• Title/Summary/Keyword: telephone call

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Design of Internet Telephone by SIP Server of Hybrid Type (Hybrid형 SIP 서버를 통한 인터넷폰 설계)

  • 김진수;양해권
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2002.11a
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    • pp.335-340
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    • 2002
  • 인터넷을 이용한 여러 응용 서비스들 중에서 저렴한 비용으로 음성을 전송할 수 있는 VoIP서비스의 발전으로 사용자의 급격한 증가가 예상된다. VoIP에 mobility, universal number, multiparty conference, voice mail, automatic call distribution과 같은 고품질의 서비스를 제공하기 위해서는 시그널링이 가능한 표준화된 프로토콜이 필요하다. 현재 IETF의 SW(Session Initiation Protocol)가 빠른 호 설정과 parsing 및 compile이 쉬운 장점으로 인해 SIP를 기반으로 한 VoIP 서비스를 제공하기 위해 국내외적으로 SIP 기반 구성요소에 대한 개발이 진행 중이다. 본 논문에서는 사용자가 보내는 request(INVITE) 메소드(method)를 처리해주는 SIP 서버의 부하 경감, 망 운용의 효율성, 많은 사용자에 대한 서비스를 제공하기 위해 새로운 서버 유형인 Hybrid형 SIP 서버를 제안하고, 이를 이용하여 새로운 타입의 인터넷폰을 설계하였다.

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Augmenting Path Algorithm for Routing Telephone Calls Problem

  • Lee, Sang-Un
    • Journal of the Korea Society of Computer and Information
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    • v.21 no.6
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    • pp.77-81
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    • 2016
  • This paper deals with the optimization problem that decides the routing of connection between multi-source and multi-sink. For this problem, there is only in used the mathematical approach as linear programming (LP) software package and has been unknown the polynomial time algorithm. In this paper we suggest the heuristic algorithm with $O(mn)^2$ time complexity to solve the optimal solution for this problem. This paper suggests the simple method that assigns the possible call flow quantity to augmenting path of ($s_i,t_i$) city pair satisfied with demand of ($s_i,t_i$). The proposed algorithm can be get the same optimal solution as LP for experimental data.

Efficient Design of a Disaster Broadcasting System using LTE Modem (이동 LTE모뎀을 활용한 재난방송시스템 설계)

  • Moon, Chaeyoung;Kim, Semin;Ryoo, Kwangki
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2018.10a
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    • pp.292-294
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    • 2018
  • Recently, damage caused by natural disasters such as fire, earthquake, heavy rains and heavy snow is increasing. In addition, traffic accidents due to freezing, fog and fire in tunnels and bridges are frequently occurring. In such a disaster situation, it is very important to take prompt action by the person in charge of managing the facility and area.To this end, a disaster broadcasting system is used, but in the existing system, the broadcasting room and the speaker are connected by a wired connection. Also, the person in charge has to be in the broadcasting room to broadcast, which has a problem of delaying the time. In this paper, we design a disaster broadcasting system using LTE modem. The designed system enables a broadcasting person to make a call to a broadcasting system from anywhere using a cellular phone and a public telephone. Broadcasting via telephone is possible only with the telephone number pre-registered in the system and can be registered / deleted by the administrator. The registered telephone number, incoming voice file, and announcement voice for automatic broadcasting are stored in the system internal SD memory for convenient management. This disaster broadcasting system is expected to contribute to quick and convenient disaster broadcasting.

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Gracilis Muscle Free Flap Transplantation Followed Up Over 10 Years (10년 이상 추시된 박 근 유리 판 이식술)

  • Yu, Chang Eun;Park, Ki Tae;Lee, Jun-Mo
    • Archives of Reconstructive Microsurgery
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    • v.20 no.2
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    • pp.108-112
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    • 2011
  • Purpose: Gracilis muscle free flap transplantation is chosen in the medium sized soft tissue defect and bone exposure from trauma and chronic osteomyelitis in the lower extremity. We set a study to search for gracilis free flaps to know whether symptoms were cured or recurred in patients that have passed over 10 years from flap transplantation. Materials and Methods: From August 1995 through September 2010, we have performed 28 cases of gracilis muscle free flap in the lower extremities. Ever since no case visited to demand any discomfort, breakdown or recurrence in the flap site on outpatient basis. We made a telephone call to patients or relatives documented in the medical record and only 2 cases visited outpatient department and 9 cases postponed the visit who satisfied with the final result but 17 cases had wrong telephone numbers. Causes, area of lower extremity, recipient vessels in the lower extremity, condition of the donor thigh and overall satisfaction of the flap transplantation in activities of daily living were investigated and written down in the medical record. Results: 11 cases were reviewed after average postoperative 13.7 years. Gracilis muscle flaps were not break down at the recipient site in all cases. The wound of donor thigh wound healed good and overall activities of daily living was satisfied in all cases. Conclusion: Gracilis muscle flaps which had performed and followed up average 13.7 years revealed confident in the medium sized soft tissue defect and bone exposure from trauma and chronic osteomyelitis in the lower extremity.

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A Target Model Development Applying Scoring Method for Sale of DATA Additional Charge Service Product in a Mobile Telephone Company A (고객 스코어링 방법을 활용한 데이터 통화료 정액제 타겟 모델 개발)

  • Chun, Heui-Ju
    • The Korean Journal of Applied Statistics
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    • v.21 no.5
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    • pp.791-799
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    • 2008
  • Ansim Flat DATA Plan is a DATA additional service product related to DATA call in a mobile telephone company A. Up to now, the company A is selling it by outbound TM after targeting customers which used data within specific price band. In this paper, we propose a targeting method applying score model combining response rate and retention rate by data mining. The suggested target model is to find customers more likely not only to respond to outbound TM but also to retain Ansim Flat DATA Plan. The proposed targeting method is expected to improve both from 23.7% to 38.8% in the response rate and from 53.2% to 61.4% in the retention rate.

A Study on the Call-Setup and Message Mapping for Interworking between H.323 and SIP (H.323과 SIP간의 상호 연동을 위한 호 설정과 메시지 매핑에 관한 연구)

  • Kim, Jeong-Seok;Tae, Won-Kwi;Kim, Jeong-Ho;Ban, Jin-Yang
    • Journal of the Korea Computer Industry Society
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    • v.5 no.9
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    • pp.1017-1024
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    • 2004
  • In this paper, we propose the progressed interworking method between H.323 and SlP, then explain the improved property. The VolP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easil)· accept the various of multimedia services from internet. Previous connectionmethod of VoIP used H.323 protocol, but it is very complex to connection establishment. so, the SIP(Session Initiation Protocol) protocol that propose in SIP-Working Group is in use recently. Therefore, we need new interworking methodology between H.323 and SIP Products. In this thesis, the progress interworking method between H.323 and SIP are Propose, then interpret unnecessary packet delay for call setup and improved feature of message exchange.

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Study on Eveluation of Performancen on Internet Phone(VoIP) using the VPN (VPN을 적용한 인터넷 전화 단말기의 성능평가에 관한 연구)

  • Lee Seong gi;Yoo Seung Sun;Lee Myeong jea;Kwak Hoon-Sung
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.30 no.6A
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    • pp.445-454
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    • 2005
  • To measure the performance of call quality, we have built the experiment environment and observed that the delay caused by encapsulation between internet and VoIP telephones is under 5ms at most. The major delay is assumed to be the time required to capsulate the packet for tunnelling of VPN. Because the difference of average delay time is under $4ms{\sim}5ms$, the difference of call quality between VoIP and VoIP telephone adopting VPN is negligible. We have concluded that the capsulation process between PAC and PNS is the major factor influencing the network load by changing the number of fames in a packet during communication Also, we have concluded that the most suitable frame numbers is tow or three by adding the frame numbers in a packet to obtain the suitable frames in a packet and setting up end-to-end delay under 150ms.

Development of Remote Management and Control System for VoIP Terminal (인터넷전화 단말기 원격관리 및 제어시스템 개발)

  • Song, Han-Chun;Ban, Ku-Ik
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.11 no.6
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    • pp.73-80
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    • 2011
  • In this paper, we design and implementation of effective remote management system for VoIP(Voice over IP) terminal. Ordinary VoIP terminal is connected inside of NAT(Network Address Translation) assigned private IP address. NAT provides address mapping between outside public IP address and inside private IP address, for outside system telephone call is connected to inside VoIP terminal. In this paper, we also design and implementation of UDP hole punching function that the outside calls pass through the NAT, for outside management system call is connected to internal VoIP terminal, and gathered to management information of the VoIP terminal. Also, we evaluated and analysed of developed system in the test environment. As a result of test, It showed that it was well performed without any data error and data loss in the connectivity, and It showed that it was well gathered management information of the VoIP terminal.

Design of User Access Authentication and Authorization System for VoIP Service (사용자 접근권한 인증을 이용한 안전한 VoIP 시스템 설계)

  • Yang, Ho-Kyung;Kim, Jin-Mook;Ryou, Hwang-Bin;Park, Choon-Sik
    • Convergence Security Journal
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    • v.8 no.4
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    • pp.41-49
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    • 2008
  • VoIP is a service that changes the analogue audio signal into a digital signal and then transfers the audio information to the users after configuring it as a packet; and it has an advantage of lower price than the existing voice call service and better extensibility. However, VoIP service has a system structure that, compared to the existing PSTN (Public Switched Telephone Network), has poor call quality and is vulnerable in the security aspect. To make up these problems, TLS service was introduced to enhance the security. In practical system, however, since QoS problem occurs, it is necessary to develop the VoIP security system that can satisfy QoS at the same time in the security aspect. In this paper, a user authentication VoIP system that can provide a service according to the security and the user through providing a differential service according to the approach of the users by adding AA server at the step of configuring the existing VoIP session is suggested. It was found that the proposed system of this study provides a quicker QoS than the TLS-added system at a similar level of security. Also, it is able to provide a variety of additional services by the different users.

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Mobile Donation Application of User Participation Base (사용자 참여 기반의 모바일 기부 어플리케이션)

  • Chung, Myoung-Beom;Ko, Il-Ju
    • Journal of the Korea Society of Computer and Information
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    • v.16 no.11
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    • pp.113-122
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    • 2011
  • In this paper, we propose an iPhone application that allows the user to pay for donations using the camera, GPS and call functions of the iPhone. As iOS version 3.0 allows the iPhone camera to detect and read bar codes and QR codes, the proposed application uses such codes to identify a product the user wishes to donation. After determining the user's location using the iPhone GPS function, the application can then perform a navigation task that guides the user to a suitable shop or store where the user can make his or her donation. In addition, the application offers an ARS call function that allows the user to make a direct donation, even if the user does not know the telephone number for making such donations. Therefore, the proposed application provides an easy means for the user to pay for donations directly or indirectly.