• Journal of KIISE:Information Networking
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• v.32 no.1
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• pp.12-19
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• 2005

Reliable transport protocols such as TCP are tuned to Perform well in traditional networks where packet losses occur mainly because of congestion. In a wireless network, however, packet losses will occur more often due to reasons such as the high bit error rate and the handoff rather than due to congestion. When using TCP over wireless network, TCP responds to losses due to the high bit error rate and the handoff by invoking congestion control and avoidance algorithms, resulting in the degraded end-to-end performance in the wireless network. There have been several schemes for improving TCP performance over wireless links. Among them, SNOOP Is a very promising scheme because of the localized retransmission. In this thesis, an efficient scheme is proposed by modifying SNOOP scheme. The invocation of congestion control mechanism is now minimized by knowing the cause of packet loss.

• 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.13C no.3 s.106
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• pp.353-358
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• 2006

Unlike TCP Reno, TCP Vegas recognizes network congestion through the measuring of RTT (Round Trip Time) and decides the main congestion control parameters, such as Windows size. But, congestion avoidance scheme of Vegas poorly reflects asymmetric characteristics of packet path because TCP Vegas uses the measuring of RTT that reflects forward/backward packet transmission delay as a forward delay. The RTT can't infer the forward/backward transmission delay variation because it only measures the packet's turn around time. In this paper, We have designed and implemented a new Vegas congestion control algorithm that can distinguish forward/backward network congestion. We have modified the source codes of TCP Vegas in Linux 2.6 kernel and verified their performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6A
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• pp.458-467
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• 2009

Long Tenn Evolution (LTE) is the next generation mobile phone technology which has being standardized by the Third Generation Partnership Project (3GPP). In the existing mobile communication networks, voice traffic is delivered through circuit switched networks. In LTE, however, all kinds of traffic are transferred through IP based packet switched networks which has best-effort characteristic. Therefore, providing QoS in LTE system is difficult. In order to provide QoS in LTE, RRM is very important. Especially, in part of RRM, call admission control (CAC) performs an important function to reduce network congestion and guarantee a certain level of QoS for on-going calls. In this paper, we propose a CAC algorithm in order to provide QoS for various kinds of services in LTE system. The performance of the proposed algorithm is evaluated with various simulation environments. The results show that the proposed algorithm provides QoS through rejections of requested calls. Especially, the proposed CAC algorithm can be satisfied with packet delay requirement defined in LTE specification.

• Journal of KIISE:Information Networking
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• v.33 no.1
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• pp.59-75
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• 2006

In this paper, we propose a congestion control scheme to control the congestion due to the mobility of ubiquitous devices on ubiquitous computing environment. Especially, this congestion control scheme provides a reverse congestion avoidance state which can classify between packet error by features of wireless network and packet dropping by congestion. Also, it provides a slow stop state which can minimize bandwidth waste due to congestion control. The proposed congestion control scheme controls more adaptive than existing congestion control schemes. The proposed congestion control scheme is designed based on DCCP(Datagram Congestion Control Protocol) being proposed by IETF(Internet Engineering Task Force) and implemented on the Linux kernel. In simulation results, the proposed congestion control scheme provides good bandwidth throughput in wireless network as well as in wired network.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.42 no.2 s.302
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• pp.1-10
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• 2005

The current implementation of TCP for the Internet is not efficient when used for Mobile Ad hoc Networks (MANENTs). This is because TCP assumes that all packet losses are caused by congestion, whereas transmission errors are a main reason for packet losses in wireless networks. To remedy this situation and increase performance, we propose an end-to-end method of using propagation delays and the differences between propagation delays to distinguish the causes for packet losses. The proposed method has two characteristics: Firstly, it is energy-efficient because this solution is only initiated when a packet loss is detected. Secondly, our approach considers only the one way propagation delay and is more accurate in determining causes for packet losses than existing methods which consider round trip time. Petri net models of the proposed TCP and of the standard TCP have been built and simulations have been performed on them. Our simulation results show that the proposed approach increases throughput and reduces propagation delay compared with standard TCP.

• Journal of Advanced Navigation Technology
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• v.20 no.4
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• pp.298-304
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• 2016

In this paper, a wavelet packet transform method is proposed for improving the altitude control performance of quadrotor UAV using an ultrasonic rangefinder. A ground tests are conducted using an ultrasonic rangefinder that is much used for vertical takeoff and landing. An ultrasonic rangefinder suffers from signal's spike due to specular reflectance and acoustic noise. The occurred spikes in short time span need to be analyzed at both sides time and frequency domain. The analyzed spikes of the ultrasonic rangefinder using a wavelet packet transform. Compared with the discrete wavelet transform, the wavelet packet decomposition can obtain more abundant time-frequency localization information, so it is more suitable for analyzing and processing ultrasonic signals spike. Experimental results show that it can effectively remove the spikes of the ultrasonic rangefinder.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.3B
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• pp.339-348
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• 2004

Packet classification is an important factor to support various services such as QoS guarantee and VPN for users in Internet. Packet classification is a searching process for best matching rule on rule tables by employing multi-field such as source address, protocol, and port number as well as destination address in If header. In this paper, we propose hardware based packet classification algorithm by employing tree bitmap of multi-bit trio. We divided prefixes of searching fields and rule into multi-bit stride, and perform a rule searching with multi-bit of fixed size. The proposed scheme can reduce the access times taking for rule search by employing indexing key in a fixed size of upper bits of rule prefixes. We also employ a marker prefixes in order to remove backtracking during searching a rule. In this paper, we generate two dimensional random rule set of source address and destination address using routing tables provided by IPMA Project, and compare its memory usages and performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.4A
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• pp.377-386
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• 2008

In this paper, a new handover management scheme has been proposed to reduce handover latency and to support fast handover without packet losses, so that it may be applicable to the wireless mobile Internet system such as IPv6-based WiBro system. To minimize the handover latency in processing of movement detection, we propose the handover management scheme which simplifies the handover message exchanging procedure between mobile subscriber station (MSS) and network by integrating layer 2 and layer 3 handovers efficiently based on the layer2 information. To reduce the processing delay from new care-of-address (NCoA) configuration during handover, we propose that NCoA is created, distributed and managed by new access control router (NACR). In addition, in order to minimize the packet transmission delay and eliminate the packet losses, the proposed scheme employs a crossover router (CR) which is upper network located over PACR and NACR and employs the packet buffering for MSS. The simulation study shows that the proposed scheme achieves loss-free packet delivery and low latency in the environment of narrow overlapped cell area or high velocity of the MSS, comparing the performance with the conventional schemes.

• Journal of IKEEE
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• v.19 no.3
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• pp.385-391
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• 2015

In this paper, we analyze the communication performance in a time synchronous multi-channel environment and deal with a packet collision avoidance technique to improve it based on IEEE1609.4 for increasing the efficiency of the control channel IEEE802.11p WAVE communication system. In previous works, they tried to solve this problem by message scheduling method on application layer software or changing the value of the random back-off optionally Contention Window. In this paper, we propose a method for adjusting the Channel Guard Interval for packet collision avoidance. The performance was evaluated by the actual vehicle test. The result was confirmed performance over 90% PDR(Packet Delivery Ratio).