• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2004.04a
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• pp.555-558
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• 2004

최근 인터넷의 발전으로 디지털 오디오 및 비디오와 같은 멀티미디어 스트림에 대한 요구가 증가하고 있다. 이러한 멀티미디어 스트리밍을 UDP로 전송할 경우 TCP와 같은 혼잡제어를 수행하지 않기 때문에 동일한 전송경로의 TCP 트래픽 궁핍을 일으켜 혼잡붕괴 및 막대한 전송지연을 초래한다. 이러한 문제점으로 인하여 실시간 멀티미디어 스트림의 전송지연과 혼잡제어를 위한 새로운 전송기법과 프로토콜에 대한 다각적인 연구가 수행되고 있다. TCP 친화적 혼잡제어 기법은 크게 일반적인 혼잡윈도우 관리기능을 이용하는 윈도우 기반 혼잡제어와 TCP 모델링 방정식 둥을 이용하여 전송율을 직접 조절하는 율 기반 혼잡제어로 나눌 수 있다. 본 논문은 윈도우 기반과 율 기반을 복합적으로 다룬 하이브리드형 TCP-friendly 혼잡제어 기법에서 Square-root 혼잡회피 알고리즘을 제안하였으며, NS를 사용하여 제안한 TEAR의 성능을 실험하였다. 실험을 통하여 제안한 기법의 적용으로 TEAR의 안정성을 개선함을 알 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.5A
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• pp.387-393
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• 2007

In this paper, we propose the load control procedure for WiBro system in order to keep the data traffic throughput maximum. The transmitter at a mobile terminal can estimate maximum available power for each user and then the maximum number of subchannels can be calculated considering the total available power. The data traffic throughput and the total throughput (the sum of signaling traffic and data traffic throughput) are considered. As the number of bandwidth requests per frame increases, the data traffic throughput can significantly decrease. Therefore, the load control procedure is indispensible to maintain the data throughput at the maximum level. So, we propose the load control procedure to prevent data traffic throughput from decreasing and evaluate the proposed procedure through the computer simulation under the multi-user environment. The maximum throughput can be maintained by applying the proposed procedure.

• The KIPS Transactions:PartC
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• v.11C no.1
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• pp.89-98
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• 2004

AQM (Active Queue Management) techniques such as RED (Random Early Detection) which be proposed to solve the congestion of internet perform congestion control effectively for TCP data. However, in the situation where TCP and UDP share the bottleneck link, they can not solve the problems of the unfairness and long queueing delay. In this paper, we proposed an simple queue management algorithm, called PSRED (Protocol Sensitive RED), that improves fairness and decreases queueing delay. PSRED algorithm improves fairness and decreases average queue length by distinguishes each type of flow in using protocol field of packets and applies different drop functions to them respectively.

• Journal of Korea Multimedia Society
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• v.8 no.8
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• pp.1099-1107
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• 2005

The Guaranteed Frame Rate(GFR) service has been designed to accomodate non-real-time applications, such as TCP/IP based traffic in ATM networks. The GFR service not only guarantees a minimum throughput at the frame level, but also supports a fair share of available resources. In this paper, we propose a cell scheduling scheme which can improve the fairness and the goodput through the traffic control in GFR service. For the evaluation of the proposed scheme, we compare the proposed scheme with the existing scheme in the fairness and the goodput. Simulation results show that proposed scheme can improve the fairness and goodput comparing with the existing buffer management scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.4
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• pp.782-792
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• 1997

Asynchronous transfer mode (ATM) provides the means to support with variety of multimedia serivices in broadband network. For the support of multimedia services, various data traffic with priortes should be controlled errectively and congestion function is required in the network to carry out the control operation. To accomplish this in a flexible and effective manner, priorty classes for the different services are commonly used. In this paper, we suggested a buffer managment algorithm in ATM network to improve overall network performance with separated buffer which consists of existing buffer with threshold andauxiliary buffer whose input consists of superposition of voice and data traffic. To evaluate the proposed buffer managment algorithms, simulations are excuted with two proiorities and the results are provied that network throughput is improved better than the existing parital buffer method.

• Journal of KIISE:Information Networking
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• v.30 no.3
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• pp.437-447
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• 2003

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) algorithm. However, RED algorithm simple but does not protect traffic from high-bandwidth flows, which include not only flows that fail to use end-to-end congestion control such as UDP flow, but also short round-trip time TCP flows. In this paper, in order to solve this problem, we propose a simple fairness queue management scheme, called AFQM(Approximate Fair Queue Management) algorithm, that discriminate against the flows which submit more packets/sec than is allowed by their fair share. By doing this, the scheme aims to approximate the fair queueing policy Since it is a small overhead and easy to implement, AFQM algorithm controls unresponsive or misbehaving flows with a minimum overhead.

• The KIPS Transactions:PartC
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• v.9C no.1
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• pp.39-44
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• 2002

In this paper, in order to improve the performance of TCP short traffic application services in wireless Internet environments, the Split-ACKs (SPACK) scheme is proposed. In wireless networks, unlike wired networks, packet losses will occur more often due to high bit error rates. Therefore, each packet loss over wireless lints results in congestion control procedure of TCP being invoked at the source. This causes severe end-to-end Performance degradation of TCP. In this paper, to alleviate the TCP Performance, the SPACK method, split acknowledgement Packets in the base station, is proposed. Using computer simulation, the performance of TCP using SPACK is analysed and shows better performance than traditional TCP Protocol.

• The KIPS Transactions:PartA
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• v.10A no.5
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• pp.487-492
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• 2003

To satisfy the requirements for QoS of Users using multimedia content stream service, it is required to control mechanism for QoS based on class priority, URFA classifies the user by two classes (super class, base class) and controls the admission ratio of user's requests by user's class information. URFA increases the admission ratio class and utilization ratio of stream server resources.

• Journal of the Korea Society of Computer and Information
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• v.8 no.3
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• pp.144-149
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• 2003

Encryption/decryption delay is minimized, because there are constraints in transporting a multimedia data through the Internet. Therefore, encryption algorithm is needed which is changed with considering network traffic and load. And during many users participate in the same multimedia service, an user who already left the service can receive and the method which decrypt the RTP payload is needed because of knowing the encryption key. Therefore in this paper, Security Control Protocol for RTP is designed and implemented for changing the encryption algorithm and the key.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.1A
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• pp.132-141
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• 2000

In this paper, we proposed the MAC protocol of APON supporting multi-class traffic such as CBBUVBR, ABR, UBR, to guarantee the required QoS of each service. For this, we analyze the performance of variousrequest mechanisms and employee the different request mechanism for each traffic classes. Upstream anddownstream frame structures to minimize transmission overhead are proposed based on our request mechanism.The proposed MAC protocol applies the different priority to permit distribution process. CBBWBR traffic, withthe stringent requirements on CDV or delay, is allocated prior to any other class. ABR traffic, which hasnon-strict CDV or delay criteria, uses flexibly the available bandwidth but ensures a minimum cell rate (MCR).UBR traffic is allocated with lowest priority for the remaining capacity. The performance of proposed protocol isevaluated in terms of transfer delay and 1-point CDV with various offered load. The result of simulation showsthat the proposed protocol guarantees the required QoS of the corresponding category, while making use of theavailable resources in both an efficient and dynamical way.