VoIP: Making Secure Calls and Maintaining High Call Quality

Modern multimedia communication tools must have high security, high availability and high quality of service (QoS). Any security implementation will directly impact on QoS. This paper will investigate how end-to-end security impacts on QoS in Voice over Internet Protocol (VoIP). The QoS is measured in terms of lost packet ratio, latency and jitter using different encryption algorithms, no security and just the use of IP firewalls in Local and Wide Area Networks (LAN and WAN). The results of laboratory tests indicate that the impact on the overall performance of VoIP depends upon the bandwidth availability and encryption algorithm used. The implementation of any encryption algorithm in low bandwidth environments degrades the voice quality due to increased loss packets and packet latency, but as bandwidth increases encrypted VoIP calls provided better service compared to an unsecured environment.Les eines modernes de comunicació multimèdia han de tenir alta seguretat, alta disponibilitat i alta qualitat de servei (QoS). Cap tipus d¿implementació de seguretat tindrà un impacte directe en la qualitat de servei. En aquest article s¿investiga com la seguretat d'extrem a extrem impacta en la qualitat de servei de veu sobre el Protocol d'Internet (VoIP). La qualitat de servei es mesura en termes de pèrdua de proporció de paquets, latència i jitter utilitzant diferents algoritmes d¿encriptació, sense seguretat i només amb l'ús de tallafocs IP en local i en xarxes d'àrea àmplia (LAN i WAN). Els resultats de les proves de laboratori indiquen que l'impacte general sobre el rendiment de VoIP depèn de la disponibilitat d'ample de banda i l'algorisme de xifrat que s'utilitza. La implementació de qualsevol algorisme de xifrat en entorns de baix ample de banda degrada la veu a causa de l'augment de la pèrdua de paquets i latència dels paquets de qualitat, però quan l'ample de banda augmenta les trucades de VoIP xifrades proporcionen un millor servei en comparació amb un entorn sense seguretat.Las herramientas modernas de comunicación multimedia deben tener alta seguridad, alta disponibilidad y alta calidad de servicio (QoS). Ningún tipo de implementación de seguridad tendrá un impacto directo en la calidad de servicio. En este artículo se investiga como la seguridad de extremo a extremo impacta en la calidad de servicio de voz sobre el Protocolo de Internet (VoIP). La calidad de servicio se mide en términos de pérdida de proporción de paquetes, latencia y jitter utilizando diferentes algoritmos de encriptación, sin seguridad y sólo con el uso de cortafuegos IP en local y en redes de área amplia (LAN y WAN). Los resultados de las pruebas de laboratorio indican que el impacto general sobre el rendimiento de VoIP depende de la disponibilidad de ancho de banda y el algoritmo de cifrado que se utiliza. La implementación de cualquier algoritmo de cifrado en entornos de bajo ancho de banda degrada la voz debido al aumento de la pérdida de paquetes y latencia de los paquetes de calidad, pero cuando el ancho de banda aumenta las llamadas de VoIP cifradas proporcionan un mejor servicio en comparación con un entorno sin seguridad

Speech prosody and remote experiments: a technical report

The aim of this paper is twofold. First, we present a review of different recording options for gathering prosodic data in the event that fieldwork is impracticable (e.g. due to pandemics). Under this light, we mimic a long-distance reading task experiment using different software and hardware synchronously. In order to evaluate the employed methodologies, we extract noise levels and frequency manipulation of the recordings. Subsequently, we examine the impact of the different recordings onto linguistic variables, such as the pitch curves and values. We also include a discussion on experimental practicalities. After balancing these factors, we decree an online platform, Zencastr, as the most affordable and practical for acoustic data collection. Secondly, we want to open up a debate on the most optimal remote methodology that researchers on speech prosody can deploy

Impact of Encryption on Qos in Voip

This paper studies the impact of different encryption algorithms on the quality of Voice over Internet Protocol (VoIP). Assuring Quality of Service (QoS) is one of the primary issues in any IP based application that examines the voice quality of VoIP. This paper examines QoS in terms of lost packet ratio, latency and jitter using different encryption algorithms along with firewalling at the IP layer. The results of laboratory tests indicate that the impact on the overall performance of VoIP depends upon the bandwidth available and encryption used. Findings include the need for the provision of bandwidth for encryption, and even when adequate bandwidth is provided encryption algorithms can increase lost packet ratios and packet latency, and reduce.Overall, the results indicate the implementation of encryption algorithms may degrade the voice quality even if bandwidth is adequate

Analysis of the quality of experience of a commercial voice-over-IP service

Voice-over-IP (VoIP) services, enabling users to make cheap phone calls using the Internet, are becoming increasingly popular, not only on desktop computers but also on mobile devices such as smartphones that are connected through mobile networks. Users' perception of the level of quality plays a key role in making a VoIP service to succeed or to fail. This paper demonstrates the influence of technical parameters (such as the audio codec, type of data network, and handovers during the call), device characteristics (such as the platform, manufacturer, model, and operating system), and application aspects (such as the software version and configuration) on the subjective quality of a commercial VoIP service. The relative influence of all these parameters is determined and a decision tree combines these results in order to assess the subjective quality. Combining a large number of objective parameters in a decision tree to determine the user's subjective evaluation of the quality of a VoIP call is a novel and complex procedure. The subjective quality, in turn, has an influence on the duration of the call, and as a result an influence on the usage behavior of the service. The users' assessment of the service quality is not evaluated by merely taking a snapshot of the perceived quality at one moment in time but rather by analyzing the perceived quality over a longer period of time during service usage, which has not been done up to now. Analyzing the VoIP service using a regression analysis over a period of 120 days showed that the perceived quality decreases slightly when the user utilizes the service more often and gets more familiar with it

A large-scale, long-term, user-centric evaluation of a commercial voice-over-IP application

To make cheap voice calls, Voice-over-IP (VoIP) services are often used as an alternative to the traditional telephone service providers. Also on the mobile platform, these VoIP services are becoming increasingly popular due to the increased capabilities and connectivity of mobile devices. Understanding the user's usage behavior and quality assessment of the VoIP service is crucial to optimize the Quality of Experience (QoE) and making the service to succeed. Whereas multimedia services are often evaluated in a controlled laboratorium setting, with a selected group of test subjects, and during a short evaluation period, this study analyzes the service usage and quality assessments in a real environment, with more than thousand users, and over a period of 120 days. The influence of various parameters (such as audio codec, handovers, platform, and manufacturer of the device) on the subjective quality of the service is validated by analyzing the quality assessments of users, provided after each voice call. The time of the day showed to have a significant influence on the number of calls, the duration, and the subjective quality assessment. Time-dependent patterns in the users' usage behavior are identified, thereby providing useful information to predict the system load. A regression analysis of the quality assessments over time shows that the perceived quality gradually decreases as users have utilized the service more, and get more familiar with it. In contrast, the mean duration of the calls increases as users get more familiar with it. This research is important in view of associating technical and contextual parameters with the QoE during service usage

An Algorithm to Evaluate the Echo Signal and the Voice Quality in VoIP Networks

Voice over the Internet Protocol (VoIP) has been increasingly popular, but reliability and voice quality remain important factors that limit the widespread adoption of VoIP systems. Providing good voice quality is of major importance for the transition from the PSTN to VoIP networks. There are several non-real-time algorithms that estimate the voice quality such as the PESQ and the E-model. In this thesis we propose a real-time fuzzy algorithm to estimate the echo quality component of the voice quality in VoIP networks. Differently from the existing algorithms, the proposed algorithm does not need a reference signal and has low computational complexity. For these reasons, the proposed algorithm can be embedded in every VoIP system of a network to monitor live calls, giving an estimate of the instantaneous voice quality to the network provider

Investigation of quality of services (QoS) support for real-time or mission critical services over IEEE 802.11e wireless networks.

Multimedia application is currently making much impact in this technological era. It has been thekey driving force behind the convergence of fixed, mobile and IP networks. Furthermore, real-timeapplications are making head way in vehicular networks, mission critical applications which usededicated short range communications (DSRC). 802.l i e standards support quality of services(QoS) guarantees in these applications. This is opposed to the problem with 802.11 legacy whichis based on distributed coordination function (DCF), and its inability to prioritized applications forservice differentiation. Simulation was done on various 802.l i e networks which use enhancedDCF (EDCF). In these simulations, it was observed that controlling low priority applicationsenhances the effectiveness of high priority applications. Different MAC and traffic generationparameters were used in various scenarios. It was actually observed that high priority applicationshave greater impacts on the performance of the network and hence performs better when itcomes to delay and throughput requirements. Even when the number of high priority applicationswere reduced, the results obtained was still able to satisfy QoS requirements for each traffic type.Results for different scenarios were taken and discussed. Also, differentiated values of delay,throughput and packet loss were recorded when same and different values of MAC and trafficgeneration parameters were used. In all results the International Telecommunications Union (ITU-T) values of these metrics parameters were kept low. These make the network design suitable forroad safety application where very low delay is required for emergency messages and tolerabledelay in routine messages. The results obtained show th at, this network can be applicable inroad safety, simply because of the low delay, and low loss which implies , messages to cars canbe successfully delivered and also good throughput. 802.11 legacy standard lacks servicedifferentiation that limits QoS support for real-time applications. These simulations were able tohandle the drawback associated with this standard and prefer a better standard which is 802.l i ethat provides differentiated access to the metrics that was used in analyzing QoS in this research

Optimización en el despliegue de servicios de Voz sobre IP (VoIP) sobre redes WiFi con restricciones de calidad de servicio

Las tecnologías de Voz sobre IP (VoIP) han permitido el despliegue de nuevos servicios de voz a través de Internet durante las dos últimas décadas. Por otro lado, las redes inalámbricas de área local (WLAN) basadas en el estándar IEEE 802.11 (i.e., WiFi) han experimentado un crecimiento de popularidad debido a su bajo coste y flexibilidad. Sin embargo, el despliegue de comunicaciones de VoIP con garantías de calidad sobre redes IEEE 802.11 implica una serie de dificultades (i.e., los paquetes pueden sufrir pérdidas, colisiones, y retardos variables) que no han sido satisfactoriamente resueltas con las técnicas y modelos disponibles en la actualidad. En esta tesis se desarrolla un nuevo modelo analítico de la sub-capa MAC de IEEE 802.11 que permite estimar la calidad y consumo energético de las conversaciones en un escenario realista de VoIP sobre WiFi (VoWiFi). Además, el modelo anterior se utiliza para plantear y resolver dos nuevas aplicaciones de despliegue y optimización de servicios VoWiFi: (a) el despliegue de vehículos aéreos no tripulados (UAVs) para proveer de un servicio de VoWiFi con garantías de calidad a un conjunto de usuarios y, (b) un nuevo mecanismo de control de admisión de llamadas en la red WiFi corporativa y unifica el acceso al servicio tanto para usuarios de terminales cableados como inalámbricos. Validamos el modelo analítico propuesto frente a simulaciones realizadas con el simulador de red ns-3. Los resultados muestran la utilidad del modelo propuesto para predecir las prestaciones (e.g., retardo, pérdidas) y el consumo energético en la tarjeta de red cuando se transmiten flujos de voz sobre IEEE 802.11 en condiciones no ideales. Esta capacidad de predicción ha sido clave en las propuestas realizadas de nuevas aplicaciones. En el caso del despliegue de drones, nos ha permitido definir un nuevo problema de posicionamiento inicial que puede resultar muy práctico en situaciones de rescate al aire libre. En el caso del control de admisión en entornos corporativos, el modelo nos ha permitido predecir la capacidad máxima de flujos de voz que puede ser admitida en la organización para garantizar calidad a las conversaciones existentes. Usando esta capacidad, hemos planteado un algoritmo nuevo que puede ser utilizado para unificar el control de acceso para usuarios WiFi y usuarios de terminales cableados y que aumenta el número de usuarios concurrentes respecto a los algoritmos existentes.Voice over IP (VoIP) technologies have enabled the deployment of new voice services over the Internet during the last two decades. Meanwhile, wireless local area networks (WLAN) based on the IEEE 802.11 standard (i.e., WiFi) have grown in popularity due to their low cost and flexibility. However, the deployment of quality-guaranteed VoIP communications over IEEE 802.11 networks implies a series of technical difficulties (i.e. lost packets, collisions, and delays) that have not been successfully addressed by the techniques and models available today. In this thesis, we develop a new analytical model for the IEEE 802.11 MAC sub-layer that allows one to estimate quality and energy consumption in a realistic VoIP over WiFi (VoWiFi) scenario. In addition, the previous model is used to propose and solve two new applications for the deployment and optimization of VoWiFi services: (a) deploying unmanned aerial vehicles (UAVs) to provide a VoWiFi service under guaranteed quality to a group of ground users and, (b) a new call admission control mechanism for WiFi corporate networks, which unifies the access to the voice service for both wired and wireless terminals. We validate the proposed analytical model against simulation results obtained with the ns-3 network simulator. Results show the accuracy of the proposed model for the prediction of the performance (e.g. delay, losses) and energy consumption of network interfaces when voice flows are transmitted over IEEE 802.11 under non-ideal conditions. This prediction capability has been a key component of the two VoWiFi applications developed. In the UAV deployment, it has allowed us to define a new initial positioning problem that can be very practical in outdoor rescue situations. Regarding admission control in corporate environments, the model has allowed us to predict the maximum capacity of voice flows that can be admitted in the organization to guarantee quality to existing conversations. Using this capability, we have proposed a new algorithm that can be used to unify access control for wireless and wired users, and that increases the number of concurrent users with respect to existing algorithm

Measures of quality of experience based on the E-model during a VoIP call

Orientador: Yuzo IanoTese (doutorado) ¿ Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de ComputaçãoResumo: A tecnologia de voz sobre protocolo de internet (voz sobre IP) ou simplesmente, VoIP (Voice Over Internet Protocol) ganha novos usuários decorrente do cenário atual de mercado de convergência de redes de dados e telecomunicações. Entretanto, o sucesso desta tecnologia depende fortemente da qualidade do sinal da fala percebida pelo usuário, uma medida subjetiva, também conhecida como Qualidade de Experiência (Quality of Experience - QoE). Esta Tese propõe e valida um mecanismo de observação de variáveis de ambiente durante uma comunicação VoIP tendo como saída um número que representa a QoE não referenciada (sem o sinal transmitido, somente o sinal recebido) ao utilizar o Modelo-E com o propósito de selecionar os melhores parâmetros disponíveis de configuração que afetam o fluxo de voz e ao mesmo tempo obter a melhor qualidade de chamada possível dentro de um determinado cenário de rede. O resultado obtido fez a ligação entre uma medida objetiva, o parâmetro R gerado pelo Modelo-E, e uma medida subjetiva estimada, o MOS (Mean Opinion Score), durante uma chamada VoIP, não se limitando a medida em si. Todavia, cobriu-se todo o cenário para a medição e a comparação com sistemas padrões de medição de qualidade, formando uma base de conhecimento com os resultados obtidos. O método utilizado de estimativa de qualidade da fala foi comparado em diferentes codecs de voz padrão ITU-T (PCMU, GSM, G.723, G.729, G.726-32), testados em uma topologia de rede que sofreu distorções, como diferentes situações de perdas de dados (0,0%, 1,0%, 2,0%, 2,5%, 3,0%, 5,0%, 7,5%, 10,0%). Uma análise de regressão foi utilizada para permitir uma melhor compreensão do impacto das condições de rede e codecs sobre a QoE do serviço VoIP medido. Foi utilizado um suíte de testes padronizados para medição da QoE nos arquivos de voz recebidos e transmitidos durante os testes baseados em testes referenciados (com os sinais transmitidos e recebidos) nos padrões ITU-T P.863 (Perceptual Objective Listening Quality Assessment - POLQA) e ITU-T P.862 (Perceptual Evaluation of Speech Quality - PESQ) e os resultados foram comparados com os obtidos pelo método não referenciado proposto para medida de QoE. Para os resultados do codec testado foi aplicado o método de regressão linear sendo a variável independente as medidas de QoE obtidas pelo método proposto e a variável dependente foram os resultados obtidos pelos algoritmos PESQ e POLQA. Para todos os codecs testados, o Coeficiente de Determinação (R2) entre o método proposto e os resultados obtidos pelo algoritmo PESQ foram superiores a 0,90 indicando uma forte correlação linear. Já entre o método proposto e o algoritmo POLQA, para os codecs PCMU, GSM e G.723, os resultados de R2 foram superiores a 0,973, indicando uma correlação muito forte. R2 para o codec G.726-32 foi de 0,88 indicando uma correlação forte. Já para o codec G.729, R2 ficou em 0,67 indicando que o modelo linear pode não ser o mais adequado para explicar a relação entre os resultados do método proposto e os valores obtidos pelo algoritmo POLQAAbstract: The technology of Voice over Internet Protocol (Voice over IP or simply VoIP) is present in our personal and professional lives. The number of VoIP users increases day after day due to the current scenario of convergence of data and telecommunications networks. However, the success of this technology depends on the speech signal quality as perceived by the user, a subjective measure as function of the user's point of view, also known as the Quality of Experience (QoE). This thesis proposes and validates an environment variable observation mechanism during a VoIP communication having as output a number that represents the QoE not referenced (without the transmitted signal, only the received signal) of the call, using the E-Model, in order to select the best available parameter settings that affect voice flow of the current VoIP call and at the same time gets the best call quality as possible within a given network scenario. The result relates an objective measurement, the R parameter generated by the E-Model, to the estimated subjective measurement, MOS, during a VoIP call, not limited to the measurement itself. However, it covered the whole scenario for measurement and comparison with quality measurement standards systems, forming a knowledge base with the results. The method of speech quality estimation was compared in different standard voice codec's ITU-T (PCMU, GSM, G.723, G.729, G.726-32) tested in a network topology that has suffered distortions, as different situations of data loss (0.0%, 1.0%, 2.0%, 2.5%, 3.0%, 5.0%, 7.5%, 10.0%). A regression analysis was used to allow a better understanding of the impact of network conditions and codec's on the VoIP service QoE measured. In this thesis it was used a suite of standardized tests for measuring QoE in voice files received and transmitted during testing based on referenced tests (with the transmitted and received signals) in the ITU-T P.863 standard (Perceptual Objective Listening Quality Assessment - POLQA) and ITU-T P.862 (Perceptual Evaluation of Speech Quality - PESQ) and the results were compared with those obtained by the no referenced measure method of QoE proposed in this thesis. The linear regression was applied in order to analyze the results of the tested codec. The independent variable was the QoE measurements obtained by the proposed method to measure QoE and the dependent variable were the results of PESQ and POLQA algorithms. For all tested codec's, the Coefficient of Determination (R2) between the proposed method and the results of the PESQ algorithm was higher than 0.90 indicates a strong linear correlation. For PCMU, G.723, GSM codec's R2 was greater than 0.973 indicating a strong correlation between the results of proposed method and the results of POLQA algorithm. R2 for G.726-32 codec was 0.88 that indicates a high correlation. For G.729 codec, R2 was 0.67 that indicates the linear model may not be the most appropriate to explain the relationship between the results of the proposed method and values obtained by POLQA algorithmDoutoradoTelecomunicações e TelemáticaDoutor em Engenharia Elétric