554 research outputs found

    An intelligent-agent approach for managing congestion in W-CDMA networks

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    PhDResource Management is a crucial aspect in the next generation cellular networks since the use of W-CDMA technology gives an inherent flexibility in managing the system capacity. The concept of a “Service Level Agreement” (SLA) also plays a very important role as it is the means to guarantee the quality of service provided to the customers in response to the level of service to which they have subscribed. Hence there is a need to introduce effective SLA-based policies as part of the radio resource management. This work proposes the application of intelligent agents in SLA-based control in resource management, especially when congestion occurs. The work demonstrates the ability of intelligent agents in improving and maintaining the quality of service to meet the required SLA as the congestion occurs. A particularly novel aspect of this work is the use of learning (here Case Based Reasoning) to predict the control strategies to be imposed. As the system environment changes, the most suitable policy will be implemented. When congestion occurs, the system either proposes the solution by recalling from experience (if the event is similar to what has been previously solved) or recalculates the solution from its knowledge (if the event is new). With this approach, the system performance will be monitored at all times and a suitable policy can be immediately applied as the system environment changes, resulting in maintaining the system quality of service

    4. generĂĄciĂłs mobil rendszerek kutatĂĄsa = Research on 4-th Generation Mobile Systems

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    A 3G mobil rendszerek szabvĂĄnyosĂ­tĂĄsa a vĂ©gĂ©hez közeledik, legalĂĄbbis a meghatĂĄrozĂł kĂ©pessĂ©gek tekintetĂ©ben. EzĂ©rt lĂ©tfontossĂĄgĂș azon technikĂĄk, eljĂĄrĂĄsok vizsgĂĄlata, melyek a következƑ, 4G rendszerekben meghatĂĄrozĂł szerepet töltenek majd be. Több ilyen kutatĂĄsi irĂĄnyvonal is lĂ©tezik, ezek közĂŒl projektĂŒnkben a fontosabbakra koncentrĂĄltunk. A következƑben felsoroljuk a kutatott terĂŒleteket, Ă©s röviden összegezzĂŒk az elĂ©rt eredmĂ©nyeket. SzĂłrt spektrumĂș rendszerek KifejlesztettĂŒnk egy Ășj, rĂĄdiĂłs interfĂ©szen alkalmazhatĂł hĂ­vĂĄsengedĂ©lyezĂ©si eljĂĄrĂĄst. SzimulĂĄciĂłs vizsgĂĄlatokkal tĂĄmasztottuk alĂĄ a megoldĂĄs hatĂ©konysĂĄgĂĄt. A projektben kutatĂłkĂ©nt rĂ©sztvevƑ Jeney GĂĄbor sikeresen megvĂ©dte Ph.D. disszertĂĄciĂłjĂĄt neurĂĄlis hĂĄlĂłzatokra Ă©pĂŒlƑ többfelhasznĂĄlĂłs detekciĂłs technikĂĄk tĂ©mĂĄban. Az elĂ©rt eredmĂ©nyek Imre SĂĄndor MTA doktori disszertĂĄciĂłjĂĄba is beĂ©pĂŒltek. IP alkalmazĂĄsa mobil rendszerekben TovĂĄbbfejlesztettĂŒk, teszteltĂŒk Ă©s ĂĄltalĂĄnosĂ­tottuk a projekt keretĂ©ben megalkotott Ășj, gyƱrƱ alapĂș topolĂłgiĂĄra Ă©pĂŒlƑ, a jelenleginĂ©l nagyobb megbĂ­zhatĂłsĂĄgĂș IP alapĂș hozzĂĄfĂ©rĂ©si koncepciĂłt. A tĂ©makörben Szalay MĂĄtĂ© Ph.D. disszertĂĄciĂłja mĂĄr a nyilvĂĄnos vĂ©dĂ©sig jutott. Kvantum-informatikai mĂłdszerek alkalmazĂĄsa 3G/4G detekciĂłra Új, kvantum-informatikai elvekre Ă©pĂŒlƑ többfelhasznĂĄlĂłs detekciĂłs eljĂĄrĂĄst dolgoztunk ki. Ehhez Ășj kvantum alapĂș algoritmusokat is kifejlesztettĂŒnk. Az eredmĂ©nyeket nemzetközi folyĂłiratok mellett egy sajĂĄt könyvben is publikĂĄltuk. | The project consists of three main research directions. Spread spectrum systems: we developed a new call admission control method for 3G air interfaces. Project member Gabor Jeney obtained the Ph.D. degree and project leader Sandor Imre submitted his DSc theses from this area. Application of IP in mobile systems: A ring-based reliable IP mobility mobile access concept and corresponding protocols have been developed. Project member MĂĄtĂ© Szalay submitted his Ph.D. theses from this field. Quantum computing based solutions in 3G/4G detection: Quantum computing based multiuser detection algorithm was developed. Based on the results on this field a book was published at Wiley entitled: 'Quantum Computing and Communications - an engineering approach'

    A novel non-intrusive objective method to predict voice quality of service in LTE networks.

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    This research aimed to introduce a novel approach for non-intrusive objective measurement of voice Quality of Service (QoS) in LTE networks. While achieving this aim, the thesis established a thorough knowledge of how voice traffic is handled in LTE networks, the LTE network architecture and its similarities and differences to its predecessors and traditional ground IP networks and most importantly those QoS affecting parameters which are exclusive to LTE environments. Mean Opinion Score (MOS) is the scoring system used to measure the QoS of voice traffic which can be measured subjectively (as originally intended). Subjective QoS measurement methods are costly and time-consuming, therefore, objective methods such as Perceptual Evaluation of Speech Quality (PESQ) were developed to address these limitations. These objective methods have a high correlation with subjective MOS scores. However, they either require individual calculation of many network parameters or have an intrusive nature that requires access to both the reference signal and the degraded signal for comparison by software. Therefore, the current objective methods are not suitable for application in real-time measurement and prediction scenarios. A major contribution of the research was identifying LTE-specific QoS affecting parameters. There is no previous work that combines these parameters to assess their impacts on QoS. The experiment was configured in a hardware in the loop environment. This configuration could serve as a platform for future research which requires simulation of voice traffic in LTE environments. The key contribution of this research is a novel non-intrusive objective method for QoS measurement and prediction using neural networks. A comparative analysis is presented that examines the performance of four neural network algorithms for non-intrusive measurement and prediction of voice quality over LTE networks. In conclusion, the Bayesian Regularization algorithm with 4 neurons in the hidden layer and sigmoid symmetric transfer function was identified as the best solution with a Mean Square Error (MSE) rate of 0.001 and regression value of 0.998 measured for the testing data set

    End-to-End Delay Distribution Analysis for Stochastic Admission Control in Multi-hop Wireless Networks

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    A Real-Time Communication Framework for Wireless Sensor Networks

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    Recent advances in miniaturization and low power design have led to a flurry of activity in wireless sensor networks. Sensor networks have different constraints than traditional wired networks. A wireless sensor network is a special network with large numbers of nodes equipped with embedded processors, sensors, and radios. These nodes collaborate to accomplish a common task such as environment monitoring or asset tracking. In many applications, sensor nodes will be deployed in an ad-hoc fashion without careful planning. They must organize themselves to form a multihop, wireless communication network. In sensor network environments, much research has been conducted in areas such as power consumption, self-organisation techniques, routing between the sensors, and the communication between the sensor and the sink. On the other hand, real-time communication with the Quality of Service (QoS) concept in wireless sensor networks is still an open research field. Most protocols either ignore real time or simply attempt to process as fast as possible and hope that this speed is sufficient to meet the deadline. However, the introduction of real-time communication has created additional challenges in this area. The sensor node spends most of its life routing packets from one node to another until the packet reaches the sink; therefore, the node functions as a small router most of the time. Since sensor networks deal with time-critical applications, it is often necessary for communication to meet real time constraints. However, research that deals with providing QoS guarantees for real-time traffic in sensor networks is still in its infancy.This thesis presents a real-time communication framework to provide quality of service in sensor networks environments. The proposed framework consists of four components: First, present an analytical model for implementing Priority Queuing (PQ) in a sensor node to calculate the queuing delay. The exact packet delay for corresponding classes is calculated. Further, the analytical results are validated through an extensive simulation study. Second, report on a novel analytical model based on a limited service polling discipline. The model is based on an M/D/1 queuing system (a special class of M/G/1 queuing systems), which takes into account two different classes of traffic in a sensor node. The proposed model implements two queues in a sensor node that are served in a round robin fashion. The exact queuing delay in a sensor node for corresponding classes is calculated. Then, the analytical results are validated through an extensive simulation study. Third, exhibit a novel packet delivery mechanism, namely the Multiple Level Stateless Protocol (MLSP), as a real-time protocol for sensor networks to guarantee the traffic in wireless sensor networks. MLSP improves the packet loss rate and the handling of holes in sensor network much better than its counterpart, MMSPEED. It also introduces the k-limited polling model for the first time. In addition, the whole sending packets dropped significantly compared to MMSPEED, which it leads to decrease the consumption power. Fourth, explain a new framework for moving data from the sink to the user, at a low cost and low power, using the Universal Mobile Telecommunication System (UMTS), which is standard for the Third Generation Mobile System (3G). The integration of sensor networks with the 3G mobile network infrastructure will reduce the cost of building new infrastructures and enable the large-scale deployment of sensor network

    Enhancement of perceived quality of service for voice over internet protocol systems

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    Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods
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