• Title/Summary/Keyword: voice communication

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Multimedia Traffic Analysis using Markov Chain Model in CDMA Mobile Communication Systems (CDMA 이동통신 시스템에서 멀티미디어 트래픽에 대한 마르코프 체인 해석)

  • 김백현;김철순;곽경섭
    • Journal of Korea Multimedia Society
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    • v.6 no.7
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    • pp.1219-1230
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    • 2003
  • We analyze an integrated voice/data CDMA system, where the whole channels are divided into voice prioritized channels and voice non-prioritized channels. For real-time voice service, a preemptivc priority is granted in the voice prioritized channels. And, for delay-tolerant data service, the employment of buffer is considered. On the other hand, the transmission permission probability in best-effort packet-data service is controlled by estimating the residual capacity available for users. We build a 2-dimensional markov chain about prioritized-voice and stream-data services and accomplish numerical analysis in combination with packet-data traffic based on residual capacity equation.

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Design of OTP based on Mobile Device using Voice Characteristic Parameter (음성 특징 파라메터를 이용한 모바일 기반의 OTP 설계)

  • Cha, Byung-Rae;Kim, Nam-Ho;Kim, Jong-Won
    • Journal of Advanced Navigation Technology
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    • v.14 no.4
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    • pp.512-520
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    • 2010
  • As the applications based on Mobile and Ubiquitous becoming more extensive, the communication security issues of those applications are appearing to be the most important concern. Therefore, every part of the system should be thoroughly designed and mutually coordinated in order to support overall security of the system. In this paper. we propose new technique which uses the voice features in order to generate Mobile One Time Passwords(OTPs). Voice is considered to be one of the powerful personal authentication factors of biometrics and it can be used for generating variable passwords for one time use. Also we performed a simulation of homomorphic variability of voice feature points using dendrogram and distribution of voice feature points for proposed password generation method.

Analysis for the Effect of Blood Pressure Increase on Vocal Cord Vibration and Voice Intensity (혈압 상승이 성대 진동 및 음성 에너지 크기에 미치는 영향 분석)

  • Kim, Bong-Hyun
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.17 no.2
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    • pp.431-437
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    • 2013
  • These days, many people live a healthy life, but suffering caused by chronic diseases. The main factors of chronic diseases are stress, blood pressure and obesity. Chronic diseases which are caused by high blood pressure are very high incidence. Therefore, this paper suggests the ways to prevent as diagnosis a phenomenon that occur rising in blood pressure consistently by analyzing the voice according th rising in blood pressure. For this, I studied some influence on voicing through rising in blood pressure by applying pitch that measure vocal fold vibration and intensity that measure voice energy size that is one of technique. That collect and analyse the voice after rising blood pressure by aerobic exercise.

A Study on the Design and Analysis of a Voice Coil Linear Force Motor for Hydraulic Valve (밸브구동용 보이스 코일 선형 포스모터 설계와 해석 연구)

  • Park, C.S.;Huh, J.Y.
    • Journal of Drive and Control
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    • v.9 no.4
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    • pp.1-7
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    • 2012
  • The voice coil linear force motor is a kind of a direct drive motion device that uses a permanent magnetic field and coil winding to produce force. In order to design a voice coil linear force motor, an exact calculations of the required force, the flux density in air gap and the flux pathway are needed. A conventional method can be used usually to calculate the flux density in air gap, but with this method it is needed to find a magnetic circuit revision constant. In this paper a voice coil linear force motor is designed by conventional design method and analyzed by 3D simulation program "Flux". For the prototype linear force motor, the results of the calculated by conventional design method and the analyzed by 3D simulation program are compared with the test result. Finally it is showed that the magnetic circuit revision constant which is found by comparing of the analyzed and the measured data can be used for the design of the voice coil type linear force motor to minimize the trial and error.

A Study of Hybrid Automatic Interpret Support System (하이브리드 자동 통역지원 시스템에 관한 연구)

  • Lim, Chong-Gyu;Gang, Bong-Gyun;Park, Ju-Sik;Kang, Bong-Kyun
    • Journal of Korean Society of Industrial and Systems Engineering
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    • v.28 no.3
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    • pp.133-141
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    • 2005
  • The previous research has been mainly focused on individual technology of voice recognition, voice synthesis, translation, and bone transmission technical. Recently, commercial models have been produced using aforementioned technologies. In this research, a new automated translation support system concept has been proposed by combining established technology of bone transmission and wireless system. The proposed system has following three major components. First, the hybrid system consist of headset, bone transmission and other technologies will recognize user's voice. Second, computer recognized voice (using small server attached to the user) of the user will be converted into digital signal. Then it will be translated into other user's language by translation algorithm. Third, the translated language will be wirelessly transmitted to the other party. The transmitted signal will be converted into voice in the other party's computer using the hybrid system. This hybrid system will transmit the clear message regardless of the noise level in the environment or user's hearing ability. By using the network technology, communication between users can also be clearly transmitted despite the distance.

A Study on the VoIP Security Countermeasure of SIP-based (SIP(Session Initiation Protocol) 기반의 VoIP 보안 대책 연구)

  • Tae, Jang-Won;Kwak, Jin-Suk
    • Journal of Advanced Navigation Technology
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    • v.17 no.4
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    • pp.421-428
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    • 2013
  • Voice over IP refers to technology that enables routing of voice conversations over the Internet or a TCP/IP network. VoIP communication costs cheaper than traditional analog phone. Phone calls can be made to anywhere / anyone: Both to VoIP numbers as well as people with normal phone numbers. VoIP protocol equipment available today follows the SIP standard. Older VoIP equipment though would follow H 323, MGCP, Megaco/H.248. A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the TCP/IP network. A SIP server is also referred to a Asterisk IP-PBX. A VoIP telephone, also known as a SIP phone or a softphone, allows the user to make phone calls to any softphone, mobile or PC by using App store. A VoIP telephone can be a simple software-based softphone. However, the SIP Server and the program is vulnerable to VoIP attacks. In this paper, eavesdropping attacks tested by using the Asterisk SIP server. Eavesdropping attacks and TLS security methods apply to VoIP system. TLS can be applied to determine whether the eavesdropping available for VoIP Environments.

Performance Analysis of Multimedia CDMA Mobile Communication System Considering Diverse Qos Requirements (멀티미디어 CDMA 이동통신 시스템에서의 다양한 QoS 요구조건을 고려한 성능 분석)

  • Kim, Baek-Hyun;Shin, Seung-Hoon;Kwak Kyung-Sup
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.27 no.1B
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    • pp.1-12
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    • 2002
  • In the multimedia CDMA mobile communication service, it is required to support various applications, such as voice, video, file transfer, e-mail, and Internet access, with guaranteed QoS. In the mixed traffic environment ,which consists of voice, stream data, and packet data, we analyze the network where preemptive priority is granted to delay-intolerant voice service and a buffer is offered to delay-tolerant stream data service. And, for best-effort packet data service, the access control by transmission permission probability is applied to obtain prominent throughput. To analyze the multimedia CDMA mobile communication system, we build a 2-dimensional markov chain model about prioritized-voice and stream data services and accomplish numerical analysis in combination with packet data traffic based on residual capacity equation.

Design and Implementation of Visual/Control Communication Protocol for Home Automated Robot Interaction and Control (홈오토메이션을 위한 영상/로봇제어 시스템의 설계와 구현)

  • Cho, Myung-Ji;Kim, Seong-Whan
    • Journal of Internet Computing and Services
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    • v.10 no.6
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    • pp.27-36
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    • 2009
  • PSTN (public switched telephone network) provides voice communication service, whereas IP network provides data oriented service, and we can use IP network for multimedia transport service (e.g. voice over IP service) with economic price. In this paper, we propose RoIP (robot on IP) service scenario, signaling call flow, and implementation to provide home automation and monitoring service for remote site users. In our scheme, we used a extended SIP (session initiation protocol) for signaling protocol between remote site users and home robots. For our bearer transport control, we implemented H.263 video codec over RTP (real-time transport protocol) and additionally DTMF (dual tone multi-frequency) transport for robot actuator control. We implemented our scheme on home robots and experimented with KTF operator network, and it shows good communication quality (average MOS = 9.15) and flexible robot controls.

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Architecture and Call Setup Latency of a Softswitch for VoIP Service (소프트스위치 시스템의 호처리 성능 향상)

  • Kim, Sung-Chul;Yoo, Byun-Hoon;Lee, Byung-Ho
    • Proceedings of the IEEK Conference
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    • 2005.11a
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    • pp.113-118
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    • 2005
  • Softswitch is the core BcN equipment which voice and multimedia switching based on the IP Technologies. It is designed to replace the Class 5(local Exchange) and Class 4(Toll Exchange) switch based on the circuit wired and wireless switching network technologies. Softswitch gets its name because typically it is a software based solution implemented on general purpose computers/servers. While the traditional PSTN switches are rely on dedicated facilities for T and S inter-connection and are designed primarily for voice communications. Packet based Softswitch is divided the control of call and bearer, very different from Public telephone network. Sometimes Call Agent or Media Gateway Controller, a key component in the VoIP solution, is also called Softswitch. This paper will suggest the software architecture of softswitch for performance in call processing part, also suggest the session management model to cover call setup latency.

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Development of Remote-Controlled Object-Recognizing Mobile Home CCTV Using Smartphone and Arduino (스마트폰과 아두이노를 이용한 원격제어 객체인식 이동형 홈 CCTV 개발)

  • Kim, Dong-Ju;Lim, Chae-Won;Choi, Hyun-Ho
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.24 no.11
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    • pp.1546-1549
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    • 2020
  • This paper introduces the development process of mobile home CCTV that enables remote control and object recognition using unused smartphones and Arduino. Clients can control motors connected to Arduino through button, enable bidirectional voice communication between client-server and receive video from the server in real time. The server sends a PUSH notification to the client when its battery is low. When the server recognizes the charger, the client's remote control allows the server to dock to the charger and charge it. It was confirmed that video and voice delivery between client and server works well without any problems, and that object recognition works smoothly.