• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.7
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• pp.1346-1363
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• 1994

Packet voice protocols have been realized in many research works. But few studies for the interconnection of LAN and PABX to facilitate the voice communication have been done. In this paper, the gateway to interconnect the Ethernet LAN with the existing PABX telephone network for voice communication has been designed and implemented. The implemented gateway protocol is a modified protocol based on CCITT`s G.764 packetized voice protocol. To accomplish this goal the hardware system has been realized, which is divided into five parts: interface part with the telephone line, voice-processing part, PC interface part, controller part, and finally DTMF part. And the gateway software is divided into three parts: interface to make use of the packet driver which drives the network card, driver to drive the PABX gateway, and the protocol handling part.

• Proceedings of the KIEE Conference
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• 2006.10c
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• pp.345-347
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• 2006

In this paper, we study the robot controller design using the voice and data communication via CDMA(Code Division Multiple Access) mobile communication network. We design the robot remote controller using the three methods, telephone call speech recognition, DTMF (Dual Tone Multiple Frequency) realization, SMS(Short Message Service) transmission/reception way via CDMA mobile communication network. We investigate the validity and effectiveness of the proposed remote controller which applied to the mobile robot.

• 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.

• Journal of Biomedical Engineering Research
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• v.19 no.6
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• pp.623-629
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• 1998

To provide sufficient sound to the hearing-impaired person(HIP) who have many difficulty in communication with others using general telephone, big amplification is needed. But big amplification can occur howling as a side effect. In this study we developed the new technique of big amplification without howling, manufactured and estimated the new hearing aid telephone. Telephone speech is divided to three frequency band, amplified respectively and fitted to HIP's hearing ability. The telephone speech frequency is monitored by counter in time domain. The counter transfers the sinusoidal sound to rectangular wave using comparator and counts the number of rectangular wave in a certain time period, that is frequency, to monitor the howling. Telephone have microphone and speaker, which are fitted in a rigid structure and frequency band of telephone sound is limited, so howling occurs in the limited frequency band. If the counter notices that howling conditions happen, microprocessor decreases quickly the gain of the related frequency band. The result of test of our new hearing aid telephone showed that we can amplifiy the sound as much as 40dB, which is meaningful level to many HIP, and make HIP increase their perception ability from 20% to 60.8% in 1 syllable test and from 28.9% to 78% in 2 syllable test.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.11
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• pp.2391-2398
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• 2011

National Information and Communication Services of Public Administration and Security organized by the 'A 'group (Line service network), 'B' group (IP service network), 'C' Group (Voice over Internet protocol(VoIP) service, IP application services) are provided by constructing the infrastructure. National Information and Communications Services 'C' group, providers are providing VoIP services. In this paper, national information and communications service 'C' group, providers of domestic calls, international calls, including calls to move we will study the basic telephone service. And text messaging, video telephony, IP-Centrex services, etc. we will study the seven value-added services. In addition, national information and communication service providers on the status of the project based on the analysis of national information and communication Internet telephone network using Internet telephony is the type of analysis. In this study, national information and communications services industry, will serve as the basis for the development.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.05a
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• pp.575-578
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• 2001

Number Portability means that one can use his original telephone number when he change telephone service provider, region and the kind of service. Number portability is not necessary when only one telephone service provider exist in korea. But it is necessary to guarantee pair competition among the service providers as new communication service provider comes. The law which is related number portability is established by government as number portability is considered a tool to guarantee pair competition between old service provider and new service provider. A national standard on number portability with which all service provider can agree is needed. We developed functional standard for that purpose. Functional standard includes call processing structure, network requirement. This paper handle functional standards.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.2
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• pp.326-332
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• 1996

In this paper, a method to design the sending loudness rating(SLR) is proposed and the desirable transmission characteristics are considered in order to specify the transmission quality, based on the loudness ratings, for the digital telephone system that is a terminal for digital speech communication. To specify the desirable SLR for digital telephone system, the subjective test defining the preferred range of inout level for CODEC was performed. From the test results, it was identified that the optimal input level for CODEC is -15dB and the range not to cause the quantization noise and the distortion of CODEC fall within -12dB and -18dB.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.2
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• pp.60-65
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• 2002

In this paper, we propose GFCC(gammatone filter frequency cepstrum coefficient) parameter which was based on the auditory characteristic for accomplishing better speech recognition rate. And it is performed the experiment of speech recognition for isolated word acquired from telephone network. For the purpose of comparing GFCC parameter with other parameter, the experiment of speech recognition are carried out using MFCC and LPCC parameter. Also, for each parameter, we are implemented CMS(cepstral mean subtraction)which was applied or not in order to compensate channel distortion in telephone network. Accordingly, we found that the recognition rate using GFCC parameter is better than other parameter in the experimental result.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.2 no.3
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• pp.359-365
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• 1998

The portable two way radio telephone amongst GMDSS equipment in the vessel is very important portable wireless one for search and rescue in distress situation. Although the portable two way radio telephone must be equipped with in the vessel by IMO & RR, other kinds of wireless tranceivers are widely used to communicate with crew members who are separately appointed at bridge forecastle and poop in leaving or arriving at the port, passing through a narrow canal or voyaging in the foggy weather, Most of those wireless tranceivers except portable two way radio telephone are allocated at 27MHz band with PTT simplex system that makes crew members difficult to work effectively in the vessel. In this research 900MHz band FDD(frequency duplex division) tranceiver system is studied to help crew members in the vessel to communicate with themselves freely and to work easily meanwhile with simple operation. 900MHz band FDD tranceiver system uses one chip FM frontend IC to realize the miniaturization and downsizing of tranceiver and small size head set attached to helmet. The credibility of this research's result is highly evaluated by circuit analysis with computer aided simulation, and partly modified 900MHz band cordless phone circuit.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.6
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• pp.1360-1364
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• 2013

Internet phone, communication cost and easy-to-use low-cost compared to the PSTN is a mobile phone of a conventional, and use of the Internet phone is spreading. Construction as part of the broadband convergence network(BCN), Internet service provider(KT, SKT, LGU+) has converted to Internet phone telephone network to all government agencies. In addition, members of the public also have an Internet phone service that you are using. In this paper, we analyze the hacking attack on IP-PBX in the IETF SIP-based that are used in Internet telephony, to the study. The test bed is constructed in the same way as the Internet telephone system to perform studies carried hacking attacks on IP-PBX, analyze the results and to extract evidence forensics. When used in crime by hacking the Internet telephone, we propose a method which can be used as evidence in forensic having legal effect.