• Title/Summary/Keyword: IP전화

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A Scalable Management Method for Asterisk-based Internet Telephony System (확장성을 고려한 Asterisk 기반 인터넷 전화 관리 방법)

  • Ha, Eun-Yong
    • Journal of Digital Convergence
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    • v.12 no.8
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    • pp.235-242
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    • 2014
  • Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. In this paper we suggested an Asterisk-based Internet telephony system which can be easily scalable. Most current systems use text files to manage their configuration: SIP users, dialplans, IVR service and etc. But we designed the management system which introduces database tables for efficiency and scalability. It also supports web-based functions developed by using Asterisk, Apache, MySQL, jQuery, PHP and open source softwares.

Enterprise Broadcasting Service Using VoIP Technology (VoIP를 이용한 기업 방송망 서비스 구현방법)

  • Lee Jong-Heon;Jang Sung-Bong;Woo Jong-Wook;Kim Hee-Dong;Whang Byung-Yeon
    • 한국정보통신설비학회:학술대회논문집
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    • 2003.08a
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    • pp.100-105
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    • 2003
  • 초고속 인터넷 망의 진화로 인하여 VoIP 기술이 대두되어 저렴한 장거리 전화로 주로 사용되었고, 기업체에서는 통신비 절감을 위하여 VoIP기술의 도입이 점차로 활발해지고 있다. 기업에서 VoIP 기술을 접목하는 서비스로는 음성전화, Fax전화, 방송 등이 있을 수 있다. 본 논문에서는 VoIP를 이용한 기업 방송 서비스를 구현 하기 위한 기본 기술들을 고찰 하고, 실제 구현 방법을 제안한다.

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Integrated Architecture of Mobile IP with H.323 (H.323과 이동 IP의 연동 구조 설계)

  • 정진우;천정훈;강현국
    • Proceedings of the Korean Information Science Society Conference
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    • 2001.04a
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    • pp.295-297
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    • 2001
  • 현재 인터넷 망에서 실시간 트래픽과 양 방향 전송을 실현하기 위한 인터넷 전화는 비용과 서비스측면에서의 장점을 바탕으로 많은 관심을 끌고 있다. 지금까지의 인터넷 전화 프로토콜의 대표적인 표준안은 ITU에서 제안된 H.323과 IETF에서 제안된 SIP(Session Initiation Protocol)이다. 그러나, 현재의 H.323 표준안은 유선망과 고정 호스트로 이루어진 패킷망에 적합하도록 설계되었기 때문에 이동 인터넷망의 특성을 고려하지 않고 있다. 본 논문에서는 이동환경에서 인터넷 전화 서비스를 제공하기 위한 Mobile IP와 H.323dml 통합 구조를 제안하였다. 현재 Mobile IP에서 제안된 표준안을 확장함으로써 인트라 도메인(intra-domain)에서의 IP 이동성을 제공하도록 하였으며, 인터 도메인(inter-domain)에서의 IP 이동성을 제공하기 위하여 게이트웨이에서의 다중 연결 유징 방법을 제안하였다. 따라서, 본 논문에서 제안한 방법은 기존의 H.323 표준의 최소 변경과 추가적인 새로운 개체의 정의가 필요 없으며, 기존의 H.323 시스템들과 호환성을 유지할 수 있다는 장점을 지니고 있다.

인터넷전화(VoIP) 제도 및 계획

  • 서성일
    • Information and Communications Magazine
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    • v.21 no.4
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    • pp.13-18
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    • 2004
  • 미래 정보통신 환경이 광대역ㆍ초고속화되고 유ㆍ무선, 음성ㆍ데이타, 통신ㆍ방송간 통합 융합화가 활발히 진행됨에 따라 VoIP기술을 이용한 융합형 멀티미디어 통신서비스가 주요 통신서비스로 자리잡을 것으로 전망됨에 따라 인터넷전화(VoIP) 제도 및 계획에 대하여 소개하고자 한다.(중략)

Development of the IP-PBX with VPN function for voice security (VPN 기능을 가진 음성 보안용 IP-PBX 개발)

  • Kim, Sam-Taek
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.10 no.6
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    • pp.63-69
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    • 2010
  • Today, Internet Telephony Services based on VoIP are gaining tremendous popularity for general user. Therefore a various demands of the user keep up increase, the most important requirements of these is voice security about telephony system. It is needed to ensure secret of voice call in a special situation. Due to the fact that many users can connect to the internet at the same time, VoIP can always be in a defenseless state by hackers. Therefore, in this paper, we have developed VPN IP-PBX for the voice security and measured conversation quality by adopting VPN IPsec based on SIP and using tunnel method in transmitting voice data to prevent eavesdrop of voice data. This VPN IP-PBX that is connected Soft-phone provide various optional services.

VoIP Planning and Evaluation through the Analysis of Speech Transmission Quality Based on the E-Model (E-모델 기반 통화 품질 분석을 통한 VoIP Planning 및 평가)

  • Bae Seong Yong;Kim Kwang Hoon
    • Journal of Internet Computing and Services
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    • v.5 no.6
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    • pp.31-43
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    • 2004
  • Voice over Internet Protocol (VoIP) is currently a popular research topic as a real time voice packet transmission method. But current Internet environment do not guarantee the quality of voice when we take a side view of delay, jitter and loss. Up to now, many voice based evaluation algorithms have been used to measure speech quality of VoIP systems. However, these algorithms have the defects that their results are different according to voice samples and some algorithms can not take network environment for speech transmission path. The E-model can be used to solve the problems of these algorithms. In this paper. we introduce VoIP planning guidelines through the various analysis of E-model which can model impairments of network quality as well as VoIP equipment quality systematically, We, also, show the evaluation method and results of speech transmission quality.

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Analysis of Business Overview and use of 'C'group's Internet phone of National Information and Communication Services (국가정보통신서비스의 'C'그룹 인터넷전화 사업현황과 이용 분석)

  • Shin, Jin;Park, Dea-Woo
    • 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.

Use and Business Analysis of the 'C'Group Internet Phone on National Information and Communication Service (국가정보통신서비스의 'C'그룹 인터넷전화 사업현황과 이용 분석)

  • Shin, Jin;Park, Dea-Woo
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2011.10a
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    • pp.249-252
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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.

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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.

인터넷전화의 발전과정을 통한 사업자 별 경쟁전략;인터넷전화 시장의 ${\pi}$ 확대를 중심으로

  • Mun, Su-Hyeon;Han, Jae-Min
    • 한국경영정보학회:학술대회논문집
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    • 2007.06a
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    • pp.652-657
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    • 2007
  • 인터넷의 발전으로 많은 시스템과 서비스가 변화를 거듭하고 있는 가운데, 당초 큰 기대를 모았던 인터넷전화는 저렴한 이용료 이외의 많은 장점에도 불구하고 활성화가 더디게 진행되어 왔다. 주요 원인으로 기존 통신사업자들의 입장 및 정책적인 규제 등이 지적되면서 다양한 측면에서의 활성화 방안이 논의되고 있다. 한편 기존의 통신사업자 이외에 인터넷포탈 및 방송사업자들이 시장에 새롭게 진입하면서 인터넷전화 시장은 새로운 변화를 맞이하고 있으며, 그로 인해 인터넷전화 시장은 보다 복잡한 경쟁관계를 보이고 있다. 본 연구에서는 인터넷전화 시장의 발전과정에 대한 분석을 통해 새롭게 변화하고 있는 인터넷전화 시장의 경쟁 속에서 인터넷전화 사업자들에게 전략적인 제언을 하고자 한다.

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