• Proceedings of the IEEK Conference
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• 2001.06d
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• pp.249-252
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• 2001

The most basic means of communication among humans is a voice. Without speaking of voice technologies, we found it is important and convenient to use a voice in everyday life. But. in consideration to speech recognition systems, we can't always desire a normal voice input as input signal to the system. Generally speaking. a pathological voice as against a normal which is a voice with a problem in the larynx. could be also special case of input voice. Of course, but the distortion of a speech signal by environmental effects i.e., noise or transmission channel was a raised problem. we will take up a pathological voices with laryngeal disease which is essential distortion factor in voice. Also, we are to find out the difference of acoustic parameters distribution between normal and pathological voice by a statistical method in our research.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2A
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• pp.187-194
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• 2004

We propose priority-based reservation code multiple access (P-RCMA) which can enhance voice traffic quality of the previous RCMA. The proposed protocol maintains two power levels and consider traffic characteristics in contending shared available codes to transmit packets. P-RCMA gives priority to the voice request packets rather than data packets by capture effect at the receiver part of base station. We show numerical results from EPA (equilibrium point analysis) analysis and simulation study in terms of voice packet dropping probability and average data packet transmission delay.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.71-76
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• 1999

In digital cellular communication systems, tandem coding occurs in communications between mobile phones with different speech codecs, resulting in poor voice quality, high computational load, and long transmission delay. In this paper, voice packet conversion technique is proposed to solve the tandem coding problems, and packet conversion algorithm from 13kbps QCELP to 8kbps QCELP is developed. Simulations using various speech data show that the proposed packet conversion method produces voice quality which is equivalent to that by the conventional tandem coding method with shorter transmission delay using about 33% computational load.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.568-570
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• 2018

Currently, with rapid progress and supply of mobile communication technology, LTE(Long Term Evolution) technology is expanded and widely used to industrial and emergency communications beyond earlier smart-phone based service. In this paper, transmission performance of voice traffic, one of railway communication service based on LTE-R as an application field of LTE technology, is analyzed. This study is performed performance analysis with level of application service and consider effects of satisfaction level for users. Computer Simulation based on ns(Network Simulation)-3 is used for analysis and VoIP(Voice over Internet Protocol) specification is used for voice traffics. Results of this paper is used to implement LTE-R networks and develope application services over LTE-R network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.4
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• pp.279-290
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• 2013

In this paper, we propose a transmission scheme of MAC protocol that enables secure voice communications by exploiting the wide spectrum and low signal strength characteristics of the ultra wide band technology. In addition, it also supports high level of terminal mobility by deploying mobile ad hoc network schemes. While most of existing UWB MAC protocols are operated as a synchronous mode, the proposed scheme operates in an asynchronous mode for supporting high mobility and sends voice packets without RTS/CTS control packets for efficient voice traffic transmission without retransmission. With simulation program, we prove that the proposed scheme satisfies the required voice quality and packet delivery time.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.394-400
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• 2002

In digital speech communications, the quality of service can be increased by speech coding scheme that is adaptive to the error rate of voice packet transmission. However, current communication protocol in cellular and internet communications does not provide the function that transmits the channel error information. To solute this problem, in this paper, new method for real-time transmission of channel error information is proposed, where channel error information is embedded in voice packet. The proposed method utilizes the pulse positions of codevector in ACELP speech codec, which results in little degradation in speech quality and low false alarm rate. The simulations with various speech data show that the proposed method meets the requirement in speech quality, detection rate, and false alarm rate.

• IEMEK Journal of Embedded Systems and Applications
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• v.2 no.1
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• pp.43-52
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• 2007

IEEE 802.15.4 specification which defines low-rate wireless personal area network(LR-WPAN) has application to home or building automation, remote control and sensing, intelligent management, environmental monitoring, and so on. Recently, it has been considered as an alternative technology to provide multimedia services such as automation via voice recognition, wireless headset and wireless camera for surveillance. In order to evaluate capability of voice traffic on the IEEE 802.15.4 LR-WPAN, we supposed two scenarios, voice traffic only and coexistence of voice and sensing traffic. For both cases we examined delay and packet loss rate in case of with and without acknowledgement, and various beacon period varying with beacon and superframe order values. In LR-WPAN with voice devices only, total 5 voice devices could be applicable and in the other case, i.e., coexisted cases of voice and sensor devices, a voice device was able to coexist with about 60 sensor devices.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.42 no.2 s.302
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• pp.11-20
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• 2005

In previous video conference system, when the number of participants in video conference increases by n, the bandwidth and memory of n2 is required. And also, it brings about increase in traffic and problem of a say during a conference in aspect of transmission of voice data. In this paper, we propose an algorithm of remote video conference using silence detection algerian to resolve the questions such as buffering method of video data in server and heavy traffic detection algorithm to the increase in participants. Video data buffering algorithm is not a method of broadcasting to other client in the server, but this algorithm uses two other methods; the buffering method of receiving compressed video data from clients and the indexing method for acquiring the video data of other participants in clients according to clients' bandwidth and network transmission speed. We apply a voice transmission algerian and a channel management algorithm to the remote video conference system. The method used in the voice transmission algorithm is a silence detection algorithm which does not send silent participants' voice data to the server. The channel management algorithm is a method allocating a say to the participants who have priority. In consideration of average 20 frames and 30ms regardless of a number of participants, we can safely conclude that the transmission of video and voice data is stable.

• Proceedings of the IEEK Conference
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• 2003.07d
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• pp.1629-1632
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• 2003

Generally an increasing number of recent applications have found TCP too limiting. There are some characteristics in the transmission of document and binary data which some transmission delay are tolerant but the content must completely be transferred. However voice signals are more sensitive with not some packet loss but some transmission delay. Therefore, Stream Control Transmission Protocol(SCTP) is proposed to minimize the delay and packet loss in the field of delivery of voice signal. SCTP is designed to transport PSTN signalling messages over IP networks, but is capable of broader applications. In this paper, the architecture of SCTP implementation is designed and some interface of SCTP software library which are implemented are specified.

• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.4
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• pp.24-30
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• 1984

Group delay distortion (phase distortion) is a characteristic which is of no account from a standpoint of voice transmission. But this distortion becomes the major source of distortion in wave form transmission such as data, FAX and others over the public switching telephone network (voice band transmission) so that it must be drastically studied. This paper makes analysis of group deray distortion of the telephone network, describs experimental and measuring results and refers also to the improvement of distortion for the purpose of opening the public switching telephone network to data transmission.