• Journal of IKEEE
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• v.27 no.4
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• pp.617-623
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• 2023

With the evolution of communication networks, there is a growing demand for stable high-speed data connections to support services relying on large-capacity data. The increasing volume of packet data aggregated from user devices underscores the significance of quality diagnostics for the backhaul network, an intermediate link transmitting data to the core network. This paper conducts empirical research on techniques to diagnose issues within the backhaul network through practical case studies, through diagnosing various factors such as circuit bandwidth, speed disparities within switches, network segment-specific buffer sizes, routing policies, among other factors that could potentially cause RTT (Round Trip Time) delays and performance degradation.

• Journal of KIISE:Information Networking
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• v.35 no.5
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• pp.409-414
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• 2008

In mobile ad hoc networks, most of on demand routing protocols such as DSR and AODV do not deal with traffic load during the route discovery procedure. To solve the congestion and achieve load balancing, many protocols have been proposed. However, the existing load balancing schemes has only considered avoiding the congested route in the route discovery procedure or finding an alternative route path during a communication session. To mitigate this problem, we have proposed a new scheme which considers the packet bursting mechanism in congested nodes. The proposed packet bursting scheme, which is originally introduced in IEEE 802.11e QoS specification, is to transmit multiple packets right after channel acquisition. Thus, congested nodes can forward buffered packets promptly and minimize bottleneck situation. Each node begins to transmit packets in normal mode whenever its congested status is dissolved. We also propose two threshold values to define exact overloaded status adaptively; one is interface queue length and the other is buffer occupancy time. Through an experimental simulation study, we have compared and contrasted our protocol with normal on demand routing protocols and showed that the proposed scheme is more efficient and effective especially when network traffic is heavily loaded.

• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.263-268
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• 2004

The end-to-end characteristic Is an important factor for QoS support. Since network users and required bandwidths for applications increase, the efficient usage of networks has been intensively investigated for the better utilization of network resources. The distributed adaptive routing is the typical routing algorithm that is used in the current Internet. The DCLC(Delay Constrained 1.east Cost) path problem has been shown to be NP-hard problem. The path cost of LD path is relatively more expensive than that of LC path, and the path delay of LC path is relatively higher than that of LD path in DCLC problem. In this paper, we investigate the performance of heuristic algorithm for the DCLC problem with new factor which is probabilistic combination of cost and delay. Recently Dr. Salama proposed a polynomial time algorithm called DCUR. The algorithm always computes a path, where the cost of the path is always within 10% from the optimal CBF. Our evaluation showed that heuristic we propose is more than 38% better than DCUR with cost when number of nodes is more than 200. The new factor takes in account both cost and delay at the same time.

• Journal of Broadcast Engineering
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• v.12 no.2
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• pp.148-158
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• 2007

Recently, IEEE 802.15.4a(low-rate UWB) technique has been paid much attention to the LR-UWB communication system for WPAN. However, there are various interferences such as MPI(Multi Path Interference) or IPI(Inter Piconet Interference) in IEEE 802.15.4a wireless channel. In order to cancel various interferences occurred to WPAN environment, in this paper, we propose a UWB wireless communication system with high QoS(Quality of Service) which is a chaotic-OOK(On-Off Keying) system using unipolar ZCD(Zero Correlation Duration) spreading code in physical layer level. Furthermore, we analyze its performance via simulations and verify the availability of proposed system with prototype implementation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1B
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• pp.42-48
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• 2002

In broadband wireless networks, the effective meeting of the QoS guarantees may strongly depend on the Contention Resolution Algorithm used in the uplink contention period. The time it takes a station to transmit a successful request to the base station, or request delay, must be kept low even during periods of high contention. If a request suffers many collisions, it cannot rely on the preemptive scheduler to receive low access delays. However, the conventional collision resolution algorithm has a problem that all collided stations are treated equally regardless of their delay from previous contention periods. Some requests may have very long request delay caused by continuous collisions. In this paper, we propose an adaptive collision resolution algorithm for fast random access in broadband wireless networks. The design goal is to provide quick access to the request with a high number of collisions. To do this, the proposed algorithm separates the whole contention region into multiple sub regions and permits access through each sub region only to the requests with equal number of collisions. The sub region is adaptively created according to the feedback information of previous random access. By simulation, the proposed algorithm can improve the performance in terms of throughput, random delay and complementary distribution of random delay by its ability to isolate higher priorities from lower ones. We can notice the algorithm provides efficiency and random access delay in random access environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7C
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• pp.691-699
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• 2006

Despite of the in-depth effort to cantrol the variability in IP networks, quality of service (QoS) is still not guaranteed in the IP networks. Thus, it is necessary to deal with the audible artifacts caused by packet lasses. To overcame the packet loss problem, most speech coding standard have their own embedded packet loss concealment (PLC) algorithms which adapt extrapolation methods utilizing the dependency on adjacent frames. Since many low bit rate CELP coders use predictive schemes for increasing coding efficiency, however, error propagation occurs even if single packet is lost. In this paper, we propose an efficient PLC algorithm with consideration about the speech characteristics of lost frames. To design an efficient PLC algorithm, we perform several experiments on investigating the error propagation effect of lost frames of a predictive coder. And then, we summarize the impact of packet loss to the speech characteristics and analyze the importance of the encoded parameters depending on each speech classes. From the result of the experiments, we propose a new PLC algorithm that mainly focuses on reducing the error propagation time. Experimental results show that the performance is much higher than conventional extrapolation methods over various frame erasure rate (FER) conditions. Especially the difference is remarkable in high FER condition.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.9 no.6
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• pp.136-150
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• 2010

Due to the effect of indoor coverage problem, the QoS of the indoor users will be degraded dramatically, with the number of indoor users. The femto cell is a popular solution for such problems. Since the price of the femto base station is usually cheap enough, one can sets up huge number of base stations in a small indoor area to reduce the size of communication cell. In this way, the QoS of the indoor users can be improved significantly. Moreover, the data rate can also be increased. However, how to decide an ideal transmitting power according to the surrounding radio environment is not a trivial problem, that still has not been addressed well. If the transmit power of femto base station is too large, the interference to the macro users will be increased. Conversely, if the transmit power of femto base station is too small; the coverage of femto base station will be reduced. To address this problem, we propose a power configuration method in femto base station using Genetic Algorithm by investigating a new fitness function. Furthermore, we adopt the weighted sum approach to improve the user performance in different modes. The simulation results show that the proposed power configuration method can not only improves the downlink SINR, but also enhance the channel capacity for both the Macro cell systems and Femto cell systems compared with some conventional methods.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.5
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• pp.19-24
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• 2016

In Long Term Evolution-Advanced (LTE-A), small cell enhancement(SCE) has been developed as a cost-effective way of supporting exponentially increasing demand of wireless data services and satisfying the user quality of service(QoS). However, due to the dense and irregular distribution of a large number of small cells, the offloading scheme should be applied in the small cell network. In this paper, we propose an user offloading scheme for SCE in LTE-Advanced system. We divide the small cells into different clusters according to the reference signal received power(RSRP) from user equipment(UE). Within a cluster, We apply the user offloading scheme with the consideration of the number of users and interference conditions. Simulation results show that proposed scheme can improve the throughput, and spectral efficiency of small cell users. Eventually, proposed scheme can improve overall cell performance.

• Journal of the Korean Institute of Intelligent Systems
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• v.20 no.1
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• pp.7-14
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• 2010

In this paper, we introduce a hierarchical internet application traffic classification system based on SVM as an alternative overcoming the uppermost limit of the conventional methodology which is using the port number or payload information. After selecting an optimal attribute subset of the bidirectional traffic flow data collected from the campus, the proposed system classifies the internet application traffic hierarchically. The system is composed of three layers: the first layer quickly determines P2P traffic and non-P2P traffic using a SVM, the second layer classifies P2P traffics into file-sharing, messenger, and TV, based on three SVDDs. The third layer makes specific classification of the entire 16 application traffics. By classifying the internet application traffic finely or coarsely, the proposed system can guarantee an efficient system resource management, a stable network environment, a seamless bandwidth, and an appropriate QoS. Also, even a new application traffic is added, it is possible to have a system incremental updating and scalability by training only a new SVDD without retraining the whole system. We validate the performance of our approach with computer experiments.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.6
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• pp.53-67
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• 2002

The multimedia synchronization is one of the key issues to be resolved in order to provide a good quality of multimedia related services, such as Video on Demands(VoD), Lecture on Demands(LoD), and tele-conferences. In this paper, we introduce an adaptive multimedia synchronization algorithm using the level of buffers and load of servers, which are modeled and analyzed by ExSpect, a Petri net based simulation tool. In the proposed algorithm, the audio and video buffers are divided to 5 different levels, and the pre-defined play-out speed controller tries to make the buffer level to be normal in different temporal relations between multimedia streams using buffer levels and server loads. Because each multimedia packet is played by the pre-defined play-out speed, the media data can be reproduced within the permissible limit of errors while preserving the level of buffers to be normal. The proposed algorithm is able to handle and support various communication restrictions between providers and users, and offers little jitter play-out to many users in networks with the limited transmission capability. The performance of the developed algorithm is analyzed in various network conditions using a Petri net simulation tool.