• 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.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.482-483
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• 2011

본 논문에서는 스마트TV향 VoIP(Voice over Internet Protocol) 컨퍼런스 기능을 위한 잡음제거 알고리즘의 성능비교 하였다. 기존에 연구 되어져 있는 Improved Minima Controlled Recursive Averaging(IMCRA)방식과 Gaussian분포 기반의 잡음제거 알고리즘, IMCRA방식과 Gamma분포 기반의 잡음제거 알고리즘, IMCRA방식과 Mel-filter를 적용한 잡음제거 알고리즘, R&L 알고리즘들의 방식을 비교하였으며, 성능 비교를 위해 각 알고리즘을 통해 나온 다양한 잡음 환경에서의 잡음이 제거된 신호의 PESQ와 연산속도를 비교한다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.335-340
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• 2002

인터넷을 이용한 여러 응용 서비스들 중에서 저렴한 비용으로 음성을 전송할 수 있는 VoIP서비스의 발전으로 사용자의 급격한 증가가 예상된다. VoIP에 mobility, universal number, multiparty conference, voice mail, automatic call distribution과 같은 고품질의 서비스를 제공하기 위해서는 시그널링이 가능한 표준화된 프로토콜이 필요하다. 현재 IETF의 SW(Session Initiation Protocol)가 빠른 호 설정과 parsing 및 compile이 쉬운 장점으로 인해 SIP를 기반으로 한 VoIP 서비스를 제공하기 위해 국내외적으로 SIP 기반 구성요소에 대한 개발이 진행 중이다. 본 논문에서는 사용자가 보내는 request(INVITE) 메소드(method)를 처리해주는 SIP 서버의 부하 경감, 망 운용의 효율성, 많은 사용자에 대한 서비스를 제공하기 위해 새로운 서버 유형인 Hybrid형 SIP 서버를 제안하고, 이를 이용하여 새로운 타입의 인터넷폰을 설계하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.05a
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• pp.578-581
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• 2014

Blackhole attack, a kinds of attacks to routing function, can cause critical effects to network transmission function, Especially, on MANET(Mobile Ad-hoc Network) which it is not easy to prepare functions to respond malicious intrusion, transmission functions of entire networks could be degraded. In this paper, effects of blackhole attack to network transmission performance is analyzed on lattice structured MANET. Specially, performance is measured for various location of blackhole attack on lattice MANET, and compared with the performance of random structured MANET. This paper is done with computer simulation, VoIP(Voice over Internet Protocol) traffic is used in simulation. The results of this paper can be used for data to deal with blackhole attack.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.11
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• pp.1077-1082
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• 2016

IDS can be used as a countermeasure for malicious attacks which cause degrade of network transmission performance by disturbing of MANET routing function. In this paper, effects of IDS for transmission performance on MANET under grayhole attacks which has intrusion objects for a part of transmissions packets, some suggestion for effective IDS will be considered. Computer simulation based on NS-2 is used for performance analysis, performance is measured with VoIP(: Voice over Internet Protocol) as an application service. MOS(: Mean Opinion Score), CCR(: Call Connection Rate) and end-to-end delay is used for performance parameter as standard transmission quality factor for voice transmission.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.7
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• pp.1048-1054
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• 2002

인터넷을 기용한 여러 응용 서비스들 중에서 저렴한 비용으로 음성을 전송할 수 있는 VoIP 서비스의 발전으로 사용자의 급격한 증가가 예상된다. VoIP에 mobility, universal number, multiparty conference, voice mail, automatic call distribution과 같은 고품질의 서비스를 제공하기 위해서는 시그널링이 가능한 표준화된 프로토콜이 필요하다. 현재 IETF의 SIP(Session Initiation Protocol)가 빠른 호 설정 과 parsing 및 compile이 쉬운 장점으로 인해 SIP를 기반으로 한 VoIP 서비스를 제공하기 위해 국내외적으로 SIP 기반 구성요소에 대한 개발이 진행 중이다. 본 논문에서는 사용자가 보내는 Request(INVITE) 메소드(method)를 처리해주는 SIP 서버의 부하 경감, 망 운용의 효율성, 많은 사용자에 대한 서비스를 제공하기 위해 새로운 서버 유형인 Hybrid형 SIP 서버를 제안하고, 이를 이용하여 새로운 타입의 인터넷폰을 설계하였다.

• MALSORI
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• no.43
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• pp.101-112
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• 2002

In real time packetized voice application, missing packets are major source of voice quality degradation. Thus packet loss concealment (PLC) algorithms are needed to guarantee QoS of VoIP. In this paper, we describe packet loss concealment scheme utilizing the next good frame which follows loss packets. When this scheme is combined with other PLC algorithms, such as G.711 pitch waveform replication recommended by ITU-T LP based PLC algorithm, additional voice quality improvement is obtained for consecutive packet loss larger than 60 msec.

• The KIPS Transactions:PartC
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• v.9C no.3
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• pp.375-384
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• 2002

In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

• The Journal of the Korea Contents Association
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• v.4 no.2
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• pp.43-49
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• 2004

SIP Caller Preference is an useful function that allows a caller to express preferences about request handling in servers. It can also feat appropriate call processing according to the callee capabilities. However, only the category of the media is considered as a criteria for target selection in the caller preference. In this case, if the callee's media information such as codec is different from the caller, an additional re­negotiation occurs for SIP call setup. Therefore, in this paper, we have suggested an extended caller preference to solve this problem. In our SIP based VoIP system, a network sewer uses detailed media informations for media stream in the session to select the target for SIP call setup. The sewer gives higher priority to the candidate which do not require re-negotiation for call setup, so that an effective call setup can be achieved in our system.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.245-248
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• 2011

Voice over Internet protocol(VoIP) is call's contents using the existing internet. Thus, in common with the Internet service has the same vulnerability. In addition, unlike traditional PSTN remotely without physical access to hack through the eavesdropping is possible. Cyber terrorism by anti-state groups take place when the agency's computer network and telephone system at the same time work is likely to get upset. In this paper is penetration testing for security threats(Call interception, eavesdropping, misuse of services) set out in the NIS in the VoIP. In addition, scenario writing and penetration testing, hacking through the Voice over Internet protocol at the examination center will study discovered vulnerabilities. Vulnerability discovered in Voice over Internet protocol presents an attack and defense plan.