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1

Tamoor-ul-Hassan, Syed, and Serkan Demir. "Fast Packet Retransmissions in LTE." Thesis, Linköpings universitet, Kommunikationssystem, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-72166.

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The cellular networks are evolving to meet the future requirements of data rate,coverage and capacity. The fourth generation mobile communication system, LTEhas been developed to meet these goals. LTE uses multiple antenna features andlarger bandwidths in order to accomplish this task. These features will furtherextend the requirements of data rate, coverage, latency and flexibility. LTE also utilizes the varying quality of the radio channel and the interferencefrom other transmitters by adapting the data rate to the instantaneous channelquality at all the time. This is typically referred to as Link Adaptation. Thelink adaptation fails from time to time due to the varying channel quality as wellas the interference from other transmitters. In order to counteract these failures,retransmission methods are employed. These methods detect the errors on thereceiver side and signals the transmitter for the retransmission of the erroneousdata. The efficiency of link adaptation increases if combined with a properly designedretransmission scheme at the expense of delays due to retransmissions. This master thesis focuses on the study of the retransmission schemes with fasterfeedback, resulting in a reduction in delay. The feedback is generated by makingan early estimate of the decoding outcome and sending it early to the transmitterresulting in faster retransmission. This is important in certain applications wherethe data transmission is intolerant to delays.The thesis work shows by system performance simulations that fast packet retransmission,precisely called Early HARQ Feedback, significantly affects the systemperformance together with the utilization of the link adaptation. The study alsoshows that the link adaptation, in certain scenarios, can be optimized to improvethe system performance. In that respect, it is also possible to increase the numberof retransmissions within the same resource utilization. That optimization is basicallycalled aggressive link adaptation. Consequently, Early HARQ Feedback incombination with aggressive link adaptation provides a large improvement in thedownlink performance of the studied cases.
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2

De, Luca Niccolò. "Impact of retransmissions in IEEE 802.11bd." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2022.

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The main goal of this thesis is to study the impact of retransmissions in the upcoming IEEE 802.11bd standard and to determine an algorithm which can, on a vehicle to vehicle basis, activate them or not depending on the channel state, using the channel busy rate (CBR) as the leading metric. The study was based on simulations performed with the WiLabV2Xsim, which is an open source discrete event simulator that can be used to simulate communication between vehicles under the rules of different protocols.
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3

Siddiquee, Intekhaab Aalam. "Throughput optimization in retransmission-based communication systems." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1996. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/nq21947.pdf.

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4

Seddigh, Nabil. "Performance analysis of TCP's retransmission timeout mechanism." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2001. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/MQ61025.pdf.

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5

Davidson, Boris. "Fundamentals of the Simplex Communication Channel With Retransmissions." Diss., Virginia Tech, 1997. http://hdl.handle.net/10919/30471.

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The need for multiple access strategies arises whenever a number of users have to share a communication resource, since it is usually either cost prohibitive or impractical to dedicate a communication channel to a particular user. A need for such algorithms arises in many instances, particularly in applications utilizing wireless systems where all users access a common channel or medium. Such random access techniques as ALOHA and slotted ALOHA have been successfully implemented in a number of wireless applications. One of the major drawbacks of these algorithms is the necessity of a return path from the central station to each system user, which makes their use both inefficient and expensive for applications where one-way communication would suffice. For such applications, a need remained for a random access algorithm which can maximize the probability of successful message transmission in a one-way communication environment. A random access technique that addresses the above-mentioned need is developed. With this technique, each user sends an original message of predetermined length to a central receiver. The user then retransmits the message a specified number of times in a predetermined interval reserved for the retransmission process. The time interval between each successive retransmission of a given message is random. Assuming total annihilation of all colliding messages, the expression for the probability of successful transmission of a given message in terms of the major channel parameters is theoretically formulated. This technique offers a significant improvement, compared to a single transmission, in ensuring that a message is successfully received. The actual message collision dynamics in this system are experimentally studied using two different types of direct-sequence spread spectrum receivers, one employing a sliding correlator and the other using a matched filter. The spreading code in such systems offers extra protection for messages against possible interferers. The results indicate that it is often possible to properly receive a given message in the presence of co-channel interferers, thus significantly improving the overall system performance. These results are subsequently incorporated with the propagation data for several different types of microcells to arrive at a more precise theory of the link.
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6

Okada, Hiraku, Takeshi Sato, Takaya Yamazato, Masaaki Katayama, and Akira Ogawa. "CDMA Unslotted ALOHA Systems with Packet Retransmission Control." IEICE, 1996. http://hdl.handle.net/2237/7202.

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7

Abualhana, Munther, and Ubaid Tariq. "Improving QoE over IPTV using FEC and Retransmission." Thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-2654.

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IPTV (Internet Protocol Television), a new and modern concept of emerging technologies with focus on providing cutting edge high-resolution television, broadcast, and other fascinating services, is now easily available with only requirement of high-speed internet. Everytime a new technology is made local, it faces tremendous problems whether from technological point of view to enhance the performance or when it comes down to satisfy the customers. This cutting edge technology has provided researchers to embark and play with different tools to provide better quality while focusing on existing tools. Our target in dissertation is to provide a few interesting facets of IPTV and come up with a concept of introducing an imaginary cache that can re-collect the packets travelling from streaming server to the end user. In the access node this cache would be fixed and then on the basis of certain pre-assumed research work we can conclude how quick retransmission can take place when the end user responds back using RTCP protocol and asks for the retransmission of corrupted/lost packets. In the last section, we plot our scenario of streaming server on one side and client, end user on the other end and make assumption on the basis of throughput, response time and traffic.
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8

Gidlund, Mikael. "On packet retransmission diversity schemes for wireless networks /." Stockholm, 2004. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-388.

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9

Tajan, Romain. "Mécanismes de retransmission Hybrid-ARQ en radio-cognitive." Phd thesis, Université de Cergy Pontoise, 2013. http://tel.archives-ouvertes.fr/tel-00967013.

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Dans les standards actuels tels que HSDPA ou LTE, des protocoles de retransmissions (ARQ: Automatic Repeat reQuest) sont utilisés conjointement au codage de canal afin de palier aux erreurs dues à l'absence ou la mauvaise de connaissance de canal à la transmission. On garantit ainsi la fiabilité du lien physique pour les couches OSI supérieures (du moins un taux d'erreur paquet faible). De tels protocoles sont appelés protocoles de retransmission hybrides (HARQ). L'objet de cette thèse est de proposer des outils permettant l'analyse et l'optimisation des systèmes de communication en présences de protocoles HARQ avec une emphase particulière sur les systèmes cognitifs.Dans la première partie, nous étudierons un système point-à-point dans lequel trois différents protocoles HARQ adaptatifs seront considérés. Dans un premier temps, nous considérerons le régime asymptotique (i.e. codes optimaux gaussiens). Nous proposerons, dans ce cas, deux optimisations possibles : la minimisation de la puissance moyenne sous la contrainte de débit moyen et la maximisation du débit moyen sous une contrainte de puissance moyenne. Nous montrerons que les Processus de Décision Markoviens (MDP) sont des outils adaptés aux problèmes d'optimisation considérés.Dans les standards actuels tels que HSDPA ou LTE, des protocoles de retransmissions (ARQ: Automatic Repeat reQuest) sont utilisés conjointement au codage de canal afin de palier aux erreurs dues à l'absence ou la mauvaise de connaissance de canal à la transmission. On garantit ainsi la fiabilité du lien physique pour les couches OSI supérieures (du moins un taux d'erreur paquet faible). De tels protocoles sont appelés protocoles de retransmission hybrides (HARQ). L'objectif de cette thèse est de proposer des outils permettant l'analyse et l'optimisation des systèmes de communication en présences de protocoles HARQ avec une emphase particulière sur les systèmes cognitifs. La radio cognitive est une approche permettant à des utilisateurs non-licenciés de communiquer dans les mêmes bandes de fréquences que des utilisateurs licenciés afin d'augmenter l'efficacité spectrale des réseaux sans fil. Les utilisateurs secondaires doivent néanmoins limiter les interférences générées sur les signaux des utilisateurs primaires. Dans ce contexte, nous étudierons les débits atteignables par un utilisateur secondaire utilisant l'observation du protocole HARQ de l'utilisateur primaire afin de contrôler son interférence.
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10

Ngbogo, Valentin. "Le droit de retransmission télévisée des évènements sportifs." Versailles-St Quentin en Yvelines, 2007. http://www.theses.fr/2007VERS007S.

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Comment naissent les droits sur les événements sportifs, sur leur retransmission télévisée et qui les détient? Les réponses à ces questions principales ont montré la complexité du sujet. Dans les années 80 le nombre des chaînes de télévision a triplé. Les diffuseurs du secteur privé qui diffusent sur des chaînes payantes ont découvert dans le sport avec ses audiences élevées, le moyen d’attirer des nouveaux abonnés et des publicitaires. L’acquisition du droit de retransmission télévisée des événements sportifs devient alors le marché où les chaînes privées, les chaînes publiques et les agences privées se livrent une concurrence acharnée par tous les moyens et les surenchères vont faire flamber le prix. Au niveau national comme au niveau de la communauté européenne, les règles de la concurrence s’imposent à tous les protagonistes. Les autorités de concurrence, les juges français et communautaire vont veiller à ce qu’il en soit ainsi
How are born the rights on the sports events, on his broadcast and which detains them? The answers to these main questions showed the complexity of the subject. In the 80s the number of television channels tripled. The diffusers of the private sector who diffuse on paying chains discovered in the sport with its high audiences, the means to attract new subscribers and advertising agents. The acquisition of the right of broadcast of the sports events becomes then the market where private channels, public channels and private agencies are engaged a fierce competition by all the means and the overbids are going to make singe the price. At the level national as at the level of the European community, the rules of the competition impose upon all the protagonists. The authorities of competition, the French and community judges are going to watch over his being there so
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11

Francis, Breeson. "Enhancing TCP Congestion Control for Improved Performance in Wireless Networks." Thèse, Université d'Ottawa / University of Ottawa, 2012. http://hdl.handle.net/10393/23254.

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Transmission Control Protocol (TCP) designed to deliver seamless and reliable end-to-end data transfer across unreliable networks works impeccably well in wired environment. In fact, TCP carries the around 90% of Internet traffic, so performance of Internet is largely based on the performance of TCP. However, end-to-end throughput in TCP degrades notably when operated in wireless networks. In wireless networks, due to high bit error rate and changing level of congestion, retransmission timeouts for packets lost in transmission is unavoidable. TCP misinterprets these random packet losses, due to the unpredictable nature of wireless environment, and the subsequent packet reordering as congestion and invokes congestion control by triggering fast retransmission and fast recovery, leading to underutilization of the network resources and affecting TCP performance critically. This thesis reviews existing approaches, details two proposed systems for better handling in networks with random loss and delay. Evaluation of the proposed systems is conducted using OPNET simulator by comparing against standard TCP variants and with varying number of hops.
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12

Liu, Xiaoyan. "Adaptive resource allocation with packet retransmissions in OFDMA systems." Thesis, University of Kent, 2011. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.587567.

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In orthogonal frequency division multiple access (OFDMA) systems, the delay of packets has a great impact on quality of service (QoS), especially for real-time transmission. The basic concept of resource allocation in an OFDMA system is to allocate a subcarrier to the user with the best channel condition in that sub carrier. However, when a retransmission technique is used, the packet delay of successfully transmitting failed packets may be quite large in conventional retransmission schemes. In this paper, the packet combining technique in the downlink OFDMA system is introduced in conjunction with frequency diversity to enhance the reliability of retransmissions and thus reduce the large time delay caused by retransmissions. The novel retransmission technique aims to reduce the maximum number of retransmissions for failed packets, while maximizing the throughput. Bit error rate (BER) and transmit power are constraints in the throughput maximization formulation. A suboptimal algorithm is proposed, in which the retransmitted packets have higher priorities to be given resources than new packets. At the receiver, failed and retransmitted packets are combined before detection by using maximal ratio combining (MRC). It is shown that the proposed retransmission scheme can reduce the maximum packet delay significantly, while maintaining the maximum throughput of the conventional retransmission schemes.
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13

Anis, Muhammad Moiz. "Cooperative Retransmission of Broadcast Data Flows Via Cellular Networks." Télécom Bretagne, 2014. https://hal.science/tel-00978769.

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Parmi les services de distribution de contenu multimédia vers des récepteurs portatifs, la télévision mobile est un service populaire. Les réseaux de diffusion vidéo numérique (DVB) offrent à la fois une grande capacité et une bonne couverture pour effectuer une distribution massive de contenus multimédias. Les réseaux DVB conventionnels sont conçus pour couvrir des récepteurs fixes avec une antenne sur le toit. Les récepteurs portatifs ont des configurations différentes qui présentent un moins bon bilan de liaison. En règle générale, un système de diffusion est unidirectionnel et il n'a pas une voie de retour pour assurer la réception des données sur le côté du récepteur. D'autre part, les réseaux cellulaires ont des liens bidirectionnels plus fiables avec les récepteurs portatifs. Lorsqu'on effectue une transmission d'un contenu populaire en propre vers chaque récepteur portable, cela génère une énorme charge dans les réseaux cellulaires. Dans l'exemple spécifique de la télévision mobile, il n'est pas efficace de régénérer la même charge de transmission pour chaque récepteur portable. Dans notre travail de thèse, nous considérons une coopération entre les systèmes de diffusion et les réseaux cellulaires pour une réception multimédia efficace et de qualité au niveau des récepteurs portatifs. Nous utilisons la liaison réseau cellulaire pour la retransmission des paquets perdus dans un flux de données de diffusion multimédia. Dans ce travail, nous analysons la couverture pour les récepteurs portatifs dans un réseau isofréquence de diffusion. Nous considérons spécifiquement plusieurs scénarios intérieurs et extérieurs dans un réseau DVB-T2. Nous analysons comment un flux de données est traité en DVB-T2 et de proposer un système pour l'identification des paquets dans la transmission DVB. La contribution de base de notre travail de thèse est la proposition de réparation d'un flux en temps réel (RFR) service basé sur le réseau cellulaire, qui répare les flux de données multimédia pour les récepteurs portatifs en temps réel. Le service RFR proposé est basé sur un protocole client-serveur, i. E. Contraint Application Protocol (CoAP). La proposition RFR est également étayée par l'analyse de la charge générée dans le réseau d'accès radio LTE a cause de service RFR
Among the multimedia content distribution services to the handheld receivers, mobile TV is a popular service. Digital Video Broadcast (DVB) networks have high capacity and coverage to perform a mass multimedia content distribution. The conventional DVB networks are designed to cover fixed receivers, which have a roof antenna. The handheld receivers have different configurations which cause weaker link budgets. Generally, a broadcast system is unidirectional and it does not have a feedback channel to ensure the reception of the data at the receiver side. On the other hand, cellular networks have more reliable bidirectional links with the handheld receivers. Individual multimedia transmission to each handheld receiver generates a huge load in the cellular networks. In the specific example of mobile TV, it is not efficient to regenerate the same transmission load for each handheld receiver. In our thesis work, we consider a cooperation between the broadcast systems and the cellular networks for an efficient multimedia reception at the handheld receivers. We use the cellular network link for the retransmission of the lost packets from a multimedia broadcast data-flow reception. In this work we analyze the coverage for handheld receivers in a single frequency broadcast network. We specifically consider several outdoor and indoor scenarios in a DVB-T2 network. We analyze how a data flow is processed in DVB-T2 and propose a scheme for the identification of packets in the DVB transmission. The core contribution of our thesis work is the proposition of a Real-time Flow Repair (RFR) Service based on cellular network, which repairs any multimedia data-flow to the handheld receivers in real-time. The proposed RFR service is based on a light client-server application protocol, namely Constrained Application Protocol (CoAP). The RFR proposal is also supported by the analysis of the load generated in LTE Radio access network due to RFR service.
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Abdourazakou, Yann. "La régulation des droits sportifs de retransmission dans l' Union européenne : efficacité ou équité? : vers une économie politique des droits sportifs de retransmission." Rouen, 2004. http://www.theses.fr/2004ROUEL487.

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Ancrée dans le champ de l’économie politique du sport professionnel, la thèse aspire à comprendre la politique de régulation des droits sportifs de retransmission dans l’Union européenne. En se référant au droit de la concurrence et aux théories de la justice sociale, notre objectif est d’identifier les arguments d’une intervention à travers deux critères traditionnels des politiques de régulation : l’efficacité et l’équité. La structure d’organisation des marchés des droits sportifs repose sur des bases anticoncurrentielles en mesure d’affecter l’intérêt des consommateurs. La protection des évènements majeurs par un système de listes nationales relève quant à elle d’un patrimoine culturel immatériel. Un dernier argument concerne la présence d’externalités positives justifiant un accès en clair. La politique de régulation révèle la double nature des droits sportifs. Biens économiques et « non économiques », ils nécessitent la combinaison d’une régulation de nature économique (efficacité) et culturelle (équité)
Dealing with political economics of professional sport, this thesis is willing to understand the regulation of sports broadcasting rights in the European Union. We refer to competition law and social justice theories to identify the reasons of an intervention through two classical criteria: efficiency and equity. Sports rights markets reveal an anticompetitive structure potentially affecting consumers’ welfare. National listing events refer to an immaterial cultural patrimony. One last argument relies on the existence of positive externalities justifying a free to air access. The regulation of sports rights markets illustrates the “economic” and “non economic” nature of sports rights that justifies a combination of an economic and cultural regulation
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15

Selig, Marc [Verfasser]. "Interference Mitigation with Selective Retransmissions in Wireless Sensor Networks / Marc Selig." Kassel : Kassel University Press, 2016. http://d-nb.info/1119922429/34.

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16

Gendreau, Ysolde 1962. "The retransmission right : copyright and the diffusion of works by cable." Thesis, McGill University, 1988. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=63946.

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17

Kirwan, John R. "Analysis of a data communication network's performance under varying retransmission disciplines." Thesis, Monterey, California. Naval Postgraduate School, 1990. http://hdl.handle.net/10945/30732.

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Approved for public release, distribution unlimited
A stochastic simulation model is developed, using the SLAM II simulation language, to study the dynamics and performance of a small data communication network. The simulation program models pertinent aspects of Defense Data Network (DDN) protocols. The effect of changes in node-to-node and host-to-host retransmission timeout intervals upon expected response time is studied using the model.
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Shakya, Rosish. "Optimal Placement of Video Caching Routers for Minimization of Retransmission Delay." University of Akron / OhioLINK, 2011. http://rave.ohiolink.edu/etdc/view?acc_num=akron1306429551.

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19

Kirwan, John R. "Analysis of a data communication networks' performance under varying retransmission disciplines." Monterey, California : Naval Postgraduate School, 1990. http://handle.dtic.mil/100.2/ADA239945.

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Thesis (M.S. in Operations Research)--Naval Postgraduate School, September 1990.
Thesis Advisor(s): Gaver, D. P. Second Reader: Jacobs, P. A. "September 1990." Description based on title screen as viewed on March 18, 2010. DTIC Identifier(s): Network Simulation, Data Networks, Program Listings, Theses. Author(s) subject terms: DDN, Network Simulation, Data Communication Network. Includes bibliographical references (p. 127). Also available in print.
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Döser, Erman. "Tradeoffs between retransmission and forward error correction in the RTP stack." Thesis, Linköpings universitet, Institutionen för datavetenskap, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-160855.

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Video conferencing applications has reached worldwide usage in recent years by the help of the improvements in network infrastructures for public services. Media data covers a significant ratio of data traffic over IP networks. However, it is challenging to ensure a decent quality of service (QoS) on public networks in terms of video and audio quality. The main factor that may cause degradation in media playback quality is packet losses. There are various techniques available to conceal packet losses in lossy channels. According to the application needs and channel characteristics such as loss patterns and round trip times, retransmission or forward error correction techniques may be applied at application level. These two techniques have different challenges which lead to tradeoffs between them, thus one might be chosen over the others. In this thesis work, retransmission’s worst case performances under considered packet loss patterns and various round trip times are compared to performances of forward error correction schemes. In addition, implementation details with respect to the relevant RFCs are provided as an example to give a better judgement on the obtained results. Results obtained under the packet loss patterns that are generated with a simple Gilbert-Elliot 2-state model shows that forward error correction techniques are a reasonable choice of error concealing in the real-time transport protocol (RTP) stack where round trip time in the channel is greater than 200 ms. In addition, bandwidth overhead revealed by forward error correction stays higher than retransmission’s bandwidth overhead in all sample runs. In cases where round trip times are high, then the choice of forward error correction scheme is bound to the packet loss pattern. In the results section, it is obtained that ReedSolomon performs well in terms of residual packet losses, which are the packets not being recovered, and bandwidth overhead when losses occur in long bursts.
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Zhang, Dan. "Impulse noise detection techniques for retransmission to reduce delay in DSL systems." Thesis, McGill University, 2012. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=107853.

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To protect Digital Subscriber Line (DSL) systems against impulse noise (IN), interleaving combined with Reed-Solomon (RS) coding is typically deployed in the conventional DSL standards. However, interleaving introduces a long delay. To reduce such delay in conventional DSL systems that are corrupted by IN, retransmission can be used instead of interleaving. For an effective retransmission, reliable detection of corruption due to IN is required. In this thesis, we consider three detection approaches. The first one is based on the RS decoding status since the decoder either detects the number of corrected errors or reports the failure of decoding when the errors exceed its correction capability. Retransmission is required when the transmitted codeword cannot be decoded. The second one uses the square distance method in which erasures are marked for unreliably received samples and retransmission is issued when the number of erased samples exceeds a certain threshold. Finally, the third one takes advantage of the unused tones in DSL systems in order to detect whether IN is present. For all the above approaches, we analyze the average retransmission delay and bit error rate (BER) and provide simulation results to validate the analysis. It is found that the "Decoding Status" approach can reliably indicate received signals corrupted by IN. We consider it a trustful way to correct the symbols and detect the errors since the probability of wrong decoding for the received symbol is very low. In the frequency selective fading channel and with the presence of Repetitive Electrical Impulse Noise (REIN), numerical results using different parameters such as various channel responses and IN powers show that retransmission offers a short delay while effectively avoiding transmission errors. Specifically, with the "Decoding Status" approach, the error caused by REIN can be completely avoided with the average retransmission delay of around 0.029ms and the maximum round-trip delay of 0.75ms.
Pour protéger les systèmes de ligne d'abonné numérique (DSL) contre le bruit impulsif (IN), les normes conventionnelles de DSL sont typiquement déployées avec de l'entrelacement combiné au codage Reed-Solomon (RS). Cependant, l'entrelacement introduit un long retard. Afin de réduire ce retard dans les systèmes conventionnels de DSL qui sont corrompus par l'IN, une retransmission peut être utilisée au lieu de l'entrelacement. Pour une retransmission efficace, une détection fiable de l'altération causée par l'IN est nécessaire. Dans cette thèse, nous considérons trois approches de détection. La première est basée sur le statut du décodage RS puisque le décodeur détecte le nombre d'erreurs corrigées ou bien signale qu'il a échoué quand le nombre d'erreurs dépassent sa capacité de correction. Une retransmission est nécessaire quand le mot codé transmis ne peut pas être décodé. La deuxième méthode emploie la méthode des distances carrées dans laquelle les suppressions sont marquées pour les échantillons reçus non fiables et une retransmission est émise quand le nombre d'échantillons supprimés dépasse un certain seuil. Enfin, la troisième méthode est de profiter des tonalités inutilisées dans les systèmes de DSL afin de détecter si l'IN est présent. Pour toutes les approches ci-dessus, nous analysons le retard moyen et le taux d'erreur sur les bits (BER) et nous fournissons des résultats de simulation pour valider l'analyse. Il est constaté que l'approche du statut de décodage peut indiquer de manière fiable les signaux reçus corrompus par IN et nous le considérons comme une manière efficace pour corriger les symboles et pour détecter les erreurs puisque la probabilité de mauvais décodage pour le symbole reçu est très basse. Dans la voie sujette à évanouissement progressif de fréquences avec la présence de bruit électrique impulsif répétitif (REIN), les résultats numériques utilisant des paramètres différents tels que de diverses réponses de voies de transmission et de diverses puissances du IN prouvent que la retransmission offre un court retard tout en évitant efficacement les erreurs de transmission. Spécifiquement, avec l'approche du statut de décodage, l'erreur provoquée par le REIN peut complètement être évitée avec un délai de retransmission moyennant 0.029ms et un retard aller-retour maximale est de 0.75ms.
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Vivier, Guillaume. "Retransmission hybride : une solution efficace pour TCP dans un environnement sans fil." Paris 6, 2003. http://www.theses.fr/2003PA066597.

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23

Swaro, James E. "A Heuristic-Based Approach to Real-Time TCP State and Retransmission Analysis." Ohio University / OhioLINK, 2015. http://rave.ohiolink.edu/etdc/view?acc_num=ohiou1448030769.

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24

Khreis, Alaa. "Cross-layer optimization of cooperative and coordinative schemes for next generation cellular networks." Thesis, Université Paris-Saclay (ComUE), 2018. http://www.theses.fr/2018SACLT011/document.

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Les demandes de haut débit, faible latence et grande fiabilité augmentent dans les nouvelles générations de systèmes de radiocommunications. Par conséquent, on propose de combiner la transmission non orthogonale avec les retransmissions HARQ afin de combattre les fluctuations de canal de transmission à haut débit. Dans la première partie de la thèse, on propose des protocoles de retransmissions HARQ avec l'aide d'un relai afin d'améliorer le débit et la fiabilité du système. Une version renforcée du protocole HARQ qui prend en compte le délai de retour est proposée dans la seconde partie de la thèse
HARQ has become an important research field in the wireless digital communications area during the last years. In this thesis, we improve the HARQ mechanisms in terms of throughput and/or latency which are the bottleneck of next generation wireless communication systems. More precisely, we improve the time-slotted HARQ systems by mimicking NOMA, which means using superposed packets in a single-user context. In the first part of the thesis, we propose HARQ protocols using the help of a relay to improve the transmission rate and reliability. An enhanced HARQ protocol adapted to delayed feedback is proposed in the second part. In this new multi-layer HARQ protocol, additional redundant packets are sent preemptively before receiving the acknowledgement, and in superposition to other HARQ processes
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25

Jin, Xiangyang. "Multi-retransmission Route Discovery Schemes for Ad Hoc Wireless Network with a Realistic Physical Layer." Thèse, Université d'Ottawa / University of Ottawa, 2011. http://hdl.handle.net/10393/20254.

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During the route discovery process, each node receiving the route request packet (RReq) will retransmit it exactly once. A distant neighbor may accidentally receive/loose the only RReq and use it to announce a new route, although that link is inferior/superior for route reply packets (RRep) or actual message routing. Overall, the constructed route may be far from the optimal. All existing route discovery schemes (including DSR/AODV) apply retransmission during route discovery exactly once (1R). Based on a realistic physical layer model, we propose two new route discovery schemes: n-retransmission (nR, retransmitting exactly n times) and n-retransmission c-reception (ncRR), retransmitting until we either reach a total of n own retransmissions or c copies from neighbors are heard. We compare our two new scheme with the traditional one, under otherwise identical conditions (same metric, same packet reception probability on each link) and the same choices about possibly retransmitting again upon discovering a better route (R+) or discarding it (R1), generating route reply packet for every received RRep (B*), or for first and better discovered routes only (B2), and retransmitting RRep exactly once (A1), up to a maximum of three times (A3), or optimally u times decided by link quality (Au). Experimental results show that the proposed ncRR scheme (for n=2 and c=3 or c=4) achieves the best tradeoff between quality of route, success rate and message overhead in the route discovery process, followed by the nR scheme, and both of them are superior to the existing traditional schemes.
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26

Kurtin, Matthew Seth. "Verification and implementation of a multiple retransmission technique for a one-way asynchronous communication channel." Thesis, Virginia Tech, 1996. http://hdl.handle.net/10919/45090.

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The subject of this thesis was the verification and the implementation of a multiple retransmission technique for the one-way asynchronous communication channel of the Interactive Video Data Service (IVDS) system. The commercial application of the IVDS project made reliability of the retransmission technique important. Verification of the reliability of the technique was done using a simulator written in the C programming language. The results of the IVDS simulator were compared to the existing statistical model of the transmission channel. Following a favorable comparison of the statistical model and simulator results, a prototype of the control section for the IVDS Consumer Control Unit (CCU) was designed and constructed. This was an important step towards the completion of the IVDS system. The CCU design defined protocols for message format and error detection and also helped to establish an IVDS test-bed system.


Master of Science
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27

Cheng, Yuelong, and Xiaoying Ma. "Cyclic Redundancy Check for Zigbee-Based Meeting Attendance Registration System." Thesis, Högskolan i Gävle, Avdelningen för elektronik, matematik och naturvetenskap, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:hig:diva-12629.

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The research accomplished in this dissertation is focused on the design of effective solutions to the problem that error codes occur in the ZigBee-based meeting attendance registration system. In this work, several different check algorithms are compared, and the powerful error-detecting Cyclic Redundancy Check (CRC) algorithm is studied. In view of the features of the meeting attendance registration system, we implement the check module of CRC-8. This work also considers the data reliability. We assume use retransmission mechanism to ensure the validity and completeness of transmission data. Finally, the potential technical improvement and future work are presented.
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28

Tamaro, Normand. "L'effet de l'absence de controverse doctrinale sur le droit d'auteur, le cas du droit de retransmission." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1995. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/nq21518.pdf.

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29

Livadas, Carolos. "Formally modeling, analyzing, and designing network protocols : a case study on retransmission-based reliable multicast protocols." Thesis, Massachusetts Institute of Technology, 2003. http://hdl.handle.net/1721.1/87451.

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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2003.
Includes bibliographical references (p. 247-250).
In this thesis, we conduct an extensive case study on formally modeling, analyzing, and designing retransmission-based reliable multicast protocols. We first present an abstract model of the communication service that several reliable multicast protocols [12, 13, 32-34] strive to provide. This model precisely specifies i) what it means to be a member of the reliable multicast group, ii) which packets are guaranteed delivery to which members of the group, and iii) how long it takes for a packet to be reliably multicast to the appropriate members of the reliable multicast group. We then model and analyze the correctness and performance of three retransmission-based reliable multicast protocols, namely the Scalable Reliable Multicast (SRM) protocol [12, 13], the novel Caching-Enhanced Scalable Reliable Multicast (CESRM) protocol [24], and the Light-weight Multicast Services (LMS) router-assisted protocol [32-34]. We show the each such protocol is correct by proving that it is a faithful implementation of our reliable multicast service model. These correctness proofs ensure the equivalence of the protocols in the sense that they guarantee the delivery of the same packets to the same members of the reliable multicast group. Under some timeliness assumptions and presuming a fixed number of per-recovery packet drops, we show that our model of SRM guarantees the timely delivery of packets. Our timeliness analysis of SRM reveals that the careless selection of SRM's scheduling parameters may introduce superfluous recovery traffic and may undermine the loss recovery process. This is an important observation that has, to date, been overlooked.
CESRM augments SRM with a caching-based expedited recovery scheme that exploits packet loss locality in IP multicast transmissions by attempting to recover from losses in the manner in which recent losses were recovered. We analytically show that the worst-case recovery latency for successful expedited recoveries in CESRM is roughly 1 round-trip time (RTT) where as that of successful first-round recoveries in SRM is 4 RTT (for typical scheduling parameter settings). Moreover, trace-driven simulations, which exhibit the packet loss locality of actual IP multicast transmissions, reveal that CESRM reduces the average recovery latency of SRM by roughly 50% and incurs less overhead in terms of recovery traffic. Finally, although LMS recovers promptly from packets in static membership and topology environments, we demonstrate several dynamic scenarios in which LMS does not perform well. Thus, CESRM is a preferable reliable multicast protocol to both SRM and LMS; CESRM inherits SRM's robustness to dynamic environments and, thanks to its caching-based expedited recovery scheme, drastically reduces the average recovery latency of SRM in static environments.
by Carolos Livadas.
Ph.D.
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30

Sonono, Tofik. "Interoperable Retransmission Protocols with Low Latency and Constrained Delay : A Performance Evaluation of RIST and SRT." Thesis, KTH, Kommunikationssystem, CoS, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-254897.

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The media industry has during the last decade migrated services from dedicated medianetworks to more shared resources and lately also the public internet and public data centers. Inorder to cater for such transition, several protocols have been designed to meet the demand forhigh-quality media transport over lossy infrastructure, protocols such as SRT and RIST. Thepurpose of Reliable Internet Stream Transport (RIST) and Secure Reliable Transport (SRT) is tohave all vendors of broadcasting equipment support an interoperable way of communication. Thelack of interoperability locks consumers into one particular vendor’s family of products - most oftenthis equipment only supports a proprietary technology. Interoperability creates a more competitivemarket space which benefits consumers and gives vendors an incentive to be more innovative intheir solutions. The purpose of this thesis is to assess the performance of these protocols by comparing theirperformance to a proprietary solution (named ÖÖÖ in this thesis and seen as an establishedsolution in the industry). The challenge is to test these protocols in a lab environment, but have theresults represent real-world use. For this, a large subset of samples is needed along with samplesmeasured over a long period. This sampling was made possible by writing a script which automatesthe sampling process. The results indicate that the versions of RIST and SRT tested in this thesis to some extentcompare well to the selected established protocol (ÖÖÖ). In many scenarios, SRT even did muchbetter, mainly when a line with a single feed was tested. For instance, when the network suffered a2% drop rate and utilized retransmission SRT performed the best and was the only protocol whichhad some samples where no packets were dropped during one hour of measurements. Whenrunning all three protocols at the same time, SRT also did the best in a network with up to 12% droprate. The results in this thesis should give a broadcaster an idea of which of these protocols willfulfill their requirements in a broadcast application.
I mediabranschen finns det en efterfrågan på utrustning som har inslag av interoperabilitet.Anledningen till detta är att någon som köper produkter från en viss återförsäljare inte vill låsas in idenna återförsäljares ”ekosystem” i flera år framöver. Då en studio sällan uppgraderar hela sinproduktionskedja på samma gång ger interoperabilitet möjligheten att köpa utrustning från andraåterförsäljare när man ska uppgradera något i produktionslinan. Detta leder till en merkonkurrenskraftig marknad samt ger incentiv till nya innovativa lösningar. Detta examensarbete går ut på att utvärdera lösningar som tagits fram för att främjainteroperabilitet och jämföra dem med en existerande proprietärlösning. Reliable Internet StreamTransport (RIST) och Secure Reliable Transport (SRT) är två protokoll som tagits fram för just dettasyfte. Utmaningen med att utvärdera dessa protokoll är att i en labbmiljö få resultat som reflekteraranvändandet av protokollen i verkligheten. Detta har gjorts med hjälp av ett program som tagitsfram i detta examensarbete. Med detta program har testandet kunnat automatiseras. Resultaten i detta examensarbete visar potential hos båda RIST och SRT. SRT är i vissascenarion till och med bättre än den proprietära lösningen. Protokollen visar något buggigtbeteende i vissa instanser, såsom att i vissa fal sluta fungera och inte kunna återgå till normalfunktion utan manuell interaktion. Allt som allt är dock protokollen i de flesta fallen testade i dettaexamensarbete ett godtyckligt alternativ till den jämförda proprietära lösningen.
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31

Panhuys, Bertrand. "Spectacle sportif, droits de retransmission et équilibre compétitif : le choix d'un modèle d'organisation : l'exemple du basket professionnel." Limoges, 2008. http://www.theses.fr/2008LIMO1008.

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L'arrivée des médias dans l'univers sportif a modifié les mécanismes de marchés et les habitudes de consommation. En transformant les outils de production et en créant de nouveaux débouchés (multiplication des chaînes câblées ou numériques), les grands groupes de communication ont fait du sport un produit plus télégénique et un vecteur de communication qui attire de nombreux partenaires financiers. Mais, cette évolution a également entraîné une modification des relations entre médias et instances sportives, rendues plus complexes en raison notamment de la multiplication des événements sportifs et de l'explosion du montants des droits de retransmission. Face au pouvoir de ces intervenants qui cherchent à maîtriser complètement le processus de production allant de l'origine jusqu'au produit final, la gouvernance du sport est mise en péril, de même que l'équilibre compétitif que les organisations tentent de préserver. Ainsi, il apparaît nécessaire de redimensionner le (s) modèle (s) d'organisation pour continuer à garantir les principes d'équité et d'incertitude propres au sport. Le basket qui a l'image d' un sport jeune et dynamique constitue une bonne ilustration de cette problématique et permet de confronter plusieurs systèmes
The arrival of the media in sports activity has modified market mechanisms and consumer habits. By transforming production means and creating new outlets (increase in cable and digital TV network), mass media firms have changed sport into a TV product and a vector of communication which attracts many more financial partners. This evolution has led to a change in the relationship between the media and the sports organizations, becoming more complex because of the increase in sport events and increases coasts of broadcasing rights. Faced with the power of these new actors who want to manage the whole supply chain, from the product's creation to its distribution, sport as a whole is in danger, as is the competitive sport balance that the organizations try to preserve. It is thus necessary to consider new management models to guarantee the principles of equity and uncertainty unique to sport. Basketball, which has a modern and dynamic image, is a good imustration of this kind of problem and a way to show the differences between several production systems of sports entertainment
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32

Xiong, Lixiang. "A Markov Chain Approach to IEEE 802.11WLAN Performance Analysis." University of Sydney, 2008. http://hdl.handle.net/2123/3679.

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Doctor of Philosopy (PhD)
Wireless communication always attracts extensive research interest, as it is a core part of modern communication technology. During my PhD study, I have focused on two research areas of wireless communication: IEEE 802.11 network performance analysis, and wireless cooperative retransmission. The first part of this thesis focuses on IEEE 802.11 network performance analysis. Since IEEE 802.11 technology is the most popular wireless access technology, IEEE 802.11 network performance analysis is always an important research area. In this area, my work includes the development of three analytical models for various aspects of IEEE 802.11 network performance analysis. First, a two-dimensional Markov chain model is proposed for analysing the performance of IEEE 802.11e EDCA (Enhanced Distributed Channel Access). With this analytical model, the saturated throughput is obtained. Compared with the existing analytical models of EDCA, the proposed model includes more correct details of EDCA, and accordingly its results are more accurate. This better accuracy is also proved by the simulation study. Second, another two-dimensional Markov chain model is proposed for analysing the coexistence performance of IEEE 802.11 DCF (Distributed Coordination Function) and IEEE 802.11e EDCA wireless devices. The saturated throughput is obtained with the proposed analytical model. The simulation study verifies the proposed analytical model, and it shows that the channel access priority of DCF is similar to that of the best effort access category in EDCA in the coexistence environment. The final work in this area is a hierarchical Markov chain model for investigating the impact of data-rate switching on the performance of IEEE 802.11 DCF. With this analytical model,the saturated throughput can be obtained. The simulation study verifies the accuracy of the model and shows the impact of the data-rate switching under different network conditions. A series of threshold values for the channel condition as well as the number of stations are obtained to decide whether the data-rate switching should be active or not. The second part of this thesis focuses on wireless cooperative retransmission. In this thesis, two uncoordinated distributed wireless cooperative retransmission strategies for single-hop connection are presented. In the proposed strategies, each uncoordinated cooperative neighbour randomly decide whether it should transmit to help the frame delivery depending on some pre-calculated optimal transmission probabilities. In Strategy 1, the source only transmits once in the first slot, and only the neighbours are involved in the retransmission attempts in the subsequent slots. In Strategy 2, both the source and the neighbours participate in the retransmission attempts. Both strategies are first analysed with a simple memoryless channel model, and the results show the superior performance of Strategy 2. With the elementary results for the memoryless channel model, a more realistic two-state Markov fading channel model is used to investigate the performance of Strategy 2. The simulation study verifies the accuracy of our analysis and indicates the superior performance of Strategy 2 compared with the simple retransmission strategy and the traditional two-hop strategy.
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33

Karayannis, Vassilios-Petros. "Liberté économique et défense de l'intérêt général: le problème de retransmission par câble des émissions télévisées dans l'Union européenne." Doctoral thesis, Universite Libre de Bruxelles, 2004. http://hdl.handle.net/2013/ULB-DIPOT:oai:dipot.ulb.ac.be:2013/211205.

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La thèse aborde le droit européen de la télévision sous deux aspects :les régimes administratifs d’accès des émissions télévisées aux réseaux câblés et les droits intellectuels.

En ce qui concerne l’accès des émissions aux réseaux câblés, la thèse met en avant le besoin de sauvegarder un service public de l’audiovisuel. Celui-ci est défini comme un ensemble des règles qui visent à la fois le paysage audiovisuel propre à chaque Etat membre (par exemple pluralisme) et le contenu des émissions proprement dit (émissions informatives, éducatives, épanouissement culturel etc.). Le droit communautaire primaire et dérivé, tel qu’interprété par la Cour de justice, fournit les moyens de conciliation entre, d’une part les intérêts généraux et, d’autre part, les exigences découlant de la libre prestation de services et de la libre concurrence.

En ce qui concerne l’application des droits intellectuels, la thèse aborde la problématique liée à l’épuisement ou la subsistance de ceux-ci. Dans le cas de la câblodistribution, la Cour a affirmé la subsistance du droit. Cette position est corroboré par la nouvelle directive européenne sur le droit d’auteur et les droits voisins dans la société de l’information. La thèse appuie la position de subsistance en considérant qu’elle constitue une condition essentielle pour la juste récompense des auteurs.

Enfin, la thèse aborde les questions plus spécifiques qui naissent à propos de la convergence technologique et juridique. Tout d’abord, il est avancé que le service public de l’audiovisuel n’est pas uniquement lié à des contraintes techniques, mais essentiellement à des objectifs qualitatifs (contenu des émissions). Ainsi, la thèse plaide en faveur de la pérpetuité du service public de l’audiovisuel dans l’ère du numérique. Par ailleurs, des questions plus spécifiques (comme l’accès à la boucle locale, l’interconnexion des réseaux et la numérisation des infrastructures) ont été examinées.


Doctorat en droit
info:eu-repo/semantics/nonPublished

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34

Wehbe, Houssein. "Transmission de flux vidéo en direct sur les réseaux pair-à-pair : optimisation de l'overlay et de la retransmission." Rennes 1, 2011. http://www.theses.fr/2011REN1S073.

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Actuellement, le service de distribution d'un flux vidéo en direct sur le réseau Internet est de plus en plus sollicité. L’utilisation de l’approche pair-à-pair pour assurer ce service a facilitée le passage à l’échelle. En même temps que le client consomme la vidéo, il peut utiliser ses propres ressources pour servir d'autres clients, augmentant ainsi la capacité du système. La garantie de la qualité de la vidéo représente le challenge principal de ce service. Ceci est dû aux caractéristiques du réseau Internet et aux comportements indéterministes des clients pouvant quitter le système à n’importe quel instant. Le but de cette thèse est d’étudier les challenges de distribution de vidéo en direct sur Internet. Nous avons mené avons mené deux contributions. Premièrement, nous avons proposé un système pair-àpair de distribution de vidéo en direct appelé P-QOLS. Il assurer une continuité d’affichage tout en réduisant autant que possible le différé et le délai d’initialisation. Nos résultats des simulations ont montré l’efficacité de P-QOLS et sa capacité à supporter un éventail de scénarios où les clients sont dynamiques et les ressources distribuées de façon hétérogène. Deuxièmement, nous avons proposé une technique de retransmission des paquets perdus. Elle consiste à demander ces paquets depuis un retransmetteur différent que leur fournisseur original. Ceci réduit la probabilité de perdre les paquets retransmis. Nos résultats ont montré que cette technique est susceptible d'être appliquée dans chaque système pair-à-pair et ce sans imposer de nouvelles contraintes sur les algorithmes utilisés par ce système
Live Streaming consists in distributing live media (video and audio) to large audiences over a computer network. Recently, a peer-to-peer (P2P) approach is used to ensure this service. It has the big advantage of seamlessly scaling to arbitrary population sizes, as every node that receives the video, while consuming resources, can at the same time offer its own upload bandwidth to serve other nodes. Providing a live streaming service based on a P2P approach, presents many challenges due to the unpredictable behavior of the system users. The aim of this thesis is to analyze these challenges and to propose a set of mechanism improving the video quality. Our contributions in this dissertation are twofold. First, we propose a P2P live streaming system called P-QOLS that ensures a smooth video display while reducing as much as possible the play-out delay. It is based on an overlay construction mechanism providing a set of optimized neighbors for each client. The results of simulations show this system efficiency in heterogeneous cases. Second, we propose a fast packet recovery mechanism ensuring a smooth video display. It consists in requesting the lost packets from a sender peer that is different from the original sender of these packets. The results of simulations show this mechanism efficiency when it is compared to the current mechanism
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35

Jiehua, Liu, Yang Dongkai, and Zhang Qishan. "An Ocean Stereo Telemetry System Based on PC104 Industrial Control Computer and Iridium Communication." International Foundation for Telemetering, 2005. http://hdl.handle.net/10150/604901.

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ITC/USA 2005 Conference Proceedings / The Forty-First Annual International Telemetering Conference and Technical Exhibition / October 24-27, 2005 / Riviera Hotel & Convention Center, Las Vegas, Nevada
To monitor ocean resources and environment, we develop an ocean stereo telemetry system built on a PC104 industrial control computer, which is carried by a buoy on the ocean. All monitoring instruments communicate with the computer by six serial ports in virtue of time division multiple access and are synchronized by GPS time to collect data. All monitoring data is archived and compressed in format of RINEX (Receiver Independent Exchange). The uploading data and downloading control command to and from monitoring center is transferred by Iridium communication in automatic retransmission request and broken-point continuing mechanism.
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36

Grinnemo, Karl-Johan. "A Study of Partially Reliable Transport Protocols for Soft Real-Time Applications." Licentiate thesis, Karlstads universitet, Avdelningen för datavetenskap, 2002. http://urn.kb.se/resolve?urn=urn:nbn:se:kau:diva-33933.

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The profileration of multimedia applications, such as streaming video, teleconferencing, and interactive gaming has created a tremendous challenge for the traditional transport protocols of the Internet – UDP and TCP. Specifically, many multimedia applications are examples of soft real-time applications. They have often relatively stringent require- ments in terms of delay and delay jitter, but typically tolerate a limited packet loss rate. In recognition of the transport service requirements of soft real-time applications, this thesis studies the feasibility of using retransmission based, partially reliable trans- port protocols for these applications. The thesis studies ways of designing retransmis- sion based, partially reliable transport protocols that are congestion aware and TCP com- patible. Furthermore, the transport protocols should provide a service that, in terms of performance metrics such as throughput, delay, and delay jitter, are suitable for soft real- time applications. The thesis work comprises the design, analysis, and evaluation of two retransmission based, partially reliable transport protocols: PRTP and PRTP-ECN. Extensive simulations have been carried out on PRTP as well as PRTP-ECN. These sim- ulations have in part been complemented by some theoretical analysis. The results of the simulations and the analysis suggest that substantial reductions in delay jitter and improvements in throughput can indeed be obtained with both PRTP and PRTP-ECN as compared to TCP. While PRTP reacted too slowly to congestion to be TCP-friendly and altogether fair, PRTP-ECN was found to be both TCP-friendly and reasonably fair. The thesis work also comprises an extensive survey on retransmission based, par- tially reliable transport protocols. Based on this survey, we have proposed a taxonomy for these protocols. The taxonomy considers two dimensions of retransmission based, partially reliable transport protocols: the transport service, and the error control scheme.
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Eshler, Todd Jacob. "Adaptive Protocols to Improve TCP/IP Performance in an LMDS Network using a Broadband Channel Sounder." Thesis, Virginia Tech, 2002. http://hdl.handle.net/10919/42190.

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Virginia Tech researchers have developed a broadband channel sounder that can measure channel quality while a wireless network is in operation. Channel measurements from the broadband sounder hold the promise of improving TCP/IP performance by trigging configuration changes in an adaptive data link layer protocol. We present an adaptive data link layer protocol that can use different levels of forward error correction (FEC) codes and link layer automatic retransmission request (ARQ) to improve network and transport layer performance.

Using a simulation model developed in OPNET, we determine the effects of different data link layer protocol configurations on TCP/IP throughput and end-to-end delay using a Rayleigh fading channel model. Switching to higher levels of FEC encoding improves TCP/IP throughput for high bit error rates, but increases end-to-end delay of TCP/IP segments. Overall TCP/IP connections with link layer ARQ showed approximately 150 Kbps greater throughput than without ARQ, but lead to the highest end-to-end delay for high bit error rate channels.

Based on the simulation results, we propose algorithms to maximize TCP/IP throughput and minimize end-to-end delay using the current bit error rate of the channel. We propose a metric, carrier-to-interference ratio (CIR) that is calculated from data retrieved from the broadband channel sounder. We propose algorithms using the carrier-to-interference ratio to control TCP/IP throughput and end-to-end delay.

The thesis also describes a monitor program to use in the broadband wireless system. The monitor program displays data collected from the broadband sounder and controls the settings for the data link layer protocol and broadband sounder while the network is in operation.
Master of Science

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Hatefi, Atoosa. "Codage distribué pour les réseaux coopératifs sans fil." Phd thesis, Supélec, 2012. http://tel.archives-ouvertes.fr/tel-00829100.

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Cette thèse est consacrée à l'étude théorique et à la conception pratique de schémas de codage conjoint réseau/canal adaptés à différents scénarii de communications dans les réseaux sans fil. Contrairement aux hypothèses conventionnelles retenues dans la littérature (accès multiple orthogonal, absence d'erreurs sur certains liens), les caractéristiques de diffusion et de superposition des signaux propres au canal radio et la présence d'évanouissements lents et de bruit sur tous les liens sont prises en compte dans la formulation du problème et exploitées. Différentes stratégies de coopération au niveau du ou des relais sont examinées et comparées. Le point commun entre toutes ces stratégies est que le système doit fonctionner même en absence de coopération. Seuls le ou les relais et la destination sont informés d'une coopération. Ni les sources, ni le ou les relais ne connaissent l'état du canal à l'émission.
Le premier volet de la thèse porte sur le canal à accès multiple avec relais unique (slow fading MARC). Le problème du codage et décodage conjoint canal/réseau (JNCC/JNCD) est étudié sur un plan théorique et pratique. Différentes hypothèses au niveau de l'accès multiple (semi-orthogonal et non-orthogonal) et différents modes de fonctionnement du relais (half-duplex et full-duplex) sont envisagés. Une nouvelle stratégie de coopération adaptative (SDF pour selective decode and forward) est définie dans laquelle le relais calcule et retransmet une fonction déterministe des messages de sources qu'il a pu décoder sans erreur. Le ré-encodage, défini sur un corps fini (corps binaire), est également conçu de manière à assurer que la performance finale au niveau de la destination atteint bien un ordre de diversité 2.
Le modèle de canal MARC est par la suite étendu à plusieurs relais (slow fading MAMRC). Une analyse théorique est conduite et des nouveaux schémas JNCC/JNCD permettant de s'approcher des limites théoriques sont décrits. Afin d'assurer la diversité pleine, nous proposons de combiner un codage canal binaire et un codage réseau non-binaire.
Pour les deux types de canaux, nous montrons que l'interférence naturellement induite par la diffusion des signaux dans un environnement sans fil, n'est pas un inconvénient mais bien un avantage dès lors qu'on est en mesure de la traiter via des techniques de codage et de décodage sophistiquées (turbo codes et leur décodage, turbo détection). Les gains en termes de capacité (rapportée à une certaine probabilité de coupure) obtenus avec un accès multiple semi-orthogonal ou non-orthogonal sont substantiels comparés à un accès multiple orthogonal (référence).
Dans la dernière partie de la thèse, la stratégie de coopération SDF est comparée à deux autres stratégies de coopération s'appuyant sur un procédé de décodage-et-retransmission "souple" (sans prise de décisions intermédiaires) : l'une basée sur les rapports logarithmiques de probabilité a posteriori sur les bits codés et l'autre basée sur l'estimation de l'erreur quadratique moyenne (MSE). Nous vérifions que la stratégie de coopération SDF fonctionne bien dans la plupart des configurations, les stratégies de coopération souples n'améliorant légèrement les performances que dans certains cas extrêmes.
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39

wang, yubing. "Modeling and Evaluating Feedback-Based Error Control for Video Transfer." Digital WPI, 2008. https://digitalcommons.wpi.edu/etd-dissertations/397.

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"Packet loss can be detrimental to real-time interactive video over lossy networks because one lost video packet can propagate errors to many subsequent video frames due to the encoding dependency between frames. Feedback-based error control techniques use feedback information from the decoder to adjust coding parameters at the encoder or retransmit lost packets to reduce the error propagation due to data loss. Feedback-based error control techniques have been shown to be more effective than trying to conceal the error at the encoder or decoder alone since they allow the encoder and decoder to cooperate in the error control process. However, there has been no systematic exploration of the impact of video content and network conditions on the performance of feedback-based error control techniques. In particular, the impact of packet loss, round-trip delay, network capacity constraint, video motion and reference distance on the quality of videos using feedback-based error control techniques have not been systematically studied. This thesis presents analytical models for the major feedback-based error control techniques: Retransmission, Reference Picture Selection (both NACK and ACK modes) and Intra Update. These feedback-based error control techniques have been included in H.263/H.264 and MPEG4, the state of the art video in compression standards. Given a round-trip time, packet loss rate, network capacity constraint, our models can predict the quality for a streaming video with retransmission, Intra Update and RPS over a lossy network. In order to exploit our analytical models, a series of studies has been conducted to explore the effect of reference distance, capacity constraint and Intra coding on video quality. The accuracy of our analytical models in predicting the video quality under different network conditions is validated through simulations. These models are used to examine the behavior of feedback-based error control schemes under a variety of network conditions and video content through a series of analytic experiments. Analysis shows that the performance of feedback-based error control techniques is affected by a variety of factors including round-trip time, loss rate, video content and the Group of Pictures (GOP) length. In particular: 1) RPS NACK achieves the best performance when loss rate is low while RPS ACK outperforms other repair techniques when loss rate is high. However RPS ACK performs the worst when loss rate is low. Retransmission performs the worst when the loss rate is high; 2) for a given round-trip time, the loss rate where RPS NACK performs worse than RPS ACK is higher for low motion videos than it is for high motion videos; 3) Videos with RPS NACK always perform the same or better than videos without repair. However, when small GOP sizes are used, videos without repair perform better than videos with RPS ACK; 4) RPS NACK outperform Intra Update for low-motion videos. However, the performance gap between RPS NACK and Intra Update drops when the round-trip time or the intensity of video motion increases. 5) Although the above trends hold for both VQM and PSNR, when VQM is the video quality metric the performance results are much more sensitive to network loss. 6) Retransmission is effective only when the round-trip time is low. When the round-trip time is high, Partial Retransmission achieves almost the same performance as Full Retransmission. These insights derived from our models can help determine appropriate choices for feedback-based error control techniques under various network conditions and video content. "
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40

Ding, Zhihong. "ARQ Techniques for MIMO Communication Systems." Diss., CLICK HERE for online access, 2006. http://contentdm.lib.byu.edu/ETD/image/etd1385.pdf.

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41

Vitkus, Donatas. "Privalomo skaitmeninės televizijos retransliavimo elektroninių ryšių tinklais teisinis reguliavimas." Master's thesis, Lithuanian Academic Libraries Network (LABT), 2012. http://vddb.laba.lt/obj/LT-eLABa-0001:E.02~2011~D_20120124_143130-17800.

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Naujųjų technologijų teisės magistro baigiamojo darbo temos aktualumas ir naujumas susijęs su besikeičiančiomis televizijos transliavimo technologijomis ir visuotiniu analoginės antžeminės televizijos išjungimu Lietuvoje ir Europos Sąjungoje. Privalomo televizijos retransliavimo elektroniniais ryšių tinklais, kitaip vadinamu „must carry“, modelis įvestas esant analoginei televizijai. Besikeičiančios technologijos verčia analizuoti teisinį reguliavimą ir ieškoti galimų geresnio ir efektyvesnio reguliavimo modelio. Skaitmeninės televizijos privalomo retransliavimo instituto svarba atskleidžiama nagrinėjant technologinę skaitmeninės televizijos pusę, pasitelkiant technologinių mokslų žinias. Išryškinami skirtumai nuo analoginės televizijos. Analizuojama privalomo retransliavimo instituto atsiradimo priežastys, aplinka, atskleidžiami instituto tikslai. Taip pat identifikuojamos transliavimo, retransliavimo, signalų siuntimo sąvokos. Analizuojant privalomo skaitmeninės televizijos retransliavimo elektroniniais ryšių tinklais teisinį reguliavimą Europos Sąjungoje pasitelkiamas loginis bei teleologinis metodai. Analizuojama Universaliųjų paslaugų direktyva, kurios 31 straipsnyje įtvirtintas privalomo televizijos retransliavimo principas. Detaliau nagrinėjama Austrijos, Vokietijos, Didžiosios Britanijos privalomo televizijos retransliavimo teisinis reguliavimas. Privalomo televizijos retransliavimo teisinis reguliavimas Lietuvoje analizuojamas nuo jo ištakų – Konstitucijoje saugomų... [toliau žr. visą tekstą]
The relevance and the novelty of the Master’s Work in New Technologies Law related to changing technology of television broadcasting and to universal analogue television turn-off in Lithuania and European Union. Television mandatory retransmission in electronic communications networks model, also called “must carry”, was created to analogue television. Changing technologies cause to analyze legal regulation and seek to create better and effective model of “must carry” regulation. Importance of the institute of digital television mandatory retransmission disclosed the analysis of digital television technology, using the knowledge of technologies science. There are highlighted differences between digital and analogue televisions. There are made analyzes of digital television mandatory retransmission establishing reasons and environment, made disclosure of the objectives of the “must carry” institute. There are also identified terms of broadcasting, retransmission, signal transporting. There are using logical ant theological methods to analyze digital television mandatory retransmission in European Union and Lithuania. In this work there is analyzed Universal Service Directive, which have “must carry” rule in 31 article. Detailed analyze made for Austrian, German, Great Britain legal regulation of digital television mandatory retransmission. Legal regulation of television mandatory retransmission in Lithuania is analyzed from the first source – from the Constitution. Comparative... [to full text]
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42

Maalel, Nourhene. "Reliability in wireless sensor networks." Thesis, Compiègne, 2014. http://www.theses.fr/2014COMP1944/document.

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Vu les perspectives qu'ils offrent, les réseaux de capteur sans fil (RCSF) ont perçu un grand engouement de la part de la communauté de recherche ces dernières années. Les RCSF couvrent une large gamme d'applications variant du contrôle d'environnement, le pistage de cible aux applications de santé. Les RCSFs sont souvent déployés aléatoirement. Ce dispersement des capteurs nécessite que les protocoles de transmission utilisés soient résistants aux conditions environnementales (fortes chaleurs ou pluies par exemple) et aux limitations de ressources des nœuds capteurs. En effet, la perte de plusieurs nœuds capteurs peut engendrer la perte de communication entre les différentes entités. Ces limitations peuvent causer la perte des paquets transmis ce qui entrave l'activité du réseau. Par conséquent, il est important d'assurer la fiabilité des transmissions de données dans les RCSF d'autant plus pour les applications critiques comme la détection d'incendies. Dans cette thèse, nous proposons une solution complète de transmission de données dans les RCSF répondant aux exigences et contraintes de ce type de réseau. Dans un premier temps, nous étudions les contraintes et les challenges liés à la fiabilisation des transmissions dans les RCSFs et nous examinons les travaux proposés dans la littérature. Suite à cette étude nous proposons COMN2, une approche distribuée et scalable permettant de faire face à la défaillance des nœuds. Ensuite, nous proposons un mécanisme de contrôle d'erreur minimisant la perte de paquets et proposant un routage adaptatif en fonction de la qualité du lien. Cette solution est basée sur des acquittements implicites (overhearing) pour la détection des pertes des paquets. Nous proposons ensuite ARRP une variante de AJIA combinant les avantages des retransmissions, de la collaboration des nœuds et des FEC. Enfin, nous simulons ces différentes solutions et vérifions leurs performances par rapport à leurs concurrents de l'état de l'art
Over the past decades, we have witnessed a proliferation of potential application domainsfor wireless sensor networks (WSN). A comprehensive number of new services such asenvironment monitoring, target tracking, military surveillance and healthcare applicationshave arisen. These networked sensors are usually deployed randomly and left unattendedto perform their mission properly and efficiently. Meanwhile, sensors have to operate ina constrained environment with functional and operational challenges mainly related toresource limitations (energy supply, scarce computational abilities...) and to the noisyreal world of deployment. This harsh environment can cause packet loss or node failurewhich hamper the network activity. Thus, continuous delivery of data requires reliabledata transmission and adaptability to the dynamic environment. Ensuring network reliabilityis consequently a key concern in WSNs and it is even more important in emergencyapplication such disaster management application where reliable data delivery is the keysuccess factor. The main objective of this thesis is to design a reliable end to end solution for data transmission fulfilling the requirements of the constrained WSNs. We tackle two design issues namely recovery from node failure and packet losses and propose solutions to enhance the network reliability. We start by studying WSNs features with a focus on technical challenges and techniques of reliability in order to identify the open issues. Based on this study, we propose a scalable and distributed approach for network recovery from nodefailures in WSNs called CoMN2. Then, we present a lightweight mechanism for packetloss recovery and route quality awareness in WSNs called AJIA. This protocol exploitsthe overhearing feature characterizing the wireless channels as an implicit acknowledgment(ACK) mechanism. In addition, the protocol allows for an adaptive selection of therouting path by achieving required retransmissions on the most reliable link. We provethat AJIA outperforms its competitor AODV in term of delivery ratio in different channelconditions. Thereafter, we present ARRP, a variant of AJIA, combining the strengthsof retransmissions, node collaboration and Forward Error Correction (FEC) in order toprovide a reliable packet loss recovery scheme. We verify the efficiency of ARRP throughextensive simulations which proved its high reliability in comparison to its competitor
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43

Xiong, Xuelin. "SCTP and Diameter Parameters for High Availability in LTE Roaming." Thesis, KTH, Radio Systems Laboratory (RS Lab), 2015. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-163254.

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Today mobile network operators utilize IP Packet exchange (IPX) carriers to interconnect their networks with other operators. Mobile network operators are free to choose one IPX carrier for their data traffic and another for their control traffic. This thesis examines the case of control traffic, specifically Stream Control Transmission Protocol (SCTP) carrying Diameter protocol traffic arising from users roaming from their home Long Term Evolution (LTE) network to another operator’s LTE network. The thesis project aims to identify a set of SCTP parameter configurations that can provide improved application/service level availability between two Diameter nodes in different network connectivity environments, specifically for IPX carriers who are Diameter service providers. These service providers provide Diameter connectivity for their customers who are mobile network operators. These mobile network operators in turn provide LTE roaming services to their customers. Unfortunately, applying the ‘One size fits all’ configuration recommendations given in the SCTP documentation is unsuitable for different network environments. In addition, the amount of Diameter signaling traffic is growing at a very rapid rate. Therefore, it is valuable to identify suitable parameter selection criteria for Diameter service providers to ensure 100% Diameter connectivity reliability for their customers. In this thesis project, author investigated how tuning SCTP parameter values affect Diameter message transmission in terms of Round Trip Delay and identified its determining parameters for packet loss recovery performance. Both IPX carriers and mobile network operators may use these values as reference when attempting to ensure high availability of Diameter transmissions under reliable, semi reliable, and unreliable network transport conditions.
Mobilnätsoperatörer använder sig av IP Packet exchange (IPX) tjänstetillhandahållare för att koppla ihop sina nät med andra operatörers nät. Mobilnätsoperatörer kan fritt välja en IPX tjänstetillhandahållare för sin datatrafik och en annan för sin kontrolltrafik. Denna uppsats undersöker fallet för kontrolltrafik, specifikt Stream Control Transmission Protocol (SCTP) kommunikationsprotokoll för Diameter protocol-trafik vid användares roaming från sitt Long Term Evolution (LTE)-hemmanät till en annan operatörs LTE-nät. Examensarbetet avser etablera en uppsättning av SCTP-parameterkonfigurationer som ger förbättrad applikations-/tjänstetillgänglighetsnivå mellan två Diameter-noder i olika nätmiljöer, särskilt för IPX tjänstetillhandahållare som är Diameter tjänstetillhandahållare.  Dessa tjänstetillhandahållare erbjuder Diameter-konnektivitet till sina kunder, som är mobilnätsoperatörer. Dessa mobilnätsoperatörer tillhandahåller i sin tur LTE-roamingtjänster till sina kunder. Tyvärr är det olämpligt att tillämpa de enhetliga konfigurationsrekommendationer, som ges i SCTP- och Diameter-protokollens dokumentation, i olika nätmiljöer. Samtidigt ökar Diameter-signaleringstrafiken mycket snabbt. Därför är det värdefullt att identifiera lämpliga parameterkriterier för Diameter-tjänstetillhandahållare att säkerställa 100% tillförlitlig Diameter-tillgänglighet för sina kunder. I detta examensarbete har författaren undersökt hur trimning av SCTP-parametervärden påverkar Diameter-meddelandeöverföring vad avser överföringstiden tur- och retur, och identifierat de avgörande parametrarna för återställande av paketförluster. Både IPX tjänstetillhandahållare och mobilnätsoperatörer kan använda dessa värden som referens för att åstadkomma hög tillgänglighet för Diameter-överföring vid tillförlitliga, halvtillförlitliga och otillförlitliga nättransportförutsättningar.
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44

Do, Minh-Tien. "Ultra-narrowband wireless sensor networks modeling and optimization." Thesis, Lyon, INSA, 2015. http://www.theses.fr/2015ISAL0065/document.

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Cette thèse a pour but de modéliser les réseaux de capteurs sans fil à faible débit (WSN) basés sur la technique ultra bande étroite. Ce réseau a déjà été déployée par la société SIGFOX et a déjà démontré sa très grande efficacité pour les applications pour l’Internet des objets (IoTs) grâce à sa capacité de communication point à point efficace en terme de puissance consommée, et de sa connectivité de longue portée. Cette étude donne quelques aperçus sur le passage à l’échelle de la technique de l’UNB pour un réseau multipoint à point pour une liaison montante. L’accès au canal spécifique qui est basé sur l’accès multiple par répartition aléatoire de fréquence et de temps (R-FTDMA) est introduit et analysé. En outre, l’impact de l’interférence due à l’absence de stratégie d’ordonnancement à la couche MAC est étudié et modélisé. Notre modèle simplifié nous permet non seulement de décrire la puissance d’interférence agrégée, mais aussi d’évaluer les performances du système d’un tel réseau en matière de taux d’erreur et de probabilité de coupure. De même, la géométrie stochastique est utilisée pour modéliser la distribution spatiale des noeuds afin d’étendre le modèle simplifié dans le canal réaliste où les dégradations de canal sont prises en compte. De plus, le mécanisme de retransmission est considéré pour ce réseau. Cette étude permet de d’identifier le nombre optimal de retransmissions. Le réseau peut être configuré avec un paramètre global unique. Et enfin, cette thèse met en évidence le fait que le réseau de l’UNB Random-FTDMA est très pertinent dans un réseau réaliste, en particulier pour les applications à faible débit, car il allége le coût élevé du réseau, le coût de la synchronisation globale, mais sans perte de performance
This thesis aims at modeling the low-throughput wireless sensor networks (WSNs) based on ultra-narrow-band technology. Such wireless network is already been deployed by Sigfox company and has proved to be ultra-efficient for the Internet of things (IoTs) applications thanks to its ability of point-to-point communication in terms of power efficiency and long range connectivity. In particular, this thesis gives some insights on the scalability of UNB technology for a multi-point-to-point network in an uplink scenario. The multiple access schemes based on random time and frequency selection are introduced and analyzed. Furthermore, the interference impact due to the lack of scheduling strategy at the MAC layer is studied and modeled. Our simplified model using rectangular function allows us not only to describe the aggregate interference power but also evaluate the system performance of such network in terms of the bit-error-rate and outage probability. Besides, the geometry stochastic is used for spatial node distribution in order to extend the simplified model in the realistic channel communication where the channel impairments are taken into account. Besides, the retransmission mechanism is considered for such network. This study argues an optimal number of retransmission. The network can be configured with a unique global parameter. Last but not least, this thesis highlights the fact that the UNB network using Random-FTDMA schemes is very relevant in a realistic network, especially for low-throughput applications, because it bypasses the high network cost, the cost of global synchronization but without loss of performance
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45

Mazel, Claude. "Evaluation des performances par simulations : application aux canaux de signalisation de systèmes radiotéléphoniques." Grenoble INPG, 1988. http://tel.archives-ouvertes.fr/tel-00327272.

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Le sujet de cette thèse est l'évaluation par simulation des performances des systèmes radiotelephoniques. En première partie, on présente les différentes techniques d'accès multiple. En deuxième partie, on applique une méthode de modélisation en deux phases inspirée des techniques d'agrégation à un réseau radiotelephonique cellulaire. La dernière partie concerne l'évaluation des performances d'un algorithme de contrôle des retransmissions dont l'intérêt est de prendre en compte le phénomène de capture
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46

Maaz, Mohamad. "Allocation de ressource et analyse des critères de performance dans les réseaux cellulaires coopératifs." Thesis, Rennes, INSA, 2013. http://www.theses.fr/2013ISAR0036/document.

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Dans les systèmes de communications sans fil, la transmission de grandes quantités d'information et à faible coût énergétique sont les deux principales questions qui n'ont jamais cessé d'attirer l'attention de la communauté scientifique au cours de la dernière décennie. Récemment, il a été démontré que la communication coopérative est une technique intéressante notamment parce qu'elle permet d'exploiter la diversité spatiale dans le canal sans fil. Cette technique assure une communication robuste et fiable, une meilleure qualité de service (QoS) et rend le concept de coopération prometteur pour les futurs générations de systèmes cellulaires. Typiquement, les QoS sont le taux d'erreurs paquet, le débit et le délai. Ces métriques sont impactées par le délai, induit par les mécanismes de retransmission Hybrid-Automatic Repeat-Request (HARQ) inhérents à la réception d'un paquet erroné et qui a un retard sur la QoS demandée. En revanche, les mécanismes HARQ créent une diversité temporelle. Par conséquent, l'adoption conjointe de la communication coopérative et des protocoles HARQ pourrait s'avérer avantageux pour la conception de schémas cross-layer. Nous proposons tout d'abord une stratégie de maximisation de débit total dans un réseau cellulaire hétérogène. Nous introduisons un algorithme qui alloue la puissance optimale à la station de base (BS) et aux relais, qui à chaque utilisateur attribue de manière optimale les sous-porteuses et les relais. Nous calculons le débit maximal atteignable ainsi que le taux d'utilisateurs sans ressources dans le réseau lorsque le nombre d'utilisateurs actifs varie. Nous comparons les performances de notre algorithme à ceux de la littérature existante, et montrons qu'un gain significatif est atteint sur la capacité globale. Dans un second temps, nous analysons théoriquement le taux d'erreurs paquet, le délai ainsi que l'efficacité de débit des réseaux HARQ coopératifs, dans le canal à évanouissements par blocs. Dans le cas des canaux à évanouissement lents, le délai moyen du mécanisme HARQ n'est pas pertinent à cause de la non-ergodicité du processus. Ainsi, nous nous intéressons plutôt à la probabilité de coupure de délai en présence d'évanouissements lents. La probabilité de coupure de délai est de première importance pour les applications sensibles au délai. Nous proposons une forme analytique de la probabilité de coupure permettant de se passer de longues simulations. Dans la suite de notre travail, nous analysons théoriquement l'efficacité énergétique (bits/joule) dans les réseaux HARQ coopératifs. Nous résolvons ensuite un problème de minimisation de l'énergie dans les réseaux coopératifs en liaison descendante. Dans ce problème, chaque utilisateur possède une contrainte de délai moyen à satisfaire de telle sorte que la contrainte sur la puissance totale du système soit respectée. L'algorithme de minimisation permet d'attribuer à chaque utilisateur la station-relai optimale et sa puissance ainsi que la puissance optimale de la BS afin de satisfaire les contraintes de délai. Les simulations montrent qu'en termes de consommation d'énergie, les techniques assistées par relais prédominent nettement les transmissions directes, dans tout système limité en délai. En conclusion, les travaux proposés dans cette thèse peuvent promettre d'établir des règles fiables pour l'ingénierie et la conception des futures générations de systèmes cellulaires énergétiquement efficaces
In wireless systems, transmitting large amounts of information with low energetic cost are two main issues that have never stopped drawing the attention of the scientific community during the past decade. Later, it has been shown that cooperative communication is an appealing technique that exploits spatial diversity in wireless channel. Therefore, this technique certainly promises a robust and reliable communications, higher quality-of-service (QoS) and makes the cooperation concept attractive for future cellular systems. Typically, the QoS requirements are the packet error rate, throughput and delay. These metrics are affected by the delay, where each erroneous packet is retransmitted several times according to Hybrid-Automatic Repeat-Request (HARQ) mechanism inducing a delay on the demanded QoS but a temporal diversity is created. Therefore, adopting jointly cooperative communications and HARQ mechanisms could be beneficial for designing cross-layer schemes. First, a new rate maximization strategy, under heterogeneous data rate constraints among users is proposed. We propose an algorithm that allocates the optimal power at the base station (BS) and relays, assigns subcarriers and selects relays. The achievable data rate is investigated as well as the average starvation rate in the network when the load, i.e. the number of active users in the network, is increasing. It showed a significant gain in terms of global capacity compared to literature. Second, in block fading channel, theoretical analyses of the packet error rate, delay and throughput efficiency in relayassisted HARQ networks are provided. In slow fading channels, the average delay of HARQ mechanisms w.r.t. the fading states is not relevant due to the non-ergodic process of the fading channel. The delay outage is hence invoked to deal with the slow fading channel and is defined as the probability that the average delay w.r.t. AWGN channel exceeds a predefined threshold. This criterion has never been studied in literature, although being of importance for delay sensitive applications in slow fading channels. Then, an analytical form of the delay outage probability is proposed which might be useful to avoid lengthy simulations. These analyses consider a finite packet length and a given modulation and coding scheme (MCS) which leads to study the performance of practical systems. Third, a theoretical analysis of the energy efficiency (bits/joule) in relay-assisted HARQ networks is provided. Based on this analysis, an energy minimization problem in multiuser relayassisted downlink cellular networks is investigated. Each user has an average delay constraint to be satisfied such that a total power constraint in the system is respected. The BS is assumed to have only knowledge about the average channel statistics but no instantaneous channel state information (CSI). Finally, an algorithm that jointly allocates the optimal power at BS, the relay stations and selects the optimal relay in order to satisfy the delay constrains of users is proposed. The simulations show the improvement in terms of energy consumption of relay-assisted techniques compared to nonaided transmission in delay-constrained systems. Hence, the work proposed in this thesis can give useful insights for engineering rules in the design of the next generation energyefficient cellular systems
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47

Thie, Johnson Electrical Engineering &amp Telecommunications Faculty of Engineering UNSW. "Optimal erasure protection assignment for scalably compressed data over packet-based networks." Awarded by:University of New South Wales. Electrical Engineering and Telecommunications, 2004. http://handle.unsw.edu.au/1959.4/19373.

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This research is concerned with the reliable delivery of scalable compressed data over lossy communication channels. Recent works proposed several strategies for assigning optimal code redundancies to elements of scalable data, which form a linear structure of dependency, under the assumption that all source elements are encoded onto a common group of network packets. Given large data and small network packets, such schemes require very long channel codes with high computational complexity. In networks with high loss, small packets are more desirable than long packets. The first contribution of this thesis is to propose a strategy for optimally assigning elements of the scalable data to clusters of packets, subject to constraints on packet size and code complexity. Given a packet cluster arrangement, the scheme then assigns optimal code redundancies to the source elements, subject to a constraint on transmission length. Experimental results show that the proposed strategy can outperform the previous code assignment schemes subject to the above-mentioned constraints, particularly at high channel loss rates. Secondly, we modify these schemes to accommodate complex structures of dependency. Source elements are allocated to clusters of packets according to their dependency structure, subject to constraints on packet size and channel codeword length. Given a packet cluster arrangement, the proposed schemes assign optimal code redundancies to the source elements, subject to a constraint on transmission length. Experimental results demonstrate the superiority of the proposed strategies for correctly modelling the dependency structure. The last contribution of this thesis is to propose a scheme for optimizing protection of scalable data where limited retransmission is possible. Previous work assumed that retransmission is not possible. For most real-time or interactive applications, however, retransmission of lost data may be possible up to some limit. In the present work we restrict our attention to streaming sources (e.g., video) where each source element can be transmitted in one or both of two time slots. An optimization algorithm determines the transmission and level of protection for each source element, using information about the success of earlier transmissions. Experimental results confirm the benefit of limited retransmission.
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48

Hatefi, Atoosa. "Distributed Coding for Wireless Cooperative Networks." Thesis, Supélec, 2012. http://www.theses.fr/2012SUPL0015/document.

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Cette thèse est consacrée à l'étude théorique et à la conception pratique de schémas de codage conjoint réseau/canal adaptés à différents scénarii de communications dans les réseaux sans fil. Contrairement aux hypothèses conventionnelles retenues dans la littérature (accès multiple orthogonal, absence d'erreurs sur certains liens), les caractéristiques de diffusion et de superposition des signaux propres au canal radio et la présence d'évanouissements lents et de bruit sur tous les liens sont prises en compte dans la formulation du problème et exploitées. Différentes stratégies de coopération au niveau du ou des relais sont examinées et comparées. Le point commun entre toutes ces stratégies est que le système doit fonctionner même en absence de coopération. Seuls le ou les relais et la destination sont informés d'une coopération. Ni les sources, ni le ou les relais ne connaissent l'état du canal à l'émission. Le premier volet de la thèse porte sur le canal à accès multiple avec relais unique (slow fading MARC). Le problème du codage et décodage conjoint canal/réseau (JNCC/JNCD) est étudié sur un plan théorique et pratique. Différentes hypothèses au niveau de l'accès multiple (semi-orthogonal et non-orthogonal) et différents modes de fonctionnement du relais (half-duplex et full-duplex) sont envisagés. Une nouvelle stratégie de coopération adaptative (SDF pour selective decode and forward) est définie dans laquelle le relais calcule et retransmet une fonction déterministe des messages de sources qu'il a pu décoder sans erreur. Le ré-encodage, défini sur un corps fini (corps binaire), est également conçu de manière à assurer que la performance finale au niveau de la destination atteint bien un ordre de diversité 2. Le modèle de canal MARC est par la suite étendu à plusieurs relais (slow fading MAMRC). Une analyse théorique est conduite et des nouveaux schémas JNCC/JNCD permettant de s'approcher des limites théoriques sont décrits. Afin d'assurer la diversité pleine, nous proposons de combiner un codage canal binaire et un codage réseau non-binaire. Pour les deux types de canaux, nous montrons que l'interférence naturellement induite par la diffusion des signaux dans un environnement sans fil, n'est pas un inconvénient mais bien un avantage dès lors qu'on est en mesure de la traiter via des techniques de codage et de décodage sophistiquées (turbo codes et leur décodage, turbo détection). Les gains en termes de capacité (rapportée à une certaine probabilité de coupure) obtenus avec un accès multiple semi-orthogonal ou non-orthogonal sont substantiels comparés à un accès multiple orthogonal (référence). Dans la dernière partie de la thèse, la stratégie de coopération SDF est comparée à deux autres stratégies de coopération s'appuyant sur un procédé de décodage-et-retransmission "souple" (sans prise de décisions intermédiaires) : l'une basée sur les rapports logarithmiques de probabilité a posteriori sur les bits codés et l'autre basée sur l'estimation de l'erreur quadratique moyenne (MSE). Nous vérifions que la stratégie de coopération SDF fonctionne bien dans la plupart des configurations, les stratégies de coopération souples n'améliorant légèrement les performances que dans certains cas extrêmes
With the rapid growth of wireless technologies, devices and mobile applications, the quest of high throughput and ubiquitous connectivity in wireless communications increases rapidly as well. Relaying is undoubtedly a key concept to provide coverage extension and capacity increase in wireless networks. Network coding, which allows the intermediate nodes to share their computation capabilities in addition to their resource and their power, has grabbed a significant research attention since its inception in information theory. It has become an attractive candidate to bring promising performance improvement, especially in terms of throughput, in relay-based cellular networks. Substantial research efforts are currently focused on theoretical analysis, implementation and evaluation of network coding from a physical layer perspective. The question is, what is the most efficient and practical way to use network coding in wireless relay-based networks, and whether it is beneficial to exploit the broadcast and multiple-access properties of the wireless medium to perform network coding. It is in such a context, that this thesis proceeds. In the first part of the thesis, the problem of Joint Network-Channel Coding (JNCC) for a Multiple Access Relay Channel (MARC) is investigated in the presence of multiple access interferences and for both of the relay operating modes, namely, half-duplex and full-duplex. To this end, three new classes of MARC, referred to as Half-Duplex Semi-Orthogonal MARC (HD-SOMARC), Half-Duplex Non-Orthogonal MARC (HD-NOMARC), and Full-Duplex Non-Orthogonal MARC (FD-NOMARC) have been introduced and studied. The relaying function in all of the classes is based on a Selective Decode-and-Forward (SDF) strategy, which is individually implemented for each source, i.e, the relay forwards only a deterministic function of the error-free decoded messages. For each class, an information-theoretic analysis is conducted, and practical coding and decoding techniques are proposed. The proposed coding schemes, perform very close to the outage limit for both cases of HD-SOMARC and HD-NOMARC. Besides, in the case of HD-NOMARC, the optimal allocation of the transmission time to the relay is considered. It is also verified that exploiting multiple access interferences, either partially or totally, results in considerable gains for MARC compared to the existing interference-avoiding structures, even in the case of single receive antenna. In the second part of the thesis, the network model is extended by considering multiple relays which help multiple sources to communicate with a destination. A new class of Multiple Access Multiple Relay Channel (MAMRC), referred to as Half-Duplex Semi-Orthogonal MAMRC (HD-SOMAMRC) is then proposed and analyzed from both information theoretic and code design perspective. New practical JNCC schemes are proposed, in which binary channel coding and non binary network coding are combined, and they are shown to perform very close to the outage limit. Moreover, the optimal allocation of the transmission time to the sources and relays is considered. Finally, in the third part of the thesis, different ways of implementing cooperation, including practical relaying protocols are investigated for the half-duplex MARC with semi-orthogonal transmission protocol and in the case of JNCC. The hard SDF approach is compared with two Soft Decode and Forward (SoDF) relaying functions: one based on log a posterior probability ratios (LAPPRs) and the other based on Mean Square Error (MSE) estimate. It is then shown that SDF works well in most of the configurations and just in some extreme cases, soft relaying functions (based on LAPPR or MSE estimate) can slightly outperform the hard selective one
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LIAO, WEI-XING, and 廖威晴. "Examining the Antecedents of Rumor Retransmission-Uses and Gratification Theory, Rumor Retransmission Model, and Basic Law of Rumor." Thesis, 2016. http://ndltd.ncl.edu.tw/handle/46sye2.

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碩士
國立中山大學
資訊管理學系研究所
104
Nowadays, people send a lot of information through social media. In addition, rumor is also an information type that spreads through people’s social media friend circles. Prior studies have focused on disastrous and catastrophic rumors that scared people, but many of today’s rumors are about our daily life. We want to investigate why people retransmit the rumor they hear and what factors will motivate them more to retransmit the rumor. To accomplish this, this study combines three theories to predict people’s retransmitting intention. The first is the Uses and Gratification Theory (U&G Theory). The second and third are the Rumor Retransmission Model and the Basic Law of Rumor. The former looks at people’s motivations to investigate what kinds of needs people want to fulfill and the latter seeks to understand what rumor characteristics will facilitate people to retransmit after receiving the rumor. The results of this study show that socializing, information seeking, and status seeking will increase the intent to retransmit; but entertainment has no significant effect. Personal involvement has no moderating effects on the relationships of socializing and information seeking to intention. Perceived credibility has a moderating effect on the relationship between socializing and intention, but is complementary with status seeking.
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"Performance analysis of poll-based retransmission schemes." Massachusetts Institute of Technology, Laboratory for Information and Decision Systems], 1993. http://hdl.handle.net/1721.1/3324.

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