• Journal of the Korea Society of Computer and Information
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• v.14 no.6
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• pp.73-81
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• 2009

TCP assumes that packet loss is always happened by congestionlike wired networks because is can not distinguish between congestion loss and transmission error loss,. This assumption results in unnecessary TCP performance degradation in wireless networks by reducing sender's congestion window size and retransmitting the lost packets. Also, repeated retransmissions loed to waste the limited battery power of mobile devices. In this paper, we propose the new congestion control scheme that add the algorithms monitoring networks states and the algorithms preventing congestion to improve TCP throughput performance and energy efficiency in wireless ad-hoc networks. Using NS2, we showd our scheme improved throughput performance and energy efficiency.

• Annual Conference of KIPS
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• 2007.11a
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• pp.995-998
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• 2007

무선 네트워크에서 사용되는 단말기는 이동성이라는 특징상 한정된 에너지를 사용하여 동작하게 된다. 따라서 무선 호스트에 의해 소모되는 에너지의 양을 감소시키기 위한 기술은 대단히 중요하다. 이러한 기술적 지원을 위해 IEEE 802.11 에서는 DCF (Distributed Coordination Function) 전력 절감 메카니즘을 제안하고 있다. 그런데, DCF 를 위한 IEEE 802.11 전력 절감 메카니즘에서는 ATIM 창 동안 노드들은 비콘 기간 동안 깨어 있는 상태로 있을 것인지를 결정하기 위해서 control packet 을 교환 하는데, 이러한 ATIM 창 크기는 각각의 노드들의 전력 절감에 상당한 영향을 미친다. 그래서 ATIM 창 크기를 효율적으로 할당하기 위해 DPSM 과 같은 기법들이 개발되었다. 본 논문은 ATIM 창 크기를 동적으로 증감시켜서 ATIM 창 시간동안 소모되는 에너지를 줄이도록 하였다. ATIM 창 크기를 동적으로 할당하기 위하여 통계적 예측 기법인 칼만 필터를 도입하여 예측시스템을 구축하였으며, 이 예측 시스템을 통해 다음 상태에서 적용할 ATIM 창 크기를 예측하여 동적으로 할당하도록 하였다. 실험 결과 네트워크 생존 시간을 28.6% 증가시켰고, ATIM 창 크기 예측값의 오차는 4.42%로 나타났다.

• The KIPS Transactions:PartC
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• v.16C no.5
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• pp.661-668
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• 2009

The Extended real-time Polling Service (ErtPS) is proposed tosupport QoS of VoIP service with silence suppression which generates variable size data packets in IEEE 802.16e systems. If the silence is suppressed, VoIP should support Comfort Noise Generation (CNG) which generates comfort noise for receiver's auditory sense to notify the status of connection to the user. CNG mode in silent-period generates a data with lower bit rate at long packet transmission intervals in comparison with talk-spurt. Therefore, if the ErtPS, which is designed to support service flows that generate data packets on a periodic basis, is applied to silent-period, resources of the uplink are used inefficiently. In this paper, we proposed the Improved ErtPS algorithm for efficient resource utilization of the silent-period in VoIP traffic supporting CNG. In the proposed algorithm, the base station allocates bandwidth depending on the status of voice at the appropriate interval by havingthe user inform the changes of voice status. The Improved ErtPS utilizes the Cannel Quality Information Channel (CQICH) which is an uplink subchannel for delivering quality information of channel to the base station on a periodic basis in 802.16e systems. We evaluated the performance of proposed algorithm using OPNET simulator. We validated that proposed algorithm improves the bandwidth utilization of the uplink and packet transmission latency

• 대한공업교육학회지
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• v.33 no.2
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• pp.287-299
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• 2008

HTTP is one of the most widely used protocols of the WWW. Currently it uses TCP as the transport layer protocol to provide reliability. The HTTP uses separate TCP connection for each file request and adds unnecessary head-of-line blocking overhead for the file retrieval. The web application is short sized and affected by the increased handover latency of TCP in wireless environment. SCTP has attractive features such as multi-streaming and multi-homing. SCTP's multi-streaming and multi-homing avoid head-of-line blocking problem of TCP and reduce handover latency of TCP in wired and wireless environment. Mean response time is the important measure in most web application. In this paper, we present the comparison of mean response time between HTTP over SCTP with that of HTTP over TCP in wired and wireless environments using NS-2 simulator. We measured mean response time for varying packet loss rate, bandwidth, RTT, and the number of web objects in wired environment and mean response time and packet loss rate for varying moving speed and region size in wireless environment. Our experimental result shows that SCTP reduces the mean response time of TCP based web traffic.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.1A
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• pp.96-103
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• 2000

The voice traffic is one of the most important real-time objects in WATM(Wireless Asynchronous Transfer Mode) networks. In this paper, we propose a new MAC(Medium Access'Control) protocol for the efficienttransmission of voice traffic over WATM networks in the multimedia environment and compare the performanceto existing similar protocols. The new protocol separates the reservation slot period for voice and that for data toguarantee some level of QoS(Quality of Service) in voice traffic. This is denoted by a slot assignment functiondepending on the frame size. According to the characteristics of voice traffic which is repeatedly in silent states,the protocol allocates voice reservation request slots dynamically with respect to the number of silent(off state)voice sources and also sends the first block of talkspurt restarted after silent period with a reservation requestslot to reduce the access delay.The simulation results show that the proposed protocol has better performance than Slotted Aloha in bandwidthefficiency, and can serve a certain level of QoS by the given slot assignment function even when the number ofvoice terminals varies dynamically. This means we can observe that the new MAC protocol is much better thanthe NC-PRMA(None Collision-Packet Reservation Multiple Access) protocol.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.149-158
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• 2005

This paper proposes a congestion avoidance algorithm which provides stable throughput and transmission rate regardless of buffer size by limiting the TCP congestion window in fixed bandwidth networks. Additive Increase, Multiplicative Decrease (AIMD) is the most commonly used congestion control algorithm. But, the AIMD-based TCP congestion control method causes unnecessary packet losses and retransmissions from the congestion window increment for available bandwidth verification when used in fixed bandwidth networks. In addition, the saw tooth variation of TCP throughput is inappropriate to be adopted for the applications that require low bandwidth variation. We present an algorithm in which congestion window can be limited under appropriate circumstances to avoid congestion losses while still addressing fairness issues. The maximum congestion window is determined from delay information to avoid queueing at the bottleneck node, hence stabilizes the throughput and the transmission rate of the connection without buffer and window control process. Simulations have performed to verify compatibility, steady state throughput, steady state packet loss count, and the variance of congestion window. The proposed algorithm can be easily adopted to the sender and is easy to deploy avoiding changes in network routers and user programs. The proposed algorithm can be applied to enhance the performance of the high-speed access network which is one of the fixed bandwidth networks.

• Journal of KIISE:Computing Practices and Letters
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• v.11 no.2
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• pp.149-156
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• 2005

The HLA is the simulation architecture standard that the civilian and military simulation communities are deeply interested in. We can find various successful practices applying HLA to constructive simulations such as war games in domestics and overseas. However, any case of real-time distributed simulations has not been reported. The reason is that a message transmission period via RTI in a network layer varies according to computing power, simulation nodes, transmission types, and packet size; further a message processing time in an application layer depends on its processing methods, thus too difficult to set up real-time constraints for the enhancement of a real-time resolution. Hence, in this paper we have studied the real-time constraints of RTI for the development of the M-SAM simulator. Thus we have developed a HLA based pilot simulator using 6 PC's in LAN and then measured and analysed the performance of the RTI. As the results of our work, we could obtain the quantitative values for message delay, RTI overhead and RTI packet transmission ratio by a real operation scenario and loads, which are not shown in the previous works. We also expect that the results can be used as a guideline to set up the number of targets, transmission frequency and message processing method in the development of the M-SAM simulator and similar applications.

• Journal of KIISE:Information Networking
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• v.29 no.4
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• pp.375-385
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• 2002

The IETF has defined the Intserv model and the RSVP signaling protocol to improve QoS capability for a set of newly emerging services including voice and video streams that require high transmission bandwidth and low delay. However, since the current Intserv model requires each router to maintain the states of each service flow, the complexity and the overhead for processing packets in each rioter drastically increase as the size of the network increases, giving rise to the scalability problem. This motivates our work; namely, we investigate and devise new control schemes to enhance the scalability of the Intesev model. To do this, we basically resort to the SCORE network model, extend it to fairly well adapt to the three services presented in the Intserv model, and devise schemes of the QoS scheduling, the admission control, and the edge and core node architectures. We also carry out the computer simulation by using ns-2 simulator to examine the performance of the proposed scheme in respects of the bandwidth allocation capability, the packet delay, and the packet delay variation. The results show that the proposed scheme meets the QoS requirements of the respective three services of Intserv model, thus we conclude that the proposed scheme enhances the scalability, while keeping the efficiency of the current Intserv model.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.8A
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• pp.1169-1176
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• 2000

In this paper, label assignment schemes considering the IP flow model for the efficient MPLS traffic engineering are proposed and evaluated. Based on the IP flow model, the IP flows are classified into transient flows and base flows. Base flows, which last for a long time, transmit data in high bit rate, and be composed of many packets, have good implications for the MPLS traffic engineering, because they usually cause network congestion. To make use of base flows for the MPLS traffic engineering, we propose two base flow classifiers and label assignment schemes where transient flows are allocated to the default LSPs and base flows to explicit LSPs. Proposed schemes are based on the traffic-driven label triggering method combined with a routing tabel. The first base flow classifier uses both flow size in packet counts and routing entries, and the other one, extending the dynamic X/Y flow classifier, is based on a cut-through ratio. Proposed schemes are shown to minimize the number of labels, not degrading the total cut-through ratio.

• Journal of Digital Contents Society
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• v.8 no.3
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• pp.323-328
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• 2007

Communication systems beyond 3G should provide more than 100 Mbps for wireless access. In addition to smart antennas, wireless multi-hop networks are proposed to increase the cell size and throughput. For example, Zigbee technology is expected to provide low cost and low power connectivity and can be implemented in wireless mesh networks larger than is possible with Bluetooth. Also, home embedded system using wireless mesh network is one of the key market areas for Zigbee applications. If the line-of-sight path is shadowed by home obstacles, a direct connection between the access point (AP) and the node is not possible at high frequencies. Therefore, by using multi-hop relay scheme the end node can be reached to AP. In this paper, the relaying of data between the AP and the end node is investigated and the throughput and PER(Packet Error Rate) are evaluated in multi-hop wireless mesh networks by using DSSS/BPSK system.