Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

Proceedings of the Third International Mobile Satellite Conference (IMSC 1993)

Satellite-based mobile communications systems provide voice and data communications to users over a vast geographic area. The users may communicate via mobile or hand-held terminals, which may also provide access to terrestrial cellular communications services. While the first and second International Mobile Satellite Conferences (IMSC) mostly concentrated on technical advances, this Third IMSC also focuses on the increasing worldwide commercial activities in Mobile Satellite Services. Because of the large service areas provided by such systems, it is important to consider political and regulatory issues in addition to technical and user requirements issues. Topics covered include: the direct broadcast of audio programming from satellites; spacecraft technology; regulatory and policy considerations; advanced system concepts and analysis; propagation; and user requirements and applications

Proceedings of the Fifth International Mobile Satellite Conference 1997

Satellite-based mobile communications systems provide voice and data communications to users over a vast geographic area. The users may communicate via mobile or hand-held terminals, which may also provide access to terrestrial communications services. While previous International Mobile Satellite Conferences have concentrated on technical advances and the increasing worldwide commercial activities, this conference focuses on the next generation of mobile satellite services. The approximately 80 papers included here cover sessions in the following areas: networking and protocols; code division multiple access technologies; demand, economics and technology issues; current and planned systems; propagation; terminal technology; modulation and coding advances; spacecraft technology; advanced systems; and applications and experiments

Adaptive implementation of turbo multi-user detection architecture

MULTI-access techniques have been adopted widely for communications in underwater acoustic channels, which present many challenges to the development of reliable and practical systems. In such an environment, the unpredictable and complex ocean conditions cause the acoustic waves to be affected by many factors such as limited bandwidth, large propagation losses, time variations and long latency, which limit the usefulness of such techniques. Additionally, multiple access interference (MAI) signals and poor estimation of the unknown channel parameters in the presence of limited training sequences are two of the major problems that degrade the performance of such technologies. In this thesis, two different single-element multi-access schemes, interleave division multiple access (IDMA) and code division multiple access (CDMA), employing decision feedback equalization (DFE) and soft Rake-based architectures, are proposed for multi-user underwater communication applications. By using either multiplexing pilots or continuous pilots, these adaptive turbo architectures with carrier phase tracking are jointly optimized based on the minimum mean square error (MMSE) criterion and adapted iteratively by exchanging soft information in terms of Log-Likelihood Ratio (LLR) estimates with the single-user’s channel decoders. The soft-Rake receivers utilize developed channel estimation and the detection is implemented using parallel interference cancellation (PIC) to remove MAI effects between users. These architectures are investigated and applied to simulated data and data obtained from realistic underwater communication trials using off-line processing of signals acquired during sea-trials in the North Sea. The results of different scenarios demonstrate the penalty in performance as the fading induces irreducible error rates that increase with channel delay spread and emphasize the benefits of using coherent direct adaptive receivers in such reverberant channels. The convergence behaviour of the detectors is evaluated using EXIT chart analyses and issues such as the adaptation parameters and their effects on the performance are also investigated. However, in some cases the receivers with partial knowledge of the interleavers’ patterns or codes can still achieve performance comparable to those with full knowledge. Furthermore, the thesis describes implementation issues of these algorithms using digital signal processors (DSPs), such as computational complexity and provides valuable guidelines for the design of real time underwater communication systems.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Adaptive implementation of turbo multi-user detection architecture

MULTI-access techniques have been adopted widely for communications in underwater acoustic channels, which present many challenges to the development of reliable and practical systems. In such an environment, the unpredictable and complex ocean conditions cause the acoustic waves to be affected by many factors such as limited bandwidth, large propagation losses, time variations and long latency, which limit the usefulness of such techniques. Additionally, multiple access interference (MAI) signals and poor estimation of the unknown channel parameters in the presence of limited training sequences are two of the major problems that degrade the performance of such technologies. In this thesis, two different single-element multi-access schemes, interleave division multiple access (IDMA) and code division multiple access (CDMA), employing decision feedback equalization (DFE) and soft Rake-based architectures, are proposed for multi-user underwater communication applications. By using either multiplexing pilots or continuous pilots, these adaptive turbo architectures with carrier phase tracking are jointly optimized based on the minimum mean square error (MMSE) criterion and adapted iteratively by exchanging soft information in terms of Log-Likelihood Ratio (LLR) estimates with the single-user’s channel decoders. The soft-Rake receivers utilize developed channel estimation and the detection is implemented using parallel interference cancellation (PIC) to remove MAI effects between users. These architectures are investigated and applied to simulated data and data obtained from realistic underwater communication trials using off-line processing of signals acquired during sea-trials in the North Sea. The results of different scenarios demonstrate the penalty in performance as the fading induces irreducible error rates that increase with channel delay spread and emphasize the benefits of using coherent direct adaptive receivers in such reverberant channels. The convergence behaviour of the detectors is evaluated using EXIT chart analyses and issues such as the adaptation parameters and their effects on the performance are also investigated. However, in some cases the receivers with partial knowledge of the interleavers’ patterns or codes can still achieve performance comparable to those with full knowledge. Furthermore, the thesis describes implementation issues of these algorithms using digital signal processors (DSPs), such as computational complexity and provides valuable guidelines for the design of real time underwater communication systems.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Physical and Link Layer Implications in Vehicle Ad Hoc Networks

Vehicle Ad hoc Networks (V ANET) have been proposed to provide safety on the road and deliver road traffic information and route guidance to drivers along with commercial applications. However the challenges facing V ANET are numerous. Nodes move at high speeds, road side units and basestations are scarce, the topology is constrained by the road geometry and changes rapidly, and the number of nodes peaks suddenly in traffic jams. In this thesis we investigate the physical and link layers of V ANET and propose methods to achieve high data rates and high throughput. For the physical layer, we examine the use of Vertical BLAST (VB LAST) systems as they provide higher capacities than single antenna systems in rich fading environments. To study the applicability of VB LAST to VANET, a channel model was developed and verified using measurement data available in the literature. For no to medium line of sight, VBLAST systems provide high data rates. However the performance drops as the line of sight strength increases due to the correlation between the antennas. Moreover, the performance of VBLAST with training based channel estimation drops as the speed increases since the channel response changes rapidly. To update the channel state information matrix at the receiver, a channel tracking algorithm for flat fading channels was developed. The algorithm updates the channel matrix thus reducing the mean square error of the estimation and improving the bit error rate (BER). The analysis of VBLAST-OFDM systems showed they experience an error floor due to inter-carrier interference (lCI) which increases with speed, number of antennas transmitting and number of subcarriers used. The update algorithm was extended to VBLAST -OFDM systems and it showed improvements in BER performance but still experienced an error floor. An algorithm to equalise the ICI contribution of adjacent subcarriers was then developed and evaluated. The ICI equalisation algorithm reduces the error floor in BER as more subcarriers are equalised at the expense of more hardware complexity. The connectivity of V ANET was investigated and it was found that for single lane roads, car densities of 7 cars per communication range are sufficient to achieve high connectivity within the city whereas 12 cars per communication range are required for highways. Multilane roads require higher densities since cars tend to cluster in groups. Junctions and turns have lower connectivity than straight roads due to disconnections at the turns. Although higher densities improve the connectivity and, hence, the performance of the network layer, it leads to poor performance at the link layer. The IEEE 802.11 p MAC layer standard under development for V ANET uses a variant of Carrier Sense Multiple Access (CSMA). 802.11 protocols were analysed mathematically and via simulations and the results prove the saturation throughput of the basic access method drops as the number of nodes increases thus yielding very low throughput in congested areas. RTS/CTS access provides higher throughput but it applies only to unicast transmissions. To overcome the limitations of 802.11 protocols, we designed a protocol known as SOFT MAC which combines Space, Orthogonal Frequency and Time multiple access techniques. In SOFT MAC the road is divided into cells and each cell is allocated a unique group of subcarriers. Within a cell, nodes share the available subcarriers using a combination of TDMA and CSMA. The throughput analysis of SOFT MAC showed it has superior throughput compared to the basic access and similar to the RTS/CTS access of 802.11

Comparative study and performance evaluation of MC-CDMA and OFDM over AWGN and fading channels environment

Η απαίτηση για εφαρμογές υψηλής ταχύτητας μετάδοσης δεδομένων έχει αυξηθεί σημαντικά τα τελευταία χρόνια. Η πίεση των χρηστών σήμερα για ταχύτερες επικοινωνίες, ανεξαρτήτως κινητής ή σταθερής, χωρίς επιπλέον κόστος είναι μια πραγματικότητα. Για να πραγματοποιηθούν αυτές οι απαιτήσεις, προτάθηκε ένα νέο σχήμα που συνδυάζει ψηφιακή διαμόρφωση και πολλαπλές προσβάσεις, για την ακρίβεια η Πολλαπλή Πρόσβαση με διαίρεση Κώδικα Πολλαπλού Φέροντος (Multi-Carrier Code Division Multiple Access MC-CDMA). Η εφαρμογή του Γρήγορου Μετασχηματισμού Φουριέ (Fast Fourier Transform,FFT) που βασίζεται στο (Orthogonal Frequency Division Multiplexing, OFDM) χρησιμοποιεί τις περίπλοκες λειτουργίες βάσεως και αντικαθίσταται από κυματομορφές για να μειώσει το επίπεδο της παρεμβολής. Έχει βρεθεί ότι οι μετασχηματισμένες κυματομορφές (Wavelet Transform,W.T.) που βασίζονται στον Haar είναι ικανές να μειώσουν το ISI και το ICI, που προκαλούνται από απώλειες στην ορθογωνιότητα μεταξύ των φερόντων, κάτι που τις καθιστά απλούστερες για την εφαρμογή από του FFT. Επιπλέον κέρδος στην απόδοση μπορεί να επιτευχθεί αναζητώντας μια εναλλακτική λειτουργία ορθογωνικής βάσης και βρίσκοντας ένα καλύτερο μετασχηματισμό από του Φουριέ (Fourier) και τον μετασχηματισμό κυματομορφής (Wavelet Transform). Στην παρούσα εργασία, υπάρχουν τρία προτεινόμενα μοντέλα. Το 1ο, ( A proposed Model ‘1’ of OFDM based In-Place Wavelet Transform), το 2ο, A proposed Model ‘2’ based In-Place Wavelet Transform Algorithm and Phase Matrix (P.M) και το 3ο, A proposed Model ‘3’ of MC-CDMA Based on Multiwavelet Transform. Οι αποδόσεις τους συγκρίθηκαν με τα παραδοσιακά μοντέλα μονού χρήστη κάτω από διαφορετικά κανάλια (Κανάλι AWGN, επίπεδη διάλειψη και επιλεκτική διάλειψη).The demand for high data rate wireless multi-media applications has increased significantly in the past few years. The wireless user’s pressure towards faster communications, no matter whether mobile, nomadic, or fixed positioned, without extra cost is nowadays a reality. To fulfill these demands, a new scheme which combines wireless digital modulation and multiple accesses was proposed in the recent years, namely, Multicarrier-Code Division Multiple Access (MC-CDMA). The Fourier based OFDM uses the complex exponential bases functions and it is replaced by wavelets in order to reduce the level of interference. It is found that the Haar-based wavelets are capable of reducing the ISI and ICI, which are caused by the loss in orthogonality between the carriers. Further performance gains can be made by looking at alternative orthogonal basis functions and finding a better transform rather than Fourier and wavelet transform. In this thesis, there are three proposed models [Model ‘1’ (OFDM based on In-Place Wavelet Transform, Model ‘2’ (MC-CDMA based on IP-WT and Phase Matrix) and Model ‘3’ (MC-CDMA based on Multiwavelet Transform)] were created and then comparison their performances with the traditional models for single user system were compared under different channel characteristics (AWGN channel, flat fading and selective fading). The conclusion of my study as follows, the models (1) was achieved much lower bit error rates than traditional models based FFT. Therefore these models can be considered as an alternative to the conventional MC-CDMA based FFT. The main advantage of using In-Place wavelet transform in the proposed models that it does not require an additional array at each sweep such as in ordered Fast Haar wavelet transform, which makes it simpler for implementation than FFT. The model (2) gave a new algorithm based on In-Place wavelet transform with first level processing multiple by PM was proposed. The model (3) gave much lower bit error than other two models in additional to traditional models

Scalable and rate adaptive wireless multimedia multicast

The methods that are described in this work enable highly efficient audio-visual streaming over wireless digital communication systems to an arbitrary number of receivers. In the focus of this thesis is thus point-to-multipoint transmission at constrained end-to-end delay. A fundamental difference as compared to point-to-point connections between exactly two communicating sending and receiving stations is in conveying information about successful or unsuccessful packet reception at the receiver side. The information to be transmitted is available at the sender, whereas the information about successful reception is only available to the receiver. Therefore, feedback about reception from the receiver to the sender is necessary. This information may be used for simple packet repetition in case of error, or adaptation of the bit rate of transmission to the momentary bit rate capacity of the channel, or both. This work focuses on the single transmission (including retransmissions) of data from one source to multiple destinations at the same time. A comparison with multi-receiver sequentially redundant transmission systems (simulcast MIMO) is made. With respect to feedback, this work considers time division multiple access systems, in which a single channel is used for data transmission and feedback. Therefore, the amount of time that can be spent for transmitting feedback is limited. An increase in time used for feedback transmissions from potentially many receivers results in a decrease in residual time which is usable for data transmission. This has direct impact on data throughput and hence, the quality of service. In the literature, an approach to reduce feedback overhead which is based on simultaneous feedback exists. In the scope of this work, simultaneous feedback implies equal carrier frequency, bandwidth and signal shape, in this case orthogonal frequency-division multiplex signals, during the event of the herein termed feedback aggregation in time. For this scheme, a constant amount of time is spent for feedback, independent of the number of receivers giving feedback about reception. Therefore, also data throughput remains independent of the number of receivers. This property of audio-visual digital transmission is taken for granted for statically configured, single purpose systems, such as terrestrial television. In the scope of this work are, however, multi-user and multi-purpose digital communication networks. Wireless LANs are a well-known example and are covered in detail herein. In suchlike systems, it is of great importance to remain independent of the number of receivers, as otherwise the service of ubiquitous digital connectivity is at the risk of being degraded. In this regard, the thesis at hand elaborates at what bit rates audio-visual transmission to multiple receivers may take place in conjunction with feedback aggregation. It is shown that the scheme achieves a multi-user throughput gain when used in conjunction with adaptivity of the bit rate to the channel. An assumption is the use of an ideal overlay packet erasure correcting code in this case. Furthermore, for delay constrained transmission, such as in so-called live television, throughput bit rates are examined. Applications have to be tolerant to a certain level of residual error in case of delay constrained transmission. Improvement of the rate adaptation algorithm is shown to increase throughput while residual error rates are decreased. Finally, with a consumer hardware prototype for digital live-TV re-distribution in the local wireless network, most of the mechanisms as described herein can be demonstrated.Die in vorliegender Arbeit aufgezeigten Methoden der paketbasierten drahtlosen digitalen Kommunikation ermöglichen es, Fernsehinhalte, aber auch audio-visuelle Datenströme im Allgemeinen, bei hoher Effizienz an beliebig große Gruppen von Empfängern zu verteilen. Im Fokus dieser Arbeit steht damit die Punkt- zu Mehrpunktübertragung bei begrenzter Ende-zu-Ende Verzögerung. Ein grundlegender Unterschied zur Punkt-zu-Punkt Verbindung zwischen genau zwei miteinander kommunizierenden Sender- und Empfängerstationen liegt in der Übermittlung der Information über erfolgreichen oder nicht erfolgreichen Paketempfang auf Seite der Empfänger. Da die zu übertragende Information am Sender vorliegt, die Information über den Erfolg der Übertragung jedoch ausschließlich beim jeweiligen Empfänger, muss eine Erfolgsmeldung auf dem Rückweg von Empfänger zu Sender erfolgen. Diese Information wird dann zum Beispiel zur einfachen Paketwiederholung im nicht erfolgreichen Fall genutzt, oder aber um die Übertragungsrate an die Kapazität des Kanals anzupassen, oder beides. Grundsätzlich beschäftigt sich diese Arbeit mit der einmaligen, gleichzeitigen Übertragung von Information (einschließlich Wiederholungen) an mehrere Empfänger, wobei ein Vergleich zu an mehrere Empfänger sequentiell redundant übertragenden Systemen (Simulcast MIMO) angestellt wird. In dieser Arbeit ist die Betrachtung bezüglich eines Rückkanals auf Zeitduplexsysteme beschränkt. In diesen Systemen wird der Kanal für Hin- und Rückweg zeitlich orthogonalisiert. Damit steht für die Übermittlung der Erfolgsmeldung eine beschränkte Zeitdauer zur Verfügung. Je mehr an Kanalzugriffszeit für die Erfolgsmeldungen der potentiell vielen Empfänger verbraucht wird, desto geringer wird die Restzeit, in der dann entsprechend weniger audio-visuelle Nutzdaten übertragbar sind, was sich direkt auf die Dienstqualität auswirkt. Ein in der Literatur weniger ausführlich betrachteter Ansatz ist die gleichzeitige Übertragung von Rückmeldungen mehrerer Teilnehmer auf gleicher Frequenz und bei identischer Bandbreite, sowie unter Nutzung gleichartiger Signale (hier: orthogonale Frequenzmultiplexsignalformung). Das Schema wird in dieser Arbeit daher als zeitliche Aggregation von Rückmeldungen, engl. feedback aggregation, bezeichnet. Dabei wird, unabhängig von der Anzahl der Empfänger, eine konstante Zeitdauer für Rückmeldungen genutzt, womit auch der Datendurchsatz durch zusätzliche Empfänger nicht notwendigerweise sinkt. Diese Eigenschaft ist aus statisch konfigurierten und für einen einzigen Zweck konzipierten Systemen, wie z. B. der terrestrischen Fernsehübertragung, bekannt. In dieser Arbeit werden im Gegensatz dazu jedoch am Beispiel von WLAN Mehrzweck- und Mehrbenutzersysteme betrachtet. Es handelt sich in derartigen Systemen zur digitalen Datenübertragung dabei um einen entscheidenden Vorteil, unabhängig von der Empfängeranzahl zu bleiben, da es sonst unweigerlich zu Einschränkungen in der Güte der angebotenen Dienstleistung der allgegenwärtigen digitalen Vernetzung kommen muss. Vorliegende Arbeit zeigt in diesem Zusammenhang auf, welche Datenraten unter Benutzung von feedback aggregation in der Verteilung an mehrere Empfänger und in verschiedenen Szenarien zu erreichen sind. Hierbei zeigt sich, dass das Schema im Zusammenspiel mit einer Adaption der Datenrate an den Übertragungskanal inhärent einen Datenratengewinn durch Mehrbenutzerempfang zu erzielen vermag, wenn ein überlagerter idealer Paketauslöschungsschutz-Code angenommen wird. Des weiteren wird bei der Übertragung mit zeitlich begrenzter Ausführungsdauer, z. B. dem sogenannten Live-Fernsehen, aufgezeigt, wie sich die erreichbare Datenrate reduziert und welche Restfehlertoleranz an die Übertragung gestellt werden muss. Hierbei wird ebenso aufgezeigt, wie sich durch Verbesserung der Ratenadaption erstere erhöhen und zweitere verringern lässt. An einem auf handelsüblichen Computer-Systemen realisiertem Prototypen zur Live-Fernsehübertragung können die hierin beschriebenen Mechanismen zu großen Teilen gezeigt werden

Adaptive PN code synchronisation in DS-CDMA systems

Spread Spectrum (SS) communication, initially designed for military applications, is now the basis for many of today's advanced communications systems such as Code Division Multiple Access (CDMA), Global Positioning System (GPS), Wireless Local Loop (WLL) , etc. For effective communication to take place in systems using SS modulation, the Pseudo-random Noise (PN) code used at the receiver to despread the received signal must be identical and be synchronised with the PN code that was used to spread the signal at the transmitter. Synchronisation is done in two steps: coarse synchronisation or acquisition, and fine synchronisation or tracking. Acquisition involves obtaining a coarse estimate of the phase shift between the transmitted PN code and that at the receiver so that the received PN code will be aligned or synchronised with the locally generated PN code. After acquisition, tracldng is now done which involves maintaining the alignment of the two PN codes. This thesis presents results of the research calTied out on a proposed adaptive PN code acquisition circuit designed to improve the synchronisation process in Direct Sequence CDMA (DS-CDMA) systems. The acquisition circuit is implemented using a Matched Filter (MF) for the correlation operation and the threshold setting device is an adaptive processor known as the Cell Averaging Constant False Alarm Rate (CA-CFAR) processor. It is a double dwell acquisition circuit where the second dwell is implemented by Post Detection Integration (PDI). Depending on the application, PDI can be used to mitigate the effect of frequency offset in non-coherent detectors and/or in the implementation of multiple dwell acquisition systems. Equations relating the performance measures - the probability of false alarm (Pra ), the probability of detection (P d) and the mean acquisition time (E {Tacq}) - of the circuit are deri ved. Monte Carlo simulation was used for the independent validation of the theoretical results obtained, and the strong agreement between these results shows the accuracy of the derived equations for the proposed circuit. Due to the combination of PDI and CA-CFAR processor in the implementation of the circuit, results obtained show that it can provide a good measure of robustness to frequency offset and noise power variations in mobile environment, consequently leading to improved acquisition time performance. The complete synchronisation circuit is realised by using this circuit in conjunction with a conventional code tracking circuit. Therefore, a study of a Non-coherent Delay-Locked Loop (NDLL) code tracking circuit is also calTied out.EThOS - Electronic Theses Online ServiceGBUnited Kingdo