• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.20-26
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• 2004

A congestion control algorithm to prevent buffer overflow in MSR(Mobility Support Router) for I-TCP is proposed. Due to high bit error rate and frequent hand-offs over wireless environment, the current congestion control scheme in TCP Reno over mixed(wired and wireless) network exhibits lower throughput than the throughput achieved over wired only network. I-TCP has been proposed to address this by splitting a TCP connection into two TCP connections over wired section and wireless section, respectively. However, buffer overflow in MSR may occur whenever there are excessive bit errors or frequent hand-offs. This may lead to the loss of packets acked by MSR(resident in buffer) to the sender, but not received by the receiver, breaking TCP end-to-end semantics. In this Paper, a new scheme is proposed to prevent the MSR buffer from overflow by introducing “flow control” between the sender and the MSR. Advertised window for the TCP connection between the sender and the MSR is tied to the remaining MSR buffer space, controlling the flow of packets to the MSR buffer before overflow occurs.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.1-10
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• 2004

This study discusses Application-layer FEC using erasure codes. Because of the simple decoding process, erasure codes are used effectively in Application-layer FEC to deal with Packet-level errors. The large number of parity packets makes the loss rate to be small, but causes the network congestion to be worse. Thus, a redundancy control algorithm that can adjust the number of parity packets depending on network conditions is necessary. In addition, it is natural that high-priority frames such as I frames should produce more parity packets than low-priority frames such as P and B frames. In this paper, we propose a redundancy control algorithm that can adjust the amount of redundancy depending on the network conditions and depending on data priority, and test the performance in simple links and congestion links.

• Journal of Korea Society of Digital Industry and Information Management
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• v.5 no.2
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• pp.141-156
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• 2009

As the TCP is the protocol designed for the wired network that packet loss probability is very low, because TCP transmitter takes it for granted that the packet loss by the wireless network characteristics is occurred by the network congestion and lowers the transmitter's transmission rate, the performance is degraded. The Snoop Protocol was designed for the wired network by putting the Snoop agent module on the BS(Base Station) that connect the wire network to the wireless network to complement the TCP problem. The Snoop agent cash the packets being transferred to the wireless terminal and recover the loss by resending locally for the error occurred in the wireless link. The Snoop agent blocks the unnecessary congestion control by preventing the dupack (duplicate acknowledgement)for the retransmitted packet from sending to the sender and hiding the loss in the wireless link from the sender. We evaluated the performance in the wired/wireless network and in various TCP versions using the TCP designed for the wired network and the Snoop designed for the wireless network and evaluated the performance of the wired/wireless hybrid network in the wireless link environment that the continuous packet loss occur.

• Journal of KIISE:Information Networking
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• v.27 no.2
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• pp.160-174
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• 2000

Nowadays, most widely used transport protocol, TCP is tuned to perform well in traditional networks where packet losses occur mostly because of congestion. TCP performs reliable end-to-end packet transmission under the assumption of low packet error rate. However, networks with wireless links suffer from significant losses due to high error rate and handoffs. TCP responds to all losses by invoking congestion control and avoidance algorithms, resulting in inefficient use of network bandwidth and degraded end-to-end performance in that system. To solve this problem, several methods have been proposed. In this paper, we analyse and compare these methods and propose appropriate model for improving TCP performance in the network with wireless links. This model uses TCP selective acknowledgement (SACK) option between TCP ends, and also uses caching method at the base station. Our simulation results show that using TCP SACK option with base station caching significantly reduces unnecessary duplicate retransmissions and recover packet losses effectively.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1372-1382
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• 1999

The major source of errors in high-speed networks such as Broadband ISDN(B-lSDN) is buffer overflow during congested conditions. These congestion errors are the dominant sources of errors in 1high-speed networks and result in cell losses. Conventional communication protocols use error detection and retransmission to deal with lost packets and transmission errors. However, these conventional ARQ(Automatic Repeat Request) methods are not suitable for the high-speed networks since the transmission delay due to retransmissions becomes significantly large. As an alternative, we have presented a method to recover consecutive cell losses using forward error correction(FEC) in ATM(Asynchronous Transfer Mode)networks to reduce the problem. The performance estimation based on the cell discard process model has showed our method can reduce the cell loss rate substantially. Also, the performance estimations in ATM networks by interleaving and IP multicast service are discussed.

• ETRI Journal
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• v.37 no.6
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• pp.1096-1107
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• 2015

This paper presents a dynamic retry adaptation scheme for H.264 HD video, called DRAS.264, which dynamically adjusts the retry limits of frames at the medium access control (MAC) layer according to the impact those frames have on the streamed H.264 HD video. DRAS.264 is further improved with a bandwidth estimation technique, better prediction of packet delays, and expanded results covering multi-slice video. Our study is performed using the Open Evaluation Framework for Video Over Networks as a simulation environment for various congestion scenarios. Results show improvements in average peak signal-to-noise ratios of up to 4.45 dB for DRAS.264 in comparison to the default MAC layer operation. Furthermore, the ability of DRAS.264 to prioritize data of H.264 bitstreams reduces error propagation during video playback, leading to noticeable visual improvements.

• Proceedings of the Korea Information Processing Society Conference
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• 2006.05a
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• pp.1147-1150
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• 2006

Stream Control Transmission Protocol (SCTP) uses the 32-bit checksum in the common header, by which a corrupted SCTP packet will be regarded as a lost packet and then discarded. This may result in degradation of SCTP's throughput performance over wireless networks. This paper proposes a new chunk checksum scheme for SCTP, in which each data chunk contains its own checksum field and SACK chunk carry corresponding Transmission Sequence Number (TSN) with timestamp for every corruption event. The proposed chunk checksum scheme is introduced with the following three purposes: 1) to distinguish the chunk corruptions from the chunk losses; 2) to avoid the unnecessary halving of the congestion window (cwnd) in the case of chunk corruption; 3) to avoid the unwanted timeouts which can be induced in conventional SCTP in the case that the retransmitted data chunks are corrupted again in wireless networks. Simulation results show that the proposed chunk checksum scheme could improve the SCTP throughput in the wireless environments with a high bit error rate.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.6
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• pp.108-117
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• 2011

Seoul metropolitan government has been operating traffic signal control system with the name of COSMOS since 2001. COSMOS uses the degrees of saturation and congestion which are calculated by installing loop detectors. At present, inductive loop detector is generally used for detecting vehicles but it is inconvenient and costly for maintenance since it is buried on the road. In addition, the estimated queue length might be influenced in case of error occurred in measuring speed, because it only uses the speed of vehicles passing by the detector. A traffic signal control algorithm which enables smooth traffic flow at intersection is proposed. The proposed algorithm assigns vehicles to the group of each lane and calculates traffic volume and congestion degree using traffic information of each group using VANETs(Vehicular Ad-hoc Networks) inter-vehicle communication. It does not demand additional devices installation such as cameras, sensors or image processing units. In this paper, the algorithm we suggest is verified for AJWT(Average Junction Waiting Time) and TQL(Total Queue Length) under single intersection model based on GLD(Green Light District) Simulator. And the result is better than Random control method and Best first control method. In case real-time control method with VANETs is generalized, this research that suggests the technology of traffic control in signalized intersections using wireless communication will be highly useful.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.12A
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• pp.975-983
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• 2011

In this paper, we investigate how the IEEE 802.16 based wireless system can adopt the network coding. To avoid the problem of overhearing, we focus on the bidirectional traffic, where each end node exchanges network coded data over a relay node. The bidirectional traffic is usually observed in Internet, where TCP makes congestion control and error recovery based on the acknowledgement from the opposite direction. Thus, enhancing the spectral efficiency of wireless Internet through the network coding is expected. Our simulation with realistic radio characteristics and TCP-like traffic shows that the network coding improves the throughput by an average of 36 percent compared to the simple relay case.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.7
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• pp.9-19
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• 2004

WLAN (Wireless Local Area Networks) needs error recovery and flow control schemes to support reliable multicast protocol. Limited wireless bandwidth, as well as queuing losses caused by the asymmetric wired/wireless interactions, demands more effective approaches for reducing packet losses. Moreover, since the wireless channel is a shared broadcast medium, if sender receives feedback information simultaneously from several receivers, the feedback delays data frame transmission of forward direction by introducing channel congestion and burden at the sender. Therefore, it is important to minimize the amount of feedback information from receivers. In this paper, we propose an ARS(Adaptive Redundancy Scheme) that combines FEC(Forward Error Correction) using channel state estimation and ARQ(Automatic Repeat Request) both to reduce the amount of feedback information and the number of retransmissions and to guarantee high data reliability in a WLAN multicast environment. Performance of the proposed scheme is evaluated by means of analysis and simulations in AWGN and Rayleigh fading channels. The results show that the proposed scheme reduces the amount of feedback information and the number of retransmissions and guarantees high data reliability, while keeping throughput efficiency similarly with the conventional FEC and ARQ scheme.