• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.9
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• pp.373-383
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• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• ETRI Journal
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• v.26 no.6
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• pp.622-634
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• 2004

In this paper, we discuss the design problem and the robustness of space-frequency trellis codes (SFTCs) for multiple input multiple output, orthogonal frequency division multiplexing (MIMO-OFDM) systems. We find that the channel constructed by the consecutive subcarriers of an OFDM block is a correlated fading channel with the regular correlation function of the number and time delay of the multipaths. By introducing the first-order auto-regressive model, we decompose the correlated fading channel into two independent components: a slow fading channel and a fast fading channel. Therefore, the design problem of SFTCs is converted into the joint design in both slow fading and fast fading channels. We present an improved design criterion for SFTCs. We also show that the SFTCs designed according to our criterion are robust against the multipath time delays. Simulation results are provided to confirm our theoretic analysis.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.5
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• pp.1108-1130
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• 2013

Wireless Body Area Networks (WBANs) are becoming increasingly important to solve the issue of health care. IEEE 802.15.6 is a wireless communication standard for WBANs, aiming to provide a real-time and continuous monitoring. In this paper, we present our development of a modified Markov Chain model and a backoff model, in which most features such as user priorities, contention windows, modulation and coding schemes (MCSs), and frozen states are taken into account. Then we calculate the normalized throughput and average access delay of IEEE 802.15.6 networks under saturation and ideal channel conditions. We make an evaluation of network performances by comparing with IEEE 802.15.4 and the results validate that IEEE 802.15.6 networks can provide high quality of service (QoS) for nodes with high priorities.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1996.06a
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• pp.55-59
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• 1996

In this paper, we first analyze the influence of interference due to the time variation and delay spread of the mobile channel on an orthogonal frequency division multiplexing (OFDM) system. With the result, we obtain the bit error rate performance of the 16-QAM OFDM system. Second, we investigate the performance of the Reed-Solomon (RS) coded 16-QAM OFDM system when the number of subcarriers varies. In the investigation, we assume that the information transmission rate and the total bandwidth expansion due to coding, guard interval, and the number of subcarriers are fixed. Under this condition, it is observed that there are optimum numbers of subcarriers that minimize the post decoding symbol error probability of RS code for various channel states.

• Journal of the Korea Computer Industry Society
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• v.7 no.5
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• pp.513-520
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• 2006

In this paper, we propose a novel real-time frame-layer rate control algorithm using sliding window method for low bit rate video coding. The proposed rate control method performs bit allocation at the frame level to minimize the average distortion over an entire sequence as well as variations in distortion between frames. A new frame-layer rate-distortion model is derived, and a non-iterative optimization method is used for low computational complexity. In order to reduce the quality fluctuation, we use a sliding window scheme which does not require the pre-analysis process. Therefore, the proposed algorithm does not produce time delay from encoding, and is suitable for real-time low-complexity video encoder. Experimental results indicate that the proposed control method provides better visual and PSNR performance than the existing TMN8 rate control method.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.9
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• pp.4272-4286
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• 2016

In resource constrained sensor networks, usage of efficient routing protocols can have significant impact on energy dissipation. To save energy, we propose an energy efficient routing protocol. In our approach, which integrates clustering and routing in sensor networks, we perform network coding during data routing in order to achieve additional power savings in the cluster head nodes. Efficacy of the proposed method in terms of the throughput and end-to-end delay is demonstrated through simulation results. Significant network lifetime is also achieved as compared with other techniques.

• Proceedings of the KAIS Fall Conference
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• 2010.11a
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• pp.438-442
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• 2010

무선 센서 네트워크 기술이 발전됨에 따라 소규모 무선 센서 네트워크에서 대규모 무선 센서 네트워크로 변하고 있으며, 이로 인하여 대규모 무선 센서 네트워크를 효율적으로 관리하기 위하여 여러 연구가 진행되고 있다. 본 논문에서는 대규모 무선 센서 네트워크를 효율적으로 관리하는 클러스터 기법을 사용한다. 또한 음성 정보를 전송하기 위해 네트워크 코딩 기법을 사용하여 수집된 자료를 목표지점까지 전달하는데 걸리는 딜레이 시간을 줄이는 방법을 제안한다.

• Journal of the Korean Operations Research and Management Science Society
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• v.30 no.2
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• pp.29-40
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• 2005

In this research, we propose packet scheduling algorithm considering different QoS characteristics of real-time traffic and non-real-time traffic in the next generation wireless communication system serving the multimedia traffic and a new efficient framing method cooperated with this packet scheduler. When the selected traffic classes of the selected users are transmitted, our proposed framing method can increase the number of serviced traffic classes by mixing the many different traffic classes within one frame considering data rate decided by the allocated AMC (Adaptive Modulation and Coding) option. Using this proposed method, the fairness among the traffic classes can be achieved and the system performance for total throughput and delay can be enhanced. Simulations are performed to analyze the performance of the proposed framing method. Our proposed packet scheduler and framing method will be applied to the next generation multimedia wireless communication system serving many traffic classes.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.6
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• pp.1089-1093
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• 2001

A H$\infty$ control theory is applied to the design problem of synthesis filters in a mutirate filter bank. We select a desired pure time-delay system as reference model, and then consider the error system between the mutirate filter bank and the reference model. 1'he synthesis filters minimize the ι$_2$-induced norm of the error system.

• Journal of Korea Multimedia Society
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• v.10 no.12
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• pp.1551-1558
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• 2007

The standard of evaluating function of speech coder for mobile telecommunication can be shown in channel capacity, noise immunity, encryption, complexity and encoding delay largely. This study is an algorithm to reduce complexity applying to CDMA(Code Division Multiple Access) mobile telecommunication system, which has a benefit of keeping the existing advantage of telecommunication quality and low transmission rate. This paper has an objective to reduce the computing complexity by controlling the frequency band nonuniform during the changing process of LSP(Line Spectrum Pairs) parameters from LPC(Line Predictive Coding) coefficients used for EVRC(Enhanced Variable-Rate Coder, IS-127) speech coders. Its experimental result showed that when comparing the speech coder applied by the proposed algorithm with the existing EVRC speech coder, it's decreased by 45% at average. Also, the values of LSP parameters, Synthetic speech signal and Spectrogram test result were obtained same as the existing method.