• Proceedings of the Korean Information Science Society Conference
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• 2003.04d
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• pp.512-514
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• 2003

본 논문은 모바일 환경에서 제한된 메모리를 가지고 있는 수신자에 의 한 TCP흐름 제어 방법을 제안한다. TCP 흐름 제어는 송신자에서 수신자에게 전달되는 Advertised 윈도우 크기를 조정하여 수행된다. 수신자는 무선 대역폭과 종단간 패킷 왕복 시간을 동적으로 측정하며 최적의 Advertised 윈도우 크기를 계산하고 송신자의 전송률을 무선 대역폭으로 제한한다. 제안된 흐름 제어 기법은 제한된 메모리를 가진 수신자를 고려하였으며 무선 네트웍의 특성을 고려 한 효율적 인 TCP 흐름 제어로 TCP의 전송 성능 향상과 종단간 패킷 왕복 시간의 지연을 줄일 수 있도록 하였다. 제안된 흐름 제어 기 법의 효율성과 성능을 구현과 실험을 통해 검증한다.

• Journal of the Korea Society of Computer and Information
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• v.10 no.4 s.36
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• pp.275-283
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• 2005

In this paper, we propose an overlay transmission method of end-to-end host to solve decrease in transmission rate caused by congestion in the application using multicast. In this proposed method, we've selected an overlay end-to-end host (OEH) for overcast transmission for each node, and the OEH can transmit duplicative packets. When the loss rate is more than the overcast threshold, the receivers of node in congestion are dropping from current layers and the OEH of lower nodes can request overcast transmission to OEH of non-congestion nodes for receiveing packets. In simulation results, it was known that the proposed method improves transmission rates over those of existing methods.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.91-100
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• 2004

This paper presents a receiver-driven TCP flow control mechanism, which is adaptive to the wireless condition, for memory constrained mobile receiver. A receiver-driven TCP flow control is, in general, achieved by adjusting the size of advertised window at the receiver. The proposed method constantly measures at the receiver both the available wireless bandwidth and the packet round-trip time. Depending on the measured values, the receiver adjusts appropriately the size of advertised window. Constrained by the adjusted window which reflects the current state of the wireless network, the sender achieves an improved TCP throughput as well as the reduced round-trip packet delay. Its implementation only affects the protocol stack at the receiver and hence neither the sender nor the router are required to be modified. The mechanism has been implemented in real environments. The experimental results show that in CDMA2000 1x networks the TCP throughput of the proposed method has improved about 5 times over the conventional method when the receiver's buffer size is limited to 2896 bytes. Also, with 64Kbytes of buffer site, the packet round-trip time of the proposed method has been reduced in half, compared the case with the conventional method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.7
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• pp.1541-1548
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• 2012

This paper proposes a new multipath-based routing protocol on MANETs with Fast-Recovery of failures. The proposed protocol establishes the primary and secondary paths between a source and a destination considering the end-to-end packet reception reliability of routes. The primary path is used to transmit messages, and the secondary path is used to recover the path when detecting failures on the primary path. If a node detects a link failure during message transmission, it can recover the path locally by switching from the primary to the secondary path. By allowing the intermediate nodes to recover locally the route failure, the proposed protocol can reduce the number of packet loss and the amount of control packets for setting up new paths. The simulation result using QualNet simulator shows that the proposed protocol was about 10-20% higher than other protocols in terms of end-to-end message delivery ratio and the fault recovery time in case of link fault was about 3 times faster than the other protocols.

• The KIPS Transactions:PartC
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• v.13C no.3 s.106
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• pp.345-352
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• 2006

In the integrated wired/wireless network which consists of the Internet and moving networks, multiple wireless links are used to connect a fixed host(FH) in the Internet to a mobile host(MH) in the moving network. For use in such an environment, we propose a scheme to overcome the TCP performance degradation due to the packet losses over the wireless links without losing the end-to-end TCP semantics. The proposed scheme in each mobile router(MR) allows to obtain the information regarding packet losses over the upstream wireless links based on the received packet sequence number and the ACK number. This information is delivered to the upstream router, which enables the upstream access router(AR) or MR to quickly retransmit the lost packets. The proposed scheme has the feature to quickly recover the packet losses incurred over the upstream wireless links and the performance of the proposed scheme is evaluated through simulation. It is shown that the significant performance gain can be obtained using the proposed scheme compared with the snoop mechanism which maintains end-to-end TCP semantics and does not require any additional features at the source and/or destination nodes.

• Proceedings of the Korean Information Science Society Conference
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• 2006.10d
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• pp.193-198
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• 2006

멀티미디어 데이터를 이종의 수신자에게 안정적으로 전달하고자 하는 것은 인터넷에서 중요한 주제 중의 하나이다. 특히, 최근에 소개된 계층형 MDC(Layered Multiple Description Coding) 기법은 오버레이 멀티캐스트에서 이 문제를 효율적으로 해결하는데 중요한 접근방법이다. 그러나, 이 과정에서 저속 수렴(slow convergence)과 참가 실험(join experiment) 동안에의 손실과 같은 문제가 새롭게 발생하게 되었다. 본 논문에서는 비디오 스트리밍 서비스에서 위의 문제를 해결할 수 있는 효율적인 계층형 멀티캐스트 혼잡 제어 기법을 소개한다. 여기서 제시하는 기법의 가장 특징적인 점은 사용자가 수신할 계층(layer)의 수를 결정하기 위하여 패킷페어(packet-pair)방식에 기반한 수신률 조절 메커니즘을 사용하는 것이다. 결과적으로 본 논문에서는 수신자가 최적의 전송률에 빠르게 수렴하면서도 손실을 최소화할 수 있는 종단-대-종단간 혼잡 제어 기법을 제시한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2014.04a
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• pp.177-179
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• 2014

컨텐츠 중심 네트워크(CCN)는 데이터 송수신을 위해 IP 주소로 종단 간 1:1 연결을 맺어야 하는 기존 인터넷의 비효율성을 해결하고자 등장한 미래 인터넷 기술 중 하나이다. CCN은 IP 주소 대신 컨텐츠의 이름을 사용하여 라우팅을 수행하며 네트워크상의 노드들이 컨텐츠를 저장하고 컨텐츠를 요청하는 인근의 소비자들에게 배포하는 방식을 사용한다. 하지만 기존의 CCN은 최초 데이터 패킷이 도착한 하나의 경로로만 컨텐츠를 수신하고 이후에 도착한 데이터 패킷은 폐기하기 때문에 전송률과 대역폭 측면에서 비효율적이다. 따라서 본 논문에서는 다수의 노드에 서로 다른 데이터 청크를 요청하여 다중경로를 통해 컨텐츠를 효율적으로 수신하는 기법을 제안한다.

• The KIPS Transactions:PartC
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• v.12C no.1 s.97
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• pp.121-128
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• 2005

Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.5A
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• pp.474-481
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• 2010

In this paper, we present an effective cross layer-based video transmission algorithm over multi-hop mobile ad hoc networks. The proposed algorithm selects the most efficient PHY mode of wireless LAN multi-rate service at each node in a distributed way based on the available information at application, MAC, and physical layers in order to minimize end-to-end delay and maintain packet loss rate in tolerable range at the receiver. Finally, experimental results are provided to show superior performance of the proposed algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.2C
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• pp.188-195
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• 2012

Many systems are developing for the realization of NCW(Network Centric Warfare). UAV(Unmanned Aerial Vehicle) Network is attracting attention in a lot of military applications. In general, UAVs have the potential to create an ad-hoc network and greatly reduce the hops from source to destination. However, UAV networks exhibit unique properties such as high mobility, high data rate, and real time service. The routing protocols are required to design the multi-hop routing protocols that can dynamically adapt to the requirements of UAV network. In this paper we analyse Geographic Routing Protocol is based on geographical distance between source and destination for efficient and reliable transmission. Geographic Routing Protocol is evaluated in video service scenarios with TDMA model in our simulation. The simulation results show that the performance of Geographic Routing Protocol is better than the MANET Routing Protocol in terms of packet received ratio, end to end delay, and routing traffic sent.