Study on high reliability multicast communication with link adaptation for multiple-access wireless networks

制度:新 ; 報告番号:甲2849号 ; 学位の種類:博士(国際情報通信学) ; 授与年月日:2009/3/15 ; 早大学位記番号:新506

Video QoS/QoE over IEEE802.11n/ac: A Contemporary Survey

The demand for video applications over wireless networks has tremendously increased, and IEEE 802.11 standards have provided higher support for video transmission. However, providing Quality of Service (QoS) and Quality of Experience (QoE) for video over WLAN is still a challenge due to the error sensitivity of compressed video and dynamic channels. This thesis presents a contemporary survey study on video QoS/QoE over WLAN issues and solutions. The objective of the study is to provide an overview of the issues by conducting a background study on the video codecs and their features and characteristics, followed by studying QoS and QoE support in IEEE 802.11 standards. Since IEEE 802.11n is the current standard that is mostly deployed worldwide and IEEE 802.11ac is the upcoming standard, this survey study aims to investigate the most recent video QoS/QoE solutions based on these two standards. The solutions are divided into two broad categories, academic solutions, and vendor solutions. Academic solutions are mostly based on three main layers, namely Application, Media Access Control (MAC) and Physical (PHY) which are further divided into two major categories, single-layer solutions, and cross-layer solutions. Single-layer solutions are those which focus on a single layer to enhance the video transmission performance over WLAN. Cross-layer solutions involve two or more layers to provide a single QoS solution for video over WLAN. This thesis has also presented and technically analyzed QoS solutions by three popular vendors. This thesis concludes that single-layer solutions are not directly related to video QoS/QoE, and cross-layer solutions are performing better than single-layer solutions, but they are much more complicated and not easy to be implemented. Most vendors rely on their network infrastructure to provide QoS for multimedia applications. They have their techniques and mechanisms, but the concept of providing QoS/QoE for video is almost the same because they are using the same standards and rely on Wi-Fi Multimedia (WMM) to provide QoS

Reliable Multicast transport of the video over the WiFi network

Le transport multicast est une solution efficace pour envoyer le même contenu à plusieurs récepteurs en même temps. Ce mode est principalement utilisé pour fournir des flux multimédia en temps réel. Cependant, le multicast classique de l IEEE 802.11 n'utilise aucun mécanisme d acquittement. Ainsi, l échec de réception implique la perte définitive du paquet. Cela limite la fiabilité du transport multicast et impact la qualité des applications vidéo. Pour résoudre ce problème, 802.11v et 802.11aa sont définis récemment. Le premier amendement propose Direct Multicast Service (DMS). D'autre part, le 802.11aa introduit GroupCast with Retries (GCR). GCR définit deux nouvelles politiques de retransmission : Block Ack (BACK) et Unsolicited Retry (UR).Dans cette thèse, nous évaluons et comparons les performances de 802.11v/aa. Nos résultats montrent que tous les nouveaux protocoles multicast génèrent un overhead de transmission important. En outre, DMS a une scalabilité très limitée, et GCR-BACK n'est pas approprié pour des grands groupes multicast. D autre part, nous montrons que DMS et GCR-BACK génèrent des latences de transmission importantes lorsque le nombre de récepteurs augmente. Par ailleurs, nous étudions les facteurs de pertes dans les réseaux sans fil. Nous montrons que l'indisponibilité du récepteur peut être la cause principale des pertes importantes et de leur nature en rafales. En particulier, nos résultats montrent que la surcharge du processeur peut provoquer un taux de perte de 100%, et que le pourcentage de livraison peut être limité à 35% lorsque la carte 802.11 est en mode d économie d'énergie.Pour éviter les collisions et améliorer la fiabilité du transport multicast, nous définissons le mécanisme Busy Symbol (BS). Nos résultats montrent que BS évite les collisions et assure un taux de succès de transmission très important. Afin d'améliorer davantage la fiabilité du trafic multicast, nous définissons un nouveau protocole multicast, appelé Block Negative Acknowledgement (BNAK). Ce protocole opère comme suit. L AP envoi un bloc de paquets suivi par un Block NAK Request (BNR). Le BNR permet aux membres de détecter les données manquantes et d envoyer une demande de retransmission, c.à.d. un Block NAK Response (BNAK). Un BNAK est transmis en utilisant la procédure classique d accès au canal afin d'éviter toute collision avec d'autres paquets. En plus, cette demande est acquittée. Sous l'hypothèse que 1) le récepteur est situé dans la zone de couverture du débit de transmission utilisé, 2) les collisions sont évitées et 3) le terminal a la bonne configuration, très peu de demandes de retransmission sont envoyées, et la bande passante est préservée. Nos résultats montrent que BNAK a une très grande scalabilité et génère des délais très limités. En outre, nous définissons un algorithme d'adaptation de débit pour BNAK. Nous montrons que le bon débit de transmission est sélectionné moyennant un overhead très réduit de moins de 1%. En plus, la conception de notre protocole supporte la diffusion scalable de lavvidéo. Cette caractéristique vise à résoudre la problématique de la fluctuation de la bande passante, et à prendre en considération l'hétérogénéité des récepteurs dans un réseau sans fil.The multicast transport is an efficient solution to deliver the same content to many receivers at the same time. This mode is mainly used to deliver real-time video streams. However, the conventional multicast transmissions of IEEE 802.11 do not use any feedback policy. Therefore missing packets are definitely lost. This limits the reliability of the multicast transport and impacts the quality of the video applications. To resolve this issue, the IEEE 802.11v/aa amendments have been defined recently. The former proposes the Direct Multicast Service (DMS). On the other hand, 802.11aa introduces Groupcast with Retries (GCR) service. GCR defines two retry policies: Block Ack (BACK) and Unsolicited Retry (UR).In this thesis we evaluate and compare the performance of 802.11v/aa. Our simulation results show that all the defined policies incur an important overhead. Besides, DMS has a very limited scalability, and GCR-BACK is not appropriate for large multicast groups. We show that both DMS and GCR-BACK incur important transmission latencies when the number of the multicast receivers increases. Furthermore, we investigate the loss factors in wireless networks. We show that the device unavailability may be the principal cause of the important packet losses and their bursty nature. Particularly, our results show that the CPU overload may incur a loss rate of 100%, and that the delivery ratio may be limited to 35% when the device is in the power save mode.To avoid the collisions and to enhance the reliability of the multicast transmissions, we define the Busy Symbol (BS) mechanism. Our results show that BS prevents all the collisions and ensures a very high delivery ratio for the multicast packets. To further enhance the reliability of this traffic, we define the Block Negative Acknowledgement (BNAK) retry policy. Using our protocol, the AP transmits a block of multicast packets followed by a Block NAK Request (BNR). Upon reception of a BNR, a multicast member generates a Block NAK Response (BNAK) only if it missed some packets. A BNAK is transmitted after channel contention in order to avoid any eventual collision with other feedbacks, and is acknowledged. Under the assumption that 1) the receiver is located within the coverage area of the used data rate, 2) the collisions are avoided and 3) the terminal has the required configuration, few feedbacks are generated and the bandwidth is saved. Our results show that BNAK has a very high scalability and incurs very low delays. Furthermore, we define a rate adaptation scheme for BNAK. We show that the appropriate rate is selected on the expense of a very limited overhead of less than 1%. Besides, the conception of our protocol is defined to support the scalable video streaming. This capability intends to resolve the bandwidth fluctuation issue and to consider the device heterogeneity of the group members.BORDEAUX1-Bib.electronique (335229901) / SudocSudocFranceF

Design of interface selection protocols for multi-homed wireless networks

This thesis was submitted for the degree of Doctor of Philosophy and was awarded by Brunel University on 10 December 2010.The IEEE 802.11/802.16 standards conformant wireless communication stations have multi-homing transmission capability. To achieve greater communication efficiency, multi-homing capable stations use handover mechanism to select appropriate transmission channel according to variations in the channel quality. This thesis presents three internal-linked handover schemes, (1) Interface Selection Protocol (ISP), belonging to Wireless Local Area Network (WLAN)- Worldwide Interoperability for Microwave Access (WiMAX) environment (2) Fast Channel Scanning (FCS) and (3) Traffic Manager (TM), (2) and (3) belonging to WiMAX Environment. The proposed schemes in this thesis use a novel mechanism of providing a reliable communication route. This solution is based on a cross-layer communication framework, where the interface selection module uses various network related parameters from Medium Access Control (MAC) sub-layer/Physical Layer (PHY) across the protocol suite for decision making at the Network layer. The proposed solutions are highly responsive when compared with existing multi-homed schemes; responsiveness is one of the key factors in the design of such protocols. Selected route under these schemes is based on the most up to date link-layer information. Therefore, such a route is not only reliable in terms of route optimization but it also fulfils the application demands in terms of throughput and delay. Design of ISP protocol use probing frames during the route discovery process. The 802.11 mandates the use of different rates for data transmission frames. The ISP-metric can be incorporated into various routing aspects and its applicability is determined by the possibility of provision of MAC dependent parameters that are used to determine the best path metric values. In many cases, higher device density, interference and mobility cause variable medium access delays. It causes creation of ‘unreachable zones’, where destination is marked as unreachable. However, by use of the best path metric, the destination has been made reachable, anytime and anywhere, because of the intelligent use of the probing frames and interface selection algorithm implemented. The IEEE 802.16e introduces several MAC level queues for different access categories, maintaining service requirement within these queues; which imply that frames from a higher priority queue, i.e. video frames, are serviced more frequently than those belonging to lower priority queues. Such an enhancement at the MAC sub-layer introduces uneven queuing delays. Conventional routing protocols are unaware of such MAC specific constraints and as a result, these factors are not considered which result in channel performance degradation. To meet such challenges, the thesis presents FCS and TM schemes for WiMAX. For FCS, Its solution is to improve the mobile WiMAX handover and address the scanning latency. Since minimum scanning time is the most important issue in the handover process. This handover scheme aims to utilize the channel efficiently and apply such a procedure to reduce the time it takes to scan the neighboring access stations. TM uses MAC and physical layer (PHY) specific information in the interface metric and maintains a separate path to destination by applying an alternative interface operation. Simulation tests and comparisons with existing multi-homed protocols and handover schemes demonstrate the effectiveness of incorporating the medium dependent parameters. Moreover, show that suggested schemes, have shown better performance in terms of end-to-end delay and throughput, with efficiency up to 40% in specific test scenarios

Coding in 802.11 WLANs

Forward error correction (FEC) coding is widely used in communication systems to correct transmis- sion errors. In IEEE 802.11a/g transmitters, convolutional codes are used for FEC at the physical (PHY) layer. As is typical in wireless systems, only a limited choice of pre-speci¯ed coding rates is supported. These are implemented in hardware and thus di±cult to change, and the coding rates are selected with point to point operation in mind. This thesis is concerned with using FEC coding in 802.11 WLANs in more interesting ways that are better aligned with application requirements. For example, coding to support multicast tra±c rather than simple point to point tra±c; coding that is cognisant of the multiuser nature of the wireless channel; and coding which takes account of delay requirements as well as losses. We consider layering additional coding on top of the existing 802.11 PHY layer coding, and investigate the tradeo® between higher layer coding and PHY layer modulation and FEC coding as well as MAC layer scheduling. Firstly we consider the joint multicast performance of higher-layer fountain coding concatenated with 802.11a/g OFDM PHY modulation/coding. A study on the optimal choice of PHY rates with and without fountain coding is carried out for standard 802.11 WLANs. We ¯nd that, in contrast to studies in cellular networks, in 802.11a/g WLANs the PHY rate that optimizes uncoded multicast performance is also close to optimal for fountain-coded multicast tra±c. This indicates that in 802.11a/g WLANs cross-layer rate control for higher-layer fountain coding concatenated with physical layer modulation and FEC would bring few bene¯ts. Secondly, using experimental measurements taken in an outdoor environment, we model the chan- nel provided by outdoor 802.11 links as a hybrid binary symmetric/packet erasure channel. This hybrid channel o®ers capacity increases of more than 100% compared to a conventional packet erasure channel (PEC) over a wide range of RSSIs. Based upon the established channel model, we further consider the potential performance gains of adopting a binary symmetric channel (BSC) paradigm for multi-destination aggregations in 802.11 WLANs. We consider two BSC-based higher-layer coding approaches, i.e. superposition coding and a simpler time-sharing coding, for multi-destination aggre- gated packets. The performance results for both unicast and multicast tra±c, taking account of MAC layer overheads, demonstrate that increases in network throughput of more than 100% are possible over a wide range of channel conditions, and that the simpler time-sharing approach yields most of these gains and have minor loss of performance. Finally, we consider the proportional fair allocation of high-layer coding rates and airtimes in 802.11 WLANs, taking link losses and delay constraints into account. We ¯nd that a layered approach of separating MAC scheduling and higher-layer coding rate selection is optimal. The proportional fair coding rate and airtime allocation (i) assigns equal total airtime (i.e. airtime including both successful and failed transmissions) to every station in a WLAN, (ii) the station airtimes sum to unity (ensuring operation at the rate region boundary), and (iii) the optimal coding rate is selected to maximise goodput (treating packets decoded after the delay deadline as losses)

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

Error and Congestion Resilient Video Streaming over Broadband Wireless

In this paper, error resilience is achieved by adaptive, application-layer rateless channel coding, which is used to protect H.264/Advanced Video Coding (AVC) codec data-partitioned videos. A packetization strategy is an effective tool to control error rates and, in the paper, source-coded data partitioning serves to allocate smaller packets to more important compressed video data. The scheme for doing this is applied to real-time streaming across a broadband wireless link. The advantages of rateless code rate adaptivity are then demonstrated in the paper. Because the data partitions of a video slice are each assigned to different network packets, in congestion-prone wireless networks the increased number of packets per slice and their size disparity may increase the packet loss rate from buffer overflows. As a form of congestion resilience, this paper recommends packet-size dependent scheduling as a relatively simple way of alleviating the buffer-overflow problem arising from data-partitioned packets. The paper also contributes an analysis of data partitioning and packet sizes as a prelude to considering scheduling regimes. The combination of adaptive channel coding and prioritized packetization for error resilience with packet-size dependent packet scheduling results in a robust streaming scheme specialized for broadband wireless and real-time streaming applications such as video conferencing, video telephony, and telemedicine

A wireless multicast delivery architecture for mobile terminals

Content delivery over the Internet to a large number of mobile users offers interesting business opportunities for content providers, intermediaries, and access network operators. A user could receive, for example, music or a digital newspaper directly to a mobile device over wireless networks. Currently, content delivery over the Internet is held back by a number of reasons. Existing network technologies, such as GPRS, have a very limited capacity to transfer large files, such as those required for good-quality pictures in a newspaper. Another problem is security. Content received over the Internet is very vulnerable to being forged. A user who cannot be certain about the source and consistency of the received stock quotes is unlikely to pay for the information. Furthermore, content providers are unwilling to distribute their valuable information over the Internet due to their fear of copyright infringements. Traditionally, content has been considered consumed as soon as it has been downloaded. Content providers have been keen on preventing their content from being transferred over peer-to-peer networks because they consider the delivery itself to be a copyright infringement. In this dissertation, content delivery is separated from content consumption by encrypting the content before delivery. When the users wishes to consume the content, a license which includes the decryption key is provided. The architecture allows content to be delivered to users' devices even before the user commits to consume the content. The user can choose to receive content whenever downloading it is the most convenient and affordable. Thus, the content providers are able to maintain control over the use of their information even after the data has been transferred to the users' terminals. In addition, content received by users can be strongly source authenticated. The architecture allows secure, efficient and reliable delivery of content to a large group of receivers. The architecture does not commit itself to any specific delivery technique, and the content can be delivered using any delivery technique including multicast, broadcast, unicast, and peer-to-peer. This dissertation focuses mostly on multicast as the delivery technique. The efficiency of the multicast delivery over unreliable heterogenous wireless access networks is thoroughly analyzed. Mobile terminals can seamlessly switch between access points and access technologies while continuing to receive data reliably from the network. The multicast delivery uses adaptive error correction and retransmissions to deliver the content as efficiently as possible to a very large number of receivers. The simulations show, that the vast majority of receivers are able to receive the content reliably with a small delay even when the radio network suffers from high packet loss probability. Although the architecture is designed to deliver content to mobile terminals, it is also suitable for delivering content to terminals with fixed Internet connectivity.Digitaalisen sisällön siirtäminen liikkuville käyttäjille Internetin yli tarjoaa uusia liiketoimintamahdollisuuksia niin sisällöntuottajille, välittäjille kuin verkko-operaattoreille. Teknikkaa voidaan käyttää esimerkiksi musiikin tai sähköisten lehtien välittämiseen käyttäjille langattoman verkon kautta. Sisällön välittämistä Internetin kautta hankaloittaa yhä usea seikka. Nykyisin laajassa käytössä olevat verkkotekniikat, kuten GPRS, ovat liian hitaita siirtämään hyvin suuria tiedostoja suurelle määrällä vastaanottajia. Lisäksi väärennetyn tiedon välittäminen Internetin kautta on erittäin helppoa. Sisältö, jonka aitoudesta ja alkuperästä ei ole varmuutta, on usein arvotonta käyttäjälle. Sisällöntuottajat puolestaan ovat haluttomia käyttämään sisältönsä levittämiseen Internetiä mikäli digitaalisesti levitettävän sisällön kopioiminen ja oikeudeton kuluttaminen on liian helppoa. Perinteisesti sisältö ajatellaankin kulutetuksi jo sillä hetkellä, kun se on siirretty käyttäjän laitteeseen. Sen vuoksi sisällön tuottajat ovatkin käyttäneet paljon resursejaan estääkseen sisältönsä välittämisen vertaisverkoissa, koska jo pelkkää sisällön siirtämistä pidetään tekijänoikeusrikkomuksena. Tässä työssä erotetaan sisällön siirtäminen sisällön kuluttamisesta suojaamalla sisältö salauksella ennen sen siirtämistä käyttäjille ja sallimalla vapaa salatun sisällön jakelu. Arkkitehtuuri mahdollistaa sisällön siirtämisen käyttäjien laitteille silloin kun sisällön siirtäminen on edullisinta ja tehokkainta. Vasta käyttäjän halutessa kuluttaa aiemmin lataamaansa sisältöä, tarkistetaan oikeis sisällön käyttöön. Arkkitehtuuri mahdollistaa myös ladatun sisällön alkuperän ja eheyden vahvan tarkistamisen. Arkkitehtuuri mahdollistaa turvallisen, tehokkaan ja luotettavan sisällön siirtämisen suurelle määrälle vastaanottajia. Arkkitehtuuri ei pakota sisällön jakelua käyttämään mitään tiettyä siirtomenetelmää vaan sisältö voidaan siirtää käyttäen esimerkiksi ryhmälähetystä (multicast), joukkolähetystä (broadcast), täsmälähetystä (unicast) tai vertaisverkkoja (peer-to-peer). Tässä työssä on keskitytty analysoimaan ryhmälähetyksen soveltuvuutta tiedon siirtomenetelmänä. Ryhmälähetysmenetelmän tehokkuutta on analysoitu siirrettäessä sisältöä heterogeenisen langattoman liityntäverkon yli. Liikkuvat päätelaitteet voivat siirtyä saumattomasti liityntäverkosta toiseen samalla kun ne vastaanottavat sisältöä. Ryhmälähetys hyödyntää adaptiivista virheenkorjausta ja uudelleenlähetyksiä siirtääkseen sisällönmahdollisimman tehokkaasti suurelle joukolle vastaanottajia. Simulaatiot osoittavat, että erittäin suuri osa vastaanottajista saa sisällön luotettavasti ja pienellä viiveellä vaikka liityntäverkossa pakettien virhetodennäköisyys olisi suuri. Arkkitehtuuri on suunniteltu siirtämään sisältöä liikkuville laitteille, mutta sitä voidaan käyttää yhtä hyvin myös kiinteään verkkoon liitettyjen laitteiden kanssa.reviewe

Optimum hybrid error correction scheme under strict delay constraints

In packet-based wireless networks, media-based services often require a multicast-enabled transport that guarantees quasi error free transmission under strict delay constraints. Furthermore, both multicast and delay constraints deeply influence the architecture of erasure error recovery (EER). Therefore, we propose a general architecture of EER and study its optimization in this thesis. The architecture integrates overall existing important EER techniques: Automatic Repeat Request, Forward Error Correction and Hybrid ARQ techniques. Each of these EER techniques can be viewed as a special case of Hybrid Error Correction (HEC) schemes. Since the Gilbert-Elliott (GE) erasure error model has been proven to be valid for a wide range of packet based wireless networks, in this thesis, we present the general architecture and its optimization based on the GE channel model. The optimization target is to satisfy a certain target packet loss level under strict delay constraints efficiently. Through the optimization for a given real-time multicast scenario, the total needed redundancy information can be minimized by choosing the best HEC scheme automatically among the entire schemes included in the architecture. As a result, the performance of the optimum HEC scheme can approach the Shannon limit as closely as possible dynamically according to current channel state information.In Paket-basierten drahtlosen Netzwerken benötigen Medien-basierte Dienste oft Multicast-fähigen Transport, der quasi-fehlerfreie Übertragung unter strikten Zeitgrenzen garantiert. Außerdem beeinflussen sowohl Multicast als auch Zeitbegrenzungen stark die Architektur von Auslöschungs-Fehlerschutz (Erasure Error Recovery, EER). Daher stellen wir eine allgemeine Architektur der EER vor und untersuchen ihre Optimierung in dieser Arbeit. Die Architektur integriert alle wichtigen EER-Techniken: Automatic Repeat Request, Forward Error Correction und Hybrid ARQ. Jede dieser EER-Techniken kann als Spezialfall der Hybrid Error Correction (HEC) angesehen werden. Da das Gilbert-Elliot (GE) Auslöschungs-Fehler-Modell für einen weiten Bereich von Paket-basierten drahtlosen Netzwerken als gültig erwiesen wurde, präsentieren wir in dieser Arbeit die allgemeine Architektur und deren Optimierung basierend auf dem GE Kanalmodell. Zweck der Optimierung ist es, eine gewisse Ziel-Paketfehlerrate unter strikten Zeitgrenzen effizient zu erreichen. Durch die Optimierung für ein gegebenes Echtzeit-Mutlicast-Szenario kann die insgesamt benötigte Redundanz-Information minimiert werden. Dies erfolgt durch automatische Auswahl des optimalen HEC Schemas unter all den Schemata, die in die Architektur integriert sind. Das optimale HEC-Schema kann die Shannon Grenze so nahe wie möglich, dynamisch, entsprechend dem derzeitigen Kanalzustand, erreichen