The emerging technologies in the field of telecommunications enable us to access high speed data services through mobile handsets and portable modems over the mobile networks. The recent statistics also shows the use of mobile broad band services are increasing and gaining popularity. In this thesis we have investigated the impact of payload size and data rate on one-way delay and packet loss in operational 3G mobile networks, through network level measurements. To collect the network level traces an experimental testbed is developed. For accurate measurement Endace DAG cards together with GPS synchronization is implemented. Results are gathered from three different commercial mobile operators in Sweden. From the results it is concluded that the combination of maximum payload size and data rate resulted in minimum one-way delay. It is also observed within the big payload size category, that the percentage of packet loss is less as compared to the smaller payload sizes. Such findings will improve efficiency of application developers to meet the challenges of UMTS network conditions.

Vu les perspectives qu'ils offrent, les réseaux de capteur sans fil (RCSF) ont perçu un grand engouement de la part de la communauté de recherche ces dernières années. Les RCSF couvrent une large gamme d'applications variant du contrôle d'environnement, le pistage de cible aux applications de santé. Les RCSFs sont souvent déployés aléatoirement. Ce dispersement des capteurs nécessite que les protocoles de transmission utilisés soient résistants aux conditions environnementales (fortes chaleurs ou pluies par exemple) et aux limitations de ressources des nœuds capteurs. En effet, la perte de plusieurs nœuds capteurs peut engendrer la perte de communication entre les différentes entités. Ces limitations peuvent causer la perte des paquets transmis ce qui entrave l'activité du réseau. Par conséquent, il est important d'assurer la fiabilité des transmissions de données dans les RCSF d'autant plus pour les applications critiques comme la détection d'incendies. Dans cette thèse, nous proposons une solution complète de transmission de données dans les RCSF répondant aux exigences et contraintes de ce type de réseau. Dans un premier temps, nous étudions les contraintes et les challenges liés à la fiabilisation des transmissions dans les RCSFs et nous examinons les travaux proposés dans la littérature. Suite à cette étude nous proposons COMN2, une approche distribuée et scalable permettant de faire face à la défaillance des nœuds. Ensuite, nous proposons un mécanisme de contrôle d'erreur minimisant la perte de paquets et proposant un routage adaptatif en fonction de la qualité du lien. Cette solution est basée sur des acquittements implicites (overhearing) pour la détection des pertes des paquets. Nous proposons ensuite ARRP une variante de AJIA combinant les avantages des retransmissions, de la collaboration des nœuds et des FEC. Enfin, nous simulons ces différentes solutions et vérifions leurs performances par rapport à leurs concurrents de l'état de l'art. / Over the past decades, we have witnessed a proliferation of potential application domainsfor wireless sensor networks (WSN). A comprehensive number of new services such asenvironment monitoring, target tracking, military surveillance and healthcare applicationshave arisen. These networked sensors are usually deployed randomly and left unattendedto perform their mission properly and efficiently. Meanwhile, sensors have to operate ina constrained environment with functional and operational challenges mainly related toresource limitations (energy supply, scarce computational abilities...) and to the noisyreal world of deployment. This harsh environment can cause packet loss or node failurewhich hamper the network activity. Thus, continuous delivery of data requires reliabledata transmission and adaptability to the dynamic environment. Ensuring network reliabilityis consequently a key concern in WSNs and it is even more important in emergencyapplication such disaster management application where reliable data delivery is the keysuccess factor. The main objective of this thesis is to design a reliable end to end solution for data transmission fulfilling the requirements of the constrained WSNs. We tackle two design issues namely recovery from node failure and packet losses and propose solutions to enhance the network reliability. We start by studying WSNs features with a focus on technical challenges and techniques of reliability in order to identify the open issues. Based on this study, we propose a scalable and distributed approach for network recovery from nodefailures in WSNs called CoMN2. Then, we present a lightweight mechanism for packetloss recovery and route quality awareness in WSNs called AJIA. This protocol exploitsthe overhearing feature characterizing the wireless channels as an implicit acknowledgment(ACK) mechanism. In addition, the protocol allows for an adaptive selection of therouting path by achieving required retransmissions on the most reliable link. We provethat AJIA outperforms its competitor AODV in term of delivery ratio in different channelconditions. Thereafter, we present ARRP, a variant of AJIA, combining the strengthsof retransmissions, node collaboration and Forward Error Correction (FEC) in order toprovide a reliable packet loss recovery scheme. We verify the efficiency of ARRP throughextensive simulations which proved its high reliability in comparison to its competitor.

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.

This master’s thesis is focused on ensuring of Quality of Service (QoS) in wireless network for standards IEEE 802.11 a/b/g/n. First part of this thesis is focused on the theory of this issue. It covers methods of data transfer in computer networks and individual transfer parameters especially on parameters needed to ensure quality of service. It also describes standards for wireless data transmission and protocols for real time data transmission. The second part is devoted to OPNET modeler and to simulations created on this program. Simulations are focused on real-time data transfer and compare the standards in terms of delay and jitter.

Diploma thesis is focused on the supporting of Quality of Services in wireless networks, especially in the Linux operation systems. The topic is connected not only with OS, but also with the wireless standard, which supports QoS in wireless networks called IEEE 802.11e. QoS is needed especially for time-consuming data transfers in real time. The theoretical part deals with the theoretical analysis of the issue of the QoS support. There are described parameters, which occurred in quality of services support. This section also deals with the division of services that are used to transmit data across computer networks. It also describes the QoS support in wireless networks according 802.11e, its implementation and methods of accessing a medium with a without possibility of traffic. This part is followed by a description of QoS support in UNIX operating systems. The chapter describes how is the QoS support designed in these operating systems. There are also characterized concrete tools which are used for control the data flow in the operating systems using Linux. At the end the theoretical part deals with different types of queues and methods used in linux OS. In the practical part of the thesis, there are various designed topologies and scenarios to verify the functionality of QoS support in wireless networks using a Unix system. These chapters show the results of different tests at selected transmission data streams that are sensitive to transmission time. There is also verified cooperation of QoS support between devices operating on the network and data link layers. The output of this work is to design a laboratory exercise for the subject Network Architecture. This exercise is focused on familiarization with the QoS support functionality in wireless networks and in Unix-like operating systems. This chapter also describes the devices and programs that are needed to measure this task. The last part of the chapter describes the procedure for the preparation of the measuring station. For this laboratory task, there is an inserted manual in the annex.

Over the course of last decade, the demand for VoIP (Voice over Internet Protocol) applications has increased significantly among enterprises and individuals due to its low cost. This increasing demand resulted in a significant increase in users who require reliable VoIP communication systems. QoS (Quality of Service) is a major issue in VoIP implementation and is a method to impel the development of real-time multimedia services like VoIP and videoconferencing. However, there are certain challenges in achieving QoS for VoIP application, which need special attentions; like latency and packet loss. The VoIP servers which are functioning on single core software/hardware model have high latency and packet loss issues due to their limited processing bandwidth. A multicore software/hardware model is the solution to cope up with the increasing demands of VoIP and yet an active research area in telecommunication. Using a multicore software/hardware model for VoIP has several challenges, one of the challenges is to design and implement QoS Benchmarking module for VoIP client and server on multicore. In this thesis the focus is on latency and packet loss of SIP messages on OpenSIPS server. This is done by performing stress testing for QoS benchmarking, where delay and call drop rate is calculated for SIP (Session Initiation Protocol) signaling messages on parallel VoIP client server model. The model is built in C for multicore and is used as a simulation tool. SIP is widely deployed protocol for call establishment, maintenance and termination in VoIP.

Mobile cloud gaming (MCG) is an emerging concept which aims to deliver video games on-demand to users with the use of cloud technologies. Cloud technology allows the offloading of computation from a less powerful user device or thin client to more robust cloud servers to minimize power consumption and provide additional cloud services such as storage. MCG is therefore very helpful that can reduce the costs of expensive hardware, but the challenge is that it requires a high Quality of Service (QoS) in order to stream and play the games where the users have a high Quality of Experience (QoE). The goal of the study is to investigate how users' QoE is affected by network conditions while playing MCG and compare the results from a previous study. A testbed was made in order to conduct subjective tests where users are going to play Counter Strike: Global Offensive (CS: GO) on a smartphone using Steam Remote Play. The testbed consists of a router, tablet, smartphone, headset, Xbox controller, USB-C multi-port adapter and four different PC's. Participants on campus, both students and non-students, were invited to participate in the experiment. A total of 24 participants completed the tests; however, results from two participants were excluded due to software issues. There were 23 network conditions that was tested for each user and included factors such as round-trip time (RTT), packet losses, bursty jitter, random jitter or combinations of different factors. A multi-platform tool, ALTRUIST, was used to control the applications and facilitate the data collection from the devices and NetEm changed the network conditions. The results showed that the network condition [bj(rtt200i15)] had the highest mean opinion score (MOS) of the QoE of 4.5 for the users with 200 milliseconds of bursty jitter every 15 seconds. The worst network condition tested with the lowest QoE rating of 1.4 was network condition [rtt25pl12] that had 25 milliseconds of RTT and 12% packet losses. There were differences between the male and female participants where the MOS of the QoE results was significantly higher with up to 1.5 MOS QoE rating differences for the females compared to the males in network conditions with RTT with packet losses. However, the sample size was low with only 5 female participants compared to 18 male participants. The MOS of the QoE results separating play time under 10 hours per week and 10 or more hours per week showed no significant changes, where the largest QoE rating difference was 0.5 points. Network condition [rtt25pl12] and [rtt2pl35] had the largest differences in the MOS QoE ratings compared to the previous study, while both was not compared to the same corresponding network condition. The largest difference comparing the same network condition to the previous study was network condition [bj(rtt200i15)] with a difference of 1.1 points higher in the MOS QoE rating.

The long-term evolution (LTE) network has been proposed to provide better network capacity than the earlier 3G network. Driven by the market, the conventional LTE (3G) network standard could not achieve the expectations of the international mobile telecommunications advanced (IMT-Advanced) standard. To satisfy this gap, the LTE-Advanced was introduced with additional network functionalities to meet up with the IMT-Advanced Standard. In addition, due to the need to minimize operational expenditure (OPEX) and reduce human interventions, the wireless cellular networks are required to be self-aware, self-reconfigurable, self-adaptive and smart. An example of such network involves transceiver base stations (BTSs) within a self-organizing network (SON). Besides these great breakthroughs, the conventional LTE and LTE-Advanced networks have not been designed with the intelligence of scalable capacity output especially in sudden demographic changes, namely during events of football, malls, worship centres or during religious and cultural festivals. Since most of these events cannot be predicted, modern cellular networks must be scalable in terms of capacity and coverage in such unpredictable demographic surge. Thus, the use of dynamic BTSs is proposed to be used in modern and future cellular networks for crowd and demographic change managements. Dynamic BTSs are complements of the capability of SONs to search, determine and deploy less crowded/idle BTSs to densely crowded cells for scalable capacity management. The mobile BTSs will discover areas of dark coverages and fill-up the gap in terms of providing cellular services. The proposed network relieves the LTE network from overloading thus reducing packet loss, delay and improves fair load sharing. In order to trail the best (least) path, a bio-inspired optimization algorithm based on swarm-particle optimization is proposed over the dynamic BTS network. It uses the ant-colony optimization algorithm (ACOA) to find the least path. A comparison between an optimized path and the un-optimized path showed huge gain in terms of delay, fair load sharing and the percentage of packet loss.

This thesis presents an empirical analysis of the performance of WebTransport and WebSocket using the Socket.IO framework under varying network conditions. The thesis investigates two primary experiments, message round trip time and multiplexing performance, both conducted under three scenarios—perfect network (0% packet loss), moderate network (5% packet loss), and poor network (15% packet loss). The experiments were conducted on a system running Fedora 40 with a 12th Gen Intel Core i5-12600K processor and 16 GB of memory. The findings demonstrate that WebTransport mitigates Head-of-Line (HOL) blocking more effectively than WebSocket, especially in environments with degraded networks. WebTransport (Reliable) provided a significant performance advantage under suboptimal network conditions when measuring message round trip time, while WebTransport (Unreliable) consistently demonstrated slightly lower latency but with some data loss. When multiplexing, WebTransport showed significant performance benefits compared to using WebSocket.

This thesis explores various Wi-Fi redundancy methods to enhance network resilience in industrial settings. Maintaining uninterrupted data transmission is crucial due to the increasing reliance on wireless technologies for industrial operations. Our research investigates the performance of different redundancy strategies, including Multi-Link Operation (MLO) and Truncated Automatic Repeat Request (TARQ), through practical prototyping and testing on specific hardware configurations. We aim to assess these methods' effectiveness in mitigating packet loss and improving transmission consistency under varying attenuation. The findings are expected to provide valuable insights into the potential of existing redundancy methods to improve Wi-Fi network robustness. This thesis evaluates two main redundancy methods: MLO and TARQ. In our findings, MLO demonstrated improved packet transmission consistency at lower levels of signal attenuation compared to standard implementations. TARQ, particularly on the 2.4 GHz band, significantly reduced packet loss across a wide range of attenuation levels, thus showing its potential to enhance network reliability.