Cooperative communication in wireless local area networks

The concept of cooperative communication has been proposed to improve link capacity, transmission reliability and network coverage in multiuser wireless communication networks. Different from conventional point-to-point and point-to-multipoint communications, cooperative communication allows multiple users or stations in a wireless network to coordinate their packet transmissions and share each other’s resources, thus achieving high performance gain and better service coverage. According to the IEEE 802.11 standards, Wireless Local Area Networks (WLANs) can support multiple transmission data rates, depending on the instantaneous channel condition between a source station and an Access Point (AP). In such a multi-rate WLAN, those low data-rate stations will occupy the shared communication channel for a longer period for transmitting a fixed-size packet to the AP, thus reducing the channel efficiency and overall system performance. This thesis addresses this challenging problem in multi-rate WLANs by proposing two cooperative Medium Access Control (MAC) protocols, namely Busy Tone based Cooperative MAC (BTAC) protocol and Cooperative Access with Relay’s Data (CARD) protocol. Under BTAC, a low data-rate sending station tries to identify and use a close-by intermediate station as its relay to forward its data packets at higher data-rate to the AP through a two-hop path. In this way, BTAC can achieve cooperative diversity gain in multi-rate WLANs. Furthermore, the proposed CARD protocol enables a relay station to transmit its own data packets to the AP immediately after forwarding its neighbour’s packets, thus minimising the handshake procedure and overheads for sensing and reserving the common channel. In doing so, CARD can achieve both cooperative diversity gain and cooperative multiplexing gain. Both BTAC and CARD protocols are backward compatible with the existing IEEE 802.11 standards. New cross-layer mathematical models have been developed in this thesis to study the performance of BTAC and CARD under different channel conditions and for saturated and unsaturated traffic loads. Detailed simulation platforms were developed and are discussed in this thesis. Extensive simulation results validate the mathematical models developed and show that BTAC and CARD protocols can significantly improve system throughput, service delay, and energy efficiency for WLANs operating under realistic communication scenarios

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

Cross-layer analysis for video transmission over COFDM-based wireless local area networks

EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Mac-Phy Cross-Layer analysis and design of Mimo-Ofdm Wlans based on fast link adaptation

The latestWLAN standard, known as IEEE 802.11n, has notably increased the network capacity with respect to its predecessors thanks to the incorporation of the multipleinput multiple-output (MIMO) technology. Nonetheless, the new amendment, as its previous ones, does not specify how crucial configuration mechanisms, most notably the adaptive modulation and coding (AMC) algorithm should be implemented. The AMC process has proved essential to fully exploit the system resources in light of varying channel conditions. In this dissertation, a closed-loop AMC technique, referred to as fast link adaption (FLA) algorithm, that effectively selects themodulation and coding scheme (MCS) for multicarriermultiantennaWLAN networks is proposed. The FLA algorithm determines the MCS that maximizes the throughput while satisfying a quality of service (QoS) constraint, usually defined in the form of an objective packet error rate (PER). To this end, FLA uses a packet/bit error rate prediction methodology based on the exponential effective SNRmetric (EESM). The FLA algorithm performance has been evaluated under IEEE 802.11n systems that thanks to the incorporation of a feedbackmechanismare able to implement closed- loop AMC mechanisms. Initially, this AMC technique relies only on physical layer information but it is subsequently extended to also take into account themediumaccess control (MAC) sublayer performance. At the physical layer, the FLA algorithm has demonstrated its effectivity by performing very close to optimality in terms of throughput, while satisfying a prescribed PER constraint. The FLA algorithm has also been evaluated using imperfect channel information. It has been observed that the proposed FLA technique is rather robust against imperfect channel information, and only in highly-frequency selective channels, imperfect channel knowledge causes a noticeable degradation in throughput. At the MAC sublayer, the FLA algorithm has been complemented with a timeout strategy that weighs down the influence of the available channel information as this becomes outdated. This channel information outdate is caused by the MAC sublayer whose user multiplexing policy potentially results in large delays between acquiring the instant in which the channel state information is acquired and that in which the channel is accessed. Results demonstrate the superiority of FLA when compared to open-loop algorithms under saturated and non-saturated conditions and irrespective of the packet length, number of users, protocol (CSMA/CA or CDMA/E2CA) and access scheme (Basic Access or RTS/CTS). Additionally, several analytical models have been developed to estimate the system performance at the MAC sublayer. These models account for all operational details of the IEEE 802.11n MAC sublayer, such as finite number of retries, anomalous slot or channel errors. In particular, a semi-analytical model that assesses the MAC layer throughput under saturated conditions, considering the AMC performance is first introduced. Then, an analytical model that allows the evaluation of the QoS performance under non-saturated conditions is presented. This model focuses on single MCS and it is able to accurately predict very important system performance metrics such as blocking probability, delay, probability of discard or goodput thanks to the consideration of the finite queues on each station. Finally, the previous non-saturated analytical approach is used to define a semi-analytical model in order to estimate the system performance when considering AMC algorithms (i.e. whenmultiple MCSs are available)La darrera versió de l’estàndard deWLAN, anomenada IEEE 802.11n, ha augmentat la seva capacitat notablement en relació als sistemes anteriors gràcies a la incorporació de la tecnologia de múltiples antenes en transmissió i recepció (MIMO). No obstant això, la nova proposta, al igual que les anteriors, segueix sense especificar com s’han d’implementar elsmecanismes de configuraciómés crucials, un dels quals és l’algoritme de codificació imodulació adaptativa (AMC). Aquests algoritmes ja han demostrat la seva importància a l’hora demaximitzar el rendiment del sistema tenint en compte les condicions canviants del canal. En aquesta tesis s’ha proposat un algoritme AMC de llaç tancat, anomenat adaptació ràpida de l’enllaç (FLA), que selecciona eficientment l’esquema demodulació i codificació adaptativa per xarxes WLAN basades en arquitectures multiportadora multiantena. L’algoritme FLA determina el mode de transmissió capaç de maximitzar el throughput per les condicions de canal actuals, mentre satisfà un requisit de qualitat de servei en forma de taxa d’error per paquet (PER). FLA utilitza una metodologia de predicció de PER basada en l’estimació de la relació senyal renou (SNR) efectiva exponencial (EESM). El rendiment de l’algoritme FLA ha estat avaluat en sistemes IEEE 802.11n, ja que aquests, gràcies a la incorporació d’unmecanisme de realimentació demodes de transmissió, poden adoptar solucions AMC de llaç tancat. En una primera part, l’estudi s’ha centrat a la capa física i després s’ha estès a la subcapa MAC. A la capa física s’ha demostrat l’efectivitat de l’algoritme FLA aconseguint un rendiment molt proper al que ens proporcionaria un esquema AMC òptim en termes de throughput, alhora que es satisfan els requisits de PER objectiu. L’algoritme FLA també ha estat avaluat utilitzant informació imperfecte del canal. S’ha vist que l’algoritme FLA proposat és robust en front dels efectes d’estimació imperfecte del canal, i només en canals altament selectius en freqüència, la informació imperfecte del canal provoca una davallada en el rendiment en termes de throughput. A la subcapa MAC, l’algoritme FLA ha estat complementat amb una estratègia de temps d’espera que disminueix la dependència amb la informació de canal disponible a mesura que aquesta va quedant desfassada respecte de l’estat actual. Aquesta informació de canal desfassada és conseqüència de la subcapa MAC que degut a la multiplexació d’usuaris introdueix grans retards entre que es determina el mode de transmissió més adequat i la seva utilització per a l’accés al canal. Els resultats obtinguts han demostrat la superioritat de FLA respecte d’altres algoritmes de llaç obert en condicions de saturació i de no saturació, i independentment de la longitud de paquet, nombre d’usuaris, protocol (CSMA/CA i CSMA/E2CA) i esquema d’accés (Basic Access i RTS/CTS). Amés, s’han desenvolupat diversosmodels analítics per tal d’estimar el rendiment del sistema a la subcapa MAC. Aquests models consideren tots els detalls de funcionament de la subcapaMAC del 802.11n, comper exemple un nombre finit de retransmissions de cada paquet, l’slot anòmal o els errors introduïts pel canal. Inicialment s’ha proposat unmodel semi-analític que determina el throughtput en condicions de saturació, considerant el rendiment dels algoritmes AMC. Després s’ha presentat un model analític que estima el rendiment del sistema per condicions de no saturació, mitjançat elmodelat de cues finites a cada estació. Aquestmodel consideramodes de transmissió fixes i és capaç de determinar de manera molt precisa mètriques de rendimentmolt importants comsón la probabilitat de bloqueig de cada estació, el retard mitjà del paquets, la probabilitat de descart o la mesura del goodput. Finalment, el model analític de no saturació s’ha utilitzat per definir un model semi-analític per tal d’estimar el rendiment del sistema quan es considera l’ús d’algoritmes AMC

A Hardware Testbed for Measuring IEEE 802.11g DCF Performance

The Distributed Coordination Function (DCF) is the oldest and most widely-used IEEE 802.11 contention-based channel access control protocol. DCF adds a significant amount of overhead in the form of preambles, frame headers, randomised binary exponential back-off and inter-frame spaces. Having accurate and verified performance models for DCF is thus integral to understanding the performance of IEEE 802.11 as a whole. In this document DCF performance is measured subject to two different workload models using an IEEE 802.11g test bed. Bianchi proposed the first accurate analytic model for measuring the performance of DCF. The model calculates normalised aggregate throughput as a function of the number of stations contending for channel access. The model also makes a number of assumptions about the system, including saturation conditions (all stations have a fixed-length packet to send at all times), full-connectivity between stations, constant collision probability and perfect channel conditions. Many authors have extended Bianchi's machine model to correct certain inconsistencies with the standard, while very few have considered alternative workload models. Owing to the complexities associated with prototyping, most models are verified against simulations and not experimentally using a test bed. In addition to a saturation model we considered a more realistic workload model representing wireless Internet traffic. Producing a stochastic model for such a workload was a challenging task, as usage patterns change significantly between users and over time. We implemented and compared two Markov Arrival Processes (MAPs) for packet arrivals at each client - a Discrete-time Batch Markovian Arrival Process (D-BMAP) and a modified Hierarchical Markov Modulated Poisson Process (H-MMPP). Both models had parameters drawn from the same wireless trace data. It was found that, while the latter model exhibits better Long Range Dependency at the network level, the former represented traces more accurately at the client-level, which made it more appropriate for the test bed experiments. A nine station IEEE 802.11 test bed was constructed to measure the real world performance of the DCF protocol experimentally. The stations used IEEE 802.11g cards based on the Atheros AR5212 chipset and ran a custom Linux distribution. The test bed was moved to a remote location where there was no measured risk of interference from neighbouring radio transmitters in the same band. The DCF machine model was fixed and normalised aggregate throughput was measured for one through to eight contending stations, subject to (i) saturation with fixed packet length equal to 1000 bytes, and (ii) the D-BMAP workload model for wireless Internet traffic. Control messages were forwarded on a separate wired backbone network so that they did not interfere with the experiments. Analytic solver software was written to calculate numerical solutions for thee popular analytic models for DCF and compared the solutions to the saturation test bed experiments. Although the normalised aggregate throughput trends were the same, it was found that as the number of contending stations increases, so the measured aggregate DCF performance diverged from all three analytic model's predictions; for every station added to the network normalised aggregate throughput was measured lower than analytically predicted. We conclude that some property of the test bed was not captured by the simulation software used to verify the analytic models. The D-BMAP experiments yielded a significantly lower normalised aggregate throughput than the saturation experiments, which is a clear result of channel underutilisation. Although this is a simple result, it highlights the importance of the traffic model on network performance. Normalised aggregate throughput appeared to scale more linearly when compared to the RTS/CTS access mechanism, but no firm conclusion could be drawn at 95% confidence. We conclude further that, although normalised aggregate throughput is appropriate for describing overall channel utilisation in the steady state, jitter, response time and error rate are more important performance metrics in the case of bursty traffic

Performance Modeling and Analysis of Wireless Local Area Networks with Bursty Traffic

The explosive increase in the use of mobile digital devices has posed great challenges in the design and implementation of Wireless Local Area Networks (WLANs). Ever-increasing demands for high-speed and ubiquitous digital communication have made WLANs an essential feature of everyday life. With audio and video forming the highest percentage of traffic generated by multimedia applications, a huge demand is placed for high speed WLANs that provide high Quality-of-Service (QoS) and can satisfy end user’s needs at a relatively low cost. Providing video and audio contents to end users at a satisfactory level with various channel quality and current battery capacities requires thorough studies on the properties of such traffic. In this regard, Medium Access Control (MAC) protocol of the 802.11 standard plays a vital role in the management and coordination of shared channel access and data transmission. Therefore, this research focuses on developing new efficient analytical models that evaluate the performance of WLANs and the MAC protocol in the presence of bursty, correlated and heterogeneous multimedia traffic using Batch Markovian Arrival Process (BMAP). BMAP can model the correlation between different packet size distributions and traffic rates while accurately modelling aggregated traffic which often possesses negative statistical properties. The research starts with developing an accurate traffic generator using BMAP to capture the existing correlations in multimedia traffics. For validation, the developed traffic generator is used as an arrival process to a queueing model and is analyzed based on average queue length and mean waiting time. The performance of BMAP/M/1 queue is studied under various number of states and maximum batch sizes of BMAP. The results clearly indicate that any increase in the number of states of the underlying Markov Chain of BMAP or maximum batch size, lead to higher burstiness and correlation of the arrival process, prompting the speed of the queue towards saturation. The developed traffic generator is then used to model traffic sources in IEEE 802.11 WLANs, measuring important QoS metrics of throughput, end-to-end delay, frame loss probability and energy consumption. Performance comparisons are conducted on WLANs under the influence of multimedia traffics modelled as BMAP, Markov Modulated Poisson Process and Poisson Process. The results clearly indicate that bursty traffics generated by BMAP demote network performance faster than other traffic sources under moderate to high loads. The model is also used to study WLANs with unsaturated, heterogeneous and bursty traffic sources. The effects of traffic load and network size on the performance of WLANs are investigated to demonstrate the importance of burstiness and heterogeneity of traffic on accurate evaluation of MAC protocol in wireless multimedia networks. The results of the thesis highlight the importance of taking into account the true characteristics of multimedia traffics for accurate evaluation of the MAC protocol in the design and analysis of wireless multimedia networks and technologies