• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.8
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• pp.1384-1390
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• 2008

The TCP protocol is originally designed for wired network, however it performs very poor in wireless network due to different nature of wireless network from wired networks. In terms of TCP performance improvement in wireless Ad-hoc network, many researches show that small congestion window size of TCP connection can improve TCP performance. We propose a new TCP algorithm to improve TCP performance in wireless Ad-hoc network. The basic idea of our approach is that TCP receiver estimates the optimum window size and then sets congestion window limit of TCP sender to an optimum value by using the advertised window field in TCP ACK packet. From extensive computer simulation, the proposed algorithm shows superior performance than traditional TCP protocols in terms of packet delivery ratio and packet loss.

• Journal of Internet Computing and Services
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• v.1 no.2
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• pp.1-10
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• 2000

ATM network technology is generally used for the solution of integrating multimedia service in high-speed Internet. In Internet protocol based on ATM services, If single cell is lost in ATM layer, the entire TCP packet will be lost. Therefore, TCP performance will be degraded. In order to reduce cell loss, when congestion occur, UBR+EPD mechanism is proposed to improve the throughput in TCP over UBR, and ER scheme is suggested in TCP over ABR. In this paper, we analyzed the performance improvement effect of UBR+EPD with FRR (Fast Retransmission and Recovery), the adjusting EPD threshold parameter (R), and variation of MTU (Maximum Transport Unit) size. As a result, through the analysis of performance, we know that the improved throughput and fairness are shown by the proposed scheme.

• Journal of Advanced Navigation Technology
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• v.6 no.1
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• pp.1-9
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• 2002

본 논문은 현재 무선 LAN 환경에서 적용되고 있는 매체접근제어 프로토콜인 IEEE 802.11 CSMA/CA를 이용함으로써, 음성 위주의 항공이동통신에서 음성과 데이터 채널을 통합하여 사용하도록 하고 있다. 또한, 표준으로 제정되어 있는 기존의 CSMA/CA 매체접근제어 프로토콜에 대한 이해를 바탕으로 성능 개선 방안을 제안하였으며, 이를 바탕으로 채널 효율을 향상시켰다. IEEE 802.11 매체접근제어(MAC, Medium Access Control) 프로토콜은 데이터 유형에 따라 채널을 두 개의 구간(PCF와 DCF 구간)으로 나누어 전송하도록 하고 있다. 본 논문에서는, 데이터 유형별로 적용되는 구간 중, DCF 구간에서의 패킷간의 충돌 가능성을 줄임으로써 데이터 전송 효율을 높이는데 주안점을 두고 있다. 제안 방안으로는 경쟁 구간에서의 전송에서 충돌에 따른 손실을 보완할 수 있도록 Wireless Window Protocol(이하, WWP)을 적용한 후 전송 과정을 제어하였다. Hidden 터미널의 영향이 적으므로 가시권내 통신에 적합한 기존의 2-Way CSMA/CA 프로토콜을 항공통신에 적용하여 상태 천이 과정을 5개에서 6개로 세분화하였고, 주어진 확률에 의하여 나누어진 두 개의 그룹별로 상태 천이 과정을 다르게 적용하였다. 따라서, 채널이 한 항공기에 의해 점유되는 PCF를 제외한, DCF 구간에서의 데이터 전송 효율을 높일 수 있었다. 시뮬레이션은 채널 대역폭과 패킷 크기 등을 파라미터 값으로 하여 수행하였으며, 시뮬레이션 결과 CSMA/CA에 비해 새로 제안한 알고리즘이 데이터 전송 과정에서 발생하는 충돌 가능성을 감소시킴으로써 성능이 향상됨을 알 수 있었다.

• Journal of KIISE:Information Networking
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• v.28 no.2
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• pp.242-250
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• 2001

This paper deals with the TCP bandwidth guarantee problem in a differentiated serviccs(Diffserv) network. The Diffserv assured s<:rvice differentiates packet drop probabilities to guarantee the promised bandwidth even under network congestion. However a token buffer marker fails to show adequate performance because TCI' generates packets according to the unique Tel' congestion control mechanism. We propose a marker that uses a data buffer as well as a token buffer. The marker with a data buffer works well with the assured service mechanism because it smooths Tel' traffic. We showed that the marker with a data buffer achieves the target throughput better than a marker with a token buffer only. We also showed that the optimal buffer size is proportional to reserved throughput and HTT.

• Journal of Advanced Navigation Technology
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• v.16 no.4
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• pp.641-648
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• 2012

The up-streams of the continuous streaming of data packets with lower importance level in the wireless sink node can cause congestion and delay, they affect on energy efficiency, memory size, buffer size, and throughput. This paper proposes a new wireless sink congestion control mechanism based on tournament scheduling. The proposed method consists of two module parts: stream decision module part and service differentiation module part. The final winner in the tournament controls congestion effectively, minimizes packet loss due to congestion, decreases energy consumption, and improves QoS. The simulation result shows that the proposed method is more effective and has better performance compared with those of congestion descriptor-based control method, reliability-based control method, and best-effort transmission control method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.4
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• pp.633-638
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• 1994

Asynchronous Transfer Model(ATM) has a fixed-length packet transport scheme. It is one of promising proposals in Broadband ISDN(B-ISDN). Since the packet length is fixed, it can be potentially to perform the various service to user. In this paper, a queueing model for an ATM Multiplexer loaded by the Continuous Bit Rate(CBR) traffic and the bursty traffic sources is analyzed. The queueing system has a finite buffer size. The cell loss probabilities with the simple and recurcive form for a performance measures of ATM multiplexer are derived, and are also evaluated through numerical examples. The results of this study may be used as guidelines in buffer design problems and performance analysis of ATM multiplexing system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12C
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• pp.261-267
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• 2001

This paper presents the architecture of multiple-queue manager for input-queued switch, which has arbitration latency, and the design of the chip. The proposed architecture of multiple-queue manager provides wire-speed routing with a pipelined buffer management, and the tolerance of requests and grants data transmission latency between the input queue manager and central arbiter using a new request control method, which is based on a high-speed shifter. The multiple-input-queue manager has been implemented in a field programmable gate array chip, which provides OC-48c port speed. It enhances the maximum throughput of the input queuing switch up to 98.6% with 128-cell shared input buffer in 16$\times$16 switch size.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.12
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• pp.32-40
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• 2015

Code division multiple access (CDMA) is one of the promising medium access control scheme for underwater acoustic sensor networks due to its beneficial features such as robustness against frequency-selective fading and high frequency-reuse efficiency. In this paper, we design a closed-loop power control scheme for the underwater CDMA, to adapt time-varying acoustic channel. In the proposed scheme, sink node sends to sensor nodes the associated path loss which is acquired by uplink-channel analysis based on received packets from the sensor nodes. Then, sensor nodes adjust their transmission power in an adaptive manner to time-varying underwater acoustic channel, according to the informations sent by the sink node.

• The KIPS Transactions:PartA
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• v.11A no.2
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• pp.143-148
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• 2004

The session management overhead on the network systems like firewalls or intrusion detection systems is getting grown as the session table is glowing. In this paper. we propose the event-based timeout management policy to increase packet processing throughput on network systems by decreasing the system's timeout management overhead that is comparable to the existing time-based timeout management policies. Through some empirical studies using a session management system implemented in this paper we probed that the proposed policy provides better packet processing throughput than the existing policies.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2003.11a
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• pp.139-142
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• 2003

무선 네트워크에서 멀티미디어 전송을 위해 Delay를 줄이는 방법을 연구하고 이에 대한 실험적 검증을 하였다. 멀티미디어 데이터는 그 특성상 주어진 시간에 도착하지 못하는 데이터는 손실로 간주되므로 Loss에 못지 않게 Delay가 중요한 QoS 파라미터가 된다. 본 연구에서는 IEEE 802.11b를 그 환경으로 하여 멀티미디어 데이터를 트랜스코더를 이용하여 미디어 전환 후 전송 학 때 Delay를 최소화 할 수 있는 방안을 연구하였다. 무선 환경에서의 SNR을 기준으로 다양한 조건에서 패킷 사이즈와 전송 간격을 조절하여 무선 네트워크에서 Delay와 Loss가 적게 발생하는 환경을 찾아내었으며 이전 상태와 현재 상태를 비교하여 미래의 상태를 미리 예측학 수 있는 알고리즘을 개발하여 적용하였다. 이러한 연구 결과는 MPEG-4의 Video racket을 가장 효율적인 크기로 인코딩 할 수 있는 판단 근거를 만들게 해 주며, 예측 된 패킷 사이즈를 통하여 다음 전송이 이루어지기 전에 트랜스코더에게 미리 인코딩 하여 준비학 수 있도록 해서 네트워크에 적응적인 전송 방식으로 구현될 수 있는 장점이 있다.