<p>With the constant development of wireless technologies, the usageof wireless devices tends to increase even more in the future.Wireless multi-hop networks (WMNs) have emerged as a keytechnology to numerous potential scenarios, ranging from disasterrecovery to wireless broadband internet access. The distributedarchitecture of WMNs enables nodes to cooperatively relay othernode's packets. Because of their advantages over other wirelessnetworks, WMNs are undergoing rapid progress and inspiringnumerous applications. However, many technical issues still existin this field. In this thesis we investigate how Voice over IP(VoIP) and peer-to-peer (P2P) application are influenced bywireless multi-hop network characteristics and how to optimizethem in order to provide scalable communication.We first consider the deployment of VoIP service in wirelessmulti-hop networks, by using the Session Initiation Protocol (SIP)architecture. Our investigation shows that the centralized SIParchitecture imposes several challenges when deployed in thedecentralized wireless multi-hop environment. We find that VoIPquality metrics are severely degraded as the traffic and number ofmultiple hops to the gateway increase. In the context ofscalability, we further propose four alternative approaches whichavoid current limitations.In the second part of this thesis we tackle the network capacityproblem while providing scalable VoIP service over wirelessmulti-hop networks. The performance evaluation shows the influenceof intra and inter-flow interference in channel utilization, whichdirect impacts the VoIP capacity. In order to avoid the small VoIPpacket overhead, we propose a new adaptive hop-by-hop packetaggregation scheme based on wireless link characteristics. Ourperformance evaluation shows that the proposed scheme can increasethe VoIP capacity by a two-fold gain.The study of peer-to-peer applicability over wireless multi-hopnetworks is another important contribution. A resource lookupapplication is realized through structured P2P overlay. We showthat due to several reasons, such as characteristics of wirelesslinks, multi-hop forwarding operation, and structured P2Pmanagement traffic aggressiveness the performance of traditionalP2P applications is rather low in wireless multi-hop environments.Therefore, we suggested that a trade-off between the P2P lookupefficiency and the P2P management traffic overhead can be achievedwhile maintaining the overlay network consistency in wirelessmulti-hop networks.</p>

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Θα περιγράψουμε τις βασικές λειτουργίες ενός VoIP δικτύου, τα συστατικά του μέρη, καθώς και τα πρωτόκολλα που είναι υπεύθυνα για την εγκατάσταση, τον έλεγχο και τον τερματισμό μιας VoIP υπηρεσίας-συνομιλίας. Τα πρωτόκολλα αυτά ονομάζονται πρωτόκολλα σηματοδοσίας. Τα πρωτόκολλα σηματοδοσίας για VoIP εφαρμογές και ιδιαίτερα το πρωτόκολλο SIP (Session Initiation Protocol) είναι το βασικό θέμα της παρούσας εργασίας. Συγκεκριμένα, έγινε ανάπτυξη ενός SIP User Agent, το λογισμικό του οποίου θα εκτελείται στο δικτυακό επεξεργαστή IXP425 της Intel, μαζί με τα απαραίτητα πρωτόκολλα για την κωδικοποίηση-αποκωδικοποίηση και μετάδοση δειγμάτων φωνής σε μορφή πακέτων δεδομένων. Το αποτέλεσμα αναμένεται να είναι ένα ολοκληρωμένο προϊόν (VoIP phone) για την πραγματοποίηση VoIP κλήσεων. / -

If not the most, one of the serious threats to data networks, particularly pervasive commercial networks such as Voice-over-IP (VoIP) providers is Denial-of-Service (DoS) attack. Currently, majority of solutions for these attacks focus on observing detailed server state changes due to any or some of the incoming messages. This approach however requires significant amount of server’s memory and processing time. This results in detectors not being able to scale up to the network edge points that receive millions of connections (requests) per second. To solve this problem, it is desirable to design stateless detection mechanisms. One approach is to aggregate transactions into groups. This research focuses on stateless scalable DoS intrusion detection mechanisms to obviate keeping detailed state for connections while maintaining acceptable efficiency. To this end, we adopt a two-layer aggregation scheme termed Advanced Partial Completion Filters (APCF), an intrusion detection model that defends against DoS attacks without tracking state information of each individual connection. Analytical as well as simulation analysis is performed on the proposed APCF. A simulation test bed has been implemented in OMNET++ and through simulations it is observed that APCF gained notable detection rates in terms of false positive and true positive detections, as opposed to its predecessor PCF. Although further study is needed to relate APCF adjustments to a certain network situation, this research shows invaluable gain to mitigate intrusion detection from not so scalable state-full mechanisms to aggregate scalable approach.

The rapid growth of the Internet and the proliferation of its new applications pose a serious challenge in network performance management and monitoring. The current Internet has no mechanism for providing feedback on network congestion to the end-systems at the IP layer. For applications and their end hosts, end-to-end measurements may be the only way of measuring network performance. Understanding the packet delay and loss behavior of the Internet is important for proper design of network algorithms such as routing and flow control algorithms, for the dimensioning of buffers and link capacity, and for choosing parameters in simulation and analytic studies. In this thesis, round trip time (RTT), one-way network delay and packet loss in the Internet are measured at different times of the day, using a Voice over IP (VoIP) device. The effect of clock skew on one-way network delay measurements is eliminated by a Linear Programming algorithm, implemented in MATLAB. Distributions of one-way network delay and RTT in the Internet are determined. It is observed that delay distribution has a gamma-like shape with heavy tail. It is tried to model delay distribution with gamma, lognormal and Weibull distributions. It is observed that most of the packet losses in the Internet are single packet losses. The effect of firewall on delay measurements is also observed.

Σκοπός της παρούσας διπλωματικής εργασίας ήταν η μελέτη των παραμέτρων που επηρεάζουν την ποιότητα εξυπηρέτησης (QoS – Quality of Service) των εφαρμογών VoIP μέσω των διαφόρων τεχνολογιών ευρυζωνικής πρόσβασης. / Aim of the present diplomatic assigment was the study of parameters that affect the quality of service (QoS) of VoIP applications via the various broadband access technologies.

Made available in DSpace on 2015-04-11T14:03:03Z (GMT). No. of bitstreams: 1 Fuad M Abinader Junior.pdf: 1012469 bytes, checksum: 83006b70c65e7b5ac0b070b259981f64 (MD5) Previous issue date: 2006-04-13 / The current wireless networks development scenario indicates that mobile VoIP applications are increasing their appeal among consumers, which creates an increasing demand for bandwidth consuption. However, bandwidth availability for VoIP applications is restricted by phisical and regulatory means. Header compression algorithms are one of the most used bandwidth optimization techniques for VoIP applications in wireless networks. This dissertation presents a performance evaluation of cooperative header compression algoritms for VoIP applications in wireless networks. The results indicate that the use of the single-channel cooperative approach leads to excelent results in terms of bandwidth optimization alied with robust context update. Also, the results indicate that the multi-channel cooperative approach has serious issues regarding parallel asyncrhonous VoIP connections. / O cenário atual do desenvolvimento das redes sem fio indica que o apelo por aplicações VoIP móveis está crescendo entre os consumidores, o que gera uma demanda cada vez maior de consumo de largura de banda. No entanto, a disponibilidade de largura de banda para aplicações VoIP é limitada tanto pelo meio físico quanto por regulamentações governamentais. O uso de algoritmos de compressão de cabeçalho é uma das técnicas mais usadas para otimização de largura de banda para aplicações VoIP em redes sem fio. Esta dissertação apresenta uma avaliação de desempenho de algoritmos de compressão de cabeçalhos cooperativos para aplicações VoIP em redes sem fio. Os resultados indicam que a utilização do algoritmo cooperativo mono-canal leva a excelentes resultados em termos de otimização de largura de banda com a manutenção das atualizações de contexto. Além disso, os resultados indicam que o uso do algoritmo cooperativo multi-canal possui sérias restrições quando utilizado em conjunto com conexões VoIP paralelas e assíncronas.

Made available in DSpace on 2016-08-17T14:52:53Z (GMT). No. of bitstreams: 1 Ricardo Henrique Bezerra Azoubel.pdf: 588389 bytes, checksum: 40c0c99f808c27d0a6967b9d18c9c22a (MD5) Previous issue date: 2004-09-24 / This work presents a platform of tests (testbed) inexpensive, constructed in a controlled environment composed by microcomputers and free softwares. It is implemented, in such platform, the differentiated service model (DiffServ), with expedited forwarding (PHB EF). It is basically considered, from the collection of metrics main of QoS (delay, jitter, loss and throughtput), the performance analysis of voice characteristic traffics, when submitted to experimental tests in some scenes and conditions. Initially, in an environment capable to differentiate traffics, flows generated by standardized voice coder/decoder (G.711 and G.726) are modeled, in which the packets size and the amount of aggregate flows are varied, in scenes with and without QoS. It is compared, after that, the behavior of flows generated by activity-silence (ON-OFF) and continuous (CBR) sources. Can be perceived in this study how much the packets size variation influence in the performance of the most priority packets. It is carried, in the sequence, a specific analysis of the aggregation factor in flows generated by ON-OFF sources, in which can be observed the action of the basic principle of the model DiffServ, where aggregate flows receive differentiated treatment. It is studied, finally, through the use of transport protocols (UDP and TCP) and of elastic flows of FTP type, how much the best effort traffic confuses the performance of voice modeled flows. / Este trabalho apresenta uma plataforma de testes (testbed) sem custos, construída num ambiente controlado composto por microcomputadores e softwares livres. Implementa-se, em tal plataforma, o modelo de serviço diferenciado (DiffServ), com encaminhamento expresso (PHB EF). Propõe-se, fundamentalmente, a partir da obtenção das principais métricas de QoS (atraso, jitter, perda e vazão), a análise do desempenho de tráfego característico de voz, quando submetido a testes experimentais em vários cenários e condições. Inicialmente, num ambiente capaz de diferenciar tráfego, modelam-se fluxos gerados por codificadores/decodificadores de voz padronizados (G.711 e G.726), em que se varia o tamanho dos pacotes e a quantidade de fluxos agregados, em cenários com e sem QoS. Compara-se, em seguida, o comportamento de fluxos gerados por fontes atividadesilêncio (ON-OFF) e contínuas (CBR). Pode-se perceber nesse estudo o quanto a variação do tamanho dos pacotes impacta no desempenho dos pacotes mais prioritários. Realiza-se, na seqüência, uma análise específica do fator agregação em fluxos gerados por fontes ONOFF e observa-se a atuação do princípio básico do modelo DiffServ, onde fluxos agregados recebem tratamento diferenciado. Estuda-se, por fim, através da utilização de protocolos de transporte (UDP e TCP) e de fluxos elásticos do tipo FTP, o quanto o tráfego de melhor esforço impacta no desempenho de fluxos modelados de voz.

Multiprotocol Label Switching (MPLS) is a new paradigm in routing architectures which has changed the way Internet Protocol (IP) packet is transferred in a Network. MPLS ensures the reliability of the communication minimizing the delays and enhancing the speed of packet transfer. One important feature of MPLS is its capability of providing Traffic Engineering (TE) which plays a vital role for minimizing the congestion by efficient load, balancing and management of the network resources. The performance evaluation is done considering the network parameters latency, jitter, packet end to end delay, and packet delay variation. Integration of QoS with the MPLS-TE network may enhance the performance of the network. Various scheduling algorithms can be used for implementing QoS on a network, which may vary the performance of the network. In our study, QoS is implemented on top of the MPLS-TE network using Differentiated Service (DiffServ) architecture. Different basic scheduling algorithms are used for the implementation of QoS and to check their impact on the network and to identify the suitable one among them. Performance evaluation is done considering the network parameters latency, jitter, packet end-to-end delay, and Packet Delay Variation. The simulation was done using OPNET modeler 16.0 and the results were analyzed. The simulation result shows that using TE along with QoS in MPLS network decreases the latency, jitter, packet delay variation and end to end packet delay compared to using TE alone for voice traffic. / +46738732963

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