• Journal of Internet Computing and Services
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• v.8 no.5
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• pp.125-132
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• 2007

Transmission mechanisms that include an available bandwidth estimation algorithm and a packet loss differentiation scheme, in general, exhibit higher TCP performance in wireless networks. TCP New Jersey, known as the best existing scheme in terms of goodput, improves wireless TCP performance using the available bandwidth estimation at the sender and the congestion warning at intermediate routers. Although TCP New Jersey achieves 17% and 85% improvements in goodput over TCP Westwood and TCP Reno, respectively, we further improve TCP New Jersey by exploring improved available bandwidth estimation, retransmission timeout, and recovery mechanisms. Hence, we propose TCP New Jersey PLUS (shortly TCP NJ+), showing that under 1% packet loss rate, it outperforms 3% by TCP New Jersey and 5% by TCP Wes1wood. In 5% packet loss rate, a characteristic of high bit-error-rate wireless network, it outperforms other TCP variants by 19% to 104% in terms of goodput even when the network is in bi-directional congestion.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.12
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• pp.54-62
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• 2007

Mobile IPv6 is the representative standard protocol proposed to offer mobility of mobile hosts, but it has drawbacks such as high handover latency and packet loss during handover. To complement these drawbacks, several protocols have been proposed and they can support faster handover than the Mobile IPv6, but there still exists inevitable packet loss during handover in those protocols. In this paper, we propose a new handover procedure of mobile host who has a dual link for minimizing packet loss rate during micro handover to improve mobility capability of Cellular Ethernet system that supports mobility at layer 2. And we derive the superiority of proposed scheme by comparing in terms of packet loss rate, handover latency and throughput of proposed scheme with those of Hierarchical Mobile IPv6 which is the representative IP-layer micro mobility protocol by simulation.

• 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.

• Journal of KIISE:Information Networking
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• v.30 no.3
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• pp.343-354
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• 2003

This paper describes algorithm that minimizes recovery time of packet loss resulting from handoff in multicast environments and guarantees reliability through interaction of FN(Foreign Network) with PMTP(Predictable Multicast Tree Protocol). To solve the problems that inefficient routing and handoff delay taking plate when using hi-directional tunneling and remote subscription independently in multicast environments, proposed algorithm uses tunneling and rejoining multicast group according to the status of an arriving FA in a foreign network. Furthermore, proposed algorithm sends packet loss information and register message to previous FA or current FA at the same time. so, MH is able to recovery packet loss in handoff delay as soon as possible. As a result of performance analysis, proposed algorithm is more efficient than previous researches and is applicable to existing handoff method without requiring additional procedures.

• Journal of Industrial Technology
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• v.24 no.B
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• pp.187-192
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• 2004

In this paper, we introduce a packet diversity scheme to increase uplink channel utilization in a wireless network where forward error correction is used. The packet diversity allows neighbor base stations to receive uplink packets from a mobile terminal in order to increase the utilization of the uplink channel. By allowing multiple base stations to receive the same packets, we can improve the error correction capability in an uplink channel. By incorporating the packet diversity we can reduce the parity overhead of each packet for a given tolerable loss probability, which improves the link efficiency.

• Journal of information and communication convergence engineering
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• v.8 no.6
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• pp.721-724
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• 2010

In a network router, packet loss may occur when it overflows due to sudden burst traffic. This paper studies how much large buffers are required to eliminate the packet losses. There are buffers on which packet arrive and one output port to which a packet is transmitted at a time. The buffer management algorithm should determine one of the buffers whose packet is transmitted to the output port at each time. The front packet belonging to the buffer determined by the algorithm is transmitted. The goal is to minimize the sum of the lengths of buffers to transmit all the packets. We provide an optimal off-line algorithm and also we show the lower bounds of on-line algorithms. The on-line algorithm has no prior information of the packets having arrived in the future. Its performance is compared to that of the optimal off-line algorithm.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.30 no.1
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• pp.59-65
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• 2007

HPI (High-speed Portable Internet) system which provides high speed internet services is going to be commercialized soon. Since HPI provides simultaneously four different service types such as UGS (Unsolicited Grant Service), rtPS (real time Polling Service), nrtPS(non-real time Polling Service), and BE (Best Effort) under different QoS (Quality of Service) requirements and limited wireless channel resources, efficient packet scheduling mechanisms are necessary to increase the utilization of channels as well as to satisfy the various QoS requirements. This study regards the traffic data to be served as time series and proposes a new packet scheduling algorithm based on the nonparametric statistical test. The performance of the newly proposed algorithm is evaluated through the simulation analysis using a simulator that can evaluate the performance of packet scheduling mechanisms under various values of system parameters and measures such as packet delay time, data transmission rate, number of loss packets, and channel utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.4B
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• pp.692-699
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• 2000

We build and analyze some types of queueing model to discuss capacity dimensioning of access links of a LAN interworking unit connected to the Internet backbone network. We assume that the IWU has a FIFO buffer to transmit IP packets to the Internet through the backbone. In order to analyze the system, we use a Poisson process and an MMPP process as input traffic models of IP packets and we use a general service time distribution as a service time model. But we use both an exponential service time and a deterministic service time in numerical examples for simple and efficient performance comparisons. As performance measures, we obtain the packet loss probability and the mean packet delay. We present some numerical results to show the effect of arrival rate, buffer size and link capacity on packet loss and mean delay.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.86-88
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• 2004

As the rapid incensement of the number of internet users has occurred recently, many multimedia application services have been emerging. To improve quality of service, traffic can be suggested to be classified with priority in EPON(Ethernet Passive Optical Network), which is supporting the multimedia application services. In this paper, multimedia application services treat bandwidth classifying device in serving both delay sensitive traffic for real-time audio, video and voice data such as VoIP(Voice over Internet Protocol), and nonreal-time traffic such as BE(Best Effort). With looking through existing mechanisms, new mechanism to improve the quality will be suggested. The delay performances and packet losses of traffic achieved by supporting bandwidth allocation of upstream traffic in suggested mechanism will be analyzed with simulation.

• Journal of Information Processing Systems
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• v.2 no.3 s.4
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• pp.183-188
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• 2006

A RMON agent system, which locates on a subnet, collects the network traffic information for management by retrieving and analyzing all of the packets on the subnet. The RMON agent system can miss some packets due to the high packet analyzing overhead when the number of packets on the subnet is huge. In this paper, we have developed a light-weight RMON agent system that can handle a large amount of packets without packet loss. Our RMON agent system has also been designed such that its functionality can be added dynamically when needed. To demonstrate the dynamic reconfiguration capability of our RMON agent system, a simple port scanning attack detection module is added to the RMON agent system. We have also evaluated the performance of our RMON agent system on a large network that has a huge traffic. The test result has shown our RMON agent system can analyze the network packets without packet loss.