Experimental Evaluation of Large Scale WiFi Multicast Rate Control

WiFi multicast to very large groups has gained attention as a solution for multimedia delivery in crowded areas. Yet, most recently proposed schemes do not provide performance guarantees and none have been tested at scale. To address the issue of providing high multicast throughput with performance guarantees, we present the design and experimental evaluation of the Multicast Dynamic Rate Adaptation (MuDRA) algorithm. MuDRA balances fast adaptation to channel conditions and stability, which is essential for multimedia applications. MuDRA relies on feedback from some nodes collected via a light-weight protocol and dynamically adjusts the rate adaptation response time. Our experimental evaluation of MuDRA on the ORBIT testbed with over 150 nodes shows that MuDRA outperforms other schemes and supports high throughput multicast flows to hundreds of receivers while meeting quality requirements. MuDRA can support multiple high quality video streams, where 90% of the nodes report excellent or very good video quality

Smart PIN: performance and cost-oriented context-aware personal information network

The next generation of networks will involve interconnection of heterogeneous individual networks such as WPAN, WLAN, WMAN and Cellular network, adopting the IP as common infrastructural protocol and providing virtually always-connected network. Furthermore, there are many devices which enable easy acquisition and storage of information as pictures, movies, emails, etc. Therefore, the information overload and divergent content’s characteristics make it difficult for users to handle their data in manual way. Consequently, there is a need for personalised automatic services which would enable data exchange across heterogeneous network and devices. To support these personalised services, user centric approaches for data delivery across the heterogeneous network are also required. In this context, this thesis proposes Smart PIN - a novel performance and cost-oriented context-aware Personal Information Network. Smart PIN's architecture is detailed including its network, service and management components. Within the service component, two novel schemes for efficient delivery of context and content data are proposed: Multimedia Data Replication Scheme (MDRS) and Quality-oriented Algorithm for Multiple-source Multimedia Delivery (QAMMD). MDRS supports efficient data accessibility among distributed devices using data replication which is based on a utility function and a minimum data set. QAMMD employs a buffer underflow avoidance scheme for streaming, which achieves high multimedia quality without content adaptation to network conditions. Simulation models for MDRS and QAMMD were built which are based on various heterogeneous network scenarios. Additionally a multiple-source streaming based on QAMMS was implemented as a prototype and tested in an emulated network environment. Comparative tests show that MDRS and QAMMD perform significantly better than other approaches

Multicast MAC extensions for high rate real-time traffic in wireless LANs

Nowadays we are rapidly moving from a mainly textual-based to a multimedia-based Internet, for which the widely deployed IEEE 802.11 wireless LANs can be one of the promising candidates to make them available to users anywhere, anytime, on any device. However, it is still a challenge to support group-oriented real-time multimedia services, such as video-on-demand, video conferencing, distance educations, mobile entertainment services, interactive games, etc., in wireless LANs, as the current protocols do not support multicast, in particular they just send multicast packets in open-loop as broadcast packets, i.e., without any possible acknowledgements or retransmissions. In this thesis, we focus on MAC layer reliable multicast approaches which outperform upper layer ones with both shorter delays and higher efficiencies. Different from polling based approaches, which suffer from long delays, low scalabilities and low efficiencies, we explore a feedback jamming mechanism where negative acknowledgement (NACK) frames are allowed from the non-leader receivers to destroy the acknowledgement (ACK) frame from the single leader receiver and prompts retransmissions from the sender. Based on the feedback jamming scheme, we propose two MAC layer multicast error correction protocols, SEQ driven Leader Based Protocol (SEQ-LBP) and Hybrid Leader Based Protocol (HLBP), the former is an Automatic Repeat reQuest (ARQ) scheme while the later combines both ARQ and the packet level Forward Error Correction (FEC). We evaluate the feedback jamming probabilities and the performances of SEQ-LBP and HLBP based on theoretical analyses, NS-2 simulations and experiments on a real test-bed built with consumer wireless LAN cards. Test results confirm the feasibility of the feedback jamming scheme and the outstanding performances of the proposed protocols SEQ-LBP and HLBP, in particular SEQ-LBP is good for small multicast groups due to its short delay, effectiveness and simplicity while HLBP is better for large multicast groups because of its high efficiency and high scalability with respect to the number of receivers per group.Zurzeit vollzieht sich ein schneller Wechsel vom vorwiegend textbasierten zum multimediabasierten Internet. Die weitverbreiteten IEEE 802.11 Drahtlosnetzwerke sind vielversprechende Kandidaten, um das Internet für Nutzer überall, jederzeit und auf jedem Gerät verfügbar zu machen. Die Unterstützung gruppenorientierter Echtzeit-Dienste in drahtlosen lokalen Netzen ist jedoch immer noch eine Herausforderung. Das liegt daran, dass aktuelle Protokolle keinen Multicast unterstützen. Sie senden Multicast-Pakete vielmehr in einer "Open Loop"-Strategie als Broadcast-Pakete, d. h. ohne jegliche Rückmeldung (feedback) oder Paketwiederholungen. In der vorliegenden Arbeit, anders als in den auf Teilnehmereinzelabfragen (polling) basierenden Ansätzen, die unter langen Verzögerungen, geringer Skalierbarkeit und geringer Effizienz leiden, versuchen wir, Multicast-Feedback bestehend aus positiven (ACK) und negativen Bestätigungen (NACK) auf MAC-Layer im selben Zeitfenster zu bündeln. Die übrigen Empfänger können NACK-Frames senden, um das ACK des Leaders zu zerstören und Paketwiederholungen zu veranlassen. Basierend auf einem Feedback-Jamming Schema schlagen wir zwei MAC-Layer-Protokolle für den Fehlerschutz im Multicast vor: Das SEQ-getriebene Leader Based Protocol (SEQ-LBP) und das Hybrid Leader Based Protocol (HLBP). SEQ-LBP ist eines Automatic Repeat reQuest (ARQ) Schema. HLBP kombiniert ARQ und paketbasierte Forward Error Correction (FEC). Wir evaluieren die Leistungsfähigkeit von ACK/NACK jamming, SEQ-LBP und HLBP durch Analysis, Simulationen in NS-2, sowie Experimenten in einer realen Testumgebung mit handelsüblichen WLAN-Karten. Die Testergebnisse bestätigen die Anwendbarkeit der Feedback-Jamming Schemata und die herausragende Leistungsfähigkeit der vorgestellten Protokolle SEQ-LBP und HLBP. SEQ-LBP ist durch seine kurze Verzögerung, seine Effektivität und seine Einfachheit für kleine Multicast-Gruppen nützlich, während HLBP auf Grund seiner hohen Effizienz und Skalierbarkeit im Bezug auf die Größe der Empfänger eher in großen Multicast-Gruppen anzuwenden ist

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

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

Service quality assurance for the IPTV networks

The objective of the proposed research is to design and evaluate end-to-end solutions to support the Quality of Experience (QoE) for the Internet Protocol Television (IPTV) service. IPTV is a system that integrates voice, video, and data delivery into a single Internet Protocol (IP) framework to enable interactive broadcasting services at the subscribers. It promises significant advantages for both service providers and subscribers. For instance, unlike conventional broadcasting systems, IPTV broadcasts will not be restricted by the limited number of channels in the broadcast/radio spectrum. Furthermore, IPTV will provide its subscribers with the opportunity to access and interact with a wide variety of high-quality on-demand video content over the Internet. However, these advantages come at the expense of stricter quality of service (QoS) requirements than traditional Internet applications. Since IPTV is considered as a real-time broadcast service over the Internet, the success of the IPTV service depends on the QoE perceived by the end-users. The characteristics of the video traffic as well as the high-quality requirements of the IPTV broadcast impose strict requirements on transmission delay. IPTV framework has to provide mechanisms to satisfy the stringent delay, jitter, and packet loss requirements of the IPTV service over lossy transmission channels with varying characteristics. The proposed research focuses on error recovery and channel change latency problems in IPTV networks. Our specific aim is to develop a content delivery framework that integrates content features, IPTV application requirements, and network characteristics in such a way that the network resource utilization can be optimized for the given constraints on the user perceived service quality. To achieve the desired QoE levels, the proposed research focuses on the design of resource optimal server-based and peer-assisted delivery techniques. First, by analyzing the tradeoffs on the use of proactive and reactive repair techniques, a solution that optimizes the error recovery overhead is proposed. Further analysis on the proposed solution is performed by also focusing on the use of multicast error recovery techniques. By investigating the tradeoffs on the use of network-assisted and client-based channel change solutions, distributed content delivery frameworks are proposed to optimize the error recovery performance. Next, bandwidth and latency tradeoffs associated with the use of concurrent delivery streams to support the IPTV channel change are analyzed, and the results are used to develop a resource-optimal channel change framework that greatly improves the latency performance in the network. For both problems studied in this research, scalability concerns for the IPTV service are addressed by properly integrating peer-based delivery techniques into server-based solutions.Ph.D

Survey of Transportation of Adaptive Multimedia Streaming service in Internet

[DE] World Wide Web is the greatest boon towards the technological advancement of modern era. Using the benefits of Internet globally, anywhere and anytime, users can avail the benefits of accessing live and on demand video services. The streaming media systems such as YouTube, Netflix, and Apple Music are reining the multimedia world with frequent popularity among users. A key concern of quality perceived for video streaming applications over Internet is the Quality of Experience (QoE) that users go through. Due to changing network conditions, bit rate and initial delay and the multimedia file freezes or provide poor video quality to the end users, researchers across industry and academia are explored HTTP Adaptive Streaming (HAS), which split the video content into multiple segments and offer the clients at varying qualities. The video player at the client side plays a vital role in buffer management and choosing the appropriate bit rate for each such segment of video to be transmitted. A higher bit rate transmitted video pauses in between whereas, a lower bit rate video lacks in quality, requiring a tradeoff between them. The need of the hour was to adaptively varying the bit rate and video quality to match the transmission media conditions. Further, The main aim of this paper is to give an overview on the state of the art HAS techniques across multimedia and networking domains. A detailed survey was conducted to analyze challenges and solutions in adaptive streaming algorithms, QoE, network protocols, buffering and etc. It also focuses on various challenges on QoE influence factors in a fluctuating network condition, which are often ignored in present HAS methodologies. Furthermore, this survey will enable network and multimedia researchers a fair amount of understanding about the latest happenings of adaptive streaming and the necessary improvements that can be incorporated in future developments.Abdullah, MTA.; Lloret, J.; Canovas Solbes, A.; García-García, L. (2017). Survey of Transportation of Adaptive Multimedia Streaming service in Internet. Network Protocols and Algorithms. 9(1-2):85-125. doi:10.5296/npa.v9i1-2.12412S8512591-