• Proceedings of the Korea Society for Simulation Conference
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• 2001.05a
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• pp.134-134
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• 2001

이동통신 망을 통한 멀티미디어 통신의 수요 급증으로, 차세대 이동통신 시스템에서는 패킷 교환에 기초한 망 구조가 사용될 것으로 예측된다. VOD(Voice Activity Detector)를 갖는 음성 단말은 데이터를 발생시키는 talk spurt(평균이 t$_1$인 지수분포를 따름)와 데이터를 발생시키지 않는 silence period(평균이 t$_2$인 지수분포를 따름)의 두가지 상태를 갖는 마코프 체인으로 모델링된다. Goodman at. al.은 음성 단말들이 talk spurt동안만 데이터를 전송하게 함으로써 더 많은 가입자들을 수용할 수 있는 PRMA(Packet Reservation Multiple Access) 기법을 제안되었다. PRMA 방식에서는 시간 축이 슬롯들로 구성되며 여러개의 슬롯들로 프레임이 형성된다. Silence period 상태에 있던 음성 단말은 talk spurt 상태가 되면 talk spurt의 첫 번째 데이터를 하나의 슬롯을 통해 전송하게 된다. 이때 단말들은 각 슬롯에서 데이터를 전송할 수 있는 확률을 나타내는 채널 접근 확률(channel access probability)에 의해 데이터를 전송하게 되며 전송에 성공하면 슬롯을 예약함으로서 다음 프레임부터는 동일한 위치의 슬롯을 통해 데이터들을 전송하게 된다. DS/CDMA(Direct Sequence/code Division Multiple Access)는 이동통신 단말의 수용 용량상의 이점, 소프트 핸드오버 능력, 보다 용이하게 셀 계획을 세울 수 있는 점 등에 의해 차세대 이동통신 망에서 채택될 예정이다. CDMA 시스템은 간섭(interference)에 의해 용량이 제한을 받게 되며, MAI(Multiple Access Interference)가 시스템의 성능에 많은 영향을 미치게 된다. Brand, et. al.은 간섭의 분산을 줄이기 위해 PRMA 개념을 DS/CDMA 환경으로 확장한 Joint CDMA/PRMA 프로토콜을 제안하였다. 이때 각 슬롯에서의 데이터 전송확률을 그 슬롯에서 예약상태에 있는 음성 단말의 수에 의존하게 하는 방식을 사용하였으며 데이터 전송확률을 나타내는 채널 접근 확률들을 시뮬레이션을 통해 유도하였다. 한편 음성 단말에게는 실시간 서비스를 제공해 주어야 하는 대신 데이터 단말에게는 실시간 서비스를 제공해 주지 않아도 되므로, 트래픽이 많을 때에는 음성 단말의 데이터 전송에 우선권을 주는 것이 바람직하다. 이를 위해서 Brand, et. al.은 채널 접근 확률을 각 슬롯의 트래픽 상태에 따라 적응적으로 산출하는 기법을 제안하였다. 본 연구에서는 Joint CDMA/PRMA의 성능이 채널 접근 함수의 효율성에 많이 의존하게 되므로 보다 효율적인 채널 접근 확률을 구하는 방법을 제안한다. 즉 채널 액세스 확률을 각 슬롯에서 예약상태에 있는 음성 단말의 수뿐만 아니라 각 슬롯에서 예약을 하려고 하는 단말의 수에 기초하여 산출하는 방법을 제안하고 이의 성능을 분석하였다. 시뮬레이션에 의해 새로 제안된 채널 허용 확률을 산출하는 방식의 성능을 비교한 결과 기존에 제안된 방법들보다 상당한 성능의 향상을 볼 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.7
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• pp.553-563
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• 2013

When the IEEE 802.11 wireless LAN-based broadcasting services, we use broadcast packets to broadcast multimedia contents to a large number of users using limited wireless resources. However, broadcast transmission is difficult to recover the loss packets compared with unicast transmission. Therefore, analysis of packet loss characteristics is required to perform efficient packet recovery. The packet loss in wireless transmissions is often bursty with high loss data rate. Even if loss patterns have the same average packet loss, they are different in the recovery rate of random loss and burst loss depending on the nature. Therefore, the analysis and research of the nature of the loss are needed to recover loss packets considering bursty characteristics. In this paper, we experimented Wi-Fi broadcast transmission according to transmission rate and analyzed bursty characteristics of loss patterns using 4-state markov model.

• Journal of KIISE:Information Networking
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• v.30 no.3
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• pp.316-328
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• 2003

In this paper, we propose and analyze two adaptive channel allocation schemes for supporting multimedia traffics in hierarchical cellular systems. It is guaranteed to satisfy the required quality of service of multimedia traffics according to their characteristics such as a mobile velocity for voice calls and a delay tolerance for multimedia calls. In the scheme 1, only slow-speed voice calls are allowed to overflow from macrocell to microcell and only adaptive multimedia calls can overflow from microcell to macrocell after reducing its bandwidth to the minimum channel bandwidth. In the scheme II, in addition to the first scheme, non-adaptive multimedia calls can occupy the required channel bandwidth through reducing the channel bandwidth of adaptive multimedia calls. The proposed scheme I is analyzed using 2-dimensional Markov model. Through computer simulations, the analysis model and the proposed schemes are compared with the fixed system and two previous studies. In the simulation result, it is shown that the proposed schemes yield a significant improvement in terms of the forced termination probability of handoff calls and the efficiency of channel usage.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.18 no.5
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• pp.112-125
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• 2019

An acoustic signal-based, tunnel-incident detection system was developed and evaluated. The system was comprised of three components: algorithm, acoustic signal collector, and server system. The algorithm, which was based on nonnegative tensor factorization and a hidden Markov model, processes the acoustic signals to attenuate noise and detect incident-related signals. The acoustic signal collector gathers the tunnel sounds, digitalizes them, and transmits the digitalized acoustic signals to the center server. The server system issues an alert once the algorithm identifies an incident. The performance of the system was evaluated thoroughly in two steps: first, in a controlled tunnel environment using the recorded incident sounds, and second, in an uncontrolled tunnel environment using real-world incident sounds. As a result, the detection rates ranged from 80 to 95% at distances from 50 to 10 m in the controlled environment, and 94 % in the uncontrolled environment. The superiority of the developed system to the existing video image and loop detector-based systems lies in its instantaneous detection capability with less than 2 s.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.12
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• pp.1919-1927
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• 1993

A new MAC protocol has been proposed and analysed to relieve the unfairness problems exhibited by the basic version of the DQDB standard. DQDB MAC protocol has the unfairness problems in throughputs. message delay and so or. And when the slots are reused or the file transmissions takes long, the unfairness problems in the system become worse. The new access protocol proposed here, which of called the Balanced DQDB, guarantees a fair bandwidth distribution by using one bit of the dual bus network protocol and keeps up all characteristics of DQDB. the DQDB analysis model introduced by Wen Jing, et al, was considered to analyse a sequential balance distribution of solts. And the probabilities of the empty in operation mode were represented to determine the probabilities for busy bits to generate on each node of the bus using the Markov chain. Through the simulations. the performances of the proposed Balanced DQDB and that of the standard DQDB of the BWB mechanism were compared at the state that the values of the RQ or CD counter on each node varied dynamically. As the results, it is shown that the Balanced DQDB has the decrement of throughputs in upstream, but the numbers of the used empty slots at each node of the Balanced DQDB had more than that of the others because the Balanced DQDB has over 0.9 throughputs in the 70~80% nodes of total node and it has constant throughputs at each node. And there results were analogous to that of the analytical model.

• Journal of KIISE:Information Networking
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• v.32 no.4
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• pp.515-524
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• 2005

As the number of the mobile nodes (MNs) increases in the networks, the signaling traffic generated by mobility management for MNs will increase explosively, and such a phenomenon will probably affect overall network performance. In this paper, we propose a novel analytical approach using a continuous-time Markov chain model and hierarchical network model for the analysis on the signaling load of representative IPv6 mobility support Protocols such as Mobile IPv6 (MIPv6) and Hierarchical Mobile IPv6 (HMIPv6). According to these analytical modeling, this paper derives the various signaling costs, which are generated by an MN during its average domain residence time when MIPv6 and HMIPv6 are deployed under the same network architecture, respectively. In addition, based on these derived costs, we investigate the effects of various mobility/traffic-related parameters on the signaling costs generated by an MN under MIPv6 and HMIPv6. The analytical results show that as the average moving speed of an MN gets higher and the binding lifetime is set . to the larger value, and as its average packet arrival rate gets lower, the total signaling cost generated during its average domain residence time under HMIPv6 will get relatively lower than that under MIPv6, and that under the reverse conditions, the total signaling cost under MIPv6 will get relatively lower than that under HMIPv6.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.552-558
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• 2002

Recently, based on the human speech recognition (HSR) model of Fletcher, the multi-band speech recognition has been intensively studied by many researchers. As a new automatic speech recognition (ASR) technique, the multi-band speech recognition splits the frequency domain into several sub-bands and recognizes each sub-band independently. The likelihood scores of sub-bands are weighted according to reliabilities of sub-bands and re-combined to make a final decision. This approach is known to be robust under noisy environments. When the noise is stationary a sub-band SNR can be estimated using the noise information in non-speech interval. However, if the noise is non-stationary it is not feasible to obtain the sub-band SNR. This paper proposes the inverse sub-band distance (ISD) weighting, where a distance of each sub-band is calculated by a stochastic matching of input feature vectors and hidden Markov models. The inverse distance is used as a sub-band weight. Experiments on 1500∼1800㎐ band-limited white noise and classical guitar sound revealed that the proposed method could represent the sub-band reliability effectively and improve the performance under both stationary and non-stationary band-limited noise environments.

• The Journal of the Acoustical Society of Korea
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• v.35 no.1
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• pp.74-82
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• 2016

In this paper, a new automatic speech recognition (ASR) was proposed where ultrasonic doppler signals were used, instead of conventional speech signals. The proposed method has the advantages over the conventional speech/non-speech-based ASR including robustness against acoustic noises and user comfortability associated with usage of the non-contact sensor. In the method proposed herein, 40 kHz ultrasonic signal was radiated toward to the mouth and the reflected ultrasonic signals were then received. Frequency shift caused by the doppler effects was used to implement ASR. The proposed method employed multi-channel ultrasonic signals acquired from the various locations, which is different from the previous method where single channel ultrasonic signal was employed. The PCA(Principal Component Analysis) coefficients were used as the features of ASR in which hidden markov model (HMM) with left-right model was adopted. To verify the feasibility of the proposed ASR, the speech recognition experiment was carried out the 60 Korean isolated words obtained from the six speakers. Moreover, the experiment results showed that the overall word recognition rates were comparable with the conventional speech-based ASR methods and the performance of the proposed method was superior to the conventional signal channel ASR method. Especially, the average recognition rate of 90 % was maintained under the noise environments.

• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.581-594
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• 2003

Recently, a location management is being more important in mobile communication systems due to an explosive increase of mobile users. Current systems have used a concept of location area. Based on this concept, a mobile user performs a location update whenever it moves into a new location area. However, this scheme can not avoid unnecessary location updates when a mobile user moves around with high movement rate as compared to call arrival rate. That results in tremendous location management cost. To overcome this drawback, our proposal divides service areas into two sets: One is a set of areas that mobile users move with high speed and another is a set of areas that they move with low speed. After establishing these two sets, this paper employs different location tracking schemes for each sets. Generally, most mobile users with high speed have a low CMR and a regular direction until they arrive at their destination. Using such the moving behavior, systems can predict a mobile user's next location area in advance. When the mobile user moves into the predicted location, our proposal does not perform a location update. Thus, it can reduce overall location management cost. The Markov model is used to analyze the performance of our proposal. Using the model, this paper compares our proposal with IS-41 and TLA. The analytic results show that as CMR grows lower, an overall cost of our proposal becomes less, particularly if a mobile user frequently moves into the specific location are predicted by mobile systems. Also, our proposal has a better performance than other two schemes when the communication cost between HLR and VLR is high.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.9
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• pp.25-37
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• 2007

The existing route re-establishment methods which intend to extend the lifetime of the network attempt to find new routes in order not to overly consume energy of certain nodes. These methods outperform other routing algorithms in the network lifetime extension aspect because they try to consume energy evenly for the entire network. However, these algorithms involve heavy signaling overheads because they find new routes based on the flooding method and route re-acquisition occurs often. Because of the overhead they often can not achieve the level of performance they intend to. In this paper, we propose a new route re-acquisition algorithm ARROW which takes into account the cost involved in the packet transmission and the route re-acquisition. Since the proposed algorithm considers future route re-acquisition costs when it first finds the route, it spends less energy to transmit given amount of data while evenly consuming the energy as much as possible. Using 2-dimensional Markov Chain model, we compare the performance of the proposed algorithm and that of other algorithms. Analysis results show that the proposed algorithm outperforms the existing route re-acquisition methods in the signaling overhead and network lifetime aspects.