Dissertations / Theses on the topic 'Packet delay'
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1
Altes, Talissa A. "Minimum delay packet length." Thesis, Massachusetts Institute of Technology, 1986. http://hdl.handle.net/1721.1/15090.
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Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1986.MICROFICHE COPY AVAILABLE IN ARCHIVES AND ENGINEERING
Bibliography: leaf 61.
by Talissa A. Altes.
M.S.
2
Husain, Samreen Riaz. "Delay-based packet scheduling for High Speed Downlink Packet Access." Thesis, Kingston, Ont. : [s.n.], 2007. http://hdl.handle.net/1974/649.
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Li, Xin. "Scheduling optical packet switches with reconfiguration delay /." View abstract or full-text, 2005. http://library.ust.hk/cgi/db/thesis.pl?COMP%202005%20LI.
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Subasingha, Subasingha Shaminda. "Quantization for Low Delay and Packet Loss." Scholarly Repository, 2010. http://scholarlyrepository.miami.edu/oa_dissertations/374.
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Quantization of multimodal vector data in Realtime Interactive Communication Networks (RICNs) associated with application areas such as speech, video, audio, and haptic signals introduces a set of unique challenges. In particular, achieving the necessary distortion performance with minimum rate while maintaining low end-to-end delay and handling packet losses is of paramount importance. This dissertation presents vector quantization schemes which aim to satisfy these important requirements based on two source coding paradigms; 1) Predictive coding 2) Distributed source coding. Gaussian Mixture Models (GMMs) can be used to model any probability density function (pdf) with an arbitrarily small error given a sufficient number of mixture components. Hence, Gaussian Mixture Models can be effectively used to model the underlying pdfs of a variety of data in RICN applications. In this dissertation, first we present Gaussian Mixture Models Kalman predictive coding, which uses transform domain predictive GMM quantization techniques with Kalman filtering principles. In particular, we show how suitable modeling of quantization noise leads to a signal-adaptive GMM Kalman predictive coder that provides improved coding performance. Moreover, we demonstrate how running a GMM Kalman predictive coder to convergence can be used to design a stationary GMM Kalman predictive coding system which provides improved coding of GMM vector data but now with only a modest increase in run-time complexity over the baseline. Next, we address the issues of packet loss in the networks using GMM Kalman predictive coding principles. In particular, we show how an initial GMM Kalman predictive coder can be utilized to obtain a robust GMM predictive coder specifically designed to operate in packet loss. We demonstrate how one can define sets of encoding and decoding modes, and design special Kalman encoding and decoding gains for each mode. With this framework, GMM predictive coding design can be viewed as determining the special Kalman gains that minimize the expected mean squared error at the decoder in packet loss conditions. Finally, we present analytical techniques for modeling, analyzing and designing Wyner-Ziv(WZ) quantizers for Distributed Source Coding for jointly Gaussian vector data with imperfect side information. In most of the DSC implementations, the side information is not explicitly available in the decoder. Thus, almost all of the practical implementations obtain the side information from the previously decoded frames. Due to model imperfections, packet losses, previous decoding errors, and quantization noise, the available side information is usually noisy. However, the design of Wyner-Ziv quantizers for imperfect side information has not been widely addressed in the DSC literature. The analytical techniques presented in this dissertation explicitly assume the existence of imperfect side information in the decoder. Furthermore, we demonstrate how the design problem for vector data can be decomposed into independent scalar design subproblems. Then, we present the analytical techniques to compute the optimum step size and bit allocation for each scalar quantizer such that the decoder's expected vector Mean Squared Error(MSE) is minimized. The simulation results verify that the predicted MSE based on the presented analytical techniques closely follow the simulation results.
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Evéquoz, Claude. "Message delay models of packet-switching networks." Thesis, McGill University, 1989. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=74223.
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The prediction of mean end-to-end delay of multiple packet messages is the focus of this thesis. The models established are of great importance to network designers as they provide the basis for meeting user requirements for time delays and for network throughput.Two distinct models are developed depending upon the traffic load requested by network users. The first model considers traffic conditions in which the traffic demand heavily loads the computer network. Mean end-to-end message delays are computed by assuming the existence of product form solution of the obtained closed queueing network model. Heuristics are developed and are validated via simulation to render the computation feasible and to extend the solution method to non product form networks.
Under light traffic conditions, clusters of packets resulting from message segmentation may enter the computer network. To this end, algorithms are developed to determine the network performance measures of closed queueing networks in which bulk transitions are possible. A second message delay model is then developed to take into account the clustered arrival of packets to the network. All models and heuristics are validated via simulation.
Finally, the appropriate length of the packets into which a message should be segmented is addressed. The performance tradeoff between message delay and throughput is discussed. Boundaries delimiting the packet size to meet user requirements are established for delay and throughput.
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VonFange, Ross. "A device for synchronous Ethernet packet delay." Thesis, Manhattan, Kan. : Kansas State University, 2009. http://hdl.handle.net/2097/1490.
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Rossi, Charles. "Delay considerations in packet radio networks with capture." Thesis, Massachusetts Institute of Technology, 1986. http://hdl.handle.net/1721.1/15062.
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Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1986.MICROFICHE COPY AVAILABLE IN ARCHIVES AND ENGINEERING
Bibliography: leaf 58.
by Charles Rossi, Jr.
M.S.
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Zhang, Yu. "Delay performance of scheduling mechanisms in packet-switched networks." Thesis, Imperial College London, 2009. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.504903.
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Fahlborg, Daniel. "Measuring one-way Packet Delay in a Radio Network." Thesis, Linköpings universitet, Kommunikationssystem, 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-148586.
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Radio networks are expanding, becoming more advanced, and pushing the limits of what is possible. Services utilizing the radio networks are also being developed in order to provide new functionality to end-users worldwide. When discussing 5G radio networks, concepts such as driverless vehicles, drones and near zero communication delay are recurrent. However, measures of delay are needed in order to verify that such services can be provided -- and measuring this is an extensive task. Ericsson has developed a platform for simulating a radio environment surrounding a radio base station. Using this simulator, this project involved measuring one-way packet delay in a radio network, and performing a Quality of Service evaluation of a radio network with a number of network applications in concern. Application data corresponding to video streams, or Voice over IP conversations, were simulated and packet delay measurements were used to calculate and evaluate the Quality of Service provided by a radio network. One of the main conclusions of this project was that packet delay variations are asymmetric in uplink, which suggests usage of non-conventional jitter measurement techniques.
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Mostafa, Ahmad A. "Packet Delivery Delay and Throughput Optimization for Vehicular Networks." University of Cincinnati / OhioLINK, 2013. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1367924037.
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Crepin-Leblond, Olivier Marie James. "Reduction of delay in ATM multiplexers." Thesis, Imperial College London, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.266067.
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Ayyorgun, Sami. "Feasibility of serving packet streams with delay and loss requirements /." Diss., Connect to a 24 p. preview or request complete full text in PDF format. Access restricted to UC campuses, 2001. http://wwwlib.umi.com/cr/ucsd/fullcit?p3031937.
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Zhu, Kai. "Statistical Delay Bounds Oriented Packet Scheduling Algorithms in High Speed Networks." NCSU, 2000. http://www.lib.ncsu.edu/theses/available/etd-20000911-100735.
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Zhu, Kai. Statistical Delay Bounds Oriented Packet Scheduling Algorithms inHigh Speed Networks. (Under the direction of Prof. Yannis Viniotis)
We first present a strategic analysis of end-to-end delay bounds and identify heuristics for scheduler design, then propose three new schedulers that are targeted at statistical delay bounds: Deadline-curve basedEarliest Deadline First (DC-EDF), Adaptive Quasi-Earliest Deadline First (AQE)and General Dynamic Guaranteed Rate Queueing (GDGRQ).
Under DC-EDF, local deadlines are assigned as strict time-shifted versions ofsource packet arrival times. This is quite different from the well-known RC-EDF(Rate-controlled EDF), which deploys traffic shaping at each switching node. Weshow that even without traffic shapers, DC-EDF provides not only end-to-enddelay bounds, but also a schedulable region as large as that of RC-EDF. DC-EDFis self-adaptive in local delay bound assignments. This property makes DC-EDFsuitable as the scheduler at intermediate switching nodes along a flow's route.
AQE is an enhancement of EDF with intelligence of adaptive scheduling. Under AQEpercentile delay bounds are the delay QoS metric. AQE behaves like EDF whenbandwidth is sufficient. When bandwidth becomes deficient, however, AQE onlyschedules a subset of flows which currently have relatively worse performance;other flows are completely blocked and will be unblocked only when bandwidthbecomes sufficient again. Essentially, AQE enforces shaping on packet delaydistributions. AQE is most suited for scheduling at the last switching nodealong a flow's route. The combination of DC-EDF and AQE provides a good solutiontowards statistical end-to-end delay bounds.
GDGQR is designed for networks with fixed packet sizes. It is a subclass of thewell-known Guaranteed Rate (GR) schedulers, thus can guarantee minimum rates toflows. The GR property of GDGRQ is retained through a sophisticated datastructure called cell transmission table. The unique feature of GDGRQ is that itallows controllable adaptive strategy of excess bandwidth distribution, which isrealized by short-term rate adjustment according to queue measurement. GDGRQ isa framework for designing schedulers that can both provide (possibly large)deterministic delay bounds and allow statistical delay bounds.
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Hendi, Sarvamangala. "Delay analysis of satellite packet broadcasting systems: a queueing theoretic approach." Thesis, Virginia Polytechnic Institute and State University, 1988. http://hdl.handle.net/10919/80104.
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This thesis develops a stochastic model for satellite packet switching networks, using results from queueing theory that have been previously explored in modeling communication networks. This thesis also analyzes message queueing delay when users of the network are generating data at moderate to high rates. Average packet delay and average number of packets in the system are formulated.
The model developed herein is applied to two cases. In the first case packet transmission and back off times are deterministic. In the second case packet transmission and back off times are exponentially distributed. The input parameters to this model are packet arrival rate, average packet transmission time, average back off time and probability of packet collision. The model yields average packet delay and average number of packets in the system. Methods to compute the probability of collision are presented.Master of Science
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Kamat, Narasinha. "A delay-efficient rerouting scheme for voice over ip traffic." [Gainesville, Fla.] : University of Florida, 2002. http://purl.fcla.edu/fcla/etd/UFE0000548.
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Valverde, Martínez David, and Otte Francisco Javier Parada. "Forward Error Correction for Packet Switched Networks." Thesis, Linköping University, Communication Systems, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-11093.
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The main goal in this thesis is to select and test Forward Error Correction (FEC) schemes suitable for network video transmission over RTP/UDP. There is a general concern in communication networks which is to achieve a tradeoff between reliable transmission and the delay that it takes. Our purpose is to look for techniques that improve the reliability while the realtime delay constraints are fulfilled. In order to achieve it, the FEC techniques focus on recovering the packet losses that come up along any transmission. The FEC schemes that we have selected are Parity Check algorithm, ReedSolomon (RS) codes and a Convolutional code. Simulations are performed to test the different schemes.
The results obtained show that the RS codes are the more powerful schemes in terms of recovery capabilities. However they can not be deployed for every configuration since they go beyond the delay threshold. On the other hand, despite of the Parity Check codes being the less efficient in terms of error recovery, they show a reasonable low delay. Therefore, depending on the packet loss probability that we are working with, we may chose one or other of the different schemes. To summarize, this thesis includes a theoretical background, a thorough analysis of the FEC schemes chosen, simulation results, conclusions and proposed future work.
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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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Marfull, Héctor. "Investigation of packet delay jitter metrics in face of loss and reordering." Thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-4289.
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Nowadays mobility is a field of great importance. The fact of travelling or moving should not mean the rupture of the connection to the Internet. And the current objective is not only to be world-wide connected, it is also to be it always through the best available connection (ABC). It means the need to perform vertical handover to switch between different networks, while maintaining the same Internet connection. All this has to be done in a transparent way to the user. In order provide the highest Quality of Experience some tools are needed to enable checking the status and performance of the different available networks, measuring and collecting statistics, in order to take advantage of each one of them. This thesis presents the theoretical base for a measurement module by describing and analysing different metrics, with special emphasis on delay jitter, collecting and comparing different methods, and discussing their main characteristics and suitability for this goal.
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Kim, Euree Y. "Packet delay and sequence number space in the radio link protocol layer." Thesis, Massachusetts Institute of Technology, 1998. http://hdl.handle.net/1721.1/46217.
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Thesis (S.B. and M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1998.Includes bibliographical references (leaf 85).
by Euree Y. Kim.
S.B.and M.Eng.
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Vellore, Padmini. "Delay, fairness and complexity of selected scheduling disciplines in broadband packet-switched networks /." Internet access available to MUN users only, 2003. http://collections.mun.ca/u?/theses,242728.
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Liu, Yau Shing. "Delay, loss and capacity utilization in single-link and ring-based packet networks." Thesis, Imperial College London, 1989. http://hdl.handle.net/10044/1/47542.
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Qiao, Li, Zhang XiaoLin, Xiong Huagang, and Fei Yuxia. "THE TIMELINESS OF ASYNCHRONOUS PACKET MULTIPLEXING IN SWITCHED ETHERNET." International Foundation for Telemetering, 2004. http://hdl.handle.net/10150/605329.
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International Telemetering Conference Proceedings / October 18-21, 2004 / Town & Country Resort, San Diego, CaliforniaPowered by single-segment switched interconnection, Ethernet can be used in time-critical data acquisition applications. Unlike synchronous time division multiple access, asynchronous packet streams result in congestions and uncertain multiplexing delays. With the delay analysis in the worst case and probabilistic guaranteeing conditions, we restrict the packet-sizes, intervals or traffic burstiness a priori to regulate delay deviations within acceptable scales. Some methods of combinatorics and stochastic theory, e.g. Cumulant Generating Function and the Large Deviation Principle, are used and verified by some simulation-based computations. The influence of time varying delay for telemetry applications is also discussed in some sense.
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Kovach, Bob. "TEMPORAL ALIGNMENT OF TELEMETRY STREAMS WITH DIVERSE DELAY CHARACTERISTICS." International Foundation for Telemetering, 2003. http://hdl.handle.net/10150/605597.
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International Telemetering Conference Proceedings / October 20-23, 2003 / Riviera Hotel and Convention Center, Las Vegas, NevadaIn many test ranges, it is often required to acquire a number of telemetry streams and to process the data simultaneously. Frequently, the streams have different delay characteristics, requiring temporal alignment before the processing step. It is desired to have the capability to align these streams so that the events in each stream are coincident in time. Terawave Communications has developed technology to perform temporal alignment for a number of streams automatically. Additionally, the algorithm performs the delay compensation independent of the source data rate of each stream. Terawave will present the algorithm and share the results of their testing in a test installation.
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Doddi, Srikar. "Empirical modeling of end-to-end delay dynamics in best-effort networks." Texas A&M University, 2003. http://hdl.handle.net/1969.1/2244.
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Quality of Service (QoS) is the ability to guarantee that data sent across a network
will be recieved by the desination within some constraints. For many advanced applications, such as real-time multimedia QoS is determined by four parameters--end-to-end delay, delay jitter, available bandwidth or throughput, and packet drop or
loss rate. It is interesting to study and be able to predict the behavior of end-to-end
packet delays in a Wide area network (WAN) because it directly a??ects the QoS of
real-time distributed applications. In the current work a time-series representation of
end-to-end packet delay dynamics transported over standard IP networks has been
considered. As it is of interest to model the open loop delay dynamics of an IP WAN,
the UDP is used for transport purposes. This research aims at developing models
for single-step-ahead and multi-step-ahead prediction of moving average, one-way
end-to-end delays in standard IP WAN??s.
The data used in this research has been obtained from simulations performed using
the widely used simulator ns-2. Simulation conditions have been tuned to enable
some matching of the end-to-end delay profiles with real traffic data. This has been
accomplished through the use of delay autocorrelation profiles. The linear system
identification models Auto-Regressive eXogenous (AR) and Auto-Regressive Moving
Average with eXtra / eXternal (ARMA) and non-linear models like the Feedforwad
Multi-layer Perceptron (FMLP) have been found to perform accurate single-step-ahead predictions under varying conditions of cross-traffic flow and source send rates.
However as expected, as the multi-step-ahead prediction horizon is increased, the
models do not perform as accurately as the single-step-ahead prediction models. Acceptable
multi-step-ahead predictions for up to 500 msec horizon have been obtained.
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Ta, Xiaoyuan. "A Quality of Service Monitoring System for Service Level Agreement Verification." Thesis, The University of Sydney, 2006. http://hdl.handle.net/2123/1005.
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Service-level-agreement (SLA) monitoring measures network Quality-of-Service (QoS) parameters to evaluate whether the service performance complies with the SLAs. It is becoming increasingly important for both Internet service providers (ISPs) and their customers. However, the rapid expansion of the Internet makes SLA monitoring a challenging task. As an efficient method to reduce both complexity and overheads for QoS measurements, sampling techniques have been used in SLA monitoring systems. In this thesis, I conduct a comprehensive study of sampling methods for network QoS measurements. I develop an efficient sampling strategy, which makes the measurements less intrusive and more efficient, and I design a network performance monitoring software, which monitors such QoS parameters as packet delay, packet loss and jitter for SLA monitoring and verification. The thesis starts with a discussion on the characteristics of QoS metrics related to the design of the monitoring system and the challenges in monitoring these metrics. Major measurement methodologies for monitoring these metrics are introduced. Existing monitoring systems can be broadly classified into two categories: active and passive measurements. The advantages and disadvantages of both methodologies are discussed and an active measurement methodology is chosen to realise the monitoring system. Secondly, the thesis describes the most common sampling techniques, such as systematic sampling, Poisson sampling and stratified random sampling. Theoretical analysis is performed on the fundamental limits of sampling accuracy. Theoretical analysis is also conducted on the performance of the sampling techniques, which is validated using simulation with real traffic. Both theoretical analysis and simulation results show that the stratified random sampling with optimum allocation achieves the best performance, compared with the other sampling methods. However, stratified sampling with optimum allocation requires extra statistics from the parent traffic traces, which cannot be obtained in real applications. In order to overcome this shortcoming, a novel adaptive stratified sampling strategy is proposed, based on stratified sampling with optimum allocation. A least-mean-square (LMS) linear prediction algorithm is employed to predict the required statistics from the past observations. Simulation results show that the proposed adaptive stratified sampling method closely approaches the performance of the stratified sampling with optimum allocation. Finally, a detailed introduction to the SLA monitoring software design is presented. Measurement results are displayed which calibrate systematic error in the measurements. Measurements between various remote sites have demonstrated impressively good QoS provided by Australian ISPs for premium services.
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Ta, Xiaoyuan. "A Quality of Service Monitoring System for Service Level Agreement Verification." University of Sydney, 2006. http://hdl.handle.net/2123/1005.
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Master of Engineering by ResearchService-level-agreement (SLA) monitoring measures network Quality-of-Service (QoS) parameters to evaluate whether the service performance complies with the SLAs. It is becoming increasingly important for both Internet service providers (ISPs) and their customers. However, the rapid expansion of the Internet makes SLA monitoring a challenging task. As an efficient method to reduce both complexity and overheads for QoS measurements, sampling techniques have been used in SLA monitoring systems. In this thesis, I conduct a comprehensive study of sampling methods for network QoS measurements. I develop an efficient sampling strategy, which makes the measurements less intrusive and more efficient, and I design a network performance monitoring software, which monitors such QoS parameters as packet delay, packet loss and jitter for SLA monitoring and verification. The thesis starts with a discussion on the characteristics of QoS metrics related to the design of the monitoring system and the challenges in monitoring these metrics. Major measurement methodologies for monitoring these metrics are introduced. Existing monitoring systems can be broadly classified into two categories: active and passive measurements. The advantages and disadvantages of both methodologies are discussed and an active measurement methodology is chosen to realise the monitoring system. Secondly, the thesis describes the most common sampling techniques, such as systematic sampling, Poisson sampling and stratified random sampling. Theoretical analysis is performed on the fundamental limits of sampling accuracy. Theoretical analysis is also conducted on the performance of the sampling techniques, which is validated using simulation with real traffic. Both theoretical analysis and simulation results show that the stratified random sampling with optimum allocation achieves the best performance, compared with the other sampling methods. However, stratified sampling with optimum allocation requires extra statistics from the parent traffic traces, which cannot be obtained in real applications. In order to overcome this shortcoming, a novel adaptive stratified sampling strategy is proposed, based on stratified sampling with optimum allocation. A least-mean-square (LMS) linear prediction algorithm is employed to predict the required statistics from the past observations. Simulation results show that the proposed adaptive stratified sampling method closely approaches the performance of the stratified sampling with optimum allocation. Finally, a detailed introduction to the SLA monitoring software design is presented. Measurement results are displayed which calibrate systematic error in the measurements. Measurements between various remote sites have demonstrated impressively good QoS provided by Australian ISPs for premium services.
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Alahari, Yeshwanth, and Prashant Buddhiraja. "Analysis of packet loss and delay variation on QoE for H.264 andWebM/VP8 Codecs." Thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-5851.
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The popularity of multimedia services over Internet has increased in the recent years. These services include Video on Demand (VoD) and mobile TV which are predominantly growing, and the user expectations towards the quality of videos are gradually increasing. Different video codec’s are used for encoding and decoding. Recently Google has introduced the VP8 codec which is an open source compression format. It is introduced to compete with existing popular codec namely H.264/AVC developed by ITU-T Video Coding Expert Group (VCEG), as by 2016 there will be a license fee for H.264. In this work we compare the performance of H.264/AVC and WebM/VP8 in an emulated environment. NetEm is used as an emulator to introduce delay/delay variation and packet loss. We have evaluated the user perception of impaired videos using Mean Opinion Score (MOS) by following the International Telecommunication Union (ITU) Recommendations Absolute Category Rating (ACR) and analyzed the results using statistical methods. It was found that both video codec’s exhibit similar performance in packet loss, But in case of delay variation H.264 codec shows better results when compared to WebM/VP8. Moreover along with the MOS ratings we also studied the effect of user feelings and online video watching experience impacts on their perception.Yeshwanth Alahari Phone : +91-9986739097 Buddhiraja Prashant Phone : +46-734897359
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Kao, Tung-Yu. "On the throughput and delay of an adaptive channel access protocol in multihop packet radio network." Ohio : Ohio University, 1993. http://www.ohiolink.edu/etd/view.cgi?ohiou1175709407.
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Fan, Wing Fai. "Admission control and scheduling for guarantee the packet loss rate and delay in IEEE 802.11e WLANs /." View abstract or full-text, 2004. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202004%20FAN.
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Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2004.Includes bibliographical references (leaves 63-66). Also available in electronic version. Access restricted to campus users.
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Toktas, Engin. "Subcarrier Allocation In Ofdma Systems With Time Varying Channel And Packet Arrivals." Master's thesis, METU, 2008. http://etd.lib.metu.edu.tr/upload/3/12610029/index.pdf.
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This study considers the average system throughput and the average delay performances of subcarrier allocation algorithms in OFDMA systems. The effects of varying the number of users, the number of subcarriers, and the statistical characteristics of incoming packets are investigated on the throughput and delay performances of the algorithms. Moreover, a new subcarrier allocation algorithm with low-order computational complexity, which performs very well almost all cases, is proposed. With the aid of the simulations, the significance of channel v.s. queue state information varying with the statistical characteristic of incoming packets is examined and reached some results which can be very valuable for channel estimation and feedback systems. Finally, the stability issue is considered in OFDMA systems and a new heuristic simulation-based method for obtaining the stability region of an OFDMA subcarrier allocation algorithm is proposed.
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Babur, Ozgur. "A Simulation Study Of Scheduling Algorithms For Packet Switching Networks." Master's thesis, METU, 2003. http://etd.lib.metu.edu.tr/upload/2/1219336/index.pdf.
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A scheduling algorithm has the primary role in implementing the quality of service guaranteed to each flow by managing buffer space and selecting which packet to send next with a fair share of network. In this thesis, some scheduling algorithms for packet switching networks are studied. For evaluating their delay, jitter and throughput performances, a discrete event simulator has been developed. It has been
seen that fair scheduling provides, fair allocation of bandwidth, lower delay for sources using less than their full share of bandwidth and protection from ill-behaved
resources.
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Cheng, Kin On. "A multi-stage optical switch with output buffer using WDM for delay lines sharing /." View Abstract or Full-Text, 2003. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202003%20CHENG.
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Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2003.Includes bibliographical references (leaves 77-79). Also available in electronic version. Access restricted to campus users.
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Ji, Kun. "Real-time control over networks." Texas A&M University, 2003. http://hdl.handle.net/1969.1/5834.
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A control system in which sensors, actuators, and controllers are interconnected over a
communication network is called a networked control system (NCS). Enhanced computational
capabilities and bandwidths in the networking technology enabled researchers to develop NCSs
to implement distributed control schemes. This dissertation presents a framework for the
modeling, design, stability analysis, control, and bandwidth allocation of real-time control over
networks. This framework covers key research issues regarding control over networks and can
be the guidelines of NCS design. A single actuator ball magnetic-levitation (maglev) system is
implemented as a test bed for the real-time control over networks to illustrate and verify the
theoretical results of this dissertation. Experimentally verifying the feasibility of Internet-based
real-time control is another main objective of this dissertation.
First, this dissertation proposes a novel NCS model in which the effects of the networkinduced
time delay, data-packet loss, and out-of-order data transmission are all considered.
Second, two simple algorithms based on model-estimator and predictor- and timeout-scheme are
proposed to compensate for the network-induced time delay and packet loss simultaneously.
These algorithms are verified experimentally by the ball maglev test bed. System stability analyses of original and compensated systems are presented. Then, a novel co-design
consideration related to real-time control and network communication is also proposed. The
working range of the sampling frequency is determined by the analysis of the system stability
and network parameters such as time delay, data rate, and data-packet size. The NCS design
chart developed in this dissertation can be a useful guideline for choosing the network and
control parameters in the design of an NCS. Using a real-time operating system for real-time
control over networks is also proposed as one of the main contributions of this dissertation.
After a real-time NCS is successfully implemented, advanced control theories such as
robust control, optimal control, and adaptive control are applied and formulated to improve the
quality of control (QoC) of NCSs. Finally, an optimal dynamic bandwidth management method
is proposed to solve the optimal network scheduling and bandwidth allocation problem when
NCSs are connected to the same network and are sharing the network resource.
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Akcasoy, Alican. "Connectionless Traffic And Variable Packet Size Support In High Speed Network Switches: Improvements For The Delay-limiter Switch." Master's thesis, METU, 2008. http://etd.lib.metu.edu.tr/upload/12609582/index.pdf.
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Quality of Service (QoS) support for real-time traffic is a critical issue in high speed networks. The previously proposed Delay-Limiter Switch working with the Framed-Deadline Scheduler (FDS) is a combined input-output queuing (CIOQ) packet switch that can provide end-to-end bandwidth and delay guarantees for connection-oriented traffic. The Delay-Limiter Switch works with fixed-size packets. It has a scalable architecture and can provide QoS support for connection-oriented real-time traffic in a low-complexity fashion. The Delay-Limiter Switch serves connectionless traffic by using the remaining resources from the connection-oriented traffic. In this case, efficient management of the residual resources plays an important role on the performance of the connectionless traffic. This thesis work integrates new methods to the Delay-Limiter Switch that can improve the performance of the connectionless traffic while still serving the connection-oriented traffic with the promised QoS guarantees. A new method that makes it possible for the Delay-Limiter Switch to support variable-sized packets is also proposed.
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Mohamed, Mahmud H. Etbega. "Some Active Queue Management Methods for Controlling Packet Queueing Delay. Design and Performance Evaluation of Some New Versions of Active Queue Management Schemes for Controlling Packet Queueing Delay in a Buffer to Satisfy Quality of Service Requirements for Real-time Multimedia Applications." Thesis, University of Bradford, 2009. http://hdl.handle.net/10454/4258.
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Traditionally the Internet is used for the following applications: FTP, e-mail and Web traffic. However in the recent years the Internet is increasingly supporting emerging applications such as IP telephony, video conferencing and online games. These new applications have different requirements in terms of throughput and delay than traditional applications. For example, interactive multimedia applications, unlike traditional applications, have more strict delay constraints and less strict loss constraints. Unfortunately, the current Internet offers only a best-effort service to all applications without any consideration to the applications specific requirements. In this thesis three existing Active Queue Management (AQM) mechanisms are modified by incorporating into these a control function to condition routers for better Quality of Service (QoS). Specifically, delay is considered as the key QoS metric as it is the most important metric for real-time multimedia applications. The first modified mechanism is Drop Tail (DT), which is a simple mechanism in comparison with most AQM schemes. A dynamic threshold has been added to DT in order to maintain packet queueing delay at a specified value. The modified mechanism is referred to as Adaptive Drop Tail (ADT). The second mechanism considered is Early Random Drop (ERD) and, iii in a similar way to ADT, a dynamic threshold has been used to keep the delay at a required value, the main difference being that packets are now dropped probabilistically before the queue reaches full capacity. This mechanism is referred to as Adaptive Early Random Drop (AERD). The final mechanism considered is motivated by the well known Random Early Detection AQM mechanism and is effectively a multi-threshold version of AERD in which packets are dropped with a linear function between the two thresholds and the second threshold is moveable in order to change the slope of the dropping function. This mechanism is called Multi Threshold Adaptive Early Random Drop (MTAERD) and is used in a similar way to the other mechanisms to maintain delay around a specified level. The main focus with all the mechanisms is on queueing delay, which is a significant component of end-to-end delay, and also on reducing the jitter (delay variation) A control algorithm is developed using an analytical model that specifies the delay as a function of the queue threshold position and this function has been used in a simulation to adjust the threshold to an effective value to maintain the delay around a specified value as the packet arrival rate changes over time. iv A two state Markov Modulated Poisson Process is used as the arrival process to each of the three systems to introduce burstiness and correlation of the packet inter-arrival times and to present sudden changes in the arrival process as might be encountered when TCP is used as the transport protocol and step changes the size of its congestion window. In the investigations it is assumed the traffic source is a mixture of TCP and UDP traffic and that the mechanisms conserved apply to the TCP based data. It is also assumed that this consists of the majority proportion of the total traffic so that the control mechanisms have a significant effect on controlling the overall delay. The three mechanisms are evaluated using a Java framework and results are presented showing the amount of improvement in QoS that can be achieved by the mechanisms over their non-adaptive counterparts. The mechanisms are also compared with each other and conclusions drawn.
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Siddiqui, Usman Ghani. "The impact of MAC service-time and route discovery time on packet queuing delay in saturated ad hoc networks." Thesis, Wichita State University, 2011. http://hdl.handle.net/10057/3978.
Full textAbstract:
This thesis presents a quantitative analysis of packet queuing delays in saturated DCF
networks. It presents the relationship between medium access control (MAC) delays, routing
delays, and the packet queuing delays, in terms of MAC service-time and route discovery time.
The effects of network size and data transmission rates on the queuing delays were also
analyzed. The simulations reveal that the MAC service-time affects the packet queuing delay in
stationary networks; whereas the route discovery time along with the MAC service-time affects
the queuing delays in mobile networks. Also, the average queuing delay increased with an
increase in the network sizes and data transmission rates, especially for network sizes of 20 and
more.Thesis (M.S.)--Wichita State University, College of Engineering, Dept. of Electrical Engineering and Computer Science.
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Chen, Y. "Mathematical modelling of end-to-end packet delay in multi-hop wireless networks and their applications to QoS provisioning." Thesis, University College London (University of London), 2013. http://discovery.ucl.ac.uk/1415093/.
Full textAbstract:
This thesis addresses the mathematical modelling of end-to-end packet delay for Quality of Service (QoS) provisioning in multi-hop wireless networks. The multi-hop wireless technology increases capacity and coverage in a cost-effective way and it has been standardised in the Fourth-Generation (4G) standards. The effective capacity model approximates end-to-end delay performances, including Complementary Cumulative Density Function (CCDF) of delay, average delay and jitter. This model is first tested using Internet traffic trace from a real gigabit Ethernet gateway. The effective capacity model is developed based on single-hop and continuous-time communication systems but a multi-hop wireless system is better described to be multi-hop and time-slotted. The thesis extends the effective capacity model by taking multi-hop and time-slotted concepts into account, resulting in two new mathematical models: the multi-hop effective capacity model for multi-hop networks and the mixed continuous/discrete-time effective capacity model for time-slotted networks. Two scenarios are considered to validate these two effective capacity-based models based on ideal wireless communications (the physical-layer instantaneous transmission rate is the Shannon channel capacity): 1) packets traverse multiple wireless network devices and 2) packets are transmitted to or received from a wireless network device every Transmission Time Interval (TTI). The results from these two scenarios consistently show that the new mathematical models developed in the thesis characterise end-to-end delay performances accurately. Accurate and efficient estimators for end-to-end packet delay play a key role in QoS provisioning in modern communication systems. The estimators from the new effective capacity-based models are directly tested in two systems, faithfully created using realistic simulation techniques: 1) the IEEE 802.16-2004 networks and 2) wireless tele-ultrasonography medical systems. The results show that the estimation and simulation results are in good agreement in terms of end-to-end delay performances.
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Karakas, Mehmet. "Determination Of Network Delay Distribution Over The Internet." Master's thesis, METU, 2003. http://etd.lib.metu.edu.tr/upload/1223155/index.pdf.
Full textAbstract:
The rapid growth of the Internet and the proliferation of its new applications pose a serious challenge in network performance management and monitoring. The current Internet has no mechanism for providing feedback on network congestion to the end-systems at the IP layer. For applications and their end hosts, end-to-end measurements may be the only way of measuring network performance.
Understanding the packet delay and loss behavior of the Internet is important for proper design of network algorithms such as routing and flow control algorithms, for the dimensioning of buffers and link capacity, and for choosing parameters in simulation and analytic studies.
In this thesis, round trip time (RTT), one-way network delay and packet loss in the Internet are measured at different times of the day, using a Voice over IP (VoIP) device. The effect of clock skew on one-way network delay measurements is eliminated by a Linear Programming algorithm, implemented in MATLAB. Distributions of one-way network delay and RTT in the Internet are determined. It is observed that delay distribution has a gamma-like shape with heavy tail. It is tried to model delay distribution with gamma, lognormal and Weibull distributions. It is observed that most of the packet losses in the Internet are single packet losses. The effect of firewall on delay measurements is also observed.
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Nygren, Johannes. "Input-Output Stability Analysis of Networked Control Systems." Doctoral thesis, Uppsala universitet, Reglerteknik, 2016. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-272344.
Full textAbstract:
The main focus of the thesis is to derive stability criteria for networked control system (NCS) models featuring imperfections such as time-varying and constant delays, quantization, packet dropouts, and non-uniform sampling intervals. The main method of proof is based on matrix algebra, as opposed to methods using Lyapunov functions or integral quadratic constraints (IQC). This work puts a particular focus on handling systems with a single integrator. This framework is elaborated in different specific directions as motivated by practical realizations of NCSs, as well as through numerical examples. A novel proof of the discrete time multivariate circle criterion and the Tsypkin criterion for systems including a single integrator is presented, as well as a stability criterion for linear systems with a single integrator subject to variable sampling periods and sector-bounded nonlinear feedback. Four stability criteria for different classes of systems subject to packet loss and time-varying delay are given. Stability criteria for a closed loop system switching between a set of linear time-invariant systems (LTIs) are proved. This result is applied to a single-link NCS with feedback subject to packet loss. Finally, necessary and sufficient conditions for delay-independent stability of an LTI system subject to nonlinear feedback are derived.
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Channe, Gowda Anushree. "Latency and Jitter Control in 5G Ethernet Fronthaul Network." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2019. http://amslaurea.unibo.it/17651/.
Full textAbstract:
With 5G technology, networks are expected to offer high speed with ultra-low latency among different users. Maintaining the current network architecture will lead to an unsustainable transport delay and jitters increase. Limiting the transport delay and the jitters have become a necessity for mobile network operators. The main requirement in 5G networks is the demand of limiting the transport delay. This, thesis proposes a novel mechanism to minimize packet delay and delay variation in 5G Ethernet fronthaul network. The goal is to achieve bounded delay aggregation of traffic ,suitable for application in fronthaul transport. Hybrid switching technology can be adopted to provide efficient fronthaul in 5G. Hybrid switches allows to multiplex traffics with different characteristics over the same wavelengths, thus increasing the network resource utilization. This thesis proposes a scheduling mechanism for hybrid switches to aggregate streams from the network, the Bypass traffic (BP), and the traffic from the fronthaul links, the ADD traffic, using an algorithm which looks for the time gaps in the BP stream for the insertion of the ADD traffic. The proposed strategy minimizes the delay of packets by making use of the available gaps during the transmission to limit the network latency. The size of the required time gaps, the time window, is suitably reduced by dividing the timeout time duration with number of intervals (N) with the Window reduction mechanism so that the delay variation or jitter of both aggregated streams are bounded. The results demonstrate that the aforementioned requirements are can be achieved by suitably tuning the parameters of the algorithm inputs, mainly the window reduction factor, timeout time duration and the number of intervals, resulting in values of packet delay and delay variation bounded at 10 microseconds or even lower up to 85-90percent carried load of aggregated flows. Hence, we show their suitability for delay sensitive future applications in 5G networking.
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Win, Htoo Aung. "BSM Message and Video Streaming Quality Comparative Analysis Using Wave Short Message Protocol (WSMP)." Thesis, University of North Texas, 2019. https://digital.library.unt.edu/ark:/67531/metadc1538706/.
Full textAbstract:
Vehicular ad-hoc networks (VANETs) are used for vehicle-to-vehicle (V2V) and vehicle-to-infrastructure (V2I) communications. The IEEE 802.11p/WAVE (Wireless Access in Vehicular Environment) and with WAVE Short Messaging Protocol (WSMP) has been proposed as the standard protocol for designing applications for VANETs. This communication protocol must be thoroughly tested before reliable and efficient applications can be built using its protocols. In this paper, we perform on-road experiments in a variety of scenarios to evaluate the performance of the standard. We use commercial VANET devices with 802.11p/WAVE compliant chipsets for both BSM (basic safety messages) as well as video streaming applications using WSMP as a communication protocol. We show that while the standard performs well for BSM application in lightly loaded conditions, the performance becomes inferior when traffic and other performance metric increases. Furthermore, we also show that the standard is not suitable for video streaming due to the bursty nature of traffic and the bandwidth throttling, which is a major shortcoming for V2X applications.
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Awan, Naser Saeed. "Characterization of SIP Signaling-Messages Over OpenSIPS Running On Multicore Server." Thesis, KTH, Skolan för informations- och kommunikationsteknik (ICT), 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-121530.
Full textAbstract:
Over the course of last decade, the demand for VoIP (Voice over Internet Protocol) applications has increased significantly among enterprises and individuals due to its low cost. This increasing demand resulted in a significant increase in users who require reliable VoIP communication systems. QoS (Quality of Service) is a major issue in VoIP implementation and is a method to impel the development of real-time multimedia services like VoIP and videoconferencing. However, there are certain challenges in achieving QoS for VoIP application, which need special attentions; like latency and packet loss. The VoIP servers which are functioning on single core software/hardware model have high latency and packet loss issues due to their limited processing bandwidth. A multicore software/hardware model is the solution to cope up with the increasing demands of VoIP and yet an active research area in telecommunication. Using a multicore software/hardware model for VoIP has several challenges, one of the challenges is to design and implement QoS Benchmarking module for VoIP client and server on multicore. In this thesis the focus is on latency and packet loss of SIP messages on OpenSIPS server. This is done by performing stress testing for QoS benchmarking, where delay and call drop rate is calculated for SIP (Session Initiation Protocol) signaling messages on parallel VoIP client server model. The model is built in C for multicore and is used as a simulation tool. SIP is widely deployed protocol for call establishment, maintenance and termination in VoIP.
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Wang, Yufeng. "Opportunistic Scheduling and Cooperative Relaying in Wireless Networks." Scholar Commons, 2012. http://scholarcommons.usf.edu/etd/4416.
Full textAbstract:
The demand for ever larger, more efficient, reliable and cost effective communication networks necessitates new network architectures, such as wireless ad hoc networks, cognitive radio, relaying networks, and wireless sensor networks. The study of such networks requires a fundamental shift from thinking of a network as a collection of independent communication pipes, to a multi-user channel where users cooperate via conferencing, relaying, and joint source-channel coding.
The traditional centralized networks, such as cellular networks, include a central controller and a fixed infrastructure, in which every node communicates with each other via a centralized based station (BS). However, for a decentralized network, such as wireless ad hoc networks and wireless sensor networks, there is no infrastructure support and no central controllers. In such multi-user wireless networks, the scheduling algorithm plays an essential role in efficiently assigning channel resources to different users for better system performance, in terms of system throughput, packet-delay, stability and fairness.
In this dissertation, our main goal is to develop practical scheduling algorithms in wireless ad hoc networks to enhance system performance, in terms of throughput, delay and stability. Our dissertation mainly consists of three main parts.
First, we identify major challenges intrinsic to ad hoc networks that affect the system performance, in terms of throughput limits, delay and stability condition.
Second, we develop scheduling algorithms for wireless ad hoc networks, with various considerations of non-cooperative relays and cooperative relays, fixed-rate transmission and adaptive-rate transmission, full-buffer traffic model and finite-buffer traffic model. Specifically, we propose an opportunistic scheduling scheme and study the throughput and delay performance, with fixed-rate transmissions in a two-hop wireless ad hoc networks. In the proposed scheduling scheme, we prove two key inequalities that capture the various tradeoffs inherent in the broad class of opportunistic relaying protocols, illustrating that no scheduling and routing algorithm can simultaneously yield lower delay and higher throughput. We then develop an adaptive rate transmission scheme with opportunistic scheduling, with the constraints of practical assumptions on channel state information (CSI) and limited feedback, which achieves an optimal system throughput scaling order. Along this work with the consideration of finite-buffer model, we propose a Buffer-Aware Adaptive (BAA) scheduler which considers both channel state and buffer conditions to make scheduling decisions, to reduce average packet delay, while maintaining the queue stability condition of the networks. The proposed algorithm is an improvement over existing algorithms with adaptability and bounded potential throughput reduction.
In the third part, we extend the methods and analyses developed for wireless ad hoc networks to a practical Aeronautical Communication Networks (ACN) and present the system performance of such networks. We use our previously proposed scheduling schemes and analytical methods from the second part to investigate the issues about connectivity, throughput and delay in ACN, for both single-hop and two-hop communication models. We conclude that the two-hop model achieves greater throughput than the single-hop model for ACN. Both throughput and delay performances are characterized.
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Devadason, Tarith Navendran. "The virtual time function and rate-based schedulers for real-time communications over packet networks." University of Western Australia. School of Electrical, Electronic and Computer Engineering, 2007. http://theses.library.uwa.edu.au/adt-WU2007.0108.
Full textAbstract:
[Truncated abstract] The accelerating pace of convergence of communications from disparate application types onto common packet networks has made quality of service an increasingly important and problematic issue. Applications of different classes have diverse service requirements at distinct levels of importance. Also, these applications offer traffic to the network with widely variant characteristics. Yet a common network is expected at all times to meet the individual communication requirements of each flow from all of these application types. One group of applications that has particularly critical service requirements is the class of real-time applications, such as packet telephony. They require both the reproduction of a specified timing sequence at the destination, and nearly instantaneous interaction between the users at the endpoints. The associated delay limits (in terms of upper bound and variation) must be consistently met; at every point where these are violated, the network transfer becomes worthless, as the data cannot be used at all. In contrast, other types of applications may suffer appreciable deterioration in quality of service as a result of slower transfer, but the goal of the transfer can still largely be met. The goal of this thesis is to evaluate the potential effectiveness of a class of packet scheduling algorithms in meeting the specific service requirements of real-time applications in a converged network environment. Since the proposal of Weighted Fair Queueing, there have been several schedulers suggested to be capable of meeting the divergent service requirements of both real-time and other data applications. ... This simulation study also sheds light on false assumptions that can be made about the isolation produced by start-time and finish-time schedulers based on the deterministic bounds obtained. The key contributions of this work are as follows. We clearly show how the definition of the virtual time function affects both delay bounds and delay distributions for a real-time flow in a converged network, and how optimality is achieved. Despite apparent indications to the contrary from delay bounds, the simulation analysis demonstrates that start-time rate-based schedulers possess useful characteristics for real-time flows that the traditional finish-time schedulers do not. Finally, it is shown that all the virtual time rate-based schedulers considered can produce isolation problems over multiple hops in networks with high loading. It becomes apparent that the benchmark First-Come-First-Served scheduler, with spacing and call admission control at the network ingresses, is a preferred arrangement for real-time flows (although lower priority levels would also need to be implemented for dealing with other data flows).
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Mulkijanyan, Nina. "Evaluation Procedure for QoS of Short Message Service : International SMS Route Analysis." Thesis, KTH, Skolan för informations- och kommunikationsteknik (ICT), 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-49828.
Full textAbstract:
Due to its ubiquitous availability, Short Message Service (SMS), first introduced in the 1980s, became not only the most popular way of communication, but also stimulated the development of SMS-based value added services. This application-to-person traffic is delivered to end users through SMS aggregators who provide the link between service providers and mobile carriers. In order to perform optimal traffic routing, the aggregators need to estimate the quality of each potential international route to the specified destination. The evaluation criteria include end-to-end delivery time, as well as correct verification of delivered data. This thesis suggests a method of quality of service (QoS) assessment for international SMS service which combines two types of tests, end-to-end delay measurements and various verification tests. A prototype of the testing system for international SMS service was developed to generate SMS traffic, collect and analyze results, and evaluate the experienced QoS of the SMS route used in accordance with the proposed approach. As a part of end-to- end delay measurement tests, SMS traffic was sent to Singtel network in Singapore along two routes. The verification tests were executed via different routes to two mobile networks: Singtel and Tele2 (Sweden). The results of the performed measurements determined the route with the highest QoS, i.e. the one with bigger bottleneck bandwidth and lower data loss rate. The prototype of the SMS testing system can be used by SMS aggregators to verify delivery of a SMS message, check the integrity of the message, figure out interconnection type of the route supplier with the destination carrier and to identify the presence of load balancers in the path. The prototype also makes it possible to compare end-to-end delay times of several routes and compute bottleneck values for each of the tested routes.
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Jogi, Mutyalu, and Madhu Vundavalli. "Evaluation of TCP Performance in 3G Mobile Networks." Thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-2411.
Full textAbstract:
With the increase in mobile broadband services the operators are gaining profits by providing high speed Internet access over the mobile network. On the other side they are also facing challenges to give QoS guarantee to the customers. In this thesis we investigate the impact of data rate and payload size on One Way Delay (OWD) and packet loss over TCP performance in 3G networks. Our goal is to evaluate the OWD and packet loss characteristics with respect to payload size and data rate from the collected network level traces. To collect these traces an experimental testbed is setup with Endace Data Acquisition and Generation (DAG) cards, for accurate measurements Endace DAG cards together with Global Positioning System (GPS) synchronization is implemented. The experiments are conducted for three different Swedish mobile operator networks and further the statistics of OWD measurements and packet loss for different data rates and payload sizes are evaluated. Our results indicate that the minimal OWD occurred at higher data rates and also shows a high delay variability. The packet loss has much impact on higher data rates and larger payload sizes, as the packet loss increases with the increase in data rate and payload size.
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Saad, Amna. "Secure VoIP performance measurement." Thesis, Loughborough University, 2013. https://dspace.lboro.ac.uk/2134/13426.
Full textAbstract:
This project presents a mechanism for instrumentation of secure VoIP calls. The experiments were run under different network conditions and security systems. VoIP services such as Google Talk, Express Talk and Skype were under test. The project allowed analysis of the voice quality of the VoIP services based on the Mean Opinion Score (MOS) values generated by Perceptual valuation of Speech Quality (PESQ). The quality of the audio streams produced were subjected to end-to-end delay, jitter, packet loss and extra processing in the networking hardware and end devices due to Internetworking Layer security or Transport Layer security implementations. The MOS values were mapped to Perceptual Evaluation of Speech Quality for wideband (PESQ-WB) scores. From these PESQ-WB scores, the graphs of the mean of 10 runs and box and whisker plots for each parameter were drawn. Analysis on the graphs was performed in order to deduce the quality of each VoIP service. The E-model was used to predict the network readiness and Common vulnerability Scoring System (CVSS) was used to predict the network vulnerabilities. The project also provided the mechanism to measure the throughput for each test case. The overall performance of each VoIP service was determined by PESQ-WB scores, CVSS scores and the throughput. The experiment demonstrated the relationship among VoIP performance, VoIP security and VoIP service type. The experiment also suggested that, when compared to an unsecure IPIP tunnel, Internetworking Layer security like IPSec ESP or Transport Layer security like OpenVPN TLS would improve a VoIP security by reducing the vulnerabilities of the media part of the VoIP signal. Morever, adding a security layer has little impact on the VoIP voice quality.
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Gui, Li. "A transport protocol for real-time applications in wireless networked control systems." Thesis, Queensland University of Technology, 2010. https://eprints.qut.edu.au/45460/1/Li_Gui_Thesis.pdf.
Full textAbstract:
A Networked Control System (NCS) is a feedback-driven control system wherein the control loops are closed through a real-time network. Control and feedback signals in an NCS are exchanged among the system’s components in the form of information packets via the network. Nowadays, wireless technologies such as IEEE802.11 are being introduced to modern NCSs as they offer better scalability, larger bandwidth and lower costs. However, this type of network is not designed for NCSs because it introduces a large amount of dropped data, and unpredictable and long transmission latencies due to the characteristics of wireless channels, which are not acceptable for real-time control systems. Real-time control is a class of time-critical application which requires lossless data transmission, small and deterministic delays and jitter. For a real-time control system, network-introduced problems may degrade the system’s performance significantly or even cause system instability. It is therefore important to develop solutions to satisfy real-time requirements in terms of delays, jitter and data losses, and guarantee high levels of performance for time-critical communications in Wireless Networked Control Systems (WNCSs). To improve or even guarantee real-time performance in wireless control systems, this thesis presents several network layout strategies and a new transport layer protocol. Firstly, real-time performances in regard to data transmission delays and reliability of IEEE 802.11b-based UDP/IP NCSs are evaluated through simulations. After analysis of the simulation results, some network layout strategies are presented to achieve relatively small and deterministic network-introduced latencies and reduce data loss rates. These are effective in providing better network performance without performance degradation of other services. After the investigation into the layout strategies, the thesis presents a new transport protocol which is more effcient than UDP and TCP for guaranteeing reliable and time-critical communications in WNCSs. From the networking perspective, introducing appropriate communication schemes, modifying existing network protocols and devising new protocols, have been the most effective and popular ways to improve or even guarantee real-time performance to a certain extent. Most previously proposed schemes and protocols were designed for real-time multimedia communication and they are not suitable for real-time control systems. Therefore, devising a new network protocol that is able to satisfy real-time requirements in WNCSs is the main objective of this research project. The Conditional Retransmission Enabled Transport Protocol (CRETP) is a new network protocol presented in this thesis. Retransmitting unacknowledged data packets is effective in compensating for data losses. However, every data packet in realtime control systems has a deadline and data is assumed invalid or even harmful when its deadline expires. CRETP performs data retransmission only in the case that data is still valid, which guarantees data timeliness and saves memory and network resources. A trade-off between delivery reliability, transmission latency and network resources can be achieved by the conditional retransmission mechanism. Evaluation of protocol performance was conducted through extensive simulations. Comparative studies between CRETP, UDP and TCP were also performed. These results showed that CRETP significantly: 1). improved reliability of communication, 2). guaranteed validity of received data, 3). reduced transmission latency to an acceptable value, and 4). made delays relatively deterministic and predictable. Furthermore, CRETP achieved the best overall performance in comparative studies which makes it the most suitable transport protocol among the three for real-time communications in a WNCS.
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Yeo, Yong-Kee. "Dynamically Reconfigurable Optical Buffer and Multicast-Enabled Switch Fabric for Optical Packet Switching." Diss., Georgia Institute of Technology, 2006. http://hdl.handle.net/1853/14615.
Full textAbstract:
Optical packet switching (OPS) is one of the more promising solutions for meeting the diverse needs of broadband networking applications of the future. By virtue of its small data traffic granularity as well as its nanoseconds switching speed, OPS can be used to provide connection-oriented or connectionless services for different groups of users with very different networking requirements. The optical buffer and the switch fabric are two of the most important components in an OPS router. In this research, novel designs for the optical buffer and switch fabric are proposed and experimentally demonstrated. In particular, an optical buffer that is based on a folded-path delay-line tree architecture will be discussed. This buffer is the most compact non-recirculating optical delay line buffer to date, and it uses an array of high-speed ON-OFF optical reflectors to dynamically reconfigure its delay within several nanoseconds. A major part of this research is devoted to the design and performance optimization of these high-speed reflectors. Simulations and measurements are used to compare different reflector designs as well as to determine their optimal operating conditions. Another important component in the OPS router is the switch fabric, and it is used to perform space switching for the optical packets. Optical switch fabrics are used to overcome the limitations imposed by conventional electronic switch fabrics: high power consumption and dependency on the modulation format and bit-rate of the signals. Currently, only those fabrics that are based on the broadcast-and-select architecture can provide truly non-blocking multicast services to all input ports. However, a major drawback of these fabrics is that they are implemented using a large number of optical gates based on semiconductor optical amplifiers (SOA). This results in large component count and high energy consumption. In this research, a new multicast-capable switch fabric which does not require any SOA gates is proposed. This fabric relies on a passive all-optical gate that is based on the Four-wave mixing (FWM) wavelength conversion process in a highly-nonlinear fiber. By using this new switch architecture, a significant reduction in component count can be expected.
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Ickin, Selim. "Implementation of Measurement Module For Seamless Vertical Handover." Thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-5104.
Full textAbstract:
Research on heterogeneous seamless handover has become popular, since the wireless networking systems were introduced with mobility. The typical current vertical handover mechanism consists of an architecture built at Layer 3 and implemented to serve for different technologies. Even though it has advantages in terms of simplicity, there still exist drawbacks such as the need of adaptation of the network architecture to different network technologies and systems. Providing transparency to the heterogeneous seamless handover can be provided by conducting the handover process at a higher layer. By that way, efficient handover decisions for vertical handover are made with more number of constraints that will lead to high performance, accessibility and low cost. Making this possible is by providing Quality of Experience (QoE) and obtaining current information of the throughput by measurements in the network. Analyzing and interpreting the statistics collected through measurements are vital in terms of decision making and to decide when to perform vertical handover. This thesis consists of the implementation of measurement module in two different approaches (Payload Dependent Approach and Payload Independent Approach) that will provide these statistics to the storage module in PERIMETER project.