• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.93-101
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• 2002

This paper presents the Internet audio broadcasting technology based on the streaming technology. In this paper, we choose the MPEG-2 AAC for multimedia data, and for the streaming of this data we use RTP/RTCP protocol. We use RTSP protocol for the control of streaming data and TCP/IP for the exchange of information between server and client. By using all of these protocols and MPEBG-2 AAC, we explain the implementation method for the unicast/multicast streaming server/client system. Our system was tested by ETRI intranet, which is connected by 2000 researchers. Experimental result show that our system can be process the packet loss and jitter by retransmission and variable length buffer. Multicast streaming server can be used for the audio broadcasting service inside the company, unicast streaming server can be used for the AOD (Audio On Demand) service.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.10 s.352
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• pp.79-89
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• 2006

TCP is the most widely used transport protocol in Internet applications that guarantees a reliable data transfer. But, in the wireless multi-hop networks, TCP performance is degraded because it is designed for wired networks. The main reasons of TCP performance degradation are contention for wireless medium at the MAC layer, hidden terminal problem, exposed terminal problem, packet losses in the link layer, unfairness problem, reordering problem caused by path disconnection, bandwidth waste caused by exponential backoff of retransmission timer due to node's mobility and so on. Specially, in the mobile ad-hoc networks, discrepancy between a station's transmission range and interference range produces hidden terminal problem that decreases TCP performance greatly by limiting simultaneous transmission at a time. In this paper, we propose a new MAC algorithm for mobile ad-hoc networks to solve the problem that a node can not transmit and just increase CW by hidden terminal. In the IEEE 802.11 MAC DCF, a node increases CW exponentially when it fails to transmit, but the proposed algorithm, changes CW adaptively according to the reason of failure so we get a TCP performance enhancement. We show by ns-2 simulation that the proposed algorithm enhances the TCP performance by fairly distributing the transmission opportunity to the failed nodes by hidden terminal problems.

• Journal of KIISE:Information Networking
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• v.29 no.5
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• pp.543-552
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• 2002

Wireless Session Protocol(WSP) which was updated and supplemented based on HyperText Transfer Protocol(HTTP) was designed by Wireless Application Protocol(WAP) forum regarding the characteristics of wireless environment. WSP improved the performance in wireless network, and introduced various facilities considering wireless environment. In this paper, we more improve the performance of WSP adding protocol message compression capability; we cail improved WSP protocol as WSP+. And, we analysis the performance of each protocol with WSP and WSP+ implementation. As a result of experiment, the capability which proposed in this paper reduced a response traffic about 45%. In $10^{-4}$ bit error rate, we also found the packet loss rate and time delay per transaction of WSP+ was improved over 40%. Finally, we found that the protocol message compression capability reduces message retransmission count in transaction layer and shorten the delay time per transaction by reducing a message size.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4A
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• pp.385-392
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• 2006

It is widely accepted that high user densities in mobile multimedia environments can only be achieved with micro- and pico-cell. The smaller cell size causes frequent handovers between cells, and decreases permissible handover processing delay. This may result in the handover failure, in addition to the loss of some packets during the handover. In these cases, re-transmission is needed in order to compensate errors, which in turn triggers a rapid degradation of throughput. In this paper, we propose a new handover scheme for the next generation mobile communication systems, in which the handover setup process is done in advance before a handover request by predicting the handover cell based on mobile terminal's current position and moving direction. Simulation is performed in order to analyze the handover failure rate and packet loss rate. Simulation results show that our proposed method provides a better performance than the conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3B
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• pp.72-84
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• 2005

Optical burst switching networks usually employ one-way reservation by sending a burst control packet with a specific offset time, before transmitting each data burst Same. Due to such a Property, burst-contentions occur when multiple bursts contend for the same wavelength in the same output link simultaneously in a node, leading to burst losses, eventually degrading the quality of service. Therefore, in this paper, we propose a dynamic routing algorithm using minimum local state information in order to decrease burst-contentions. In this proposed scheme, if burst loss rate exceeds a threshold value at a certain node, a new alternative routing path is chosen according to burst priority and location of burst generation, which enables the contending bursts to detour around the congested link. Moreover, for reducing the effect of sending bursts on the primary path due to the alternative path, we also apply a minimum path-cost based routing on link-cost concept. Our simulation results show that proposed scheme improves the network performance in terms of burst loss probability and throughput by comparing with conventional one.

• Journal of the Korea Society of Computer and Information
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• v.24 no.11
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• pp.11-19
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• 2019

Recently, hybrid CDN/P2P video streaming architecture is specially designed and deployed to achieve the scalability of P2P networks and the desired low delay and high throughput of CDNs. In this paper, we propose a cooperative video streaming and active node buffer management technique in hybrid CDN/P2P architecture. The key idea of this streaming strategy is to minimize network latency such as jitter and packet loss and to maximize the QoS(quality of service) by effectively and efficiently utilizing the information sharing of file location in CDN's proxy server which is an end node located close to a user and P2P network. Through simulation, we show that the proposed cooperative video streaming and active node buffer management technique based on CDN and P2P network improves the performance of realtime video streaming compared to previous methods.

• 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.

• Journal of Advanced Navigation Technology
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• v.11 no.1
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• pp.31-36
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• 2007

TCP (Transmission Control Protocol) is one of the protocols which are widely used from the Internet environments. Through the flow control of TCP, it could be increased efficiency for the loss and a re-transmission of data and the flow control become accomplished through window technique which puts the limit of size. By the flow control, TCP divided in various versions. In this paper, it is analyzed the simulation result which applies the error model in the Newreno which is an improved model of the representative Tahoe, Reno.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.118-120
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• 2003

이동 컴퓨팅 환경이란 사용자가 어떤 장소로 이동하더라도 동일한 환경에서 네트워크 서비스를 지속적으로 제공받을 수 있는 환경을 말하며, Mobile IP는 이러한 특성들을 고려한 무선 네트워크 프로토콜이다. 하지만 이 프로토콜도 이동 노드의 빈번한 핸드오프시 위치 등록과정에서 반드시 홈 에이전트를 거쳐야 하므로 네트워크 자원과 서비스 지연을 초래하는 단점을 가진다. 이를 극복하고자 본 논문에서는 지역적 FA 기법을 도입하여 방문 도메인에서의 등록을 지역화 하여 이동 노드의 빈번한 핸드오프시 등록과정에 대한 지연 및 네트워크 부하를 줄일 수 있다. 또한 부하분산 요소를 부가하여, 모든 FA가 상황에 따라 동적으로 루트권한 FA가 될 수 있는 기법을 도입하여 중앙으로만 집중되는 등록 메시지를 분할하여 네트워크 자원 낭비를 줄이고 빠른 응답 시간을 얻음 수 있으며 고전적인 문제점인 패킷의 지연 및 손실 그리고 재전송을 예방하고자 한다.

• The Journal of Engineering Research
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• v.6 no.2
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• pp.39-46
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• 2004

In this paper, we carry out a performance study related to the Broadband Network. For this network, it has been proposed to use the leaky bucket as a way of controlling congestion within the network. On the top of leaky bucket type rate based congestion control scheme for high speed networks, a user will typically operate an error control scheme for retransmitting lost and erroneous packets. We propose a performance model in order to study the interaction between a user's error control scheme and the leaky bucket congestion control scheme for high speed networks. Simulation results show that parameters such as the window size and the token generation rate in the leaky bucket are key factors affecting the end-to-end delay.