• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.6B
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• pp.382-389
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• 2007

In this paper we report the development of mobility test bed, the realization of basic protocol that supports L3 mobility and the measurement of performance parameters that can affect handover performance. Previously proposed mobility management system, BBM (Break Before Make) method such as MIPV6 and Fast MIPV6, uses one interface in wireless network. In this method, to connect to new AP, it first disconnects existing AP and tries to connect to new AP. This can cause packet loss for the traffic vulnerable to delay such as VoIP. To provide seamless handover between different wireless networks, we propose MBB (Make Before Break) handover method having two network interfaces. Comparative study of previous method and proposed method on mobility and handover was conducted under simulated real environment on the test bed. Almost no packet loss was found with newly proposed method. In conclusion, it is shown that mobility protocol with proposed handover method can be applied to the application services sensitive to delay and packet loss.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.70-80
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• 2004

In order to reduce the increasing packet loss rates caused by an exponential increase in network traffic, the IETF(Internet Engineering Task Force) is considering the deployment of active queue management techniques such as RED(Random Early Detection). But, RED configuration has been a problem since its first proposal. This problem is that proposed configuration is only good for the particular traffic conditions studied, but may have detrimental effects if used in other conditions. While active queue management in routers and gateways can potentially reduce packet loss rates in the Internet, this paper has demonstrated the inherent weakness of current techniques and shows that they are unstable for tile various traffic conditions. The inherent problem with these queue management algorithms is that they all use static parameter setting. In this paper, in order to solve this problem, a new active queue management algorithm called SQM(Stabilized Queue Management) is proposed. This paper shows that it is easy to parameterize SQM algorithm to perform well under different congestion scenarios. This algorithm can effectively reduce packet loss while maintaining high link utilizations and is good for the various traffic conditions.

• 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 KIISE:Information Networking
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• v.28 no.4
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• pp.517-526
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• 2001

In order to reduce the increasing packet loss rates caused by an exponential increase in network traffic, the IETF(Internet Engineering Task Force) is considering the deployment of active queue management techniques such as RED(Random Early Detection). While active queue management in routers and gateways can potentially reduce total packet loss rates in the Internet, this paper has demonstrated the inherent weakness of current techniques and shows that they are ineffective in preventing high loss rates. The inherent problem with these queue management algorithms is that they all use queue lengths as the indicator of the severity of congestion. In this paper, in order to solve this problem, a new active queue management algorithm called MRED(Modified Random Early Detection) is proposed. MRED computes the packet drop probability based on our heuristic method rather than the simple method used in RED. Using simulation, MRED is shown to perform better than existing queue management schemes. To analyze the performance, we also measure throughput of traffics under the FIFO control, and compared the performance with that of this MRED system.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.2 s.332
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• pp.39-48
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• 2005

An HMIPv6 provides micro mobility management using MAP for decreasing handover delay and network load in a mobile IP networks. An HMIPv6 uses distance based algorithm for MAP selection when a mobile host enters a new network domain. However, since every mobile hosts select a farthest router as a MAP, a handover delay and packet loss will be increased. A new MAP selection scheme is herein proposed to solve the problems caused by the distance based MAP selection algorithm by using secondary MAP. We executed the performance evaluation by simulation about handover delay and packet loss of an HMIPv6 and our proposed scheme. The simulation results show that the performance of our proposed scheme is better than that of HMIPv6.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.6
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• pp.1063-1079
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• 2010

Vehicle ad hoc networks (VANET) are one of the most important technologies to provide various ITS services. While VANET requires rapid and reliable transmission, packet transmission in VANET is unstable because of high mobility. Many routing protocols have been proposed and assessed to improve the efficiency of VANET. However, topology-based routing protocols generate heavy overhead and long delay, and position-based routing protocols have frequent packet loss due to inaccurate node position. In this paper, we propose a position-based routing repair algorithm to improve the efficiency of VANET. This algorithm is proposed based on the premise that AODV (-PGB) can be used effectively in VANET, if the discovery, maintenance and repair mechanism of AODV is optimized for the features of VANET. The main focus of this algorithm is that the relay node can determine whether its alternative node exits and judge whether the routing path is disconnected. If the relay node is about to swerve from the routing path in a multi-hop network, the node recognizes the possibility of path loss based on a defined critical domain. The node then transmits a handover packet to the next hop node, alternative nodes and previous node. The next node repairs the alternative path before path loss occurs to maintain connectivity and provide seamless service. We simulated protocols using both the ideal traffic model and the realistic traffic model to assess the proposed algorithm. The result shows that the protocols that include the proposed algorithm have fewer path losses, lower overhead, shorter delay and higher data throughput compared with other protocols in VANET.

• 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.

• Journal of the Korean Institute of Intelligent Systems
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• v.19 no.2
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• pp.185-190
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• 2009

In this paper, an observer-based intelligent controller for the nonlinear networked control systems with packet loss is proposed for wireless sensor network. For the intelligent control of the nonlinear system, it uses the fuzzy system with Takagi-Sugeno (T-S) fuzzy model. The observer is designed for the fuzzy networked control system, and the output feedback controller is proposed for the stability of estimates and errors. The stability condition of the closed-loop system with the proposed controller is represented to the linear matrix inequality (LMI) form, and the observer and control gain are obtained by LMI. An example is given to show the verification discussed throughout the paper.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.707-715
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• 2003

In this paper, we propose a new MAC(Medium Access Control) protocol using Movable-boundary, which tries to guarantee Qos for multimedia traffic in the slotted CDMA system. In this scheme, the traffic scheduler assigns channel resource according to the packet priority per service class and adapts the Movable-boundary concept in which the minimum resource is assigned to each traffic class; the remaining resource if it is available can be assigned dynamically according to the temporal demand of other traffic classes. For performance analysis, we performed computer simulations to obtain throughput and packet loss rate and compared the results with Fixed-boundary system. We observed that the error rate of voice traffic could be maintained below a prescribed value while bursty traffic such as video source shares the same channel. In comparison with Fixed-boundary scheme, our protocol exhibits better throughput and packet loss rate performance.

• 대한공업교육학회지
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• v.33 no.1
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• pp.233-248
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• 2008

Stream Control Transmission Protocol(SCTP) is a new transport protocol that is known to provide improved performance than Transmission Control Protocol(TCP) in multi-homing environment that is having two and more IP addresses. But currently single-homed computer is used primarily that is having one IP address. To identify whether mean transfer time for SCTP is faster that for TCP in single-homed environment considering packet loss, we make up real testbed regulating the bandwidth, delay time and packet loss rate on router and observe the transfer time. We write server and client applications to measure SCTP and TCP mean transfer time by C language. Analysis of these experimental results from the testbed implementation shows that mean transfer time of SCTP is not better than performance of TCP in single homed environment exceptional case. Main reasons of performance are that SCTP compared to TCP stops transmitting data by timeout and data transmission is often delayed when SACK congestion happens. The result of study shows that elaborate performance tuning is required in developing a new SCTP module or using a implemented SCTP module.