• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4B
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• pp.255-269
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• 2006

The differentiated services(DiffServ) architecture provides packet level service differentiation through the simple and predefined Per-Hop Behaviors(PHBs). The Assured Forwarding(AF) PHB proposed as the assured services uses the RED-in/out(RIO) approach to ensusre the expected capacity specified by the service profile. However, the AF PHB fails to give good QoS and fairness to the TCP flows. This is because OUT(out- of-profile) packet droppings at the RIO buffer are unfair and sporadic during only network congestion while the TCP's congestion control algorithm works with a different round trip time(RTT). In this paper, we propose an Adaptive Regulating Drop(ARD) marker, as a novel dropping strategy at the ingressive edge router, to improve TCP fairness in assured services without a decrease in the link utilization. To drop packets pertinently, the ARD marker adaptively changes a Temporary Permitted Rate(TPR) for aggregate TCP flows. To reduce the excessive use of greedy TCP flows by notifying droppings of their IN packets constantly to them without a decrease in the link utilization, according to the TPR, the ARD marker performs random early fair remarking and dropping of their excessive IN packets at the aggregate flow level. Thus, the throughput of a TCP flow no more depends on only the sporadic and unfair OUT packet droppings at the RIO buffer in the core router. Then, the ARD marker regulates the packet transmission rate of each TCP flow to the contract rate by increasing TCP fairness, without a decrease in the link utilization.

• Journal of Communications and Networks
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• v.10 no.3
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• pp.338-350
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• 2008

The differentiated services (DiffServ) is proposed to provide packet level service differentiations in a scalable manner. To provide an end-to-end service differentiation to users having a connection over multiple domains, as well as a flow marker, an intermediate marker is necessary at the edge routers, and it should not be operated at a flow level due to a scalability problem. Due to this operation requirement, the intermediate marker has a fairness problem among the transmission control protocol (TCP) flows since TCP flows have intrinsically unfair throughputs due to the TCP's congestion control algorithm. Moreover, it is very difficult to resolve this problem without individual flow state information such as round trip time (RTT) and sending rate of each flow. In this paper, to resolve this TCP fairness problem of an intermediate marker, we propose a fair scalable marker (FSM) as an intermediate marker which works with a source flow three color marker (sf-TCM) operating as a host source marker. The proposed fair scalable marker improves the fairness among the TCP flows with different RTTs without per-flow management. Through the simulations, we show that the FSM can improve TCP fairness as well as link utilization in multiple domain DiffServ networks.

• Journal of KIISE:Computing Practices and Letters
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• v.16 no.2
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• pp.227-231
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• 2010

The TFRC is a congestion control mechanism which supports the requirements of video streaming applications and controls its sending rate by using the information such as loss event rate and RTT (round-trip time). However, TFRC has a performance degradation in wireless networks because it performs congestion control by judging all the losses occurred in wireless network as a congestion signal. In this paper, we propose new loss discrimination mechanism which is using ECN in order to solve the performance degradation of TFRC. Through the results of simulation, we proved that the proposed mechanism can improve the performance of TFRC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10B
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• pp.638-643
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• 2007

Interconnecting wireless local area networks (WLANs) with third generation (3G) cellular networks has become an issue of great interest. However, a Vertical Handover (VHO) causes an abrupt change in link bandwidth. Due to such a change, TCP triggers unnecessary fast retransmission during a Downward VHO (DVHO) from a cellular network to a WLAN, causing throughput degradation. Thus, we propose a new reordering mechanism for DVHO that suppresses unnecessary retransmission due to the spurious duplicate acknowledgments. We analytically investigate the throughput of TCP in the literature and our proposed scheme. Through the numerical and simulation results, it is shown that our proposed TCP achieves better performance in terms of throughput, compared with Nodupack with SACK.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1683-1690
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• 2005

This study examines dynamic flow control method in UDP, analyzes packet loss which is frequently used element in measuring existing dynamic flow control and characteristics of round trip time, and proposes a new method of measurement, RSST. The algorithm that uses the proposed RSST enables accurate measurement of network status by considering both the currently measured network status and the past history of network status in controlling the transmission rate. For comparison study, a network status measurement software program that displays detailed information about volume of transmission generation of network status, and flow pattern of network was developed. The performance test shows that the proposed algorithm can better adjust to network condition in terms of low pack loss rate over existing algorithms.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1345-1348
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• 2002

기존의 TCP/IP 네트워크 상에서 혼잡을 제어하는 일반적인 방법은 end-to-end 피드백 방식이다. 혼잡이 발생했는지의 여부는 수신측의 ACK 신호로 예측할 수밖에 없으므로 RTT(round trip time)가 길어질 경우 혼잡처리 시간이 길어지게 된다. Active Network 기술을 이용한 ACC(Active Congestion Control)의 경우 혼잡이 발생한 라우터에서 전송자에게 직접적으로 정보를 전송하기 때문에 보다 빠른 혼잡처리가 가능하다. 비록 코어 라우터에서 패킷을 누락시키는 방법에 있어 상태정보를 필요로 하는 단점이 있지만 모의 실험 결과 향상된 처리량을 보여주었다. 혼잡처리에 있어 새로운 처리 방식은 AQM(Active Queue Management)이다. 이 방식은 RED 를 기본 모델로 변형된 것이 대부분이며 RED 가 갖는 단점을 개선하고 혼잡처리와 공평성(fairness)을 향상 시키는데 목적이 있다. 본 논문에서는 ACC 의 피드백 메커니즘과 AQM 의 융통성 있는 큐 관리 기법을 이용하여 보다 향상된 혼잡제어 기법을 보이겠다.

• Journal of Korea Society of Industrial Information Systems
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• v.15 no.2
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• pp.23-30
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• 2010

This paper presents implementation details of car communications system based on V2I2V model over broadband wireless networks including WCDMA, HSDPA and WiBro. Also, we measures the major performance metrics such as Round Trip Time, end-to-end latency and others between in-vehicle client and location management system by driving across urban areas in Daegu city. Analysis of experimental results would be helpful to consider the applicability of traffic safety services using V2I2V-based car communications over various broadband wireless networks.

• 대한공업교육학회지
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• v.33 no.2
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• pp.287-299
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• 2008

HTTP is one of the most widely used protocols of the WWW. Currently it uses TCP as the transport layer protocol to provide reliability. The HTTP uses separate TCP connection for each file request and adds unnecessary head-of-line blocking overhead for the file retrieval. The web application is short sized and affected by the increased handover latency of TCP in wireless environment. SCTP has attractive features such as multi-streaming and multi-homing. SCTP's multi-streaming and multi-homing avoid head-of-line blocking problem of TCP and reduce handover latency of TCP in wired and wireless environment. Mean response time is the important measure in most web application. In this paper, we present the comparison of mean response time between HTTP over SCTP with that of HTTP over TCP in wired and wireless environments using NS-2 simulator. We measured mean response time for varying packet loss rate, bandwidth, RTT, and the number of web objects in wired environment and mean response time and packet loss rate for varying moving speed and region size in wireless environment. Our experimental result shows that SCTP reduces the mean response time of TCP based web traffic.

• Journal of the Korea Society of Computer and Information
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• v.4 no.4
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• pp.148-154
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• 1999

TCP has improved by many papers which suggest the new algorithms and modify the previous algorithms. This paper compares Tahoe, Reno, New-Reno, Vegas, and SACK. The first version is Tahoe and is globally used. Reno has optimal performance during occurring one packet loss within a window of data, but can suffer from performance when multiple packets are dropped from a window of data. New-Reno avoids some of the performance problems of Reno TCP when multiple packets are dropped from a window of data. but is occurring the problem of the necessary retransmission. SACK resolves the all above problems and is used in bandwidth delay product environment. Vegas uses network bandwidth more efficiently and is a new implementation of TCP that achieves between 40 and 70 better throughput, with one-fifth to one-half the losses, as compared to the implementation of Reno TCP.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.1
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• pp.58-65
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• 2016

Satellite communication is a wide area network (WAN) which provides communication service worldwide. However, the performance of TCP can be seriously degraded in the satellite networks due to limited bandwidth, long round-trip time (RTT) and high bit error rate (BER) over satellite links. In order to improve the performance of TCP, this paper proposes cross-layer Performance Enhancing Proxy (PEP) in digital video broadcasting-return channel via satellite (DVB-RCS) networks. The proposed protocol sets TCP Congestion Window (CWND) size by using satellite resource allocation information exchanged between TCP and the link-layer. we implement PEP testbed based on Linux to evaluate the performance of the proposed protocol. The simulation results show that the proposed protocol performs better than standard TCP both in single and multiple sessions in variant BER, because the proposed protocol sets TCP CWND size by using satellite resource allocation.