• Title/Summary/Keyword: PSTN (Public Service Telephone Network)

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A Study of Call Service Mechanism on SIP for Emergency Communication Services (긴급통신서비스 제공을 위한 SIP에서의 호 서비스 메커니즘에 관한 연구)

  • Lee, Kyu-Chul;Lee, Jong-Hyup
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.11 no.2
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    • pp.293-300
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    • 2007
  • As the development of the various IP-based services, it is expected that Internet telephony service will gradually replace the traditional PSTN-based telephony service. But there are many issues resolved to spread the Internet telephony service. One of them is to support the emergency services in the Internet telephony. In the case of USA, it has been regulated that 911 services should be supported in the Internet telephony services using VoIP on the similar performance level to PSTN 911 service. According to the regulation, basic VoIP 911 calls should be routed to the general access line of LEA without the location information or the callback number, but the enhanced VoIP 911 calls with the location information and callback number should be routed on the dedicated 911 network and destined to the local 911 distribution center such as PSAP. But, in the current VoIP-based Internet telephony network, the emergency call service has not been handled as one of the special services as well at has a worse performance in comparison to it on PSTN. Moreover, the service has a critical problem that it can not be destined to the nearest PSAP because of the insufficient information about the location information and the call back number. In this paper, we suggest the SIP-based emergency call service mechanism in order to resolve the problems above mentioned. This suggested mechanism is implemented to show its effectiveness and efficiency.

Transmission Performance of ADSL for High Speed Multimedia Service Using Unshield Twisted Pair (동선에 의한 고속 멀티미디어 서비스를 위한 ADSL 전송성능)

  • Kim, Jin-Tae;Yang, Sung-Mo;So, Woon-Sup;Kwak, Kyung-Sup;Choi, Byung-Ha
    • The Transactions of the Korea Information Processing Society
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    • v.4 no.12
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    • pp.3111-3122
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    • 1997
  • Since the advent of an information era, the traffic of communication has been remarkably increasing and the information service has gradually become speedy and popular. With these trends, it is necessary to extend the utilization of existing twist pair cables for PSTN(Public Switched Telephone Network) to provide high speed data services for telephone subscribers, In this paper, we have studied on a structure of discrete multi-tone ADSL(Asymmetric Digital Subscriber Lines) system which can support high speed multimedia service using a twist pair cable, and analyzed the transmission performance by computer simulations.

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A Study on the Development of Remote Fault Diagnosis and Maintenance System for Machine Tool (공작기계에서의 원격고장진단 시스템 개발에 관한 연구)

  • 현웅근;신동수;박인준
    • Proceedings of the Korean Society of Precision Engineering Conference
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    • 1997.04a
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    • pp.708-713
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    • 1997
  • A remote data communication system for monitoring of NC machine fault diagnosis and status is developed. This system communicates with host PC by using dial-up communication method on PSTN. The developed system consists of (1)remote communication module among NC's and host PC using PSTN, (2) 8 channels analog data sensing module, (3) digital I/O module for control of NC machine, (4) communication module between NC machine and remote data communication system using RS-232c, and (5) Software man-machine interface. This system may be applied for remote sensing of the status in Fms. To show the veridity of the developed system, several examples are illustrated.

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Implementation of Extended Automatic Callback Service in SIP-based VoIP System (SIP 기반의 VoIP 시스템에서의 확장된 자동 콜백 서비스의 구현)

  • Jo Hyun-Gyu;Lee Ky-Soo;Jang Choon-Seo
    • The KIPS Transactions:PartC
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    • v.12C no.2 s.98
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    • pp.251-260
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    • 2005
  • On the internet phone or PSTN(Public Switched Telephone Network), the automatic callback is an useful service in the case of busy state when one user calls the other. By using this service, automatic redial is possible when the other party hangs up. However, in the basic automatic callback service, the user who wants callback should wait until the other party hangs up even in the case of emergency. Therefore in this paper, to solve this problem we have extended CPL(Call Processing Language) and, within user system we have included and linked this extended CPL processing module and dialog event package which processes SIP INVITE initiated dialog state informations. We have implemented this system for being used in SIP(Session Initiation Protocol)-based VoIP(Voice over IP) system.

Remote Fault Diagnosis and Maintenance System for NC Machine Tools (공작기계용 원격 고장진단 및 보수 시스템)

  • 신동수;현웅근;정성종
    • Journal of the Korean Society for Precision Engineering
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    • v.15 no.1
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    • pp.19-25
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    • 1998
  • Remote fault diagnosis and maintenance system using general telecommunication network is necessary for an effective fault diagnosis and higher productivity of NC machine tools. In order to monitor machine tool condition and diagnose alarm states due to electrical and mechanical faults, a remote data communication system for monitoring of NC machine fault diagnosis and status is developed. The developed system consists of (1) remote communication module among NC's and host PC using PSTN. (2) 8 channels analog data sensing module, (3) digital I/O module for control or NC machine, (4) communication module between NC machine and remote data communication system via RS-232C, and (5) software man-machine interface. Diagnostic monitoring results generated through a successive type inference engine are displayed in user-friendly graphics. The validity and reliability of the developed system is verified to be a powerful commercial version on a vertical machining center through a series of experiments.

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Extraction of Forensic Evidence and Hacking Attacks about IP-PBX (IP-PBX에 대한 해킹공격과 포렌식 증거 추출)

  • Park, Dea-Woo
    • 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.

Extraction of Forensic Evidence and Hacking Attacks about IP-PBX (IP-PBX에 대한 해킹공격과 포렌식 증거 추출)

  • Park, Dea-Woo
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2013.05a
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    • pp.197-200
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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, LG) 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.

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Design and Performance Analysis of EU Directory Service (ENUM 디렉터리 서비스 설계 및 성능 평가)

  • 이혜원;윤미연;신용태;신성우;송관우
    • Journal of KIISE:Information Networking
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    • v.30 no.4
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    • pp.559-571
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    • 2003
  • ENUM(tElephon NUmbering Mapping) is protocol that brings convergence between PSTN Networks and IP Networks using a unique worldwide E.164 telephone number as an identifier between different communication infrastructure. The mechanism provides a bridge between two completely different environments with E.164 number; IP based application services used in PSTN networks, and PSTN based application services used in IP networks. We propose a new way to organize and handle ENUM Tier 2 name servers to improve performance at the name resolution process in ENUM based application service. We build an ENUM based network model when NAPTR(Naming Authority PoinTeR) resource record is registered and managed by area code at the initial registration step. ENUM promises convenience and flexibility to both PSTN and IP users, yet there is no evidence how much patience is required when users decide to use ENUM instead of non-ENUM based applications. We have estimated ENUM response time, and proved how to improve performance up to 3 times when resources are managed by the proposed mechanism. The proposition of this thesis favorably influences users and helps to establish the policy for Tier 2 name server management.

User Authentication Mechanism for SIP Call Signaling (SIP Call Signaling을 위한 사용자 인증 기법)

  • Choi, Kyoung-Ho;Im, Eul-Gyu
    • Proceedings of the Korean Information Science Society Conference
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    • 2008.06d
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    • pp.110-115
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    • 2008
  • 음성 데이터를 IP기반의 패킷망을 통해 전송하는 기술인 VoIP(Voice over Internet Protocol) 기술은 음성 데이터를 기존의 PSTN(Public Switched Telephone Network)망을 통해 전송하는 방식에 비해 비용 절감 등의 장점을 가지고 있다. 그러나 VoIP가 기존의 PSTN망을 대체하기 위해서는 QoS(Quality of Service)의 보장과 보안이 제공되어야 한다는 문제점을 가지고 있다. VoIP망에서 보안을 위해서는 사용자간에 전송되는 음성 데이터에 대한 보안과 초기의 세션 연결 시 사용자를 인증하는 과정이 고려되어져야 한다. 실질적인 대화 내용인 음성 데이터의 보안도 중요한 부분이지만 대화에 참여하는 사용자를 인증하는 과정이 선행되어야 한다. VoIP에서는 세션 연결 설정을 위해 H.323과 SIP를 사용하고 있으며, 최근에는 H.323에 비해 간단한 SIP가 주목을 받고 있다. RFC3261에서는 SIP를 이용해 세션 연결을 하는 과정에서 사용자를 인증하기 위한 몇 가지 인증 메커니즘을 제시하고 있다. 본 논문에서는 SIP를 이용하여 세션을 연결하는 과정에서 사용자의 인증을 위해 사용되는 인증 메커니즘 중 한 가지인 HTTP Digest Authentication의 취약점을 분석하고, 이를 보완하기 위한 새로운 인증 메커니즘을 제시한다.

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Value of Number Portability on Internet Phones

  • Ku, Jung-Eun;Lee, Sang-Woo;Hyun, Tchang-Hee
    • ETRI Journal
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    • v.32 no.1
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    • pp.169-171
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    • 2010
  • Even though Internet phone service (IPS) continues to evolve in terms of quality and competitive pricing compared with the public switched telephone network (PSTN), the number of Internet phone subscribers is still low in South Korea. The Korean government decided to apply number portability to the IPS market in 2008. This study expresses its value in monetary terms with a contingent valuation approach. With this methodology, this study investigates the willingness of 316 current PSTN subscribers to pay for IPS, using a specific survey method called double-bounded dichotomous choices, to obtain more accurate data. The results show that the calculated value of Internet phone number portability positively impacts consumer welfare.