• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.409-411
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• 2002

VoIP(Voice over IP)는 저렴한 비용과 다양한 응용서비스 기능으로 기존의 전화서비스를 보조 또는 대체할 서비스로 인식되고 있다. 특히 SIP(Session Initiation Protocol)는 텍스트 기반으로 다양한 애플리케이션과 쉽게 연동될 수 있다는 장점 때문에 차세대 VoIP프로토콜로 급성장하고 있으며 이를 이용한 VoIP 망 구성 및 부가서비스에 대한 관심이 증대되고 있다. 부가서비스는 콜의 발생 여부에 따라 사용자의 콜을 처리하여 서비스를 제공하는 형태와 서버가 직접 콜을 발생시켜 서비스를 제공하는 형태의 두 가지 유형으로 나눌 수 있다. 현재 SIP기반 VoIP에서는 사용자의 콜을 처리하는 서비스를 위한 CPL(Call Processing Language)이 표준화 중이며 이를 기반으로 여러 서비스도 구현되어 있다. 그러나 특정시간에 콜을 발생시켜 사용자에게 서비스를 제공하는 형태의 서비스는 논의되지 않고 있으며 구현된 사례도 아직 없는 상황이다. 따라서 본 논문에서는 이러한 유형의 새로운 서비스의 하나로 예약통화 시스템을 제안한다. 예약통화 시스템은 예약된 시간에 콜을 발생시켜 두 사용자간의 통화를 연결해 주는 예약 시간 기반의 서비스를 개발하는데 이용될 수 있다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.5
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• pp.95-100
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• 2011

The next generation IP-PBX system are demanding multimedia facility to carry out UC(Unified Communication) and voice security also. Therefore, in this paper, we have developed the integrated solution of IP-PBX for the voice security by adopting VPN IPsec based on SIP using tunnel method in transmitting voice data to prevent eavesdrop of voice data and have shared between communication system based on PC and PSTN terminals. In particular, We have developed a video conference, private switching, distributed processing and measured telephone conversation quality. This IP-PBX that is connected Soft-phone provide various optional services.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.367-370
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• 2011

Tactical communication environment evolving for the purpose of providing services such as voice, video and text based on ALL-IP. Therefore, To be able to guarantee QoS level which meets the required level of subscriber for these services in the constrained tactical communication infrastructure, it is required to take the characteristics such as wireless transmission link, mobility of troops or personal into service quality scheme. In this paper, to support differentiated QoS for each individual or mission in the tactical communication environment, we presents a technique that can provide same QoS level which was served originally regardless of the situation to user's move through dynamically determining the QoS level to be provided at the time of the service request on VoIP-Switch.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.12
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• pp.2321-2328
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• 2006

Today, SIP standard (The signalling protocol for the Internet phone service) raises to be the standard technique because the expandability is high and complexity is low. It is widely investigated and actively advocated to use Si81a1 ring protocol for SIP in VoIP service. SIP service can be applied even outside the Internet phone service; instance messaging and various multimedia technology are just an example. This paper proposed an embodiment proxy server for rambling support to use JAIN SIP API. It provides standard interface for testing the Proxy server for SIP and embodiment of user agent that transfer instant massaging and voice communication.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.172-175
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• 2003

In this paper we propose a method fur the evaluation of the quality of service for VoIP services in NGN. Specifically, let us anatomize the elements of delay of a voice connection in the network in an end-to-end manner and investigate expected value at each point. We extract the delay time in each element in the network such as gateway, network node, and terminal equipment, and estimate an upper bound fur the tolerable delay in each element.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.7
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• pp.773-778
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• 2014

IDS(Intrusion Detection System) can be used as a countermeasure for blackhole attacks which cause degrade of transmission performance by causing of malicious intrusion to routing function of networks. In this paper, effects of IDS for transmission performance based on mobility patterns is analyzed for MANET(Mobile Ad-hoc Networks), a suggestion for effective countermeasure is considered. Computer simulation based on NS-2 is used in performance analysis, VoIP(Voice over Internet Protocol) as an application service is chosen for performance measure. MOS(Mean Opinion Score), call connection ratio and end-to-end delay is used as performance parameter.

• Proceedings of the Korea Contents Association Conference
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• 2009.05a
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• pp.884-888
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• 2009

Vehicular Adhoc Networks (VANET), a Vehicle-to-Infrastucture or Vehicle-to-Vehicle communication technology, is an area that makes more specific use of Mobile Adhoc Networks(MANET). VANET's Quality of Service(QoS) focuses on preventing possible emergencies like car crash from happening by immediately transmitting information to the cars around, while MANET's QoS is being studied for the quality of multimedia data such as Video on Demand(VoD), Video streaming, Voice over IP(VoIP), etc. In this paper, I structure the actual network configuration using Link State Routing(LSR), implement QoS Provisioning Model using Common Open Policy Service(COPS), and suggest more effective k-hop Cluster and inter-domain policy negotiation which fit better to the characteristics of VANET.

• Annual Conference of KIPS
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• 2023.11a
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• pp.947-948
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• 2023

본 논문은 보이스피싱 취약 계층을 위해 통화 내용을 신속하게 처리하여 실시간으로 범죄 여부를 판별하는 VoIP 에 특화된 시스템을 제안하였다. 실제 보이스 피싱 통화 유형을 학습한 탐지 모델을 개발하여 API 로 배포하였다. 또한 보이스피싱 위험도가 일정 수준에 도달할 경우 사용자에게 보이스피싱 가능성을 경고하는 장치를 제작하였다. 본 연구는 보이스피싱을 사전에 탐지함으로써 개인정보의 유출 및 금융 피해를 예방하고 정보 보안을 실천하는 데 기여할 것으로 기대된다.

• Proceedings of the Korea Multimedia Society Conference
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• 2012.05a
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• pp.353-354
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• 2012

광대역 VoIP 네트워크 환경에서는 불안정한 네트워크 환경으로 인해 음성패킷이 불규칙적으로 수신되어 음성데이터의 재생이 원활하지 못하다. 이러한 문제점을 해결하기 위해 본 논문에서는 네트워크 상태에 따라 원활하게 음성패킷을 재생시키는 스케줄링 방식을 제안한다. 제안하는 방식은 수신단에 도착한 패킷 헤더정보를 이용해 네트워크 지터를 추정하고, 추정된 지터와 지터버퍼와 음성프레임버퍼에 존재하는 패킷수 및 음성프레임 개수, 음성클래스정보에 따라 음성프레임의 길이를 변화시켜 재생시킴으로써 수신단의 버퍼링 지연을 줄이고 출력신호의 음성왜곡을 최소화한다. 제안하는 스케줄링 방식의 성능측정을 위해 버퍼링 지연과 PESQ를 기존 음성패킷 재생 스케줄링 방식과 비교한다.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.759-763
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• 2004

While the needs for VoIPs(Voice over IP) encourage the commercial trials for VoIP services, there are many problems such as user authentication, blocking of illegal user and eavesdropping. In this paper, a management algorithm of registration of VoIP terminals is explained and security methods for tolling and data encryption module is designed and built up. The module structure will have the advantages of the entire development of secured gatekeeper without whole modification of gatekeeper. In order to secure the ordinary gatekeeper based on H.323 standard, user authentication and data encryption technologies are developed based on the H.235 standard and simply located over the plain H.323 stacks. The data structures for secured communications are implemented according to ASN.1 structures by H.235.