• Title/Summary/Keyword: VoIP Conference

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VoIP service performance improvements of MANET based on Diffserv (Diffserv기반 MANET에서 VoIP 서비스 성능 개선)

  • Yoon, Tong-il;Kim, Young-Dong
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2009.10a
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    • pp.777-779
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    • 2009
  • Study for VoIP in MANET is mainly focused on transmission performance or voice quality of VoIP traffic as single traffic. In this study, VoIP performance is analyzed of MANET which has several types of traffics. Diffserv scheme is used to process the traffics. It is showed that VoIP traffic performance is improved by differv scheme, as a result of analysis for VoIP and FTP traffic. Simulator, which is implemented with NS-2 package and VoIP patch, is used.

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Design and Implementation of Multipoint VoIP using End-point Mixing Model (단말혼합 방법을 이용하는 다자간 VoIP의 설계 및 구현)

  • Lee, Sung-Min;Lee, Keon-Bae
    • Journal of Korea Multimedia Society
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    • v.10 no.3
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    • pp.335-347
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    • 2007
  • VoIP (Voice over IP) is a technology to transport video and voice traffic over IP networks such as Internet. Today, the VoIP technology is viewed as the right choice for providing voice, video, and data communication among various terminals over the next generation network. This paper discusses a multipoint VoIP implementation with end-point mixing model which can support multipoint conference without a conference bridge. The multipoint VoIP is implemented with SIP (Session Initiation Protocol), and supports STUN (Simple Traversal of UDP Through NATs) since it works in an asymmetric NAT (Network Address Translator) environment. The characteristics of this paper are as follows. It is possible that all terminals in the hierarchical conference don't receive the duplicated media information because we use the end-point mixing model with the new media processing module. And, the paper solves the problem that the hierarchical conference session should be separated into several sessions when a mixing terminal terminates the hierarchical conference session.

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A Study on the Next Generation VoIP Network Architecture (차세대 VoIP 망 구조)

  • 윤태상;정성호;이일진;강신각
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2002.11a
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    • pp.839-843
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    • 2002
  • In this paper, we present a next generation VoIP network architecture. Specifically, we present key components such as SoftSwitch and media gateway, and important protocols for VoIP services. We also present basic components and mechanisms for VoIP QoS. The architecture presented in this paper is able to support open interfaces and multimedia traffic, and therefore various multi-services can be supported efficiently using the architecture.

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The development of a VoIP system with Mobile IP in all-IP networks (유무선통합 All-IP망에서 Mobile IP기반 VoIP시스템 개발)

  • 김경수;조계호;김원태;박용진
    • 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.

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A Study on Countering SIP-based VoIP Spam using VoIP-RBL (VoIP-RBL을 이용한 SIP기반 VoIP스팸 차단 방법)

  • Yoon, Seok-Ung;Jung, Hyun-Cheol;Park, Hae-Ryoung;Won, Yoo-Jae;Yoo, Hyeong-Seon
    • Proceedings of the Korean Information Science Society Conference
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    • 2011.06d
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    • pp.135-136
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    • 2011
  • The more VoIP service is widely used, the more VoIP spam becomes threatened. Both VoIP spam violates the user's privacy and VoIP spam can cause money trouble. Therefore, it is important to reduce the VoIP spam but it is not easy to adopt some useful techniques to counter e-mail spam due to VoIP characteristics. We propose a technique using VoIP-RBL for countering SIP-based VoIP spam.

Study of the Simulation of VoIP Traffic Generation with Considering Self-Similiarity (자기유사성을 고려한 VoIP 트래픽 생성 시뮬레이션 방법 의 연구)

  • 김윤배;이계신;김재범
    • Proceedings of the Korea Society for Simulation Conference
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    • 2004.05a
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    • pp.25-29
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    • 2004
  • VoIP는 인터넷 프로토콜(IP)를 이용하여 음성을 데이터 packet처럼 전송하는 것을 의미한다. 최근 VoIP 기술의 도입으로 기존 망 성능 관리에 대한 관심이 높아지고 있다. 보다 원활한 기술 구현을 위해서는 VoIP 트래픽에 대한 체계적인 분석과 위험성 검증을 할 수 있는 도구가 필요하다. 또한 기존의 트래픽 시뮬레이션 기법에서 실제 망에서의 자기유사성을 적용한 사례가 적다는 것 또한 본 연구가 행하여진 동기이다. 본 연구에서는 자기유사성을 반영하여 소량의 샘플을 갖고 전체 VoIP 망 트래픽을 생성할 수 있는 방법론을 개발하고자 시도하였다.

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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.

An Agent Based IP Transcript System in VoIP Network (VoIP망에서 Agent 기반 IP 녹취 시스템)

  • Lim Jae-Jin;Kim Soo-Hee;Jung In-Sang;Jung In-Hwan
    • 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.

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QoS Guaranteed System for Multi-functional VoIP End Terminal (복합 기능 VoIP 단말을 위한 음성 품질 보장 시스템)

  • 김대호
    • Proceedings of the IEEK Conference
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    • 2003.11c
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    • pp.153-156
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    • 2003
  • In this paper, we propose QoS guarantee system fur multi-functional VoIP end Terminal. This system guarantees low delay of voice data for Internet telephony in VoIP end terminal that has various kinds of Internet dependant application. QoS system we propose support low delay transmission in VoIP terminal interface.

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Improvement of VoIP Service over Mobile Ad-Hoc Network (MANET 기반 VoIP 서비스 성능 개선)

  • Ming, Li;Kim, Young-Dong
    • 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.

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