• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.7C
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• pp.624-632
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• 2005

We consider an adaptive bit-loading technique for bit interleaved coded modulation-orthogonal frequency division multiplexing(BICM-OFDM) systems. By adjusting transmission parameter of each subcarrier adaptively depending on the subchannel state, the performance of OFDM system can be improved dramatically. In this paper, the number of bits for each subcarrier is allocated to minimize bit error rate keeping the constant throughput for the adaptive transmission technique of BICM-OFDM system which can be applied to real time transmission. Also, We use the discrete Lagrange multiplier method to get the optimum solution under the integer bit allocation constraint. Simulation results show that computational amount of the proposed bit allocation technique is not high and BICM-OfDM system using the proposed technique can get the SNR gain by 2$\~$3 dB over nonadaptive one.

• Journal of Korea Multimedia Society
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• v.2 no.3
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• pp.339-346
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• 1999

When a multicast video conference system utilizes RTP (Real Time Protocol) and RTCP (Real Time Control Protocol), the loss rate and the synchronization of transfer in RTCP affect the scalability of the system. The random delay technique introduced to resolve the problems is so simple that leads the network to meet some congestion in synchronizing feedback information when lots of people transfer the feedback information simultaneously, which reduces the scalability of system. In this paper, we propose a new feedback control algorithm that provides priority levels with the RTCP packet, which cuts down the feedback delay and increases the scalability. The criteria of providing priority Based on the decided priority level, Agent forced the session participants to provide much more RTCP packets, positively controlled, and the possible bandwidth can be measured. The simulation on this technique decreases the delay, and the feedback messages are equally distributed on a given time period.

• Journal of the Korea Computer Industry Society
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• v.3 no.8
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• pp.1027-1038
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• 2002

Multimedia data services have been widely developed with the recent progression of the Internet technology. Multimedia services, especially, must guarantee Quality of Service with restricted environment resources. The abrupt increase of transmission quanta, so called burst, and the number of client that access multimedia server, with limited network bandwidth will demage the quality of service and the utilization of network resource. Smoothing techniques can reduce the burst of a variable bit rate stream by transmitting data at a series of fixed rates. In this paper, we present a new smoothing algorithm using MPEG's characteristics in transmitting stored video data with VBR. The smoothing algorithm we present will be verified with assorted existing techniques in diverse environments.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1683-1690
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• 2005

This study examines dynamic flow control method in UDP, analyzes packet loss which is frequently used element in measuring existing dynamic flow control and characteristics of round trip time, and proposes a new method of measurement, RSST. The algorithm that uses the proposed RSST enables accurate measurement of network status by considering both the currently measured network status and the past history of network status in controlling the transmission rate. For comparison study, a network status measurement software program that displays detailed information about volume of transmission generation of network status, and flow pattern of network was developed. The performance test shows that the proposed algorithm can better adjust to network condition in terms of low pack loss rate over existing algorithms.

• Proceedings of the Korea Information Processing Society Conference
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• 2008.05a
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• pp.995-998
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• 2008

무선 통신 기술의 발전과 함께 802.11n과 같은 광대역 네트워크 인프라의 구축이 확대됨에 따라, 이에 적합한 높은 성능을 낼 수 있는 전송 계층의 프로토콜이 필요하다. 유선 환경에서 개발된 TCP(Transmission Control Protocol)는 현재 가장 널리 이용하는 전송 프로토콜이다. 하지만 높은 에러율과 긴 딜레이가 발생하는 무선 네트워크상에서는 패킷 손실의 원인을 구별하지 못하기 때문에 성능이 저하된다. 또한, 광대역폭을 지원하는 초고속 네트워크상에서 TCP가 동작할 경우 기존 무선 네트워크에서 처리하는 혼잡윈도우 값보다 커진 값을 제대로 처리하지 못하기 때문에 가용대역폭을 활용하지 못하는 문제점이 발생한다. 따라서, 본 논문에서는 초고속 네트워크 안에서 slowstart threshold에 기반하여 가용대역폭을 동적으로 획득하고, 높은 전송률을 보장하는 TCP VEST(TCP Vegas-based Estimator with Slowstart Threshold)를 제안한다. TCP VEST는 기본적으로 송신자기반 무선네트워크에서 우수한 성능을 지원하는 TCP WestwoodVT의 패킷 손실원인 구분법을 사용한다. 그리고 TCP VEST는 패킷 손실의 원인을 구분하고 각각의 경우에 따라서 전송량을 slowstart threshold 값의 변화에 맞춰 조절한다. NS-2를 통한 성능평가에서 VEST는 WestwoodVT와 비교했을 때 링크 에러율이 1%인 환경에서는 20% 이상의 전송률 향상을 나타내며, Westwood와 비교했을 때는 60%의 성능향상이 있다. 또한 초고속 네트워크에서 5%와 10% 사이의 에러율을 가질 때에는 WestwoodVT와는 50%, Westwood와는 300% 이상의 성능향상이 나타남을 보인다. 이를 통해 VEST가 기존의 무선 네트워크와 더불어 초고속 네트워크에서도 동적으로 대역폭을 획득함으로써 높은 전송률을 지원하는 것을 입증한다.

• Journal of KIISE:Computing Practices and Letters
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• v.16 no.7
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• pp.804-808
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• 2010

In this paper, we propose Restricted Angle-control Flooding considered density network of sensor nodes and node‘s limit energy. Restricted Angle-control Flooding, increase energy efficiency by reducing unnecessary candidate nodes involved in forwarding closer to the destination. And The Hole when faced with our proposal to raise rates data through Hole Detection which is sender‘s forwarding area is extended or broadcast to the entire network. Compared to the traditional flooding, we show the superiority at the node’s energy consumption, data rate and network lifetime through the performance.

• Proceedings of the Korean Information Science Society Conference
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• 1998.10a
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• pp.456-458
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• 1998

최근 인터넷 상에서의 효율적인 영상/음성 시스템을 개발하려는 연구가 활발히 진행되고 있다. 이러한 연구중의 중요한 분야로는 인터넷의 대역폭을 효율적으로 사용하려는 즉 가변적인 인터넷의 사용가능한 대역폭을 예측하고, 예측된 대역폭에 알맞게 영상과 음성 정보를 압축하는 연구일 것이다. 본 논문에서는 사용 가능한 대역폭 예측에 알맞게 영상정보의 압축률을 조절하는 기법을 소개한다. 본 논문에서는 영상 압축 알고리즘의 하나인 H.261에서 압축률에 관련된 변수 값과 다양한 영상에서의 압축율 변화의 상관관계를 측정하였다. 또한 본 논문에서는 측정된 상관 관계를 수식화되고, 이 수식을 어떻게 전송량 조절 (flow control)알고리즘에 사용할 것인가를 제안한다.

• The KIPS Transactions:PartC
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• v.9C no.3
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• pp.375-384
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• 2002

In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1B
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• pp.154-161
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• 2010

In wireless sensor networks, due to the many-to-one convergence of upstream traffic, congestion more probably appears. The existing congestion control protocols avoid congestion by controlling incoming traffic, but the duty-cycle operation of MAC(Medium Access Control) layer has not considered. In this paper, we propose DCA(Duty-cycle Based Congestion Avoidance), an energy efficient congestion control scheme using duty-cycle adjustment for wireless sensor networks. The DCA scheme uses both a resource control approach by increasing the packet reception rate of the receiving node and a traffic control approach by decreasing the packet transmission rate of the sending node for the congestion avoidance. Our results show that the DCA operates energy efficiently and achieves reliability by its congestion control scheme in duty-cycled wireless sensor networks.

• Journal of KIISE:Information Networking
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• v.33 no.5
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• pp.369-379
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• 2006

In recent years, there have been many researches about Ad-hoc Networks which is available to communicate freely between mobile devices by using multi-hop without any support of relay base or access point. TCP that used the most widely transport protocol in the Internet repeats packet loss and retransmission because it increases congestion window size by using reactive congestion control until packet loss occurs. As a result of this, energy of mobile device is wasted unnecessarily. In this paper, we propose TCP-New Veno in order to improve the energy efficiency of mobile device. According to the state of network, the scheme adjusts appropriate size of congestion window. Therefore, the energy efficiency of mobile device and utilization of bandwidth are improved by the scheme. From the simulation by using ns-2, we could see more improved energy efficiency with TCP-New Veno than those with TCP in Ad-hoc Networks.