• Journal of KIISE:Information Networking
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• v.34 no.1
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• pp.27-32
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• 2007

Next-generation wireless networks are expected to support a wide range of services, including high-rate data applications, Various service types request differentiated QoSs(Qualities of Service) such as minimum data rate, accuracy, fairness and so on. Although resources of radio systems are limited, for many applications, it is important that certain QoS targets are required to be met. In this paper, we propose a QoS based scheduling algorithm for next generation systems, based on analyzing previous researches, and we develop the proposed QoS algorithm only for MIMO(multi-Input Multi-Output) systems. Moreover, we subsequently prove that the proposed algorithm optimize throughput relative to prespecified target values and converge to certain throughput.

• Journal of Institute of Control, Robotics and Systems
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• v.6 no.4
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• pp.313-319
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• 2000

As the interest of flexible manufacturing systems and computer integrated manufacturing systems increase, the distribution of centralized control systems using industrial control networks is getting more attention. In this paper, we investigate the rate-based traffic control of industrial control networks to improve the performance regarding the throughput, fairness, and error rates. Especially, we consider the protocol of Lon-$Works^{(TM)}$ which consists of all OSI 7-layers and supports various communication media at a low cost. Basically, the proposed rate-based traffic control system is closed loop by utilizing the feedback channel errors, which shows improved performance when compared with other industrial control networks commonly operated in open loop. To this end, an additional network node called monitoring node is introduced to check the channel status without increasing the channel load. The Proposed control loop is in effect whenever the feedback channel error becomes greater than an admittable value. We demonstrate the improved performance of the controlled network system in view of throughput and fairness measures by implementing the lab-scale network system based on LonWorks and through the experimentation upon it.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.9A
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• pp.2173-2183
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• 1998

We consider a frequency-hopped multiple-access communication system that employs reed-solomon code over GF(Q) and M-ary FSK signaling ($M{\leq}Q$) in rayleigh fading channel. We investigate the tradeoff among the modulation symbol size (M), the number of frequency slots, and the code rate in maximizing the average number of successfully transmitted information bits per unit time and unit bandwidth (called normalized throughput). We find that it is desirabel to use a large M in noise-limited environment. In interference-limited environment, it is more improtant to prevent errors (hits) by increasing the number of frequency slots than to correct them with formward error correction techniques or to reduce the error rate by increasing M.

• ETRI Journal
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• v.26 no.5
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• pp.375-383
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• 2004

This paper proposes a common interleaving method associated with independent channel-encoding among transmitter antenna branches in orthogonal frequency and code division multiplexing based on multiple-input multiple-output (MIMO) multiplexing to achieve an extremely high throughput such as 1 Gbps using a 100 MHz bandwidth. This paper also investigates the average packet error rate performance as a function of the average received signal energy per bit-to-background noise power spectrum density ratio $(E_b/N_0)$. We found that the loss in the required average received $E_b/N_0$ of the proposed method is only within approximately 0.3 dB in up to a 12-path Rayleigh fading channel, using 16QAM and Turbo coding with a coding rate of 5/6. We also clarify that even for a large fading correlation among antenna branches, 1 Gbps is still possible by increasing the transmission power. Therefore, the proposed method reduces the processing rate to 1/4 in the turbo decoder with only a slight loss in the required average received $E_b/N_0$.

• Journal of the Korea Society for Simulation
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• v.18 no.3
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• pp.91-102
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• 2009

The average window size is most closely related to average throughput. In order to improve fairness, the proposed dropper tries to control the window size of each flow to equal level by intentional packet drop. Intentional packet drop is performed only to the flows that have been occupied bandwidth in a large amount. Because of intentional packet drop, this flow cut down its transmission rate to a half. Accordingly, somewhat capacity of core link comes into existence. And other flow can use this new capacity of this link. Hence other flows have more throughput than before. In this paper, we propose the TRC (Transmission Rate Control) Dropper improving the fairness between individual flows of aggregated sources on DiffServ network. It has the fairness improvement mechanism mentioned above paragraph.

• Proceedings of the IEEK Conference
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• 2002.06a
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• pp.85-88
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• 2002

Maintaining a proper level of cell lead, system throughput can be maximized by a transmission rate control over the uplink in DS/CDMA cellular system to support integrated services of real-time and delay-tolerant traffic. We find that the cell load-based rate control scheme can be further enhanced by taking the varying channel condition into account In conjunction with some fair scheduling algorithm. Our simulation results show that the proposed scheme outperforms the original cell load-based rate control with the round-robin sharing scheduling scheme.

• Journal of Satellite, Information and Communications
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• v.12 no.3
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• pp.8-14
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• 2017

In 4G mobile communication, data services of mobile communication began to be provided in earnest. As a result, services such as the internet and multimedia including the video have become the main contents in the recent wireless traffic. Accordingly, research on 5G mobile communication with higher transmission rate has been actively carried out. 5G mobile communication is expected to be based on 4G for compatibility with existing terminals. Therefore, the simulation of this paper is based on 4G and we propose a new user scheduling scheme based on the Max throughput scheduling algorithm to improve system performance. This paper derives the frequency efficiency and fairness to compare the existing user scheduling algorithm with the proposed user scheduling algorithm. The proposed scheme shows better frequency efficiency and fairness than Max throughput in all situations. This paper contributes to the research for improving the system performance of 5G mobile communication technology, and I hope that it will help some of the ongoing standardization work.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.3
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• pp.303-308
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• 2016

Recently, as the rapid development of network technology and the increase of services required high bandwidth such as multimedia service, the network traffic dramatically increases. This massive increase of network traffic causes some problems such as the degradation of QoS and the lack of network resources and, to solve these problems, various research to guarantee QoS have been performing. Currently, The most representative method to guarantee the QoS is the DiffServ(: Differentiated Service). The DiffServ defines the AF(: Assured Forwarding) PHB(: Per Hop Behavior) and statistically ensures the throughput over the certain level of data rate. However, the TCP congestion control method that make up the majority of the Internet traffic is not fundamentally suitable to the DiffServ that guarantees the throughput without managing the individual flow. Therefore, in this paper, we present this mismatch through the simulation as an example and propose the solution by controlling the TCP of the terminal in the network. The proposed scheme utilizes the information of the reception window size included in the ACK frame and does not require any modification of the TCP algorithms currently in use.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12B
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• pp.1665-1676
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• 2001

The effect of spreading gain control on the throughput of a CDMA slotted ALOHA system is considered. Mobile users transmit packets over a shared channel, and the packets transmitted in the same time slot over the shared channel act as simultaneous access interference (SAI). When using spread-spectrum signal, a CDMA slotted ALOHA channel achieves high probability of capture due to the property of high title resolution, and the bit rate of user information is determined by spreading gain. When the SAI level gets larger, the high value of spreading gain enhances the packet throughput by increasing the probability of a successful packet transmission, while it degrades the of the effective throughput by reducing the user information bits carried within a packer. To solve the problem, we investigated the effect of the capture probability and the SAI level on these system throughputs, and evaluated the throughput performance of the system for each spreading gain control scheme. The results showed that the maximum effective throughput could be achieved with an unified method despite the variation of the SAI level by deriving an optimal value of the spreading gain according to 171e system states.

• Journal of Internet Computing and Services
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• v.8 no.1
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• pp.15-23
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• 2007

Wireless LAN (IEEE 802.11) uses traditional TCP for reliable data transmission, But it brings the unintentional packet loss which is not congestion loss caused by handoff, interference, and fading in wireless LAN. In wireless LAN, TCP experiences performance degradation because it consumes that the cause of packet loss is congestion, and it decrease the sending rate by activating congestion control algorithm. This paper analyzes that correlation of throughput and buffer size for wireless buffer tuning. We find MBT (Maximum Buffer Threshold) which does not increase the throughput through the analysis, For calculation of MBT, we experiment the throughput by using high volume music data which is creased by real-time performance of piano. The experiment results is shown that buffer tuing based on MBT shows 20.3%, 21.4%, and 45.4% throughput improvement under 5ms RTT, 10ms RTT, and 20ms RTT, respectively, comparing with the throughput of operation system default buffer size, In addition, we describe that The setting of TCP buffer size by exceeding MBT does not have an effect on the performance of TCP.