. Este trabalho estuda os efeitos que a existência de tráfego de voz e dados em redes sem fio padrão IEEE 802.11 causa no desempenho da rede e por consequência no desempenho das aplicações. Para realizar este estudo foram executadas simulações baseadas em uma rede sem fio com topologia ad hoc, com variações no número de estações e quantidade de tráfego gerado. Foram simulados os padrões 802.11g e 802.11e com o objetivo de analisar o desempenho dos mecanismos de qualidade de serviço e os benefícios que estes mecanismos geram para a transmissão de tráfego em tempo real em redes sem fio padrão IEEE 802.11. Verificou-se que o padrão IEEE 802.11e apresenta um ganho de desempenho para aplicações de tempo real, porém ele apresenta limitações que devem ser consideradas nos projetos de redes sem fio. / The use of wirelles networks in most various sectors has been growing drastically in past years Allied to the wireless networks use, the diversification of applications and services provided can be directly verified whether by home or corporate users. This alliance contributes significantly to the needs of technology development which offer the quality of service mechanisms, allowing satisfactory use of real-time applications like voice and video This paper studies the effects that coexistent voice and data traffic on a IEEE 802.11 standard wireless network cause in the network performance and, consequently, in the applications performance. To make this study a reality, it was necessary to run simulations of a wireless ad hoc topology network, with variations in the number of workstations and the quantity of generated traffic. The 802.11g and 802.11e standards were used in the simulations with the purpose of analyzing the performance of quality of service mechanisms and the benefits they create for the real-time transmissions in IEEE 802.11 standard wireless networks. It was verified that the IEEE 802.11e standard presents a perfomance gain for the real-time applications, but it has limitations that should be considered in wireless networks design.

A utilização das redes sem fio nos mais diversos segmentos vem crescendo vertiginosamente nos últimos anos. Aliado ao crescimento da utilização das redes, está a diversificação de aplicações utilizadas por todos os usuários, sejam eles corporativos ou domésticos. Isto tem contribuído significativamente para o desenvolvimento de tecnologias que ofereçam mecanismos de qualidade de serviço, de forma a permitir o uso satisfatório de aplicações como voz e vídeo em tempo real. Este trabalho estuda os efeitos que a existência de tráfego de voz e dados em redes sem fio padrão IEEE 802.11 causa no desempenho da rede e por consequência no desempenho das aplicações. Para realizar este estudo foram executadas simulações baseadas em uma rede sem fio com topologia ad hoc, com variações no número de estações e quantidade de tráfego gerado. Foram simulados os padrões 802.11g e 802.11e com o objetivo de analisar o desempenho dos mecanismos de qualidade de serviço e os benefícios que estes mecanismos geram para a transmissão de tráfego em tempo real em redes sem fio padrão IEEE 802.11. Verificou-se que o padrão IEEE 802.11e apresenta um ganho de desempenho para aplicações de tempo real, porém ele apresenta limitações que devem ser consideradas nos projetos de redes sem fio. / The use of wirelles networks in most various sectors has been growing drastically in past years Allied to the wireless networks use, the diversification of applications and services provided can be directly verified whether by home or corporate users. This alliance contributes significantly to the needs of technology development which offer the quality of service mechanisms, allowing satisfactory use of real-time applications like voice and video This paper studies the effects that coexistent voice and data traffic on a IEEE 802.11 standard wireless network cause in the network performance and, consequently, in the applications performance. To make this study a reality, it was necessary to run simulations of a wireless ad hoc topology network, with variations in the number of workstations and the quantity of generated traffic. The 802.11g and 802.11e standards were used in the simulations with the purpose of analyzing the performance of quality of service mechanisms and the benefits they create for the real-time transmissions in IEEE 802.11 standard wireless networks. It was verified that the IEEE 802.11e standard presents a perfomance gain for the real-time applications, but it has limitations that should be considered in wireless networks design.

This work presents a novel Peer to Peer (P2P) framework for Session Initiation Protocol (SIP) on Mobile Ad Hoc Network (MANET). SIP is a client-server model of computing which can introduce a single point of failure problem. P2P SIP addresses this problem by using a distributed implementation based on a P2P paradigm. However, both the traditional SIP and P2P SIP architectures are not suitable for MANETs because they are initially designed for infrastructured networks whose most nodes are static. We focus on distributed P2P resource lookup mechanisms for SIP which can tolerate failures resulting from the node mobility. Our target application is SIP-based multimedia communication in a rapidly deployable disaster emergency network. To achieve our goal, we provide four contributions as follows. The first contribution is a novel P2P lookup architecture based on a concept of P2P overlay network called a Structured Mesh Overlay Network (SMON). This overlay network enables P2P applications to perform fast resource lookups in the MANET environment. SMON utilizes a cross layer design based on the Distributed Hashing Table (DHT) and has direct access to OLSR routing information. Its cross layer design allows optimizing the overlay network performance during the change of network topology. The second contribution is a distributed SIP architecture on MANET providing SIP user location discovery in a P2P manner which tolerates single-point and multiple-point of failures. Our approach extends the traditional SIP user location discovery by utilizing DHT in SMON to distribute SIP object identifiers over SMON. It offers a constant time on SIP user discovery which results in a fast call setup time between two MANET users. From simulation and experiment results, we find that SIPMON provides the lowest call setup delay when compared to the existing broadcast-based approaches. The third contribution is an extended SIPMON supporting several participating MANETs connected to Internet. This extension (SIPMON+) provides seamless mobility support allowing a SIP user to roam from an ad hoc network to an infrastructured network such as Internet without interrupting an ongoing session. We propose a novel OLSR Overlay Network (OON), a single overlay network containing MANET nodes and some nodes on the Internet. These nodes can communicate using the same OLSR routing protocol. Therefore, SIPMON can be automatically extended without modifying SIPMON internal operations. Through our test-bed experiments, we prove that SIPMON+ has better performance in terms of call setup delay and handoff delay than MANET for Network Mobility (MANEMO). The fourth contribution is a proof-of-concept and a prototype of P2P multimedia communication based on SIPMON+ for post disaster recovery missions. We evaluate our prototype and MANEMO-based approaches through experimentation in real disaster situations (Vehicle to Infrastructure scenarios). We found that our prototype outperforms MANEMO-based approaches in terms of call setup delay, packet loss, and deployment time.

O projeto VoIPFix surgiu da necessidade de uma ferramenta que complementasse as demais existentes no ramo de análise de redes de computadores para telefonia IP. Ele foi construído para ser uma ferramenta de gerenciamento eficiente e exclusiva para VoIP, com funcionalidades necessárias para dar suporte ao profissional de rede de computadores e telefonia IP a observar e diagnosticar problemas de VoIP. / The VoIPFix project arose from the need for a tool to complement similar tools in the analysis of computer networks for IP telephony. It was built to be an efficient and unique management tool for VoIP, with advanced features required to support the computer network and IP telephony professionals to observe and diagnose problems related to VoIP.

Les communications téléphoniques en voix sur IP souffrent de la perte de paquets causée par les problèmes d'acheminement dus aux nœuds du réseau. La perte d'un paquet de voix induit la perte d'un segment de signal de parole (généralement 10ms par paquet perdu). Face à la grande diversité des codeurs de parole, nous nous sommes intéressés dans le cadre de cette thèse à proposer une méthode de masquage de pertes de paquets générique, indépendante du codeur de parole utilisé. Ainsi, le masquage de pertes de paquets est appliqué au niveau du signal de parole reconstruit, après décodage, s'affranchissant ainsi du codeur de parole. Le système proposé repose sur une modélisation classique de type « modèles de Markov cachés » afin de suivre l'évolution acoustique de la parole. À notre connaissance, une seule étude a proposé l'utilisation des modèles de Markov cachés dans ce cadre [4]. Toutefois, Rødbro a utilisé l'utilisation de deux modèles, l'un pour la parole voisée, l'autre pour les parties non voisées, posant ainsi le problème de la distinction voisée/non voisée. Dans notre approche, un seul modèle de Markov caché est mis en œuvre. Aux paramètres classiques (10 coefficients de prédiction linéaire dans le domaine cepstral (LPCC) et dérivées premières) nous avons adjoint un nouvel indicateur continu de voisement [1, 2]. La recherche du meilleur chemin avec observations manquantes conduit à une version modifiée de l'algorithme de Viterbi pour l'estimation de ces observations. Les différentes contributions (indice de voisement, décodage acoutico-phonétique et restitution du signal) de cette thèse sont évaluées [3] en terme de taux de sur et sous segmentation, taux de reconnaissance et distances entre l'observation attendue et l'observation estimée. Nous donnons une indication de la qualité de la parole au travers d'une mesure perceptuelle : le PESQ. / Packet loss due to misrouted or delayed packets in voice over IP leads to huge voice quality degradation. Packet loss concealment algorithms try to enhance the perceptive quality of the speech. The huge variety of vocoders leads us to propose a generic framework working directly on the speech signal available after decoding. The proposed system relies on one single "hidden Markov model" to model time evolution of acoustic features. An original indicator of continuous voicing is added to conventional parameters (Linear Predictive Cepstral Coefficients) in order to handle voiced/unvoiced sound. Finding the best path with missing observations leads to one major contribution: a modified version of the Viterbi algorithm tailored for estimating missing observations. All contributions are assessed using both perceptual criteria and objective metrics.