MMS-enabled terminals on the market today are very complicated to use. It takes several steps to create a multi-slide MMS-message with images and text. This discourages users from using it. To increase usage of MMS, several companies provide web-based or stand-alone programs that allow users to create and send MMS-messages from a regular computer. However these editors have many limitations and are not user-friendly. This thesis describes the design and implementation of a user-friendly web-based MMS-portal where users can create, edit and send MMS-messages. The portal is integrated into Densitet’s system for development of mobile services. Conclusions that can be draw from this work are that problems with MMS interoperability have mostly the poor standardization to blame. Different terminals support different types of images and sound formats, and to make the MMS-portal user-friendly, format conversions of uploaded content had to be implemented. Also the MMS-portal only supports basic MMS-functionality. If the MMS-specification includes more audio and image formats and if the MMS-terminals are upgraded to handle these formats, sending MMS-messages will be easier and mobile messaging will continue to grow.

ireless Communication is one of the fields of Telecommunications which is growing with the tremendous speed. With the passage of time wireless communication devices are becoming more and more common. It is not only the technology of business but now people are using it to perform their daily tasks, be it for calling, shopping, checking their emails or transfer their money. Wireless communication devices include cellular phones, cordless phones and satellite phones, smart phones like Personal Digital Assistants (PDA), two way pagers, and lots of their devices are on their way to improve this wireless world. In order to establish two way communications, a wireless link may be using radio waves or Infrared light. The Wireless communication technologies have become increasingly popular in our everyday life. The hand held devices like Personal Digital Assistants (PDA) allow the users to access calendars, mails, addresses, phone number lists and the internet. Personal digital assistants (PDA) and smart phones can store large amounts of data and connect to a broad spectrum of networks, making them as important and sensitive computing platforms as laptop PCs when it comes to an organization’s security plan. Today’s mobile devices offer many benefits to enterprises. Mobile phones, hand held computers and other wireless systems are becoming a tempting target for virus writers. Mobile devices are the new frontier for viruses, spam and other potential security threats. Most viruses, Trojans and worms have already been created that exploit vulnerabilities. With an increasing amount of information being sent through wireless channels, new threats are opening up. Viruses have been growing fast as handsets increasingly resemble small computers that connect with each other and the internet. Hackers have also discovered that many corporate wireless local area networks (WLAN) in major cities were not properly secured. Mobile phone operators say that it is only a matter of time before the wireless world is hit by the same sorts of viruses and worms that attack computer software.

During the recent last years, transport of multimedia sessions, such as audio streams and video conferences, over IP has acquired a lot of attention since most of communication technologies are migrating to work over IP. However, sending media streams over IP networks has encountered some problems related to signaling issues. The ongoing research in this area has produced some solutions to this subject. Internet Engineering Task Force (IETF) has introduced Session Initiation Protocol (SIP), which has proved to be an efficient protocol for controlling sessions over IP. While a great deal of research performed in evaluating the performance of SIP and comparing it with its competent protocols such as H.323, studying the delay caused by initiating the session has acquired less attention. In this document, we have addressed the SIP session setup delay problem. In the lab, we have built up a test bed for running several SIP session scenarios. Using different models for those scenarios, we have measured session setup delays for all used models. The analysis performed for each model showed that we could propose some models to be applied for SIP session setup delay components.

The purpose of this thesis project is to study and evaluate a UWB Synthetic Aperture Radar (SAR) data image formation algorithm, that was previously less familiar and, that has recently got much attention in this field. Certain properties of it made it acquire a status in radar signal processing branch. This is a fast time-domain algorithm named Local Backprojection (LBP). The LBP algorithm has been implemented for SAR image formation. The algorithm has been simulated in MATLAB using standard values of pertinent parameters. Later, an evaluation of the LBP algorithm has been performed and all the comments, estimation and judgment have been done on the basis of the resulting images. The LBP has also been compared with the basic time-domain algorithm Global Backprojection (GBP) with respect to the SAR images. The specialty of LBP algorithm is in its reduced computational load than in GBP. LBP is a two stage algorithm — it forms the beam first for a particular subimage and, in a later stage, forms the image of that subimage area. The signal data collected from the target is processed and backprojected locally for every subimage individually. This is the reason of naming it Local backprojection. After the formation of all subimages, these are arranged and combined coherently to form the full SAR image.

The aim of this thesis is to investigate the possibility to form high resolution Synthetic Aperture Radar (SAR) images using the Global Navigation Satellite System (GNSS) Galileo, GPS and Glonas, In particular the thesis study the GPS signal and evaluate its properties for bistatic case. The report is based on the fact that Galileo and GPS are both positioning systems with similar characteristics. The difference is mainly that Galileo System uses a larger number of satellites and a different modulation scheme to improve the efficiency of the system, resulting in a better accuracy. On the topic of GNSS SAR, the report will be described with modes, resolution, geometry and algorithms. It is also explained the Space Surface Bi-static Radar and within two particular cases: parallel and non parallel paths

Abstract: The channel characterization of a mobile satellite communication which is an important and fast growing arm of wireless communication plays an important role in the transmission of information through a propagation medium from the transmitter to the receiver with minimum barest error rate putting into consideration the channel impairments of different geographical locations like urban, suburban, rural and hilly. The information transmitted from satellite to mobile terminals suffers amplitude attenuation and phase variation which is caused by multipath fading and signal shadowing effects of the environment. These channel impairments are commonly described by three fading phenomena which are Rayleigh fading, Racian fading and Log-normal fading which characterizes signal propagation in different environments. They are mixed in different proportions by different researchers to form a model to describe a particular channel. In the thesis, the general overview of mobile satellite is conducted including the classification of satellite by orbits, the channel impairments, the advantages of mobile satellite communication over terrestrial. Some of the major existing statistical models used in describing different type of channels are looked into and the best out of them which is Lutz model [6] is implemented. By simulating the Lutz model which described all possible type of environments into two states which represent non-shadowed or LOS and shadowed or NLOS conditions, shows that the BER is predominantly affected by shadowing factor.

The advancements in broadband and mobile communication has given many privileges to the subscribers for instance high speed data connectivity, voice and video applications in economical rates with good quality of services. WiMAX is an eminent technology that provides broadband and IP connectivity on “last mile” scenario. It offers both line of sight and non-line of sight wireless communication. Orthogonal frequency division multiple access is used by WiMAX on its physical layer. Orthogonal frequency division multiple access uses adaptive modulation technique on the physical layer of WiMAX and it uses the concept of cyclic prefix that adds additional bits at the transmitter end. The signal is transmitted through the channel and it is received at the receiver end. Then the receiver removes these additional bits in order to minimize the inter symbol interference, to improve the bit error rate and to reduce the power spectrum. In our research work, we investigated the physical layer performance on the basis of bit error rate, signal to noise ratio, power spectral density and error probability. These parameters are discussed in two different models. The first model is a simple OFDM communication model without the cyclic prefix, while the second model includes cyclic prefix.

In this thesis, an unequal error protection scheme for transmitting JPEG2000 images over wireless channel is investigated. The rapid growth of wireless communication has resulted in a demand for robust transmission of compressed images over wireless networks. The challenge of robust transmission is to protect the compressed image data against the impairments of the radio channel, in such a way as to maximize the received image quality. However, for highly compressed images, it would be beneficial that regions of interest (ROI) are prioritized for interpretability. The thesis addresses this problem; investigating unequal error protection for transmitting JPEG2000 compressed images. More particularly, the results reported in this thesis provide guidance concerning the implementation of stronger error correction coding schemes (Golay code) for ROI and comparatively weaker coding (Hamming code) for non-ROI image spaces. Such unequal error protection can be utilized by the base station for transmitting JPEG2000-encoded images over next generation wireless networks.

With the increase use of portable devices such as Personal Digital Assistants (PDA), laptops, voice and data integrated cell phones and many more, there is a need of wireless communication method using air as the medium to transmit and receive information between terminals. Radio waves propagate from transmitting antenna and travel through free space undergoing reflections, diffractions and scattering. They are greatly affected by ground terrain, the atmosphere and the objects in their path like buildings, bridges, hills etc. Nowadays, the existence of a direct line of sight path between the transmitter and the receiver is unlikely. These multiple phenomena are responsible for most of the characteristic features like the quality of the received signal. In the above case propagation is mainly due to reflection and cattering from the buildings and by diffraction. So, in practice the transmitted signal arrives at the receiver via several paths with different time delays creating a multi path situation at the receiver, these multipath waves with randomly distributed amplitudes and phases combine to give a resultant signal that fluctuates in time and space. This phenomenon of random fluctuations in received signal level is termed as fading. The existing demodulation techniques like FM, AM will determine the signal from the received signal based on the mean distance method, which cannot provide the desired level of BER, which fails in proper estimation under high fading and high Doppler-Shift effect. SOLUTION: This project provides the implementation of an enhancement to the demodulation technique using Bayesian approach for the physical layer simulation of a General Packet Radio System (GPRS) considering variable Rician fading and variable Doppler-Shift effect for an AWGN channel. The system performance is evaluated based on Bit Error Rate (BER) and Signal to Noise Ratio (SNR) for the realized GPRS system. Matlab platform is used for the implementation, analysis of the proposed system with for functional verification in terms of BER and SNR. We have showed the comparative difference between the theoretical calculation of QPSK signal and to the values obtained by our program. The values show difference up to 0.4 db for a 1000 bit random vector. Moreover, we also compared with QAM demodulation technique in MATLAB code to show difference up to 1.4 db for a 1000 bit vector. These results signify better performance of the system as it has saved bandwidth.

Radio resources in wireless communication systems, implementing different multiple access techniques, must be wisely managed. This perspective is pivotal since the variations in propagation channel are very fast and the system is highly complex due to random and unpredictable movement of mobile users continuously. This complexity in the cellular system periodically contributes to different interference levels, high or low, resulting in the degradation of the system capacity. Transmitter power control is an efficient technique to mitigate the effect of interference under fading conditions, combat the Near-Far problem and conserve the battery life. Thus, an effective implementation of different power control algorithms in cellular radio communication systems can offer a significant improvement in the Quality of Service (QoS) to all the users. Choice of an appropriate power control algorithm is of prime importance, as it should aim at increasing the overall efficiency of the system. In this thesis different distributed power control algorithms, each suited for implementation under different cellular technologies, were studied extensively. Specifically, six distributed power control algorithms are compared through simulations on the basis of performance metrics like Carrier to Interference Ratio (CIR) and Outage for the downlink case. The work involves in finding the link gain matrix by modeling the cellular system in MATLAB and simulating different power control algorithms. The results obtained from the simulation work are used to evaluate the efficiency of the Distributed Power Control (DPC), Fully Distributed Power Control (FDPC), Improved Fully Distributed Power Control (FDPC+) and Balanced Distributed Power Control (BDPC) algorithms on the basis of convergence speed and at the same time evaluating the limitations of the different algorithms. Also, with the results obtained on the basis of outage comparison between Fixed Step Power Control (FSPC) and Augmented Constant Improvement Power Control (ACIPC) algorithms, the quality of active link protection and cell removal procedures are demonstrated.

Due to the complex diversity of contemporary Internet-services, computer network measurements have gained considerable interest during recent years. Since they supply network research, development and operations with data important for network traffic modelling, performance and trend analysis, etc. The quality of these measurements affect the results of these activities and thus the perception of the network and its services. This thesis contains a systematic investigation of computer network measurements and a comprehensive overview of factors influencing the quality of performance parameters obtained from computer network measurements. This is done using a novel network performance framework consisting of four modules: Generation, Measurement, Analysis and Visualization. These modules cover all major aspects controlling the quality of computer network measurements and thus the validity of all kinds of conclusions based on them. One major source of error is the timestamp accuracy obtained from measurement hardware and software. Therefore, a method is presented that estimates the timestamp accuracy obtained from measurement hardware and software. The method has been used to evaluate the timestamp accuracy of some commonly used hardware (Agilent J6800/J6830A and Endace DAG 3.5E) and software (Packet Capture Library). Furthermore, the influence of analysis on the quality of performance parameters is discussed. An example demonstrates how the quality of a performance metric (bitrate) is affected by different measurement tools and analysis methods. The thesis also contains performance evaluations of traffic generators, how accurately application-level measurements describe network behaviour, and of the quality of performance parameters obtained from PING and J-OWAMP. The major conclusion is that measurement systems and tools must be calibrated, verified and validated for the task of interest before using them for computer network measurements. A guideline is presented on how to obtain performance parameters at a desired quality level.

Due to the complex diversity of contemporary Internet applications, computer network measurements have gained considerable interest during the recent years. Since they supply network research, development and operations with data important for network traffic modelling, performance and trend analysis etc., the quality of these measurements affect the results of these activities and thus the perception of the network and its services. One major source of error is the timestamp accuracy obtained from measurement hardware and software. On this background, we present a method that can estimate the timestamp accuracy obtained from measurement hardware and software. The method is used to evaluate the timestamp accuracy of some commonly used measurement hardware and software. Results are presented for the Agilent J6800/J6830A measurement system, the Endace DAG 3.5E card, the Packet Capture Library (PCAP) either with PF_RING or Memory Mapping, and a RAW socket using either the kernel PDU timestamp (ioctl) or the CPU counter (TSC) to obtain timestamps.

In this paper we describe a distributed passive measurement infrastructure. Its goals are to reduce the cost and configuration effort per measurement. The infrastructure is scalable with regards to link speeds and measurement locations. A prototype is currently deployed at our university and a demo is online at http://inga.its.bth.se/projects/dpmi. The infrastructure differentiates between measurements and the analysis of measurements, this way the actual measurement equipment can focus on the practical issues of packet measurements. By using a modular approach the infrastructure can handle many different capturing devices. The infrastructure can also deal with the security and privacy aspects that might arise during measurements.

Quality of Service is major factor for a successful business in modern and future network services. A minimum level of services is assured indulging quality of Experience for modern real time communication introducing user satisfaction with perceived service quality. Traffic engineering can be applied to provide better services to maintain or enhance user satisfaction through reactive and preventive traffic control mechanisms. Preventive traffic control can be more effective to manage the network resources through admission control, scheduling, policing and traffic shaping mechanisms maintaining a minimum level before it get worse and affect user perception. Accuracy, dynamicity, uniformity and reproducibility are objectives of vast research in network traffic. Real time tests, simulation and network emulation are applied to test uniformity, accuracy, reproducibility and dynamicity. Network Emulation is performed over experimental network to test real time application, protocol and traffic parameters. DummyNet is a network emulator and traffic shaper which allows nondeterministic placement of packet losses, delays and bandwidth changes. KauNet shaper is a network emulator which creates traffic patterns and applies these patterns for exact deterministic placement of bit-errors, packet losses, delay changes and bandwidth changes. An evaluation of KauNet with different patterns for packet losses, delay changes and bandwidth changes on emulated environment is part of this work. The main motivation for this work is to check the possibility to delay and drop the packets of a transfer/session in the same way as it has happened before (during the observation period). This goal is achieved to some extent using KauNet but some issues with pattern repetitions are still needed to be solved to get better results. The idea of history and trace-based traffic shaping using KauNet is given to make this possibility a reality.

Fler och fler människor handlar på Internet idag. Det är därför inte konstigt att butikerna vill nå ut till dessa kunder också. Grandpa AB, är en liten butik i Stockholm som säljer framför allt kläder men även andra saker, är just en av dessa butikerna.

This application is created to provide valid and relevant travel information to travelers on their journey. The application is proposed as a demonstration for seamless communications in the VINNOVA-sponsored project PIITSA (Personal Information Intelligent Transport systems through Seamless communications and Autonomous decisions). The PIITSA-project is a cooperation between Blekinge Institute of Technology (BTH), Saab AB, SP Swedish National Testing and Research Institute and Swedish Road Administration region Stockholm. The project is going on under three years, and the goal is to define communication and decision functions which will make it easier for ITS-applications (Intelligent Transport Systems and Services) that have the need of supplying personalised information. This project presents a solution to the question “How do I get from point A to point B in any case”. All that the user needs is a PDA with the application installed and being able to connect to the service via a mobile network. If the user has specified a route in the application, then the user will receive up-to-date information on how to get to his destination. This information is presented on the user’s PDA. If something happens on the current route, then the user will get an automatic message sent to him presenting the problem and give him a new travel route. Thanks to this the user won’t be stuck in the traffic problem and will probably reach its destination in time.

The usage of Internet is rapidly increasing and a large part of the Internet traffic is generated by the World Wide Web (WWW) and the associated protocol HiperText Transfer Protocol (HTTP). Several important parameters that affect the performance of the WWW are bandwidth, scalability and latency. To tackle these parameters and to improve the overall performance of the system, it is important to understand and to characterize the application level characteristics. This article is reporting on the measurement and analysis of HTTP traffic collected on the student access network at the Blekinge Institute of Technology in Karlskrona, Sweden. The analysis is done on various HTTP traffic parameters, e.g., inter-session timings, inter-arrival timings, request message sizes, response code and number of transactions. The reported results can be useful for building synthetic workloads for simulation and benchmarking purposes.

In this paper we present a novel framework supporting distributed network management using a self-organizing peer-to-peer overlay network. The overlay consists of several Distributed Network Agents which can perform distributed tests and distributed monitoring for fault and performance management. In that way, the concept is able to overcome disadvantages that come along with a central management unit, like lack of scalability and reliability. So far, little attention has been payed to the quality of service experienced by the end user. Our self-organizing management overlay provides a reliable and scalable basis for distributed tests that incorporate the end user. The use of a distributed, self-organizing software will also reduce capital and operational expenditures of the operator since fewer entities have to be installed and operated.

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In recent years there has been a rapid growth in the number of wireless users, particularly in the area of mobile communications. This rapid growth in mobile communications demands for more system capacity through efficient utilization of frequency spectrum and also keeping the Interference as low as possible. In today’s Radio resource management, Adaptive array antennas have an important role in increasing the system capacity and controlling the interference in mobile communications. Adaptive array antenna is an antenna system, which uses spatially separated antennas called array antennas and processes the received signals with a digital signal processor. Simply speaking these array antennas can reduce the co channel interference and effectively utilize the bandwidth by steering a high gain in the direction of interest and low gains in the undesired directions, technically this is called adaptive beamforming. This adaptive beamforming enables the base station to form narrow beam towards desired user and nulls towards the interfering user, hence improving the signal quality.

SWOT analysis is performed on IMS security authentication schemes to demonstrate the effectiveness of each schemes in terms of multiple parameters and associated dependencies for users and network operators. Due to SWOT tool we have been able to describe strengths, weakness, opportunities and threats in each authentication scheme separately. The analysis performed mostly based on the state of art studies, SWOT tool itself approximates authentication schemes but in the discussion chapter we illustrated clearly what is necessary to adopt individually between Non SIM and SIM base authentication schemes. It will help MOBICOME project and network operators to choose the most appropriate authentication technology/ technologies for design and implementation.

Dagens teknik har bidragit till att mobiltelefonens användningsområden har expanderat avsevärt de senaste åren. Detta har lett till att alltfler mobila tjänster har dykt upp på marknaden. Uppgiften i detta examensarbete består i att utveckla en version anpassad för mobiltelefoner av den existerande produkten TelMe, som är en webbtjänst för vikarietillsättning inom olika verksamheter. Den mobilanpassade versionen kommer att gå under namnet TelMe Mobile. Problemställningen var att implementera de väsentliga funktionerna i produkten med avsikt för att göra dessa tillgängliga för mobiltelefoner. För att hitta en lösning till problemställningen har följande frågeställningar formulerats; hur kan webbtjänsten TelMe anpassas till en mobiltelefon, hur kan innehållet presenteras i en mobiltelefon på ett användarvänligt sätt, hur kan dataflödet reduceras samt hur implementeras de tekniska lösningarna för TelMe Mobile? Eftersom utveckling för mobiltelefoner kräver ett helt annat tankesätt, samlades information om mobila webbapplikationer från officiella organisationer. Begränsningar i mobiltelefoner som t.ex. skärmstorlek, bandbredd och förhållandevis underlägsen hårdvara innebär att webbapplikationen måste vara enkel och inte resurskrävande. Med hänsyn till ovanstående restriktioner måste TelMe designas om och följa denna enkla struktur. Webbapplikationen måste ha en enkelkolumnig layout på grund av mobiltelefoners relativt mindre skärmstorlekar. På grund av den begränsade bandbredden och minneskapaciteten för mobiltelefoner är det viktigt att datamängden inte är för stor. För att reducera datamängden för varje sida måste antalet rader med information minskas. Användning av bilder för layouten kan bidra till en ökning av datamängden, därför bör mindre eller inga bilder utnyttjas. Den tekniska lösningen för TelMe Mobile har implementerats som en separat sida för mobiltelefoner medan TelMe är avsedd för datorer. Implementeringen genomfördes med hjälp av XHTML Mobile Profile 1.0 och eFrog Builder. W3C:s riktlinjer för mobil webbutveckling har följts för att säkerställa TelMe Mobiles kompatibilitet med ett brett utbud av mobiltelefoner. Kompatibiliteten kan dock inte garanteras med alla mobiltelefoner på grund av den breda variationen av mobila plattformar. TelMe Mobile har testats på ett antal olika mobiltelefoner med de vanligast förekommande plattformarna. Webbapplikationen har även testats på Opera Mini, som är en populär tredjepartswebbläsare med bra stöd för den mobila webben. Mobilvänligheten för TelMe Mobile har validerats på http://ready.mobi, vilket kontrollerar webbapplikationen gentemot W3C:s standarder för mobil webbutveckling.

The main goals of the paper are towards an understanding of the delay process in best-effort Internet for both non-congested and congested networks. A dedicated measurement system is re-ported for delay measurements in IP routers, which follows specifications of the IETF RFC 2679. The system is using both passive measurements and active probing. Dedicated application-layer software is used to generate UDP traffic with TCP-like characteristics. Pareto traffic models are used to generate self-similar traffic in the link. The reported results are in the form of several impor-tant statistics regarding processing delay of a router, router delay for a single data flow, router delay for more data flows as well as end-to-end delay for a chain of routers. We confirm results reported earlier about the fact that the delay in IP routers is generally influenced by traffic characteristics, link conditions and, at some extent, details in hardware implementation and different IOS releases. The delay in IP routers usually shows heavy-tailed characteristics. It may also occasionally show extreme values, which are due to improper functioning of the routers.

Ethernet is one of the most common link layer technologies, used in local area networks, wireless networks and wide area networks. There is however a lack of traffic models for Ethernet that is usable in performance analysis. In this paper we use such a model. The model operates on matching multiple moments of the bit rate at several timescales. In order to match the model parameters to measured traffic, five methods have been developed. We use this to model three different links; the BCpOct89 Bellcore trace, an Internet access link and an ADSL link. Our results show that, as the number of sources present on an Ethernet link grows, the model becomes better and less complex.

The mobile operators are seeking for opportunities to create differentiation and increase profit. One powerful way is to provide personalized mobile services. A good example of personalisation is by location. Services based on position are called Location Based Services – LBS. To realise LBS, some sort of positioning method is needed. The two most common positioning methods today are Global Positioning System - GPS and network based positioning. GPS is not fully suited for LBS because you need an additional handset to receive the satellite signals. In network positioning however, you only need a mobile phone, but on the other hand, the accuracy is far less, only between 100 metres up to several kilometres. What technology would be a good positioning technology for location based services? Could A-GPS be such technology? A-GPS is a positioning system which uses the same satellites as GPS, but besides that, it also uses a reference network. The reference network tracks the receiver and the satellites. It also makes some of the heavy calculations that the handsets are doing in the GPS system. That makes the A-GPS receivers less power consuming and more suited to be implemented into mobile phones. Furthermore, A-GPS are more sensitive, meaning that it easier can receive signals when using indoor, for example. The question is if A-GPS technology holds its promises? Does A-GPS really work well in mobile phones? Is the accuracy and availability as good as the theory says and is it possible to implement an own, well working, location based service into an A-GPS mobile phone?

This report describes two important investigations which formed part of an overall project aimed at separating overlapping speech signals. The first investigation uses chirp signals to measure the acoustic transfer functions which would typically be found in the speaker separation project. It explains the behaviour of chirps in acoustic environments that can be further used to find the room reverberations as well, besides their relevance to measuring the transfer functions in conjunction with speaker separation. Chirps that have been used in this part are logarithmic and linear chirps. They have different lengths and are analysed in two different acoustic environments. Major findings are obtained in comparative analysis of different chirps in terms of their cross-correlations, specgrams and power spectrum magnitude. The second investigation deals with using automatic speech recognition (ASR) system to test the performance of the speaker separation algorithm with respect to word accuracy of different speakers. Speakers were speaking in two different scenarios and these were nonoverlapping and overlapping scenarios. In non-overlapping scenario speakers were speaking alone and in overlapping scenario two speakers were speaking simultaneously. To improve the performance of speaker separation in the overlapping scenario, I was working very close with my fellow colleague Mr. Holfeld who was improving the existing speech separation algorithm. After cross-examining our findings, we improved the existing speech separation algorithm. This further led to improvement in word accuracy of the speech recognition software in overlapping scenario.

UMTS is an emerging cell phone technology and it is basically another name for 3G mobile communication systems. It provides an enhanced range of multimedia services like video conferencing and high speed internet access. Sometimes UMTS is marketed as 3GSM emphasizing the both 3G nature and GSM standards which it was designed to succeed. UMTS is also European term for wireless systems based on the IMT-2000 standards. To utilize various merits in mobile telecommunication system which consist of various radio access networks, UMTS as Third Generation wireless technology utilizes a wideband CDMA or TD/CDMA transceiver and also cover large area. Handover is basically a function which continues the communication between users without any gaps when the hardware goes to a place where it finds no network coverage. When we talk in terms of cellular communications systems, handover is a process which is referred to the transfer of a connection from one cell to another. Handover time is generally between 200 and 1,200 milliseconds (ms), which accounts for the delay. In this thesis we are going to find the reasons for these factors which affect the Quality of service of handover. The main focus of this research is to study the some factors which really affect the handover phenomenon in UMTS that basically affect the overall quality of mobile network. For this we intend to find the solution for problems which born during the handover. Handover provides the mobility to users which are the main theme of wireless technology and it is also make the interoperability between different network technologies.

As Wireless ad-hoc Networks are becoming an increasingly emerging technology taking a centre stage in exciting research areas which has attracted many research attentions in communication systems, the need for efficient intrusion detection techniques to enhance network security and privacy has gained significant prominence. Intrusion detection systems(IDS) in ad- hoc networks provides an effective method to improve the security, energy efficiency of transmission nodes allowing better spatial reuse of the wireless channel and control over network resources in order to optimize node security and the transmit power thereby maintaining network connectivity. However these global properties have been falsified due to lack of appropriate intrusion detection system leading to severe network failures which is the bane of next generation ad-hoc networks. In this thesis, we investigate few wireless ad-hoc security attacks and vulnerabilities relative to topology control schemes and evaluate their performance under hostile environments. We propose a novel Distributed Intrusion Detection System (DIDS) that incorporates rule- based cluster topology relevant to both Wireless sensor networks (WSNs) and Mobile ad hoc networks (MANETs) to determine their security/performance in application- specific environments. Our DIDS draws inferences of intrusion by comparing anomalous patterns from packet traces of transmit and receive signal powers, ratio of packet arrival rates and anomaly in radio receiver packet power thresholds using buffer window count. Hence we evaluate our intrusion detection mechanism on a jammer attack and observe the effect on the network throughput. Our approach is simulated using the OPNET® simulator. Simulation results show that the detection capabilities of our scheme under a denial of service (DoS) (jammer) attack, increases the bit error rates, increase in transmit delay responses and considerable decrease in both the signal to noise powers and the average network throughput due to the presence of jammer attack which forms the baseline for our analysis required to maintain energy efficiency and improve security in ad hoc network.

Wireless and Mobile Internet have changed the way people and businesses operate. Communication from any Internet access point, including wireless networks such as UMTS, GPRS or WLAN has enabled organizations to have a mobile workforce. However, networked applications such as web, email, streaming multimedia etc. rely upon the ability of timely data delivery. The achievable throughput is a quality measure for the very task of a communication system, which is to transport data in time. Throughput is thus one of the most essential enablers for networked applications. While in general, throughput is defined on network or transport level, the application-perceived throughput reflects the Quality of Service from the viewpoints of the application and user. The focus of the thesis is on the influence of the network on the applicationperceived Quality of Service and thus the user perceived experience. An analysis of application based active measurements mimicking the needs of streaming applications is presented. The results reveal clear influence of the network on the application-perceived Quality of Service seen from variations of application-perceived throughput on small time scales. Results also indicate that applications have to cope with considerably large jitter when trying to use the nominal throughputs. It was observed that the GPRS network had considerable problems in delivering packets in the downstream direction even when the nominal capacity of the link was not reached. Finally, the thesis discusses the suitability of wireless networks for different mobile services, since the influence of the network on the application-perceived Quality of Service is of great significance when it comes to customer satisfaction. Therefore, application-perceived Quality of Service in wireless networks must also be considered by the mobile application programmer during the application development.

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Communication services are expected to deliver the desired information in a timely manner without challenging the user’s patience. The corresponding offers by networks are reflected in user- or applications-perceived throughput, which is measured end-to-end for different mobile and wireless networks, namely Digital Audio Broadcast (DAB), General Packet Radio Service (GPRS), Universal Mobile Telecommunication Systems (UMTS), and Wireless Local Area Networks (WLAN). The measurements are based on passive monitoring of throughput statistics for active test traffic and reveal amongst others different kinds of changes of traffic burstiness induced by mobile up- and downlinks and by buffering, respectively. The results will be used for seamless communication, i.e. automatic network selection within the project PIITSA (Personal Information for Intelligent Transport Systems through Seamless communications and Autonomous decisions) aiming at maintaining user-perceived performance and security at the desired levels while keeping communication cost as small as possible.

Wideband Code Division Multiple Access (WCDMA) is one of the main technologies for the implementation of third-generation (3G) cellular systems. It is based on radio access technique proposed by ETSI Alpha group and the specifications was finalised 1999. WCDMA is also known as UMTS and has been adopted as a standard by the ITU under the name “IMT-2000 direct spread”. The implementation of WCDMA will be a technical challenge because of its complexity and versatility. The complexity of WCDMA systems can be viewed from different angles: the complexity of each single algorithm, the complexity of the overall system and the computational complexity of a receiver. In WCDMA interface different users can simultaneously transmit at different data rates and data rates can even vary in time. WCDMA increases data transmission rates in GSM systems by using the CDMA air interface instead of TDMA. WCDMA is based on CDMA and is the technology used in UMTS. WCDMA is the dominating 3G technology, providing higher capacity for voice and data and higher data rates. The gradual evolution from today's systems is driven by demand for capacity, which is required by new and faster data based mobile services. WCDMA enables better use of available spectrum and more cost-efficient network solutions. The operator can gradually evolve from GSM to WCDMA, protecting investments by re-using the GSM core network and 2G/2.5G services. Orthogonal Frequency Division Multiplexing (OFDM) - technique for increasing the amount of information that can be carried over a wireless network uses an FDM modulation technique for transmitting large amounts of digital data over a radio wave. OFDM works by splitting the radio signal into multiple smaller sub-signals that are then transmitted simultaneously at different frequencies to the receiver. OFDM reduces the amount of crosstalk in signal transmissions. 802.11a WLAN, 802.16 and WiMAX technologies use OFDM. It's also used in the ETSI's HiperLAN/2 standard. In addition, Japan's Mobile Multimedia Access Communications (MMAC) WLAN broadband mobile technology uses OFDM. In frequency-division multiplexing, multiple signals, or carriers, are sent simultaneously over different frequencies between two points. However, FDM has an inherent problem: Wireless signals can travel multiple paths from transmitter to receiver (by bouncing off buildings, mountains and even passing airplanes); receivers can have trouble sorting all the resulting data out. Orthogonal FDM deals with this multipath problem by splitting carriers into smaller subcarriers, and then broadcasting those simultaneously. This reduces multipath distortion and reduces RF interference allowing for greater throughput. In this paper we have discussed about these two methods of third generation radio transmission system which are WCDMA and OFDM with various aspects. In between these two radio transmission technique, a better choice will be investigated.

Wideband Code Division Multiple Access (WCDMA) is one of the main technologies for the implementation of third-generation (3G) cellular systems. It is based on radio access technique proposed by ETSI Alpha group and the specifications was finalised 1999. WCDMA is also known as UMTS and has been adopted as a standard by the ITU under the name “IMT-2000 direct spread”. The implementation of WCDMA will be a technical challenge because of its complexity and versatility. The complexity of WCDMA systems can be viewed from different angles: the complexity of each single algorithm, the complexity of the overall system and the computational complexity of a receiver. In WCDMA interface different users can simultaneously transmit at different data rates and data rates can even vary in time. WCDMA increases data transmission rates in GSM systems by using the CDMA air interface instead of TDMA. WCDMA is based on CDMA and is the technology used in UMTS. WCDMA is the dominating 3G technology, providing higher capacity for voice and data and higher data rates. The gradual evolution from today's systems is driven by demand for capacity, which is required by new and faster data based mobile services. WCDMA enables better use of available spectrum and more cost-efficient network solutions. The operator can gradually evolve from GSM to WCDMA, protecting investments by re-using the GSM core network and 2G/2.5G services. Orthogonal Frequency Division Multiplexing (OFDM) - technique for increasing the amount of information that can be carried over a wireless network uses an FDM modulation technique for transmitting large amounts of digital data over a radio wave. OFDM works by splitting the radio signal into multiple smaller sub-signals that are then transmitted simultaneously at different frequencies to the receiver. OFDM reduces the amount of crosstalk in signal transmissions. 802.11a WLAN, 802.16 and WiMAX technologies use OFDM. It's also used in the ETSI's HiperLAN/2 standard. In addition, Japan's Mobile Multimedia Access Communications (MMAC) WLAN broadband mobile technology uses OFDM. In frequency-division multiplexing, multiple signals, or carriers, are sent simultaneously over different frequencies between two points. However, FDM has an inherent problem: Wireless signals can travel multiple paths from transmitter to receiver (by bouncing off buildings, mountains and even passing airplanes); receivers can have trouble sorting all the resulting data out. Orthogonal FDM deals with this multipath problem by splitting carriers into smaller subcarriers, and then broadcasting those simultaneously. This reduces multipath distortion and reduces RF interference allowing for greater throughput. In this paper we have discussed about these two methods of third generation radio transmission system which are WCDMA and OFDM with various aspects. In between these two radio transmission technique, a better choice will be investigated.

The main goals of this thesis are towards an understanding of the delay process in best-effort Internet for both non-congested and congested networks. A novel measurement system is reported for delay measurements in IP routers, which follows specifications of the IETF RFC 2679. The system employs both passive measurements and active probing and offers the possibility to measure and analyze different delay components of a router, e.g., packet processing delay, packet transmission time and queueing delay at the output link. Dedicated application-layer software is used to generate UDP traffic with TCP-like characteristics. Pareto traffic models are used to generate self-similar traffic in the link. The reported results are in form of several important statistics regarding processing and queueing delays of a router, router delay for a single data flow, router delay for multiple data flows as well as end-to-end delay for a chain of routers. They confirm results reported earlier about the fact that the delay in IP routers is generally influenced by traffic characteristics, link conditions and, to some extent, details in hardware implementation and different IOS releases. The delay in IP routers may also occasionally show extreme values, which are due to improper functioning of the routers. Furthermore, new results have been obtained that indicate that the delay in IP routers shows heavy-tailed characteristics, which can be well modeled with the help of several distributions, either in the form of a single distribution or as a mixture of distributions. There are several components contributing to the OWTT in routers, i.e., processing delay, queueing delay and service time. The obtained results have shown that, e.g., the processing delay in a router can be well modeled with the Normal distribution, and the queueing delay is well modeled with a mixture of Normal distribution for the body probability mass and Weibull distribution for the tail probability mass. Furthermore, OWTT has several component delays and it has been observed that the component delay distribution that is most dominant and heavy-tailed has a decisive influence on OWTT.

Today, the telecommunication industry is undergoing two important developments with implications on future architectural solutions. These are the irreversible move towards Internet Protocol (IP)-based networking and the deployment of broadband access. Taken together, these developments offer the opportunity for more advanced and more bandwidth-demanding multimedia applications and services, e. g., IP television (IPTV), Voice over IP (VoIP) and online gaming. A plethora of Quality of Service (QoS) requirements and facilities are associated with these applications, e. g., multicast facilities, high bandwidth and low delay/jitter. Moreover, the architectural solution must be a unified one, and be independent of the access network and content management. An interesting solution to these challenges is given by overlay multicast networks. The goal of these networks is to create and to maintain efficient multicast topologies among the multicast participants as well as to minimize the performance penalty involved with application layer multicasting. Since they operate at the application layer, they suffer from two main drawbacks: higher delay and less efficient bandwidth utilization. It is therefore important to assess the performance of overlay multicast networks in “real- world”-like conditions. For this purpose, we first performed an in-depth measurement and modeling study of the packet delay at the network layer. The reported results are in the form of several important statistics regarding processing and queueing delays of a router. New results have been obtained that indicate that the delay in IP routers shows heavy-tailed characteristics, which can be well modeled with the help of several distributions, in the form of a single distribution or as a mixture of distributions. There are several components contributing to the delay in routers, i. e., processing delay, queueing delay and service time. It was observed that the component delay distribution that is most heavy-tailed has a decisive influence on delay. Furthermore, we selected three representative categories of overlay multicast networks for study, namely Application Level Multicast Infrastructure (ALMI), Narada and NICE is the Internet Cooperative Environment (NICE). The performance of these overlay multicast protocols was evaluated through a comprehensive simulation study with reference to a detailed set of performance metrics that captured application and network level performance. A particular interest was given to the issues of scalability, protocol dynamics and delay optimization as part of a larger problem of performance-aware optimization of the overlay networks. The simulations were configured to emulate “real-world”-like characteristics by implementing a heavy-tailed delay at the network level and churn behavior of the overlay nodes. A detailed analysis of every protocol is provided with regard to their performance. Based on our study, significant conclusions can be drawn regarding the scalability of the protocols with reference to overlay multicast group management, resource usage and robustness to churn. These results contribute to a deeper understanding of the requirements for such protocols targeted at, e. g., media streaming.

This report is a contribution towards a better understanding of traffic measurements associated with e2e delays occurring in best-effort networks. We describe problems and solutions associated with OWTT delay measurements, and give examples of such measurements. A dedicated measurement system is reported for delay measurements in IP routers, which follows specifications of the IETF RFC 2679. The system uses both passive measurements and active probing. Dedicated application-layer software is used to generate traffic. Pareto traffic models are used to generate self-similar traffic in the link. Both packet inter-arrival times and packet sizes are matching real traffic models. A passive measurement system is used for data collection that is based on using several so-called Measurement Points, each of them equipped with DAG monitoring cards. Hashing is used for the identification and matching of packets. The combination of passive and active measurements, together with using the DAG monitoring system, gives us an unique possibility to perform precise traffic measurements as well as the flexibility needed to compensate for the lack of analytic solutions. The real value of our study lies in the hop-by-hop instrumentation of the devices involved in the transfer of IP packets. The mixture of passive and active traffic measurements used, allows us to study changes in traffic patterns relative to specific reference points and to observe different contributing factors to the observed changes. This approach offers us the choice of better understanding diverse components that may impact on the performance of packet delay as well as to to measure queueing delays in operational routers.

The paper reports on a dedicated measurement system for delay measurements in IP routers, which follows specifications of the IETF RFC 2679. The system uses both passive and active measurements. Dedicated application-layer software is used to generate traffic. Pareto traffic models are used to generate self-similar traffic in the link. Both packet inter-arrival times and packet sizes are matching real traffic models. A passive measurement system is used for data collection that is based on using several so-called Measurement Points, each of them equipped with DAG monitoring cards. Hashing is used for the identification and matching of packets. The combination of passive measurements and active probing, together with using the DAG monitoring system, gives us an unique possibility to perform precise traffic measurements as well as the flexibility needed to compensate for the lack of analytic solutions.

In recent years, Internet has known an unprecedented growth, which, in turn, has lead to an increased demand for real-time and multimedia applications that have high Quality-of-Service (QoS) demands. This evolution lead to difficult challenges for the Internet Service Providers (ISPs) to provide good QoS for their clients as well as for the ability to provide differentiated service subscriptions for those clients who are willing to pay more for value added services. Furthermore, a tremendous development of several types of overlay networks have recently emerged in the Internet. Overlay networks can be viewed as networks operating at an inter-domain level. The overlay hosts learn of each other and form loosely-coupled peer relationships. The major advantage of overlay networks is their ability to establish subsidiary topologies on top of the underlying network infrastructure acting as brokers between an application and the required network connectivity. Moreover, new services that cannot be implemented (or are not yet supported) in the existing network infrastructure are much easier to deploy in overlay networks. In this context, multicast overlay services have become a feasible solution for applications and services that need (or benefit from) multicast-based functionality. Nevertheless, multicast overlay networks need to address several issues related to efficient and scalable congestion control schemes to attain a widespread deployment and acceptance from both end-users and various service providers. This report aims at presenting an overview and taxonomy of current solutions proposed that provide congestion control in overlay multicast environments. The report describes several protocols and algorithms that are able to offer a reliable communication paradigm in unicast, multicast as well as multicast overlay environments. Further, several error control techniques and mechanisms operating in these environments are also presented. In addition, this report forms the basis for further research work on reliable and QoS-aware multicast overlay networks. The research work is part of a bigger research project, "Routing in Overlay Networks (ROVER)". The ROVER project was granted in 2006 by EuroNGI Network of Excellence (NoE) to the Dept. of Telecommunication Systems at Blekinge Institute of Technology (BTH).

The paper reports on a performance study of several Application Layer Multicast (ALM) protocols. Three categories of overlay multicast networks are investigated, namely Application Level Multicast Infrastructure (ALMI), Narada and NICE is the Internet Cooperative Environment (NICE). The performance of the overlay multicast protocols is evaluated with reference to a set of performance metrics that capture both application and network level performance. The study focuses on the control overhead induced by the protocols under study. This further relates to the scalability of the protocol with increasing number of multicast participants. In order to get a better assessment of the operation of these protocols in "real-life"-like conditions, we implemented in our simulations a heavy-tailed delay at the network level and churn behavior of the overlay nodes. Our performance study contributes to a deeper understanding and better assessment of the requirements for such protocols targeted at, e.g., media streaming.

Folkets Hus och Parker gav under våren 2008 en förfrågan om att under sommaren uppmäta den ljudtrycksdos en normal festivaldeltagare tillskansar sig under ett festivalbesök. De mätningar som genomfördes skedde på tre olika festivaler, Hultsfredsfestivalen, Peace & Love och Storsjöyran. Mätningarna skulle ske över hela festivalbesöket och inte endast innefatta mätningar av konserter, för att kunna ge en rättvisare bild av den ljudtrycksdos som en person kan tänkas utsättas för under en festival. De två av Socialstyrelsen uppsatta gränsvärdena som ska följas är att evenemanget ska hålla sig under ett medelvärde på max 100 dB(A)LEQ och dess toppvärde får vara högst 115 dB(A). I överlag kunde vi se att ljudnivåerna hölls relativt bra inom riktvärdena på de tre festivalerna. Resultaten för festivalerna var att mätningarna på Hultsfred gav ett ljudtrycksmedelvärde på 86,1 dB(A)LEQ, Peace & Love 88,8 dB(A)LEQ och Storsjöyran 85,7 dB(A)LEQ. Detta kan jämföras med arbetsmiljöverkets uppsatta gräns på 85 dB(A) för daglig bullerexponering. Viss forskning tyder dock på att musik och buller inte kan likställas och att människor normalt kan utsättas för 5 dB högre ljudnivåer av musik än buller, vilket i så fall skulle innebära att den dagliga gränsen för musikexponering skulle ligga på 90 dB(A).

Movies, songs, conferences, video streams and other multimedia services are spreading in everybody’s life, needs and work. Different types of audio and video codecs of various qualities appear to be used for different purposes. However, each codec can be suitable for a special usage, as for example a codec that can be use for high quality film may not be suitable for streaming it into network. The main purpose of this thesis is to compare the performance of different codecs in one particular scenario and indicate which codec performance is the best. The thesis work includes the investigation of several codec aspects (audio and video) and problems related to choosing a multimedia codec (coder/decoder) suitable for large-scale multimedia distribution over the Internet. More specifically, the work focuses on various features of modern media codecs, compares them and finds the best application for each of them. Furthermore, the thesis provides an overview of available research results related to this work and codec comparison tables for codecs and their features. Additionally, as part of this work, a testbed was developed to measure the performance of codecs when used to stream media over a network. The testbed enabled the collection of various QoS parameters, such as bandwidth usage and burst, size for each codec considered in this thesis.

IP Television (IPTV) and other media distribution applications are expected to be one of the next Internet killer applications. One indication of this is the corporate backing that the IP Multimedia Subsystem (IMS) is getting. However, the bandwidth utilization of these applications is still an issue, as the volume of multimedia grows due to larger image resolution and higher bitrate audio. One way of managing this increase in bandwidth requirements is to use existing endhost bandwidth to decrease the load on the content server in a Peer-to-Peer (P2P) fashion. One of the most successful P2P applications is BitTorrent (BT), a swarming file transfer system. This paper presents an implementation of a BT simulator intended for future use in investigating modifications to the BT system to provide streaming media capabilities. The simulator is validated against real-world measurements and a brief performance analysis is provided.

Network performance management is facing the challenge of provisioning advanced services with stringent delay and throughput requirements. For this reason, shortage of network capacity implying delay or loss, so-called bottlenecks, have to be identified and to be classified. The latter tasks imply the need for tractable analytical performance models. As we are particularly interested in the traffic flow properties, we focus on a discrete-time fluid flow traffic model. The fluid model considers averages of bit rates on small time scales. These averages can easily be measured, and thus, bit rate statistics for data streams entering and leaving a bottleneck can be obtained. The fluid flow model helps us to obtain analytical estimates on the impact of network entities on data streams and their statistics, i.e., efficiently predict the effects of bottlenecks on data streams. In this work we derive formulae for an iterative calculation of summary statistics such as the bit rate distribution and autocorrelation function. In addition, a corresponding simulation has been implemented, and the simulation results agree closely with the exact solutions. With the aid of the derived formulae we are able to match the model to observations.

Increasingly many work processes depend on Information and Communication Technology (ICT) services. These define the efficiency of how tasks are solved. The perceived usability of a service is heavily linked to the performance of the service, for instance the service responsiveness. Users do not like to wait; long response times (RTs) will interrupt their flow of thoughts and finally result in a loss of interest. Quality aspects, seen from the user’s point of view, have gained importance. While the rather traditional notion Quality of Service (QoS) is mostly related to technical quality parameters of the actual data transport, the more recently established notion of Quality of Experience (QoE) extends the notion of quality to include user perception and expectations. Obviously, QoE has strong subjective components and is also connected to the situation and context in which the user finds him- or herself. This work aims at finding quantitative relationships between QoE and QoS in order to provide application designers with means to adapt their application to available networks and the related conditions. We have particularly focused on a Geographical Information System (GIS) Map Service, with a web-based client-server application, used amongst others by professional users for tasks of urban planning. We have conducted user experiments to derive the relationship between user-perceived QoE, expressed qualitatively (through comments) and quantitatively (on the well-known scale from 1 = worst to 5 = best), and the response time, i.e. the time the user has to wait for the execution of a command. We have then established quantitative relationships between the response time and network problems such as losses and delays that have been generated in a controlled way using a traffic shaper between client and server.

Road User Charging (RUC) solutions based on the concept of thin clients rely upon timely delivery of Heavy Goods Vehicles (HGV) positions towards a RUC provider, which then performs the map matching and the calculation of the road user charges. In order to avoid data loss and late delivery, a proper dimensioning of the system is required. For the Swedish case, we investigate the feasibility of legacy mobile and wireless systems (in particular GPRS, UMTS and WLAN) for both streaming and bulk transfer of positioning data, and we determine sensible combinations of sampling and reporting intervals in terms of efficient and economic reporting. We also highlight a couple of traps regarding the dimensioning, in particular of sending patterns that might cause overload at the server side of the system. It becomes obvious that mobile networks can well cope with the extra load caused by reporting HGVs, while bulk transfer in wireless hot-spots can be used as a complementary solution.

Purpose - This paper aims to provide a solution for allowing an arbitrary number of weights and weight values to be used in an electronic voting system. Design/methodology/approach - Robust Electronic Voting System (REVS) was chosen, a voting system designed to support internet voting processes, as the start point for studying the introduction of vote weights. Findings - The work resulted in an internet voting system supporting vote weights, based on REVS. Research limitations/implications - In the solution presented, there is still a problem regarding anonymity, for the case of a single voter with a particular weighted vote. This is a fundamental problem, since it also exists in paper-based elections. Practical implications - This version of REVS has the same properties as the original REVS, apart from the anonymity issue above. Apart from this the process of registering voters got more complicated, since now it includes an extra action for binding a weight. Originality/value - To the best of the authors' knowledge, this modified version of REVS is the first electronic voting system supporting vote weights. Another novelty of this work is the modulus sharing optimization used for the RSA key pairs of each entity.

This paper addresses the evaluation of image quality in the context of wireless systems using feature-based objective metrics. The considered metrics comprise of a weighted combination of feature values that are used to quantify the extend by which the related artifacts are present in a processed image. In view of imaging applications in mobile radio and wireless communication systems, reduced-reference objective quality metrics are investigated for quantifying user-perceived quality. The examined feature-based objective metrics provide options for controlling the processing complexity and implementation resources required for performing the quality evaluation. Furthermore, the metrics support the scaling of overhead needed for communicating the image features over the wireless link and the accuracy for predicting the image quality.

In this paper, we present an on-going research project in which we are focusing on examining how users of map-based services on-line experience the quality of these services when the traffic load is high, and how the users' experiences of acceptable or not acceptable quality can be related to measurable parameters which can be used to manage network traffic and improve technical solutions. The project is a multi- and interdisciplinary project in cooperation between researchers within human work science and informatics, and researchers within telecommunication systems. Additionally, there are two external partners in the project: a provider of Internet-based map services, and a municipality which uses this provider's map services regularly. One of the main methodological issues addressed in the project is how laboratory based, quantitative research methods from research on Quality of Service in the telecommunication systems area can be related to qualitative research methods focusing on workplace- or other live-world based use situations and Quality of Expereince as defined by users of the services. How can experiments and studies be designed, and results shared, such that both network traffic measuring and evaluation of user experiences retain their own paradigmatic validity and relevance, while fruitfully informing service design?

The Internet has experienced two major revolutions. The first was the emergence of the World Wide Web, which catapulted the Internet from being a scientific and academic network to becoming part of the societal infrastructure. The second revolution was the appearance of the Peer-to-Peer (P2P) applications, spear-headed by Napster. The popularity of P2P networking has lead to a dramatic increase of the volume and complexity of the traffic generated by P2P applications. P2P traffic has recently been shown to amount to almost 80% of the total traffic in a high speed IP backbone link. One of the major contributors to this massive volume of traffic is BitTorrent, a P2P replication system. Studies have shown that BitTorrent traffic more than doubled during the first quarter of 2004, and still amounts to 60% of all P2P traffic in 2005. This thesis reports on measurement, modelling and analysis of BitTorrent traffic collected at Blekinge Institute of Technology (BTH) as well as at a local ISP. An application layer measurement infrastructure for P2P measurements developed at BTH is presented. Furthermore, a dedicated fitness assessment method to avoid issues with large sample spaces is described. New results regarding BitTorrent session and message characteristics are reported and models for several important characteristics are provided. Results show that several BitTorrent metrics such as session durations and sizes exhibit heavy-tail behaviour. Additionally, previously reported results on peer reactivity to new content are corroborated.

Large-scale, real-time multimedia distribution over the Internet has been the subject of research for a substantial amount of time. A large number of mechanisms, policies, methods, and schemes have been proposed for media coding, scheduling, and distribution. Internet Protocol (IP) multicast was expected to be the primary transport mechanism for this, though it was never deployed to the expected extent. Recent developments in overlay networks have reactualised the research on multicast, with the consequence that many of the previous mechanisms and schemes are being re-evaluated. This thesis provides a brief overview of several important techniques for media broadcasting and stream merging, as well as a discussion of traditional IP multicast and overlay multicast. Additionally, we propose a number of modifications and extensions to the BitTorrent (BT) distribution and replication system to make it suitable for use in providing a streaming video delivery service, and implement parts of these in a simulator. Also, we report on a simulation study of the implemented extensions to the BT system, as well as a detailed validation study of the BT simulator itself. Furthermore, we present a comprehensive set of BT models for several important traffic characteristics, at both session and message levels.