• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.12
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• pp.5843-5848
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• 2011

To improve the efficiency of the channel spectrum and to reduce the power consumption of the system in EVRC, the voice signal is compressed and transmitted only when the user speaks to. In addition to this, voice frames are divided into three rates 1, 1/2 and 1/8 and each frame is handled differently. For example, we assumed that the input is silence region if the 1/8 rate is used. In this paper, the sections are firstly separated into the voiced speech signal region, unvoiced speech signal region, and silence region by using distribution characteristics of LSP parameters. Then the paper suggested to encode 1 rate for the voiced speech signal, 1/2 rate for the unvoiced speech signal region, 1/8 rate for the silence region. In other words, traditional way of transmission is used when sending full rate in the EVRC. However, when sending half rate, the voice is firstly distinguished between voiced and unvoiced. If the voice is distinguished as voiced, voice is converted into full rate before the transmission. If it is distinguished as silence, EVRC's basic rate is applied. In the experimental results with SNR, ASDM, transmission bit rate measurement, we have demonstrated that voice quality was improved by using the proposed algorithm.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.20 no.2
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• pp.750-762
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• 2019

The importance of the call center as a contact point for the enterprise is growing. However, call centers have difficulty with their operating agents due to the agents' lack of knowledge and owing to frequent agent turnover due to downturns in the business, which causes deterioration in the quality of customer service. Therefore, through an N-bank call center case study, we developed a system to reduce the burden of keeping up business knowledge and to improve customer service quality. It is a "real-time agent advisor" system that provides agents with answers to customer questions in real time by combining AI technology for speech recognition, natural language processing, and questions & answers for existing call center information systems, such as a private branch exchange (PBX) and computer telephony integration (CTI). As a result of the case study, we confirmed that the speech recognition system for real-time call analysis and the corpus construction method improves the natural speech processing performance of the query response system. Especially with name entity recognition (NER), the accuracy of the corpus learning improved by 31%. Also, after applying the agent advisor system, the positive feedback rate of agents about the answers from the agent advisor was 93.1%, which proved the system is helpful to the agents.

• Journal of KIISE
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• v.44 no.8
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• pp.749-759
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• 2017

The weapon system based on voice call can cause delay, error or damage to the message during the exchange of information. Furthermore, since the weapon system has a unique message format, it has limited data distribution. Therefore, a Korea Variable Message Format(KVMF) has been developed in this study to utilize a standard sized data format to guarantee the transmission quality and minimize the transmission amount. The ground tactical data link system quickly and accurately shares tactical information by incorporating a field management system that utilizes the KVMF standard message in the mobile weapon system. In this study, we examine the possibility of performing the mission immediately by recovering the state of network joining when a normal/abnormal termination situation of the battlefield management system occurs.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.5
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• pp.1131-1136
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• 2015

VoIP and IP Telephony telephony technology development is a growing by easy to using IP-PBX by using phone from using existing lines rather than the internet. IP-PBX do not use the phone line from phone work for many companies and institutions of management costs reduce as provides similar to regular phone line quality. But IP-PBX to introduce for need to be the initial cost on is should buy for expensive hardware equipment or commercial software. In this paper, suggest way to introduce IP-PBX do not buy expensive hardware equipment or commercial software. Suggest IP-PBX on designed and implement for IP-PBX server using Raspberry Pi and Asterisk. And verification treatise on the suitability of conducted by voice calls based on IP-PBX between PC and a Smartphone

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.8
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• pp.9-16
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• 2007

Currently, the measurement of RF parameters for a base station in operation is typically limited to easily measured forward path items. In this paper, the forward monitoring ports of base stations are used to measure the reverse RF performance. The system has been implemented and effectiveness has been proven on an operating base station. The receiver sensitivity is measured using an internal CDMA modem which is used to monitor the output power based on closed loop power control when the modem is connected to the base station via a voice call. In order to improve accuracy, in addition to the modem Tx adjust(TxAdj) parameter, the detector's actual measurement is used. For accurate receiver sensitivity, the measurement should be made when there is no traffic which is not possible on an operating base station. Therefore, pilot channel chip energy to received signal power spectral density ratio$(E_c/I_o)$ compensation method is used to offset the receiver sensitivity degradation with voice traffic increase.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.5
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• pp.85-90
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• 2015

Work environments within the firm with a concept of mobile office is growing divided into two sections. It's Wi-Fi FMC(Fixed Mobile Convergence) field Which are implemented in a telephone service available from existing fixed-line service in the center of the smart phones and the EMS(Enterprise Mobility Service) field to make people will be able to handle PC the center of the information system within an enterprise using a smart phone instead of terminal facility connects to a system in the workplace and external. This paper developed FMC controller to allow execution IMS(IP Multi-Media Subsystem) services to complement the issues of the FMC corporation, telephony service associated. The controller includes FMC automatic enrollment services, voice quality enhancement of the mobile phone, anywhere within the firm on his mobile phone calls can provide mobility and is also implemented FMC LCR function that use status information from mobile soft-phone within the IP-PBX.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2E
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• pp.32-40
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• 1994

It is the purpose of this paper to describe the methods for establishing loudness ratings and talker echo out of transmission quality of ISDN telephone connected to fully digital network. In order to design the desirable loudness ratings and talker echo for ISDN telephone, the model system of digital speech communication for subjective tests is developed. Using this model system, opinion tests which decide the optimal CODEC input level, the range of overall loudness rating, sidetone masking rating and talker echo are performed. From the results of tests, we decided that the loudness ratings are 6 to 8dB for sending, 0 to 2dB for receiving, and 8 to 12dB for sidetone masking rating. And, the terminal coupling loss of TCLw of at least 40dB is necessary to provide echo-free telephone communications to telophone users when the overall loudness rating of ISDN telephone is normalized to 10dB.