• Proceedings of the Korea Contents Association Conference
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• 2007.11a
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• pp.60-64
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• 2007

Increasing using multimedia services as VoIP, Video conference, DMB, IPTV, etc, it is necessary to increase network traffics and develop the mechanism about a flow control for real time traffic transmission. In order to transfer realtime multimedia data, the transfer rate can be control on network state data measuring packet losses of a receiver and delay time of packets through getting periodical feedback RTP/RTCP packet. This paper describes using efficiant flow control on multicast that can reduce errors according to getting feedback tranfer delay and proposes the mechanism that can adapt dynamic change of network. In simulation, the transfer rate can efficiently be control on dynamic change of network and it makes the maximum of the use of a bandwidth and the minimum of packet losses.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.12
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• pp.39-44
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• 2010

In LTE-Advanced systems, every sector uses the entire range of the frequency resource, and when the UEs are located at a cell edge, user throughputs degrade due to the interferences from the adjacent cells. In this paper, we propose a combined scheme for inter-cell interference avoidance and power control. In the proposed algorithm, the sectors consist of the right edge, the left edge and the center for resource allocation and we control the transmission power to improve the user throughputs at the edge of each cell. Using a system level simulation, we analyze low 5th percentile and average user throughputs of the UEs who are located the cell, center and edge when the inter-cell interference avoidance and the power control are combined.

• Proceedings of the Korean Information Science Society Conference
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• 2006.10d
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• pp.765-768
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• 2006

센서 네트워크에서 데이터 병합 시 타임아웃 스케줄링은 각 센서노드가 자식노드로부터 메시지를 받기 위해 기다려야하는 시간을 조절하는 것이다. 본 논문에서는 데이터 패턴의 변화에 따라 타임아웃을 적응적으로 조절하는 새로운 타임아웃 스케줄링 기법인 ATS-DA(Adaptive Timeout Scheduling for Data Aggregation)를 제안한다. ATS-DA는 자식노드로부터 수신한 데이터가 특별한 이벤트 없이 기준 데이터 변동률 범위 내에 있게 되면 타임아웃 값을 줄인다. 타임아웃을 작게 하면 파워 소모가 감소하고 전송 지연도 짧아진다. 센싱 데이터가 기준 데이터 변동률을 벗어나면 보다 정확한 데이터 병합 값을 얻기 위해 타임아웃 값을 늘린다. 시뮬레이션 결과에서 제안하는 ATS-DA 기법이 기존 캐스케이딩 타임아웃 스케줄링 기법에 비해 데이터 정확도와 파워 소모 측면에서 우수한 성능을 보여주었다.

• The KIPS Transactions:PartC
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• v.9C no.5
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• pp.679-686
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• 2002

The transmission of prerecorded and compressed video data without degradation of picture quality requires video servers to cope with large fluctuations in bandwidth requirement. Bandwidth smoothing techniques can reduce the burst of a variable-bit rate stream by prefetching data at a series of fixed rates and simplifying the allocation of resources in the video servers and the network. In this paper, the proposed smoothing algorithm results in the optimal transmission plans for (1) the smallest bandwidth requirements, (2) the minimum number of changes in transmission rate, and (3) the minimum amount of the server process overhead. The advantages of the proposed smoothing algorithm have been verified through the comparison with the existing smoothing algorithms in diverse environments.

• KIPS Transactions on Computer and Communication Systems
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• v.5 no.5
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• pp.103-108
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• 2016

In this paper, we propose a resource allocation scheme that guarantees transmission rate for each mobile stations by mitigating interference between a base station-to-mobile station link and a relay station-to-mobile station link. Specifically, we dynamically adjust the boundary between access zone and relay zone using signal to interference plus noise ratio. Moreover, we cluster the mobile stations under sever interference and manage the channel quality of these mobile stations by allocating additional radio resource. Our simulation results show that the proposed scheme can improve the efficiency of radio resources and ensure fairness among mobile stations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.7
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• pp.1817-1823
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• 1998

V.42bis, a type of LZW(Lempel-Ziv-Welch) code, is well-known as theinter national standard is asynchronous data compression. In this paper, we analyze several undesirable phenomena arising from the application of v.42bis to high speed data transmission, and we propose a modified technique to overcome them. the proposed technique determines the proper size of the dictionary, one of important factors affecting the compression ratio, and improves the method of dictionary generation for a higher compression ratio. Furthermore, we analyze the problem of excessive mode changes and solve it to a certain degree by adjusting the threshold for mode change. By doing this, we can achieve smiller variation of the compression ratio in time. This improvement chtributes to easier and better design and control of the buffer in high speed data transmission.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.11B
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• pp.969-978
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• 2006

Recently, there has been explosive growth in Internet service contents, and we have witnessed the emergence of many real-time applications on the Internet. These types of applications require special support from the underlying network such as reliability, timeless, and guaranteed delivery, as well as different levels of service quality. Estimation network status and transcoding data rate are researching for QoS of multimedia service by using the RTP/RTCP. We propose two schemes that the way to calculate the one-way delay by expanding the structure of RTP/RTCP in Internet that provides us 'best effort' service and to estimate the network condition by using this in this paper. And we propose the dynamic RTCP mechanism based on one-way delay to deal with rapidly changed network situation immediately. In simulation, proposed monitoring schemes using the one-way delay has similar in performance to monitoring schemes using the GPS, proposed dynamic RTCP scheme obtain the desired result of decreased packet loss rate by sending RTCP RR packet lower than other schemes.

• Journal of Internet Computing and Services
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• v.8 no.5
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• pp.125-132
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• 2007

Transmission mechanisms that include an available bandwidth estimation algorithm and a packet loss differentiation scheme, in general, exhibit higher TCP performance in wireless networks. TCP New Jersey, known as the best existing scheme in terms of goodput, improves wireless TCP performance using the available bandwidth estimation at the sender and the congestion warning at intermediate routers. Although TCP New Jersey achieves 17% and 85% improvements in goodput over TCP Westwood and TCP Reno, respectively, we further improve TCP New Jersey by exploring improved available bandwidth estimation, retransmission timeout, and recovery mechanisms. Hence, we propose TCP New Jersey PLUS (shortly TCP NJ+), showing that under 1% packet loss rate, it outperforms 3% by TCP New Jersey and 5% by TCP Wes1wood. In 5% packet loss rate, a characteristic of high bit-error-rate wireless network, it outperforms other TCP variants by 19% to 104% in terms of goodput even when the network is in bi-directional congestion.

• Journal of KIISE:Information Networking
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• v.35 no.5
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• pp.402-408
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• 2008

This paper considers the problem of minimizing network-wide broadcast latency in multi-rate wireless mesh networks where a node can dynamically adjust its link layer transmission rates to its neighbors. We propose a broadcast algorithm that complements existing broadcast construct algorithm which chooses a multicast node randomly when each candidate node has same metric. We consider the currently accumulated broadcast latency from source node to the each candidate node so far to choose the next broadcast node. The proposed broadcast algorithm for minimizing latency in a multi-rate mesh networks which exploit wireless advantage and the multi-rate nature of the network. Simulation based on current 802.11 parameters shows that proposed MinLink_WCDS algorithm improves overall latency than the previous existing broadcast algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.12B
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• pp.1183-1191
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• 2010

In this paper, I propose an adaptive OFDM CP length algorithm for in PLC systems for smart grid. The proposed scheme calculates the channel delay information at the CP controller of the receiver by taking correlation between a received data frame and the following delayed one. The CP controller, immediately, feeds back the channel delay information to the transmitter. Then, the transmitter adapts CP length for next data frame. As an impulsive noise model, Middleton Class A interference model was employed. The performance is evaluated in terms of packet data rate, cumulative packet data rate, and bit error rate (BER). The simulation results showed data gain (which is the amount of the reduced bits) gets larger as the number of packets increase, but the amount of data gain reduced as the number of branches ($N_{br}$) increase. In respects of BER for the cases $N_{br}$ is 3, 4, and 5, performance of the adaptive CP length algorithm and the fixed CP scheme are similar. Therefore, it is confirmed the proposed scheme achieved data rate increment without BER performance reduction compared to the conventional fixed CP length scheme.