• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.1
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• pp.7-16
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• 2012

In this paper we describe a VoIP transcript system which is able to record call conversation between counselor and customer in an IP contact center based on IP telephony environment. The transcript system, designed and implemented in this paper, uses packet sniffering to capture packets without imposing network overhead on overall system. It can decode H.323 and SIP which are used to setup call sessions in VoIP environment and captures voice data and record without any loss of contents. Implemented transcript system can be integrated with CTI system in that it can manage and record call more effectively. It is designed generically so that it is implemented both on Windows and Linux environment.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.2
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• pp.245-252
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• 2011

VoIP transmits voices over IP-based networks and it is the abbreviation of Voice over Internet Protocol. Recently, VoIP provides various services in addition to voices. Since VoIP services' provision is extending, VoIP service quality management is becoming an important issue. Therefore, in this paper, we study VoIP service quality management. We examine VoIP technology, service types, and network architecture. Then, we investigate key quality indicators(KQIs)/key performance indicators(KPIs) in terms of customers, not network service providers. Toward this, we also study the concept of general service quality management as well as the concept of telecommunication related service quality management. Moreover, we apply $\bar{x}$ and R charts to show how to use statistical quality control techniques in real telecommunication companies with one KQI.

• Journal of the Korea Convergence Society
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• v.3 no.3
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• pp.7-12
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• 2012

mVoIP (mobile Voice over Internet Protocol) service is a technology to transmit voice data through an IP network using mobile device. mVoIP provides various supplementary services with low communication cost. It can maximize the availability and efficiency by using IP-based network resources. In addition, the users can use voice call service at any time and in any place, as long as they can access the Internet on mobile device easily. However, SIP on mobile device is exposed to IP-based attacks and threats. Observed cyber threats to SIP services include wiretapping, denial of service, and service misuse, VoIP spam which are also applicable to existing IP-based networks. These attacks are also applicable to SIP and continuously cause problems. In this study, we analysis the threat and vulnerability on mVoIP service and propose several possible attack scenarios on existing mobile VoIP devices. Based on a proposed analysis and vulnerability test mechanism, we can construct more enhanced SIP security mechanism and stable mobile VoIP service framework after eliminating its vulnerability on mobile telephony system.

• Electronics and Telecommunications Trends
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• v.17 no.2 s.74
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• pp.7-17
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• 2002

오늘날 대부분의 신기술은 통신규제를 우회하므로, 각국 정부는 규제를 포기하고 기술과 시장이 통신업종의 발전을 지향하는 방향으로 나아가고 있다. 그러나, IP Telephony와 관련된 최근의 동향은 기존의 법률과 규제가 각국에서 IP Telephony의 발전을 결정하는 강력한 요소임을 보여주고 있다. 본 고는 IP Telephony와 관련한 기술발전 추세와 표준화동향, 시장도입효과를 알아보고 특히, IP Telephony와 관련한 구체적인 도입방안을 위한 논의가 있었던 유럽연합과 프랑스를 중심으로 규제동향을 살펴보았다.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.10-12
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• 2002

WWW의 급격한 팽창은 인터넷을 사용하는 사용자들의 급격한 증가를 가져 왔으며, 인터넷 사용자의 증가로 IP Address의 부족문제가 발생하게 되었다. 이에 대한 해결책으로 서브넷, 사설 IP Address, Proxy Server, NAT등의 다양한 해결책들이 제시되어 부분적으로 적용하고 있으나, 사설 IP Address를 사용하는 경우 인터넷 서비스를 이용하는 것은가능하나 인터넷 서버로서의 역할을 수행할 수 없다. 본 논문에서 설계 구현한 Gate-P 시스템은 사설 IP Address를 가지는 시스템을 인터넷 서버로서 활용할 수 있도록 한다.

• Convergence Security Journal
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• v.10 no.3
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• pp.69-75
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• 2010

At present, deterioration in service quality, which caused by increased network traffic from a lot of users in the same network, makes smooth communication service difficult. In this paper, we present policy-based QoS management system for VoIP service in order to supply smooth VoIP service to users. The presented system can guarantee quality of VoIP service even in huge traffic situation, and makes it possible to operate whole network in an efficient manner.

• Proceedings of the Korea Electromagnetic Engineering Society Conference
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• 2000.11a
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• pp.20-24
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• 2000

VoIP는 Packet Netwark(ATM, xDSL, Frame Relay, Cable Network)망을 이용하여 음성데이터를 전송 하는 기술로서 PSTN을 통해 음성데이터를 전송하는 것보다 비용절감의 효과가 크다. 본 연구에서는 최적의 VoIP 서비스 제공을 위한 음성/데이터 통합 네트워크 망을 설계하기 위하여 IP와 ATM을 이용한 서로 다른 2개의 망을 설계하여 지연과 Routing 정책, Traffic 추가 후 지연현상에 대하여 살펴보았다. 지연은 순수한 VoIP 망을 구성 할 경우 8-10ms. VoIP+ATM으로 망을 구성 할 경우 2ms로 나타났고, 라우팅 정책(RIP, IGRP, OSPF 적용)에서는 IP 또는 IP+ATM으로 망을 구성 할 경우 RIP는 25ms, IGRP는 22ms로 나타났고, OSPF를 이용할 경우 14ms로 평가되어 OSPF를 이용한 라우팅 정책을 설정하는 것이 바람직하다고 볼 수 있다. 결론적으로 본 연구의 결과 VoIP망을 구성 할 경우 IP+ATM을 기반으로 구축하면 보다 더 효과적인 인터넷 망을 구성할 수 있음을 확인하였다.

• Journal of Korea Society of Digital Industry and Information Management
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• v.14 no.4
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• pp.117-125
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• 2018

Recently, VoIP technology is widely used in our daily life. Even VoIP has become a technology that can be easily accessed from services such as home phone as well as KakaoTalk.[1] Most of these Internet telephones use the RTP protocol. However, there is a vulnerability that the audio data of users can be intercepted through packet sniffing in the RTP protocol. So we want to create a tool to check the security level of a VoIP network using the RTP protocol. To do so, we capture data packet from and to these VoIP networks. For this purpose, we first configure a virtual VoIP network using Raspberry Pi and show the security vulnerability by applying our developed sniffing tool to the VoIP network. We will then analyze the captured packets and extract meaningful information from the analyzed data using the Google Speech API. Finally, we will address the causes of these vulnerabilities and possible solutions to address them.

• Proceedings of the Korea Information Processing Society Conference
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• 2009.04a
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• pp.1463-1465
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• 2009

인터넷망을 이용하여 음성 및 영상통화를 제공하는 VoIP(인터넷전화)서비스가 확대되고, VoIP 서비스가 인터넷망을 이용함에 따라 기존의 인터넷망에서 제기된 보안위협이 증가하고 사이버 공격에 노출되고 있다. 이에 따라 본 고에서는 국제 표준인 공통평가기준(ISO 15408)에서 규정된 방법에 따라 VoIP 서비스를 제공하는 IP Phone 및 IP PBX시스템이 노출되는 사이버 상의 보안위협사항을 도출하고, 보안목적과 VoIP 및 IP PBX 시스템 보안위협에 대응할 수 있는 보안요구사항을 정의하였다.

• Journal of IKEEE
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• v.18 no.3
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• pp.341-348
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• 2014

When several resuable semiconductor IPs are connected and implemented into an integrated chip, each semiconductor IP should provide code files for synthesis and interface modeling files for simulation and verification. However, description methods and levels of abstraction of interface modeling files are different because these semiconductor IPs are designed by different designers, which makes some problems in simulation and verification. This paper proposes a standardized modeling method of semiconductor IP interfaces. It restricts semiconductor IP interfaces to several predefined level of abstraction. The proposed method helps the chip integration designer to easily connect different semiconductor IPs and to simulate and verify them.