• Journal of KIISE:Computing Practices and Letters
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• v.14 no.1
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• pp.101-105
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• 2008

This paper proposes an analytical model to calculate VoIP (Voice of IP) capacity over wireless LANs with frequent bit errors. Since the traditional analytical models for VoIP capacity have not included the effect of bit errors, simulations ould only evaluate VoIP capacity over erroneous channels. For analytically accurate estimation of VoIP capacity over noisy channels, we extend the conventional model to include the effect of propagation errors, end-to-end delay, voice quality, the waiting time in AP(Access Point). The experiments show that our model predicts the VoIP capacity of a given network within the range from 3% to 9% difference comparing with the simulation results.

• Journal of Advanced Navigation Technology
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• v.12 no.1
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• pp.28-33
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• 2008

VoIP provided voice service using Internet is receiving footlights when it escapes an initial curiosity. VoIP interest became larger, because it can transfer existing phone service and deliver voice data through internet technology. Is inheriting as it is threats of IP base because a VoIP technology provides audiocommunication service taking advantage of an existing IP technology, and have new threats that happen from new know-hows for VoIP service offer. In this paper, presented about problem and consideration and countermeasure of examines about security threat of VoIP service, and applies security.

• Proceedings of the Korean Information Science Society Conference
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• 2003.04d
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• pp.109-111
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• 2003

본 논문은 최근 활성화되고 있는 VoIP 서비스 기술에 이동성을 보장하기 위해 Mobile IP기술을 접목시킨 시스템을 설계하고 개발한다. PDA와 같은 임베디드 시스템의 제한된 자원을 효율적으로 이용하기 위한 VoIP 서비스용 시스템에 필요한 요구사항과 문제점들을 분석하고 해결책을 제시한다. 또 이동환경에서의 VoIP 서비스를 제공하기 위해 Mobile IP 통신 모듈을 선계하고 구현한다. 유무선 통합 실험망을 이용하여, 개발된 Mobile IP 기반 VoIP시스템의 기능을 검증하고 향후 방안을 제시한다.

• Journal of the Korea Society of Computer and Information
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• v.11 no.4 s.42
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• pp.155-164
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• 2006

VoIP technology is Eight New Services among Ubiquitous-IT839 strategies. This paper tested wiretapping or VoIP service in connected a soft phone and LAN and WAN sections, Internet telephones and a device. IP PBX, a banner operator network to have been connected to VoIP Internet network. As a result of having experimented on wiretapping of VoIP networks, Vulnerability was found. and a wiretapping by attacks of a hacker was succeeded in a terminal and proxy and attachment points of a VoIP network like a hub to follow a CVE list. Currently applied a security plan of an each wiretapping section in viewpoints of 6 security function of Access Control. Confidentiality, Authentication. Availability, Integrity. Non-repudiation in VoIP networks named to 070. Prevented wiretapping of contents by the results, the AES encryption that executed wiretapping experiment about a packet after application of a security plan. Prevented wiretapping, and kept security and audit log. and were able to accomplish VoIP information protection to network monitoring and audit log by an access interception and qualification and message hash functions and use of an incoming refusal.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6B
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• pp.575-584
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• 2009

In this paper, we design and implement a VoIP traffic generator for simulating call processing in IP contact center systems. Creating a VoIP call based on H.323 and SIP and generating RTP traffic which uses G.711 codec, the generator lets many users simulate situations on which they call each other. With this tool, which is named VoIPTG, users can combine H.323 or SIP session control protocol, the number of users, time variation, and voice codecs and then direct various situations for simulation. This traffic generator can be used for testing functions of an IP contact center and especially it is necessary for testing the quality of IP based call recording systems.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.795-797
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• 2009

Voice over Internet Protocol(VoIP) service becomes more and more popular nowadays. As such, it is developed over many kinds of network models, especially wireless networks. Mean Opinion Score(MOS) computes the QoS of VoIP service which should be supported by robust network environment. However, MANET is not stable enough to supply high MOS values for VoIP service. In this paper, VoIP service over MANET is simulated using ns-2(Network Simulation 2). In oder to get different MOS values in the results, we differentiate between network environments by adjusting the parameters of MANET.Through comparing the results we can know how to improve the QoS.

• Journal of KIISE:Information Networking
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• v.37 no.4
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• pp.263-271
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• 2010

SIP/RTP-based VoIP services are being popular. Recently, however, VoIP anomaly traffic such as delay, interference and termination of call establishment, and degradation of voice quality has been reported. An attacker could intercept a packet, and obtain user and header information so as to generate an anomaly traffic, because most Korean VoIP applications do not use standard security protocols. In this paper, we propose three VoIP anomaly traffic generation methods for CANCEL;BYE DoS and RTP flooding, and a detection method through flow-based traffic measurement. From our experiments, we showed that 97% of anomaly traffic could be detected in real commercial VoIP networks in Korea.

• Proceedings of the Korea Information Processing Society Conference
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• 2006.05a
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• pp.1243-1246
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• 2006

초고속 통신망의 확대 적용으로 인터넷의 빠른 성장과 함께 음성과 비디오 그리고 데이터를 통합하고자 하는 노력이 시도되고 있다. VoIP(Voice over IP)는 IP를 이용하여 음성과 데이터를 패킷 형태로 통합하여 실시간으로 전송하는 기술이다[1]. 패킷 네트워크에서 VoIP 시그널링 기술을 이용하면 망 자원으 효율적 이용 및 PSTN에 가까운 음질 그리고 인터넷과 연계한 다양한 음성서비스 지원이 가능하다. 콜센터에서도 VoIP를 사용하게 됨에 따라 VoIP망에서의 녹취 시스템이 필요하다. VoIP 녹취 시스템은 상담원과 고객 간의 통화 내용을 자동으로 녹음하여 보관함으로써 고객의 요구사항을 명확하게 파악할 수 있으며 녹취 데이터의 통계 자료 제공으로 효율적인 관리가 가능하고, 선택 녹취, 스케쥴링 녹취, 상담원의 평가 자료를 제공하여 고객 관리의 질적인 향상을 지원한다. 본 논문에서는 성능에 큰 영향을 주지 않고 기존의 VoIP 녹취 시스템의 문제점을 해결한 에이전트를 포함한 VoIP 녹취 시스템을 제안한다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.414-418
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• 2003

VoIP is a core technology used to transmit both voice and data in an integrated packet' form. Within this technology, SIP is the signaling protocol used for 'real time' call services; particularly those where H323 is used. Yet, when considering the needs of mobile users, it is essential we integrate VoIp within the mobile technology so the mobile host is able to receive the 'packet' transported and by, and connected to, any available internet-address. For all this to occur, we need to improve Network Delay by reducing transmission problems associated with mobile services. If we are to obtain an optimal service then we must reduce any network delays which may arise from joining Mobile IP and VoIp services. This paper, therefore, considers how, unlike previous research, these delays may be improved through the use of the signaling technology\ulcorner SIP. It also considers how this research may be introduced into current wired and wireless integrated services enabling them to use the IP 'packet'.

• 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.