<p>The trends in todays tele- and datacommunication market point toward using IP for all sorts of service delivery ranging from voice calls to TV. The next natural step in this evolution is to provide the same set of services to the end users independent of the access technology and device used. The IP Multimedia Subsystem (IMS) is an IP based telecommunications platform which targets this and lets the operators develop new services once which can then be used on many different devices.</p><p>This thesis examines the integration of IPTV and IMS. Can IMS be used to deliver TV services and can the IPTV set-top-boxes of today be used as clients in IMS? Since this is a new and previously unexamined area an explorative approach is taken. The aim is to identify how such an integration could be performed and the possible problems which have to be solved. To assist in this exploration a TV-push service based on IMS technology is constructed.</p><p>Based on the experiences from this service a general architecture for IPTV in IMS is suggested.</p><p>A number of problems crucial to solve for a successful integration are identified and possible solutions to these are discussed.</p>

<p>Voice over Internet Protocol (VoIP) has become very popular in recent days andbecome the first choice of small to medium companies for voice and data integration inorder to cut down the cost and use the IT resources in much more efficient way. Anotherpopular technology that is ruling the world after the year 2000 is 802.11 wirelessnetworks. The Organization wants to implement the VoIP on the wireless network. Thewireless medium has different nature and requirement than the 802.3 (Ethernet) andspecial consideration take into account while implementing the VoIP over wirelessnetwork.One of the major differences between 802.11 and 802.3 is the bandwidthavailability. When we implement the VoIP over 802.11, we must use the availablebandwidth is an efficient way that the VoIP application use as less bandwidth as possiblewhile retaining the good voice quality. In our project, we evaluated the differentcompression and decompression (CODEC) schemes over the wireless network for VoIP.To conduct this test we used two computers for comparing and evaluatingperformance between different CODEC. One dedicated system is used as Asterisk server,which is open source PBX software that is ready to use for main stream VoIPimplementation. Our main focus was on the end-to-end delay, jitter and packet loss forVoIP transmission for different CODECs under the different circumstances in thewireless network. The study also analyzed the VoIP codec selection based on the MeanOpinion Score (MOS) delivered by the softphone. In the end, we made a comparisonbetween all the proposed CODECs based on all the results and suggested the one Codecthat performs well in wireless network.</p>

The Internet grows extremely fast in terms of number of users and traffic volume, as well as in the number of services that must be supported. This development results in new requirements on routers—the main building blocks of the Internet. Existing router designs suffer from architectural limitations that make it difficult to meet future requirements, and the purpose of this thesis is to explore new ways of building routers. We take the approach to investigate distributed and modular router designs, where routers are composed of multiple modules that can be mapped onto different processing elements. The modules communicate through open well-defined interfaces over an internal network. Our overall hypothesis is that such a combination of modularization and decentralization is a promising way to improve scalability, flexibility, and robustness of Internet routers—properties that will be critical for new generations of routers. Our research methodology is based on design, implementation, and experimental verification. The design work has two main results: an overall system design and a distributed router control plane. The system design consists of interfaces, protocols, and internal mechanisms for physically separation of different components of a router. The distributed control plane is a decomposition of control software into independent modules mapped onto multiple distributed processing elements. Our design is evaluated and verified through the implementation of a prototype system. The experimental part of the work deals with two key issues. First, transport mechanisms for communication of internal control information between processing elements are evaluated. In particular, we investigate the use of reliable multicast protocols in this context. Results regarding communication overhead as well as overall performance of routing table dissemination and installation are presented. The results show that even though there are certain costs associated with using reliable multicast, there are large performance gains to be made when the number of processing elements increases. Second, we present performance results of processing routing information in a distributed control plane. These results show that the processing time can be significantly reduced by distributing the workload over multiple processing elements. This indicates that considerable performance improvements can be made through the use of the distributed control plane architecture proposed in this thesis. / QC 20100616

Det här arbetet handlar om att ta fram beteendemodeller av analog till digital omvandlare avsedda för tillämpningar i radiofrekvensområdet. Det gäller tillämpningar inom telekommunikation men även in test- och mätinstrument där omvandlingen från analoga till digitala signaler ofta är en prestandamässig flaskhals. Modellerna är avsedda att användas för att efterbehandla utdata från omvandlaren och på så sätt förbättra prestanda på den digitala signalen. Genom att skapa modeller av verkliga omvandlare och hur dessa avviker från ett idealt beteende kan ofullständigheter korrigeras genom så kallad postkorrigering. Beteendemodeller innebär att genererar en lämplig insignal, mäta utdata och beräkna en modell. För omvandlare i radiofrekvensområdet ställs höga krav på instrumentering. Den testutrustningen som används är baserad på moderna högprestanda instrument som har kompletterats med specialbyggd utrustning för signalkonditionering och datainsamling. I avhandlingen har även olika insignaler utvärderats med såväl teoretisk som experimentell analys. Det finns ett flertal olika varianter av modeller för att modulera ett olinjär, dynamisk system. För att få en parametereffektiv modell har utgångspunkten varit att utgå från en Volterramodell som på ett optimalt sätt beskriver svagt olinjära dynamiska system, så som analog till digital omvandlare, men som är alltför omfattande i antal parametrar. Volterramodellens har sedan reducerats till en mindre parameterintensiv, modellerstruktur på så sätt att Volterrakärnans symmetriegenskaper jämförts med symmetrierna hos andra modeller. En alternativ metod är att använda en Kautz-Volterramodell. Den har samma generella egenskaper som Volterramodellen, men är inte lika parameterkrävande. I den här avhandlingen redovisas experimentella resultat av Kautz-Volterramodellen som i framtiden kommer att vara intressanta att använda för postkorrigeringen. För att kunna beskriva beteenden som en dynamiska olinjära modellen inte klarar av har modellen kompletterats med en statisk styckvis linjär modellkomponent. I avhandlingen presenteras en sluten lösning för att identifiera samtliga paramervärden i modellen. Vidare har det i avhandlingen genomförs en analys av hur respektive komponent påverkar prestanda på utsignalen. Därigenom erhålls ett mått på den maximala prestandaförbättring som kan uppnås om felet kan elimineras. / This work considers behavior modeling of analog to digital converters with applications in the radio frequency range, including the field of telecommunication as well as test and measurement instrumentation, where the conversion from analog to digital signals often is a bottleneck in performance. The models are intended to post-process output data from the converter and thereby improve the performance of the digital signal. By building a model of practical converters and the way in which they deviate from ideal, imperfections can be corrected using post-correction methods. Behavior modeling implies generation of a suitable stimulus, capturing the output data, and characterizing a model. The demands on the test setup are high for converters in the radio frequency range. The test-bed used in this thesis is composed of commercial state-of-the-art instruments and components designed for signal conditioning and signal capture. Further, in this thesis, different stimuli are evaluated, theoretically as well as experimentally. There are a large number of available model structures for dynamic nonlinear systems. In order to achieve a parameter efficient model structure, a Volterra model was used as a starting-point, which can describe any weak nonlinear system with fading memory, such as analog to digital converters. However, it requires a large number of coefficients; for this reason the Volterra model was reduced to a model structure with fewer parameters, by comparing the symmetry properties of the Volterra kernels with the symmetries from other models. An alternative method is the Kautz-Volterra model, which has the same general properties as the Volterra model, but with fewer parameters. This thesis gives experimental results of the Kautz-Volterra model, which will be interesting to apply in a post-correction algorithm in the future. To cover behavior not explained by the dynamic nonlinear model, a complementary piecewise linear model component is added. In this thesis, a closed form solution to the estimation problem for both these model components is given. By gradually correcting for each component the performance will improve step by step. In this thesis, the relation between a given component and the performance of the converter is given, as well as potential for improvement of an optimal post-correction. / QC 20100629

This dissertation addresses the problem of providing a®ordable mobile mul-timedia services in wide area wireless networks. The approach is to con-sider novel system architectures, based on reusing and sharing of the ex-isting network infrastructure for cellular and terrestrial TV broadcastingsystems. The focus has been on the radio resource management techniques,and the evaluation of the potential cost savings, compared to traditionalevolution tracks of the cellular and broadcasting systems.The studies show that deployment cost of a wide area broadcastingnetwork, using DVB-H technology, is very large if high data rate and fullarea coverage is targeted. For this reason we propose to avoid the broad-casting infrastructure dimensioning for full area coverage, and use insteadthe cellular systems to enable error correction for broadcasting transmis-sions. For the special case of mobile users, the chosen approach is to tradesystem's cost and capacity for improved perceived coverage. This trade-o®is enabled by the use of application layer forward error correction, usingRaptor coding.The general purpose Ambient Networks technology was chosen to en-able a platform for inter-operability between cellular and broadcasting sys-tems, especially the necessary interfaces. Under the Ambient Networksframework, we investigate the achievable cost savings o®ered by a hybridcellular-broadcasting system when combinations of broadcast and point-to-point transmissions are jointly utilized to provide ¯le transfers and stream-ing services. Two cases were investigated: one where the cellular systemacts as a replacement and deliver the data in the areas where broadcastingtransmissions cannot reach, and another one where cellular system carriesparity data to users that experience temporary outage in the broadcastingsystem. The results are encouraging, as they show that ¯le transfer costcan be reduced by more than 50%, but only under certain conditions.On a short term, hybrid cellular-broadcasting systems based on 3G andDVB-H, o®er a good platform for testing new and innovative mobile TVservices, enriched with interactivity and content personalization. From atechnical perspective, the outcomes of the presented studies indicate thatfuture systems built on hybrid cellular-broadcasting infrastructures are ableprovide a long term and cost e±cient solution for delivery of a®ordablebroadband multimedia services to mobile users. However, today's cellularand broadcasting systems live in di®erent worlds, are driven by di®erentrevenue models, and they are now starting to compete, instead of cooperate,for controlling the multimedia delivery channels to mobile users. / QC 20100708

Future wireless networks will combine multiple radio technologies and subsystems, possibly managed by competing network providers. For such systems it may be advantageous to let the end nodes (terminals) make some or all of the resource management decisions. In addition to reducing complexity and costs, increasing redundancy, and facilitating more timely decisions; distributed resource sharing regimes can decouple the individual subsystems. Decoupled subsystems could be desirable both because competing operators can be business-wise separated and because it allows new technologies to be added (removed) in a modular fashion. However, distributed regimes can also lead to “selfish” wireless nodes who only try to maximize their own performance. The first part of this dissertation studies if selfish nodes can make efficient use of wireless resources, using multiaccess and network layers as examples. The related problems are formulated as noncooperative games between nodes. To maintain tractability nodes are confined to simple strategies that neither account for future payoffs nor allow for coordination. Yet, it is demonstrated that selfish nodes can achieve comparable performance to traditional protocols. These results should be interpreted as an argument in favor of distributed regimes. The second part of this dissertation evaluates the effects of multi-provider network architectures where users can roam freely across all networks. From a supply side perspective the benefits are improved path gain statistics and the fact that different networks may have non-overlapping busy hours. Several network configurations are analyzed and it is shown that cooperation between symmetric providers can yield significant capacity gains for both downlink and uplink; even if the providers have nearly collocated sites. When the providers have different site densities the gains from cooperation are reduced and the provider with a sparse network always gains more from cooperating. This suggests that initially, voluntary cooperation may be limited to some special cases. Lastly, the architecture is analyzed in a context where the providers compete for users on a per session basis by offering access at different prices. Although such architectures currently only exist in a few special cases, they could emerge in domestic markets where the costs to switch and search for new networks are low. Based on a game theoretic formulation it is shown that a competitive market for wireless access can be advantageous for both users and providers. The results presented suggest that the advantages of cooperation of competing providers occur in more than just a few cases. / QC 20100812

The problem of detecting a vector of symbols, drawn from a finite alphabet and transmitted over a multiple-input multiple-output (MIMO) channel with Gaussian noise, is of central importance in digital communications and is encountered in several different applications. Examples include, but are not limited to; detection of symbols spatially multiplexed over a multiple-antenna channel and the multiuser detection problem in a code division multiple access (CDMA) system. Two algorithms previously proposed in the literature are considered and analyzed. Both algorithms have their origin in other fields of science but have gained mainstream recognition as efficient algorithms for the detection problem considered herein. Specifically, we consider the sphere decoder and semidefinite relaxation detector. By incorporating assumptions applicable in the communications context the performance of the two algorithms is addressed. The first algorithm, the sphere decoder, offers optimal performance in terms of its error probability. Further, the algorithm has proved extremely efficient in terms of computational complexity for moderately sized problems at high signal to noise ratio (SNR). Although it is recognized that the algorithm has an exponential worst case complexity, there has been a widespread belief that the algorithm has a polynomial average complexity at high SNR. A contribution made herein is to show that this is incorrect and that the average complexity, as the worst case complexity, is exponential in the number of symbols detected. Instead, another explanation of the observed efficiency of the algorithm is offered by deriving the exponential rate of growth and showing that this rate, although strictly positive for finite SNR, is small in the high SNR regime. The second algorithm, the semidefinite relaxation (SDR) detector, offers polynomial complexity at the expense of suboptimal performance in terms of error probability. Nevertheless, previous numerical observations suggest that error probability of the SDR algorithm is close to that of the optimal detector. Herein, the near optimality is of the SDR algorithm is given a precise meaning by studying the diversity of the SDR algorithm when applied to the (real valued) i.i.d.~Rayleigh fading channel and it is shown that the SDR algorithm achieves the same diversity order as the optimal detector. Further, criteria under which the SDR estimates coincide with the optimal estimates are derived and discussed. / Ett grundläggande problem som påträffats inom digital kommunikation är detektering av en symbolvektor, tillhörande ett ändligt symbolalfabet, som sänts över en MIMO (från engelskans multiple-input multiple-output) kanal med Gausiskt brus. Detta problem påträffas bland annat då symboler sänts över en trådlös kanal med flera antenner hos mottagaren och sändaren samt då flera användare i ett CDMA system simultant skall avkodas. In denna avhandling behandlas två mottagaralgoritmer konstruerade för detta ändamål. Algoritmerna har sin bakgrund i andra forskningsområden men kan i nuläget sägas vara mycket välkända inom kommunikationsområdet. De benämns vanligtvis som sfäravkodaren (eng. sphere decoder) samt den semidefinita relaxeringsdetektorn (eng. semidefinite relaxation detector). Algoritmerna analyseras i denna avhandling matematiskt genom att införa förenklande antaganden som är relevanta och applicerbara för de kommunikationsproblem som är av intesse. Den första algoritmen, sfäravkodaren, löser dessa detektionsproblem på ett optimalt sätt i betydelsen att den minimerar sannolikheten för att detektorn fattar ett felaktigt beslut rörande det sända meddelandet (symbolvektorn). Också vad gäller algoritmens komplexitet har simuleringar visat att den är oväntat låg, åtminstone vid höga signalbrusförhållanden (SNR). Trots att det är allmänt känt att algoritmen i sämsta fall har exponentiell komplexitet så har detta lett till den allmänt spridda uppfattningen att medelkomplexiteten (eller den förväntade komplexiteten) endast är polynomisk vid höga signalbrusförhållanden. Ett av huvudbidragen i denna avhandling är att visa att denna uppfattning är felaktig och att också medelkomplexiteten växer exponentiellt i antalet symboler som simultant detekteras. Ytterligare ett bidrag ligger i att ge en alternativ förklaring till den observerat låga medelkomplexiteten. Det visas att den exponentiella hastighet med vilken komplexiteten växer beror på signalbrusförhållande, och att den är låg för höga SNR. Den andra algoritmen, den semidefinita relaxeringsdetektorn, erbjuder polynomisk komplexitet vid en något högre felsannolikhet. Intressant nog har dock felsannolikheten tidigare, genom simuleringar, visat sig vara endast marginellt högre än felsannolikheten hos den optimala mottagaren. Bidraget som relaterar till den semidefinita relaxeringsmottagaren ligger i att både förklara och i att ge en specifik kvatifierbar mening åt uttalandet att felsannolikheten endast är marginellt högre. I syfte att åstadkomma detta studeras diversitetsordningen för detektorn, och det bevisas att diversitetsordningen för den semidefinita relaxeringsdetektorn är densamma som för den optimala mottagaren. Utöver detta karakteriseras också de krav som måste uppfyllas för att den detektorn skall finna den optimala lösningen. / QC 20100901

The Internet has rapidly evolved from being a scientific experiment to a commercial network connecting millions of hosts that carries traffic generated by a large amount of applications with diverse requirements. Its architecture was however designed to enable efficient point-to-point delivery of bulk data, and can not provide statistical guarantees on the timely delivery of delay sensitive data such as streaming and real-time multimedia. Thus, applications that require low loss probabilities in today's Internet have to use some end-to-end error recovery mechanism. For delay sensitive applications the introduced latency by the applied schemes has to be low as well. Traffic control functions such as delay limited shaping and forward error correction (FEC), and multiple description coding (MDC) have been proposed for variable bitrate video. Their major drawback is, however, that it is difficult to predict their efficiency, as it depends on many factors like the characteristics of the stream itself, the characteristics of the traffic in the network and the network parameters. Consequently, it is difficult to decide which control mechanisms to employ, how to combine them and to choose the right parameters (e.g. block length, code rate) for optimal performance. In this thesis we present results on the efficiency of traffic control functions and MDC for video transmission based on mathematical models and simulations. We investigate the efficiency of delay limited traffic shaping and the trade-offs in the joint use of traffic shaping and forward error correction. We identify the packet size distribution of the traffic in the network as an additional factor that may influence the efficiency of FEC, and present a thorough analysis of its possible effects. We present an analytical comparison of MDC versus media-dependent FEC and media-independent FEC, and based on the results we conclude that MDC is a promising error control solution for multimedia communications with very strict delay bounds in an environment with bursty losses. We combine the analytical results with traces from measurements performed on the Internet to evaluate how efficient these error control schemes are under real loss patterns. We compare the efficiency of MDC and media-dependent FEC in the presence of channel estimation errors; we propose a new rate allocation method, which is robust to mis-estimations of the channel state and which improves error resilience on non-stationary channels. Finally we present an analytical model of the performance of an end-point-based multimedia streaming architecture based on multiple distribution trees and forward error correction, and analyze the behavior of the architecture for a large number of nodes. / QC 20101115

In order to be competitive with emerging mobile systems and to satisfy the ever growing request for higher data rates, the 3G consortium, 3rd Generation Partnership Project (3GPP), is currently developing concepts for a long term evolution (LTE) of the 3G standard. The LTE-concept at Ericsson is based on Orthogonal Frequency Division Multiplexing (OFDM) as downlink air interface. OFDM enables the use of frequency domain link adaptation to select the most appropriate transmission parameters according to current channel conditions, in order to maximize the throughput and maintain the delay at a desired level. The purpose of this thesis work is to study, implement and evaluate different link adaptation algorithms. The main focus is on modulation adaptation, where the differences in performance between time domain and frequency domain adaptation are investigated. The simulations made in this thesis are made with a simulator developed at Ericsson. Simulations show in general that the cell throughput is enhanced by an average of 3% when using frequency domain modulation adaptation. When using the implemented frequency domain power allocation algorithm, a gain of 23-36% in average is seen in the users 5th percentile throughput. It should be noted that the simulations use a realistic web traffic model, which makes the channel quality estimation (CQE) difficult. The CQE has great impact on the performance of frequency domain adaptation. Throughput improvements are expected when using an improved CQE or interference avoidance schemes. The gains with frequency domain adaptation shown in this thesis work may be too small to motivate the extra signalling overhead required. The complexity of the implemented frequency domain power allocation algorithm is also very high compared to the performance enhancement seen.

Interaktiv mikro-TV (iµTV) handlar om att utnyttja kameran i en mobiltelefon för att kommunicera, informera och/eller underhålla. I den här uppsatsen uttrycks visionen att varje individ i framtiden skulle bära runt på en nästan komplett sändningsbuss i fickan i form av en mobiltelefon. Interaktivitet uppstår genom möjligheten att koppla samman flera användares sändningar, där de deltagande personerna kan påverka det som sänds. Det erbjuds också interaktivitet för tittarna som kan välja vilken sändning de vill följa. Flera olika scenarier beskrivs för att visa hur användandet skulle se ut och för att tänka sig in i situationer där en individ skulle ha en vilja och ett behov av att använda en sådan teknologi. Med hjälp av de här scenarierna har jag tagit fram förslag till olika gränssnitt och funktioner som alla har en praktisk nytta. Att ha funktioner som underlättar individens vardag kommer hjälpa till att skapa ett behov och en vilja till att använda teknologin. För att ta hänsyn till olika miljöer kan man använda sig av inställningsprofiler och kan då ta hänsyn till var användaren befinner sig.