• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.6
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• pp.1060-1064
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• 2006

Data collected by sensors in field are transmitted to the base station gathering all of data. Because sensors have to gather data in surroundings and periodically transmit data to the base station, it makes energy consumed much. In this paper, we mose the scheme that is to avoid traffic congestion with achievement of energy efficiency, so collected data is transmitted efficiently. This is to adjust transmission rate differently in case of increasing or decreasing traffic and minimize the energy consumption with setting ideal options up basic CSMA(Carrier Sense Multiple Access) protocol in each sensor. Through the simulation, we find the ideal CSMA options and apply the proposed scheme of traffic control mechanism to them and analyze them, then show energy efficiency and effective traffic control mechanism.

• Journal of Communications and Networks
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• v.16 no.3
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• pp.344-354
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• 2014

Transmission control protocol friendly rate control (TFRC) is designed to mainly provide optimal service for unicast applications, such as multimedia streaming in the best-effort Internet environment. However, high bandwidth networks with large delays present an environment where TFRC may have a problem in utilizing the full bandwidth. TFRC inherits the slow-start mechanism of TCP Reno, but this is a time-consuming process that may require many round-trip-times (RTTs), until an appropriate sending rate is reached. Another disadvantage inherited from TCP Reno is the RTT-unfairness problem, which severely affects the performance of long-RTT flows. In this paper, we suggest enhanced TFRC for high quality video streaming over high bandwidth delay product networks. First, we propose a fast startup scheme that increases the data rate more aggressively than the slow-start, while mitigating the overshooting problem. Second, we propose a bandwidth estimation method to achieve more equitable bandwidth allocations among streaming flows that compete for the same narrow link with different RTTs. Finally, we improve the responsiveness of TFRC in the presence of severe congestion. Simulation results have shown that our proposal can achieve a fast startup and provide fairness with competing flows compared to the original TFRC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.6B
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• pp.895-900
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• 2010

As the diverse telecommunication services have been developed, network designers need to prevent congestion which may be caused by properties of timecorrelation and burstiness, and unpredictable statistical fluctuation of traffic streams. This paper considers the leaky bucket scheme with combined control of arrival and token rates, in which the arrival rate and the token generation interval are controlled according to the queue length. By using the embedded Markov chain and the supplementary variable methods, we obtain the queue length distribution as well as the loss probability and the mean waiting time.

• Proceedings of the Korean Information Science Society Conference
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• 2006.06d
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• pp.307-309
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• 2006

본 논문에서는 MPLS (Multiprotocol Label Switching)망의 네트워크 자원을 효율적으로 사용 하기위한 MPLS 망의 단계별 리라우트 기법을 제안한다. MPLS 망 내에서 특정 노드가 혼잡할 경우 노드의 단계별 등급을 증가 시켜서 리라우트하는 기법이다. 이는 LSP (Label Switched Path)를 통해 입구노드에서 출구노드까지의 최소 비용 경로가 정해진 상태에서 경로상의 어느 특정 노드가 트래픽의 증가로 인해 혼잡이 발생하면 자신이 가지고 있는 일정 등급 이상의 값을 가질 경우 경로 재설정 메시지를 통하여 새로운 대체 복구 경로를 찾아 리라우트한다. 또한 혼잡이 발생한 노드가 혼잡에서 벗어나 등급이 원래의 값으로 돌아올 경우 최초 정해진 경로상의 최소 비용을 갖는 최적 경로로 다시 리라우트 된다. 이는 제안하는 단계별 리라우트 기법이 입 출구 LER에 집중되는 스트레스를 모든 망 내의 자원이 분담함으로써 효율적으로 사용할 수 있는 방안을 보여줄 수 있다.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.281-284
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• 2004

This paper presents the design and implementation of a QoS based enhanced collaboration system in MDO. This is an efficient distributed communication tool between designers. It supports text communication, audio/video communication, file transfer and XML data sending/receiving. Specially, this system supports a dynamic QoS self-adaptation by using the improved direct adjustment algorithm (DAA+). The original direct adjustment algorithm adjusts the transmission rate according to the congestion level of the network, based on the end to end real time transport protocol (RTP), and controls the transmission rate by using the information of loss ratio in real time transport control protocol (RTCP). But the direct adjustment algorithm does not consider when the RTCP packets are lost. We suggest an improved direct adjustment algorithm to solve this problem. We apply our improved direct adjustment algorithm to our of QoS (Quality of Service) [1] based collaboration system and show the improved performance of transmission rate and loss ratio.

• Proceedings of the KAIS Fall Conference
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• 2007.05a
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• pp.143-146
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• 2007

컴퓨터 네트워크에서 전송제어 프로토콜 상에서 다중시간 간격으로 혼잡제어의 지연을 피드백 루프의 시간간격을 조정하여 명시적 예측을 실행해 제어의 단점을 개선하는데 연구한다. 먼저 TCP의 모둘 형 확장에서 Tahoe, Reno, Vegas등 다양한 버전의 TCP에 적용되는 간단한 인터페이스를 통한 함수 호출을 정의하고, 이것이 성능을 크게 향상시는 것을 입증한다. 두 번째로, 광대역 WAN에서 지연-대역폭의 곱이 높을 경우 더욱 심각해지는 사후 제어의 불확실성 차이를 해소함으로써 MTS TCP가 기반 피드백 제어에 사전성을 부여한다는 것을 입증한다. 세번째는 트래픽 제어의 3가지 차원인, 즉 추적 능력, 연결 지속 기간, 공정성이 성능에 미치는 영향 등을 비교 분석하여 입증한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2B
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• pp.263-271
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• 2004

In this paper, we proposed a new buffer management algorithm using perVC-Queuing discipline. Proposed algorithm uses service rate estimation and assigns dynamic perVC threshold to each VCs. Service rate estimation and dynamic perVC threshold combined with global threshold provide nearly optimal throughput and improve fairness performance of network resource among GFR VCs as well as guarantee MCR of all VCs. From simulation results, we demonstrate the proposed scheme fulfills the requirement of GFR service as well as improves the TCP throughput.

• Journal of Navigation and Port Research
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• v.28 no.4
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• pp.299-303
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• 2004

In the Internet, to guarantee transmission rate and delay and to achieve fair bandwidth allocation, many per-flow scheduling algorithms, such as fair queueing, which have many desirable properties for congestion control, have been proposed and designed. However, algorithms based on per-flow need maintain rate state, buffer management and packet scheduling, so that it cost great deal ： implement. Therefore, in this paper, to implement routers cost-effectively, we propose CS-FNE algorithm based on FNE in Core-Stateless network We evaluate CS-FNE comparing with four additional algorithms i.e., CSFQ, FRED, RED and DRR, in several different, configurations and traffic sources. Through simulation results, we show that CS-FNE algorithm can allocate fair bandwidth approximately and is simpler and easier to implement than other per-flow basis queueing mechanisms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.1
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• pp.56-65
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• 1994

As a method to control the congestion, priority mechanism which gives two-level priority to various traffics using CLP bit in cell haeder is carried in ATM network. In this paper, we proposed PBS(partial buffer sharing) mechanism with two threshold for provide three-level priority using CLP bit and Res. bit. By performance analysis proposed mechanism, we knew that provides flexibility to satisfy requirements of various quality of service, compare to existing mechanism with two-level priority.

• Proceedings of the Korean Information Science Society Conference
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• 2006.10d
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• pp.245-248
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• 2006

최근 UDP를 이용한 실시간 어플리케이션이 증가하고 있지만, 대부분 혼잡제어를 사용하지 않고 있다. 이러한 문제를 해결하기 위하여 IETF에서는 UDP를 대신하면서 실시간 트래픽 전송을 위하여 혼잡제어 메커니즘을 포함한 DCCP(Datagram Congestion Control Protocol)를 제안하였다. DCCP는 윈도우 기반 흐름제어를 사용하여 실시간 트래픽을 지원하기 위하여 제안된 차세대 전송 프로토콜이다. 본 논문에서는 DCCP의 기본 개념과 혼잡제어 기법을 소개하고, 유선과 무선에서 다양한 전송 프로토콜을 이용하여 DCCP의 성능평가를 한다. DCCP를 기반으로 유선과 무선 네트워크 환경에서 시뮬레이션 성능분석을 위해 Network Simulator 2(NS-2)를 이용한다.