• Journal of KIISE:Information Networking
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• v.34 no.6
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• pp.466-474
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• 2007

Asymmetric networks that the downlink bandwidth is larger than the uplink bandwidth may cause the degradation of the TCP performance due to the uplink congestion. In order to solve this problem, this paper designs and implements the Dynamic Segment Size Control mechanism which offers a suitable segment size for current networks. The proposed mechanism does not require any changes in customer premises but suppress the number of ACKs using segment reassembly technique to avoid the uplink congestion. The gateway which adapted the Dynamic Segment Size Control mechanism, detects the uplink congestion condition and dynamically measures the bandwidth asymmetric ratio and the packet loss ratio. The gateway reassembles some of segments received from the server into a large segment and transmits it to the client. This reduces the number of corresponding ACKs. In this mechanism, the SACK option is used when occurs the bit error during the transmission. Based on the simulation in the GEO satellite network environment, we analyzed the performance of the Dynamic Segment Size Control mechanism.

• Proceedings of the Korea Society for Industrial Systems Conference
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• 1999.12a
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• pp.121-129
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• 1999

When continuous media are transmitted over the communication networks, asynchrony which can not maintain temporal relationships among packets may occur due to a random transit delay. There exist two types of synchronization schemes ; for guaranteed or non-guaranteed resource networks. The former which applies a resource reservation technique maintains delay characteristics, however, the latter supply a best-effort service. In this paper, I propose a intra-media synchronization scheme to transmit continuous media on general networks not guaranteeing a bounded delay tome. The scheme controls transmission times of the packets by estimating next delay time with the delay distribution. So, the arriving packets may be maintained within a limited delay boundary, and playout will be performed after buffering to smoothen small delay variations. The continually increasing delay due to network overload causes buffer underflow at the receiver. To solve it, the transmitter is required to speed up instantaneously. Too much increase of transmission-rate may cause network congestion. At that time, the transmitter drops the current packet when informed excessive delay from the receiver.

• The Journal of the Korea Contents Association
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• v.8 no.2
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• pp.67-73
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• 2008

Recently, due to the rapid advances of mobile communication, multimedia service can be provided by mobile devices. Cellular network tends to uses SIP as a call setup protocol in order to provide various multimedia service to consumers. Cellular network holds the characteristic of higher BER(Bit Error Rate), narrower bandwidth than the wired network. The value of SIP RTI(Retransmission Time interval) based on the wired network decreases a network efficiency and increases a call setup delay over cellular network. By using NS-2 simulator, we show new SIP RTI value adequate over cellular network. We design and implement home network by using the modified SIP that is suitable for cellular network.

• Journal of KIISE:Computing Practices and Letters
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• v.16 no.11
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• pp.1066-1070
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• 2010

In wireless network, energy efficiency is an important design consideration for continuous multimedia streaming service. This paper proposes a new streaming system, called BACASS (Buffer-Aware and Content-Aware Streaming System) that working on the 802.11e U-APSD (Unscheduled Automatic Power Save). The BACASS leads the DP (Doze Period) of U-APSD for improving energy efficiency by utilizing the PSNR based on content-aware and client buffer occupancy that is hinged on a network-aware streaming system using SVC. The simulation results demonstrate the effectiveness of the proposed streaming system.

• The Transactions of the Korea Information Processing Society
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• v.4 no.10
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• pp.2605-2614
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• 1997

Recently, we are concerned with effective service according as the demand increase for high speed data service. We can use high speed transfer and multiple traffic service on the ATM networks, so we concentrate on preventive-control method rather than reactive-control one. But it is possible to have low QoS and traffic congestion due to unpredictable traffic and burst traffic. Specially, ATM Forum has discussed to standardization of traffic management of ABR(Available Bit Rate) service. Because ABR traffic controls the flow of traffic using the feedback information and the current status information of cell, it allocates bandwidth systematically and dynamically to the user. In this paper, we propose a new Rate-based flow control scheme which adapted double threshold buffer idea. The double threshold buffer controls the traffic control by establishing two threshold in buffer.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1B
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• pp.154-161
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• 2010

In wireless sensor networks, due to the many-to-one convergence of upstream traffic, congestion more probably appears. The existing congestion control protocols avoid congestion by controlling incoming traffic, but the duty-cycle operation of MAC(Medium Access Control) layer has not considered. In this paper, we propose DCA(Duty-cycle Based Congestion Avoidance), an energy efficient congestion control scheme using duty-cycle adjustment for wireless sensor networks. The DCA scheme uses both a resource control approach by increasing the packet reception rate of the receiving node and a traffic control approach by decreasing the packet transmission rate of the sending node for the congestion avoidance. Our results show that the DCA operates energy efficiently and achieves reliability by its congestion control scheme in duty-cycled wireless sensor networks.

• The Journal of Korean Association of Computer Education
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• v.9 no.1
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• pp.107-118
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• 2006

Load redistribution algorithm is a critical factor in computer system. In a receiver-initiated load redistribution algorithm, receiver(underloaded processor) continues to send unnecessary request messages for load transfer until a sender(overloaded processor) is found while the system load is light. Therefore, it yields many problems such as low cpu utilization and system throughput because of inefficient inter-processor communications until the receiver receives an accept message from the sender in this environment. This paper presents an approach based on genetic algorithm(GA) for dynamic load redistribution including self-adjustable in heterogeneous distributed systems. In this scheme the processors to which the requests are sent off are determined by the proposed GA to decrease unnecessary request messages.

• Journal of the Institute of Convergence Signal Processing
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• v.6 no.4
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• pp.176-181
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• 2005

This paper presents an equal bit rate control algorithm utilizing the statistical change between the previous frame and the current frame. The previous studies on the model-based rate control have focused on the models of bit rate and distortion in types of coders, in terms of the quantization parameter. The proposed algorithm improves the typical model-based rate control by updating a model parameter instead of modeling a better model of the rate and distortion. The proposed algorithm updates this model parameter by recognizing the change in statistics between the previous frame and the current frame. We implement the proposed algorithm in MPEG-4 coders and verify its performance while comparing it to the TMN8's approach (up to 0.6dB of improvement).

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.672-686
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• 2002

현재 단대단(End-to-End) 비디오 서비스의 질을 높이기 위해 많은 연구가 각 계층별로 진행되고 있다. 전송계층에서의 오류 제어(Error Control), 네트워크계층에서의 QoS (Quality of Service)모델, 표현 및 응용계층에서의 오류 강인성(Error resilience)/오류 은닉(Error concealment) 등이 연구 개발되고 있다. 그러나 계층 간의 연관성이 높은 부분에서의 통합을 통한 성능향상에 관한 연구는 그 필요성과 효율에 비해 아직도 미흡하다. 본 논문은 QoS 서비스 모델하에서의 적응적 FEC(Forward Error Correction) 적용 및 우선순위에 따른 비디오패킷(VP ,VideoPacket)을 통하여 효율적인 계층화 비디오 스트리밍을 단대단 QoS성능의 향상에 관점을 맞추어 제안한다. 제안하는 방식은 최소 화질 보장과 같은 효율에서 보다 적은 가격에서의 서비스를 제공할 수 있다. 이를 위하여 통합형 서비스(IntServ, IS, Integrated Service) 의 자원예약을 사용하는 방법과 높은 가격의 자원 예약을 사용하지 않는 차별화 서비스(DiffServ, DS, Differentiated Service)를 적용했으며 이에 보장형 서비스의 특징을 공통을 가지기 위해 계층화 FEC를 적용하였으며 적절한 가격의 조절을 위하여 비디오패킷을 통한 데이터 분할을 적용하였다. 본 논문은 또한 최종 사용자의 만족도를 PSNR(Picture Signal to Noise Ration)과 PSNR에서 표현하지 못하는 부분의 평가를 위해 손상프레임율(DFR, Damaged Frame Ratio)과 오류프레임율(EFR,Error Frame Ratio)을 제안 이를 통해 평가하고자 한다. 제안하는 방식의 실험 결과는 비디오 코딩계층과 전송 계층, 네트워크 계층의 결합된 성능이며 이는 또한 화질의 개선뿐만 아니라 사용자의 가격문제에 대하여서도 비교 분석하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.12
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• pp.2315-2320
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• 2006

A real-time frame-layer rate control algorithm with a token bucket traffic shaper is proposed for low bit rate video coding. The proposed rate control method uses a non-iterative optimization method for low computational complexity, and performs bit allocation at the frame level to minimize the average distortion over an entire sequence as well as variations in distortion between frames. In order to reduce the quality fluctuation, we use a sliding window scheme which does not require the pre-analysis process. Therefore, the proposed algorithm does not produce time delay from encoding, and is suitable for real-time low-complexity video encoder. Experimental results indicate that the proposed control method provides better visual and PSNR performances than the existing rate control method.