• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.235-244
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• 2004

TCP Vegas version provides better performance and more stable services than TCP Tahoe and Reno versions, which are widely used in the current Internet. However, in the situation where TCP Vegas and Reno share the bottleneck link, the performance of TCP Vegas is much smaller than that of TCP Reno. This unfairness is due to the difference of congestion control mechanisms of each TCP use. Several studies have been executed in order to solve this unfairness problem. In this paper, we analyze the minimum window size to maintain the maximum TCP performance of link bandwidth. In addition, we propose an algorithm which maintains the TCP performance and improves fairness by selective packet drops in order to allocate proper window size of each TCP connections. To evaluate the performance of the proposed algorithm, we have measured the number of data bytes transmitted between end-to-end systems by each TCP connections. The simulation results show that the proposed algorithm maintains the maximum TCP performance and improves the fairness.

• Journal of KIISE
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• v.43 no.8
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• pp.919-926
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• 2016

In a wireless network, TCP causes the performance degradation because of mistaking packet loss, which is caused by characteristics of wireless link and throughput oscillation due to change of devices connected on a limited bandwidth. Delay based TCP is not affected by packet loss because it controls window size by using the RTT. Therefore, it can solve the problem of unnecessary degradation of the rate caused by misunderstanding reason of packet loss. In this paper, we propose an algorithm for improving the remaining problems by using delay based TCP. The proposed scheme can change throughput adaptively by adding the RTT, which rapidly reflects the network conditions to BaseRTT. It changes the weight of RTT and the increases and decreases window size based on the remaining amount of the buffer. The simulation indicated that proposed scheme can alleviate the throughput oscillation problem, as compared to the legacy TCP Vegas.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.7
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• pp.543-552
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• 2013

For the safety messages in IEEE 802.11p/WAVE vehicles network environment, strict periodic beacon broadcasting requires status advertisement to assist the driver for safety. In crowded networks where beacon message are broadcasted at a high number of frequencies by many vehicles, which used for beacon sending, will be congested by the wireless medium due to the contention-window based IEEE 802.11p MAC. To resolve the congestion, we consider a MAC scheme based on slotted p-persistent CSMA as a simple non-cooperative Bayesian game which involves payoffs reflecting the attempt probability. Then, we derive Bayesian Nash Equilibrium (BNE) in a closed form. Using the BNE, we propose new congestion control algorithm to improve the performance of the beacon rate under saturation condition in IEEE 802.11p/WAVE vehicular networks. This algorithm explicitly computes packet delivery probability as a function of contention window (CW) size and number of vehicles. The proposed algorithm is validated against numerical simulation results to demonstrate its stability.

• 대한공업교육학회지
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• v.32 no.2
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• pp.199-216
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• 2007

Stream Control Transmission Protocol(SCTP) is a transport layer protocol to support the data transmission. SCTP is similar to Transmission Control Protocol(TCP) in a variety of aspects. However, several features of SCTP including multi-homing and multi-streaming incur the performance difference from TCP. This paper highlights the data transfer during the initial slow start phase in SCTP congestion control composed of slow start phase and congestion avoidance phase. In order to compare the mean transfer time between SCTP and TCP, we experiment with different performance parameters including bandwidth, round trip time, and data length. By varying data length, we also measure the corresponding initial window size, which is one of factors affecting the mean transfer time. For the experiment, we have written server and client applications by C language using SCTP socket API and have measured the transfer time by ethereal program. We transferred data between client and server using round-robin method. Analysis of these experimental results from the testbed implementation shows that larger initial window size of SCTP than that of TCP brings the reduction in the mean transfer time of SCTP compared with TCP by 15 % on average during the initial slow start phase.

• Journal of Internet of Things and Convergence
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• v.1 no.1
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• pp.1-4
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• 2015

This paper aims to present the mathematical model to find the mean web object transfer latency in the slow-start phase of TCP congestion control mechanism, which is one of the main control techniques of Internet. Mean latency is an important service quality measure of end-user in the network. The application area of the proposed latency model is the narrowband environment including multi-hop wireless network and Internet of Things(IoT), where packet loss occurs in the slow-start phase only due to small window. The model finds the latency considering initial window size and the packet loss rate. Our model shows that for a given packet loss rate, round trip time and initial window size mainly affect the mean web object transfer latency. The proposed model can be applied to estimate the mean response time that end user requires in the IoT service applications.

• Proceedings of the Korean Information Science Society Conference
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• 2005.11a
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• pp.340-342
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• 2005

최근 기지국(Base Station)의 도움 없이 이동 단말기(Mobile Device) 간의 다중 무선 홉을 사용하여 송수신자 간의 데이터 전송을 가능하게 하는 Ad-hoc 통신망에 관한 연구가 활발히 진행되고 있다. Ad-hoc 통신망을 구성하는 이동 단말은 이동성을 고려하여 한정된 배터리로 동작을 하게 된다. Ad-hoc 통신망에서 이동 단말은 종단(End Host)으로만 동작하는 것이 아닌 중계 노드(Intermediate Node)로도 동작을 하기 때문에 통신상에 경로를 제공하는 중요한 역할도 한다. 하지만, 현재 인터넷 상에서 광범위하게 사용되고 있는 전송 규약인 TCP(Transmission Control Protocol)는 수동적인(Reactive)혼잡 제어(Congestion Control)방식으로 망의 혼잡으로 인한 패킷 손실 발생 이전까지 전송 윈도우의 크기를 증가 때문에 반복적인 혼잡과 그로 인한 패킷 손실로 인해 불필요한 재전송을 반복하게 된다. 이와 같이 기존 TCP는 무선 통신망에서 동작하는 이동 단말의 한정된 배터리 전원을 고려하지 않고 동작하기 때문에 이동 단말의 에너지를 불필요하게 낭비하는 문제를 가지게 된다. 본 논문은 Ad-hoc 망에서 이동 단말의 에너지 효율을 개선하기 위해 불필요한 재전송 방지 및 망 상태에 따라 전송률을 적절하게 조절하는 TCP의 새로운 혼잡 제어 기법을 제시하였다. 또한 ns-2 시뮬레이터를 이용한 실험을 통해 이동 단말의 에너지 효율이 제안된 혼잡 제어 기법에 의해 향상되었음을 확인하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.7
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• pp.1599-1609
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• 2014

TCP does not ensure the bandwidth and delay bound required for multimedia streaming services in broadband wireless network environments. In this paper, we propose a new congestion control scheme for efficient multimedia transmission, called COLO TCP (Concave Increase Slow Start Logarithmic Increase Congestion Avoidance TCP). The COLO TCP prevents the burst packet loss by applying the concave increase algorithm in slow start phase. In the congestion avoidance phase, COLO TCP uses the logarithmic increase algorithm that quickly recovers congestion window after packet loss. To highly utilize network bandwidth and reduce packet loss ratio, COLO TCP uses additive increase algorithm and adaptive decrease algorithm. Through simulation results, we prove that our COLO TCP is more robust for random loss. It is also possible for efficient multimedia transmission.

• The KIPS Transactions:PartC
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• v.15C no.4
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• pp.281-288
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• 2008

Recently, the advance of multimedia hardware has fostered the development of wireless multimedia sensor network which is able to ubiquitously obtain multimedia content such as image or audio from the environment. The multimedia data which has several characteristics such as large size and correlation between the data requires reliability in transmission. However, the existing solution which take the focus on the efficiency of network mainly, is not appropriate to transmit the multimedia data. In the paper, we proposes a reliable asynchronous image transfer protocol, RAIT. RAIT applies double sliding window method in node-to-node image tansfer to prevent the packet loss caused by network congestion. The double sliding window consists of one sliding window for the receiving queue, which is used for prevention of packet loss caused by communication failure between nodes and the other sliding window for the sending queue which prevents the packet loss caused by network congestion. the routing node prevents the packet loss and guarantees the fairness between the nodes by scheduling the packets based on the image non-preemptively. The RAIT implements the double sliding window method by cross layer design between RAIT layer, routing layer, and queue layer. The experiment shows that RAIT guarantees the reliability of image transmission compared with the existing protocol.

• Proceedings of the Korean Information Science Society Conference
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• 2004.04a
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• pp.526-528
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• 2004

본 논문은 유무선 환경에서 TCP를 이용한 데이터 전송 시 에이젼트를 이용하여 패킷 손실의 원인을 분석, 무선 링크에서 발생한 패킷 손실에 대해서는 혼잡 윈도우 크기를 유지하고, 유선 링크에서 발생한 패킷 손실에 대해서는 지역 재전송을 수행하는 저 전력 전송기법을 제안한다. 제안하는 저 전력 전송기법은 전송 후 WNIC를 저 전력 모드로 전환시킴으로써 WNIC 전력소비를 최소화한다. NS2 시뮬레이션 결과 기존 TCP 보다 무선 링크에서 에러 발생시 67~177(%) 성능향상과 22~44(%) 에너지 감소효과를 보였고, 유선 링크에서 에러 발생시 3~22(%)의 성능 향상과 2~13(%) 에너지 감소 효과를 나타냈다.

• Proceedings of the Korea Information Processing Society Conference
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• 2001.10a
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• pp.259-262
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• 2001

휴대용 컴퓨터와 무선통신 기술 발달로 인해 사용자들은 시간과 장소의 구애를 받지 않고 컴퓨팅 환경을 제공받을 수 있게 되었다. 그러나 일반적인 무선 환경에서는 핸드오프나 비트오류 등 단시간 단절이 매우 흔한 일이지만 TCP는 이를 유선상의 혼잡 신호로 인식하여 전송 윈도우의 크기를 줄이고 불필요한 대기시간을 설정하는 등의 혼잡제어 정책을 수행하게 된다. 따라서 이러한 문제를 해결하기 위해 여러 계층에서 시도하는 노력이 있었으나 링크 상태를 파악하여 응용계층에서 자료 전송 여부를 미리 결정할 수 있다면 불필요한 전송을 막을 수 있다. Remote Method Invocation(RMI)는 대화형 응용 프로그램과 서비스를 쉽게 만들 수 있도록 해주는 Java 기반의 하부 구조로써, 응용계층과 전송계층의 중간에 위치하여 사용자의 자료를 전달하는 역할을 하므로 이를 수정하는 것이 가장 바람직하다. 이 논문에서는 무선 상황에 적절하게 적응하는 RMI를 위해 동작과정을 확장하여 구현한다.