• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.3
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• pp.275-280
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• 2010

In this paper, VoIP(Voice over Internet Protocol) transmission performance for MANET(Mobile Ad-hoc Networks) is improved and analyzed with packet aggregation scheme which is aggregating some of short length packets to one large packet and sending to networks. VoIP simulator based on NS(Network Simulator)-2 is implemented and used to measure performance of VoIP traffic transmission. In this simulation, VoIP traffics are generated with parameters of some codes such as G.711, G.729A, GSM.AMR and iBLC. MOS(Mean Opinion Score), end-to-end network delay, packet loss rate and transmission bandwidth are measured. Performance improvements of 98% for MOS, 6.4times for end-to-end network delay, 32times for packet loss rate is shown as simulation results. On the other hand, transmission bandwidth is increased about maximum 10%. Finally, VoIP implementation guide for the performance with packet aggregation is suggested.

• Proceedings of the Korea Society for Industrial Systems Conference
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• 1999.12a
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• pp.121-129
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• 1999

When continuous media are transmitted over the communication networks, asynchrony which can not maintain temporal relationships among packets may occur due to a random transit delay. There exist two types of synchronization schemes ; for guaranteed or non-guaranteed resource networks. The former which applies a resource reservation technique maintains delay characteristics, however, the latter supply a best-effort service. In this paper, I propose a intra-media synchronization scheme to transmit continuous media on general networks not guaranteeing a bounded delay tome. The scheme controls transmission times of the packets by estimating next delay time with the delay distribution. So, the arriving packets may be maintained within a limited delay boundary, and playout will be performed after buffering to smoothen small delay variations. The continually increasing delay due to network overload causes buffer underflow at the receiver. To solve it, the transmitter is required to speed up instantaneously. Too much increase of transmission-rate may cause network congestion. At that time, the transmitter drops the current packet when informed excessive delay from the receiver.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.7
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• pp.1659-1666
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• 1998

The multicode CDMA systems that are widely studied as an effective transmission methodology in the IMT-2000 systems, employ orthogonal codes to transform high rate data into parallel, low rate data for simultaneous transmission. In this paper, we propose a new multicode CDMA system which achieves the same data rate and processing gain of the conventional systems, while significantly improves bit error rate performance by exploiting a convolutional code with code rate r=1/2 and a bi-orthogonal code. The simulation results for synchronous systems using maximal ratio combining Rake receivers under additive white Gaussian noise and multi-path fading channels, show significant improviements by the proposed system.

• Journal of Korea Society of Industrial Information Systems
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• v.19 no.1
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• pp.53-60
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• 2014

IEEE 802.15.4 is one of the most widely adopted physical layer standards in the area of LR-WPAN(Low-Rate Wireless Personal Area Network). Numerous previous researches have contributed to deep insights on energy efficiency, transmission throughput, and reliability that IEEE 802.15.4 delivers to the LR-WPAN. As a research that is orthogonal and complementary to previous researches, we explore the implementation and practical performance evaluation of IEEE 802.15.4 MAC software. We implement the MAC software from the perspective of the networking stack, exploring the issues raised when the MAC software serves as a functional component in a complete networking stack consisting of MAC, network as well as well as application support layers. The performance is evaluated on a realistic experimental software environment integrated with operating system, networking stack, and applications.

• Journal of Korea Game Society
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• v.7 no.2
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• pp.33-40
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• 2007

Smoothing is a transmission plan where variable rate video data is converted to a constant bit rate stream. Among them are CBA, MCBA, MVBA and others. However, these smoothing algorithms produce a transmission plan where extra bandwidth in server is not considered. This may cause difficulty in providing videos to many clients in a server. In this paper, we propose the smoothing algorithm with monotonic transmission rate increase and abrupt transmission rate decrease, in order to provide videos to as many clients in server with limited network bandwidth as possible. In order to show the performance of our proposed algorithm, various evaluation factors were used such as the maximum number of clients, the average number of clients, and so on. Experiments demonstrated that the proposed algorithm outperformed other algorithms in evaluation factors such as the maximum number of clients and the average number of clients.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4B
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• pp.323-332
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• 2006

In this paper, we consider a WDM based metro ring connecting broadband access networks by using FT-TR (Fixed Transmitter-Tunable Receiver) type access nodes and a new MAC protocol for fair transmission is proposed. The basic channel access scheme is CSMA/CA(Carrier Sense Multiple Access with Collision Avoidance) and the transferred optical slot is changed to be empty slot by SS(Source-Stripping) which drops a packet from ring-rotated slot. This empty slot can be either used to send head-of-line packet in buffer or transmitted to next access node. In former cases, there is the improvement of network throughput, but latter case leads unfairness problem of transmission. Thus the proposed MAC protocol exploits the advantages form tow cases. It gives downstream access nodes chance to use empty slots and limits the unconditional usage of empty slots at upstream access nodes with probability as called p-Persistent MAC protocol. We use a numerical analysis to evaluate bandwidth efficiency and maximum node throughput and compares simulation results in terms of node throughput, fairness factor, transfer delay depending on probability. Under two different FT-TR type architectures for metro ring network the proposed MAC protocol is evaluated and compared.

• Proceedings of the KIEE Conference
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• 2008.04a
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• pp.83-85
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• 2008

현행 지상파 DMB(Digital Multimedia Broadcasting)에서는 최대 CIF급 해상도의 동영상을 384 Kbps 정도의 비트율로 압축하여 비디오 CD급의 화질을 제공한다. 따라서 큰 화면을 통해 DMB 방송을 시청할 경우 해상도 부족 등으로 인한 화질 열화를 느낄 수 있다. 이와 같은 문제를 해결하기 위한 방안으로 계층적으로 부호화된 비트스트림을 서로 다른 채널로 전송하는 고품질 AT-DMB 시스템이 연구되고 있다. 더불어, 계층변조 기술을 이용하여 추가적인 전송대역폭을 확보하고, 계층적 비디오 부호화 기술을 이용하여 고품질 이동 멀티미디어 방송을 제공하는 고품질 T-DMB 방식이 개발되고 있다. 이와 같은 AT-DMB 방식으로 사용자에게 안정된 고품질 비디오 서비스를 제공하기 위해서는 계층적 비디오 부호화를 고려한 효율적인 AV다중화 및 동기화 구조에 대한 연구가 필수적으로 요구 된다. 본 논문에서는, 이와 같은 고품질 AT-DMB 시스템을 위해서 각 계층 ES(Elementary stream)을 MPEG-4 over MPEG-2 시스템과 Eureka-147 DAB 시스템을 통해 서로 다른 채널로 전송한다. 계층별 전송으로 인한 시간 지연에 대처하여 수신입력부에서는 각각의 계층으로 전송된 ES들이 동기화 되어 복호화 되도록 SVC의 계층적 비트스트림 구조와 적응적으로 동작되는 다중화 시스템구조를 제작하였다.

• Proceedings of the KAIS Fall Conference
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• 2009.05a
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• pp.805-809
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• 2009

본 논문에서는 IEEE 802.11 WLAN의 MAC인 DCF의 성능을 개선하는 알고리즘을 제안하고 이를 해석적으로 분석한다. IEEE 802.11 WLAN의 MAC에서는 데이터 전송을 제어하기 위한 방법으로 DCF와 PCF를 사용하며, DCF의 경우 CSMA/CA를 기반으로 한다. DCF는 경쟁 스테이션이 적은 상황에서는 비교적 우수한 성능을 보이나 경쟁 스테이션의 수가 많은 경우 처리율, 지연 관점에서 성능이 저하되는 문제점이 있다. 본 논문에서는 패킷 전송 후 충돌이 발생하면 윈도우 값을 최대 CW로 증가시키고 연속적으로 c번 패킷전송에 성공하면 CW를 감소함으로써 패킷 충돌 확률을 낮추는 알고리즘을 제안하고 이를 수학적으로 분석한다. 제안하는 알고리즘의 효율성을 입증하기 위해 시뮬레이션을 수행하여 그 타당성을 제시하였다.

• Proceedings of the KAIS Fall Conference
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• 2010.05a
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• pp.467-470
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• 2010

본 논문에서는 IEEE 802.11 WLAN의 MAC인 DCF의 성능을 개선하는 알고리즘을 제안하고 이를 해석적으로 분석한다. IEEE 802.11 WLAN의 MAC에서는 데이터 전송을 제어하기 위한 방법으로 DCF와 PCF를 사용하며, DCF의 경우 CSMA/CA를 기반으로 한다. DCF는 경쟁 스테이션이 적은 상황에서는 비교적 우수한 성능을 보이나 경쟁 스테이션의 수가 많은 경우 처리율, 지연 관점에서 성능이 저하되는 문제점이 있다. 본 논문에서는 패킷 전송 후 충돌이 발생하면 윈도우 값을 최대 CW로 증가시키고 연속적으로 c번 패킷전송에 성공하면 CW를 감소함으로써 패킷 충돌 확률을 낮추는 알고리즘을 제안하고 이를 수학적으로 분석한다. 제안하는 알고리즘의 효율성을 입증하기 위해 시뮬레이션을 수행하여 그 타당성을 제시하였다.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.12 no.5
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• pp.810-816
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• 2001

This paper proposes a Truncated Type-II Hybrid ARQ scheme using an adaptive modulation system to achieve high throughput data transmission systems for mobile communication systems. In this paper, the adaptive modulation system analyzed in Nakagami (m-distribution) fading channel environment. The adaptive modulation system controls the modulation level and symbol rate according to the Nakagami fading parameter(m). When the received $E_bN_0$ is high or the Nakagami fading parameter m is high, the propose system selects higher modulation level and higher symbol rate to increase throughput. On the other hand, this system selects lower modulation level and lower symbol rate to prevent throughput performance degradation when the received $E_bN_0$ is low. The modulation method have been adopted QPSK(Quadrature Phase Shift Keying), 16 QAM(Quadrature Amplitude Modulation), 64 QAM, 256 QAM. Therefore, Adaptive Modulation Systems with Truncated Type-II Hybrid ARQ Scheme is proper for mobile and radio for mobile and radio data communication system that require high reliability and delay-limited applications.