• Title/Summary/Keyword: IP-PBX server

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Study on Design of IP PBX of Distribute Base on SIP Protocol Stack (SIP프로토콜 스텍을 기반으로 하는 분산형 IP PBX 단말기 설계)

  • Yoo Seung-Sun;Yoo Gi-Hyoung;Lim Pyung-Jong;Hyun Chul-Ju;Kwak Hoon-Sung
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.4A
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    • pp.377-384
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    • 2006
  • According to fast VoIP technology development, more and more companies change voice network into IP based network among branch offices. IP PBX, which is deployed up to now, composed of IP phone and VoIP Gateway. Every telphone has replaced with If phone which support VoIP and VoIP gateway is installed in PBTN connection point to relay voice data. It can reduce the communication expense of International call, long distance call and call between a headquater and a trance because it uses internet line. In this paper, IP PBX is implemented that can distribute call using PBX network only usig personal terminal without Proxy Server. Depending on Role, terminal can be registered Master, Server and Client and it is verified in terms of performance and validation.

Development of a VoWLAN Terminal based on Open Source Software (공개 소스 소프트웨어 기반의 VoIP 서비스를 위한 무선단말 개발)

  • Suh, Hyo-Joong;Lee, Byung-Ho;Kim, Tae-Hyoun
    • The KIPS Transactions:PartD
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    • v.14D no.5
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    • pp.565-572
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    • 2007
  • In this paper, we developed a VoWLAN(Voice over WLAN) system based on an open source software. The system aims to provide VoIP service over wireless LAN with an IP-PBX server. The features of system presented in this paper are as follows. First, the initial cost for the development is reduced since the system is developed based on open source software. Second, the system provides various additional services such as Voice Mail, Conference Call, and Interactive Voice Response with a software IP-PBX server. Third, the VoWLAN terminal provides high-level user applications with minimal system resources using lightweight open software solutions. Finally, it is highly scalable since it is based on the open source software.

IP-PBX System of RasPBX-Based (RasPBX 기반의 IP-PBX 시스템)

  • Jeong, Dae-Jin;Song, Hyun-Ok;Jung, Hoe-kyung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.19 no.5
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    • pp.1131-1136
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    • 2015
  • VoIP and IP Telephony telephony technology development is a growing by easy to using IP-PBX by using phone from using existing lines rather than the internet. IP-PBX do not use the phone line from phone work for many companies and institutions of management costs reduce as provides similar to regular phone line quality. But IP-PBX to introduce for need to be the initial cost on is should buy for expensive hardware equipment or commercial software. In this paper, suggest way to introduce IP-PBX do not buy expensive hardware equipment or commercial software. Suggest IP-PBX on designed and implement for IP-PBX server using Raspberry Pi and Asterisk. And verification treatise on the suitability of conducted by voice calls based on IP-PBX between PC and a Smartphone

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.

Implementation of Safety management broadcasting system for IoT based in IP PBX (IP PBX기반 안전관리 IoT 방송 시스템 구현)

  • Kim, Sam-Taek
    • Journal of the Korea Convergence Society
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    • v.10 no.8
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    • pp.9-14
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    • 2019
  • Currently, with the success of 5G commercialization, a server system that integrates various Internet public safety services should be developed. In this paper, we developed a public safety integrated server, which is an IoT platform connecting IoT device and IoT gateway based on IP PBX. This server is based on embedded OS and various IoT services are executed in one system and call processing / broadcasting server function that processes emergency call and emergency broadcasting in public places is built in. This system collects IoT sensor data and emergency bell information and automatically sends out emergency alarms, emergency evacuation broadcasts, etc. at an accident site in an emergency situation, and transmits the daily information to the upper IoT service server, Provide public safety management services.

Design of B-ISDN Protocol embedded Internet-Phone PBX System (B-ISDN 프로토콜 내장의 인터넷 폰 교환시스템 설계)

  • Choi Jae-Won
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.10 no.5
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    • pp.821-831
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    • 2006
  • In this paper we researched on the development methods and design of the B-ISDN protocol embedded Internet-Phone PBX System for the previous telephone exchange and communication. We designed the structure of the user terminal md exchange server of the Internet-Phone PBX System, defined message and data structure for call control, and designed call control message flows between sender and receiver terminal for user's communication.

Design and Verification Test of Virtualized VoIP to support Secured Voice Communication (음성 보안을 제공하기 위한 가상화 기반의 VoIP 설계 및 검증 테스트)

  • Cha, Byung-Rae;Park, Sun;Kim, Jong-Won
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.18 no.10
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    • pp.2462-2472
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    • 2014
  • Recently, the computing paradigm has been changing and VoIP technology is being revisited to support various services. In this paper, we have designed and implemented the system of software PBX open source Asterisk, hardware platform, and mobile devices to support secured voice service based on VoIP. Specially, we designed the various platform from single board to servers based on XenServer in hardware platform. And we verified the delay test of network traffics and the secured voice communication test based on this platform.

UC(Unified Communication) Systems Development using Mobile Application (Mobile Application을 이용한 UC(Unified Communication) 시스템 개발)

  • Kim, Hee-Chul
    • The Journal of the Korea institute of electronic communication sciences
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    • v.8 no.6
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    • pp.873-879
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    • 2013
  • In this paper, high-quality business-type communications(UC) capabilities of the communication activities overlap, waste, reducing rework process improvement provides for high efficiency. Messages sent via UC app development, FMC calling features, schedule management organization for the development and deployment DataBase UC server deployment, the search for the JSP implementation, XMPP is using the messaging system. IP-PBX running on the IP network, on the basis of UC applications in real life, improve utilization of the infrastructure necessary to provide services to the system design and implementation.

Voice and Video Call Continuity for Enterprise Users (기업형 사용자들을 위한 음성/영상 서비스 이동성 제공 방안)

  • Jung, Chang-Yong;Kim, Hyeon-Soo;Moon, Jeong-Hyeon;Kim, Hee-Dong
    • 한국정보통신설비학회:학술대회논문집
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    • 2009.08a
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    • pp.99-103
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    • 2009
  • Recently, as wired and wireless communication services have rapidly developed and multimodal mobile devices which have various characteristics have widely spread, the need for new convergence services increases. The growing population of VoIP technologies and the high communication expense yield that the market of IP based telephony such as WiFi phone and IP phone is substituted for one of the conventional PSTN telephony. With the help of this trend, the wireline network operators desire to find a market in mobile networks. Therefore, they focus on Fixed Mobile Convergence (FMC) service as one of the key factors to accomplish this goal. FMC services are able to provide the mobility of voice services between circuit switched and packet switched networks. IP Multimedia Subsystem (IMS) based Voice Call Continuity (VCC) is one of the schemes to embody FMC services. As Application Server (AS) which has this VCC function provides seamless handover of services between heterogeneous networks, FMC subscribers can communicate seamlessly with others m WiFi domain and COMA domain using WiFi-COMA dual phone. Most of enterprises have already introduced IP network infrastructure and IP-PBX (Private Branch eXchange) for telephony. However, the problems of high communication cost and work inefficiency due to frequent outside jobs or business trips have remained. In order to solve these problems, demands for enterprise FMC services increase. In this paper, we introduce a new IP-PBX based VCC model that can provide seamless handover of voice services between WiFi and COMA networks for enterprise users and we investigate some interworking and security issues between Soft Switch (SSW) and IMS, or between IMSs. In addition, we introduce a new service that can provide the continuity of voice sessions as well as video sessions using Multimedia Session Continuity (MMSC) technology which has evolved from VCC. This service is expected to be one of the next-generation personalized services based on user's context.

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