• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.95-104
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• 2002

The interworking for voice service between next generation mobile network and traditional network can be deployed better flexible and expansible network conditions with providing efficiency and economy of network at the same time. So it is essential to develop the interworking strategies together with a evolved network and traditional network. This paper describes a design and verification of internetworking protocol for CC of next generation mobile network and SIP of IP network for applying interworking technology to next generation mobile network, which can harmoniously expropraise voice service from traditional network.

• Proceedings of the KIEE Conference
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• 1988.07a
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• pp.514-518
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• 1988

The switching system and transport will be developed and serve as hybrid switching system and link respectively according to the needs of mixed voice and data service in ISDN era. This paper describes a theory of optimal band width allocation in multiplexing voice and nonvoice traffic, and analyzes traffic performances on a model network.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.795-797
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• 2009

Voice over Internet Protocol(VoIP) service becomes more and more popular nowadays. As such, it is developed over many kinds of network models, especially wireless networks. Mean Opinion Score(MOS) computes the QoS of VoIP service which should be supported by robust network environment. However, MANET is not stable enough to supply high MOS values for VoIP service. In this paper, VoIP service over MANET is simulated using ns-2(Network Simulation 2). In oder to get different MOS values in the results, we differentiate between network environments by adjusting the parameters of MANET.Through comparing the results we can know how to improve the QoS.

• Speech Sciences
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• v.14 no.1
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• pp.151-162
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• 2007

This study focuses on the classification of pathological voice using GMM (Gaussian Mixture Model) and compares the results to the previous work which was done by ANN (Artificial Neural Network). Speech data from normal people and patients were collected, then diagnosed and classified into two different categories. Six characteristic parameters (Jitter, Shimmer, NHR, SPI, APQ and RAP) were chosen. Then the classification method based on the artificial neural network and Gaussian mixture method was employed to discriminate the data into normal and pathological speech. The GMM method attained 98.4% average correct classification rate with training data and 95.2% average correct classification rate with test data. The different mixture number (3 to 15) of GMM was used in order to obtain an optimal condition for classification. We also compared the average classification rate based on GMM, ANN and HMM. The proper number of mixtures on Gaussian model needs to be investigated in our future work.

• Journal of the Korean Society for Industrial and Applied Mathematics
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• v.3 no.2
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• pp.69-80
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• 1999

Recently in telecommunication, BISDN ( Broadband Integrated Service Digital Network ) has received considerable attention for its capability of providing a common interface for future communication needs including voice, data and video. Since all information in BISDN are statistically multiplexed and are transported in high speed by means of discrete units of 53-octet ATM ( asynchronous Transfer Mode ) cells, appropriate traffic control needs. For traffic shaping of voice, the output cell discarding scheme has been proposed. We analyze the scheme with a MMPP/$D_1$, $D_2$/1/B queueing system to obtain performance measures such as loss probability and waiting time distribution.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2E
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• pp.98-104
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• 2002

The present study describes a combination method to recognize the human affective states such as anger, happiness, sadness, or surprise. For this, we extracted emotional features from voice signals and facial expressions, and then trained them to recognize emotional states using hidden Markov model (HMM) and neural network (NN). For voices, we used prosodic parameters such as pitch signals, energy, and their derivatives, which were then trained by HMM for recognition. For facial expressions, on the other hands, we used feature parameters extracted from thermal and visible images, and these feature parameters were then trained by NN for recognition. The recognition rates for the combined parameters obtained from voice and facial expressions showed better performance than any of two isolated sets of parameters. The simulation results were also compared with human questionnaire results.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.298-300
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• 2005

There are several types of communication method which is used for providing the supply of electric power stably. But the communication methods used in KEPCO have weak points in the viewpoint of economy, technology and management. Therefore the power plant located in the island could not be provided the communication service because of above reasons. Because the Satellite communication systems have competitive power in price and technology nowdays, the utility could provide the communication service in even back land. This paper presents some of design efforts for the satellite communication systems as the voice network in the island.

• ETRI Journal
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• v.28 no.3
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• pp.311-319
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• 2006

This paper presents a terminal-assisted frame-based packet reservation multiple access (TAF-PRMA) protocol, which optimizes random access control between heterogeneous traffic aiming at more efficient voice/data integrated services in dynamic reservation TDMA-based broadband access networks. In order to achieve a differentiated quality-of-service (QoS) guarantee for individual service plus maximal system resource utilization, TAF-PRMA independently controls the random access parameters such as the lengths of the access regions dedicated to respective service traffic and the corresponding permission probabilities, on a frame-by-frame basis. In addition, we have adopted a terminal-assisted random access mechanism where the voice terminal readjusts a global permission probability from the central controller in order to handle the 'fair access' issue resulting from distributed queuing problems inherent in the access network. Our extensive simulation results indicate that TAF-PRMA achieves significant improvements in terms of voice capacity, delay, and fairness over most of the existing medium access control (MAC) schemes for integrated services.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.3
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• pp.277-287
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• 1992

In this paper, we study the performance eualuations of the FDDI Network, by mathmeticlal analysis and simulation, in which the Integrated Voice/Data transmission system with voice traffic in synchronous mode and data traffic inasynchronous mode.For the mean waiting times of Voice/Data packet, we use two-state of Marcov models for voice traffic with talkspurt and silenci state, and the data traffic would traffic would transmit at the silence state of voice traffic. By the mean wating times, we analyze the relations between synchronous and asynchronous mode. As a result, using Sync/Async mode with voice and data, voice was not under influnece of data traffic. and in the same time,data can be tanaxmitted in a short waiting time, too.

• The KIPS Transactions:PartD
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• v.15D no.1
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• pp.131-138
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• 2008

This paper describes the implementations of the first application service based on embedded Linux; Mobile Teller which uses WiBro network for data communications and CDMA network for voice communications. Currently, with the appearance of WiBro service, dual-mode terminals which support two heterogeneous networks are available. But, the development of applications which effectively use these networks for providing better service to user is rarely prepared. At Mobile Teller, when a sender on a dual-mode terminal types texts, the texts are transmitted to a TTS server located in the Internet through WiBro network. Subsequently, the TTS server converts the texts into voices and transmits the voice data to the dual-mode terminal. At last the dual-mode terminal sends the voice to the receiver through the CDMA network. In case of noisy environment or when a user has difficulty in speaking, Mobile Teller makes voice communication possible