• Title/Summary/Keyword: VoIP(Voice over IP) Service

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The comparison of SIP Quality of Service Mechanism using RSVP (RSVP를 이용한 SIP QoS 메커니즘 연구)

  • Park, Jung-Jin;Youn, Hyung-Un;Kahng, Hyun-Kook
    • Proceedings of the Korea Information Processing Society Conference
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    • 2002.04b
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    • pp.1547-1550
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    • 2002
  • VoIP(Voice over IP) 기술을 이용한 음성 통신은 H.323과 SIP(Session Initiation Protocol)를 기반으로 빠른 확산과 기술의 발달로 점점 더 보편화되어 가고 있다. H.323에 비해 보다 간편한 SIP 프로토콜이 각광을 받고 있으나, 이것 또한, PSTN망의 전화보다는 통화품질이 떨어진다. 따라서, 실시간 음성 트래픽을 사용하는 SIP의 통화 품질의 향상을 위해 현재 나와있는 QoS 메커니즘 중에, RSVP를 이용한 SIP QoS 메커니즘을 연구하였다. 본 연구에서는 RSVP를 이용한 메커니즘 중에, RSVP 시그널링을 우선해주는 QoS Assured 확장 메커니즘에 대해서 살펴보고, 이것의 구현 방안을 제시한다.

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Implementation of Watcher Information Service in SIP-based Presence System (SIP기반의 Presence 시스템에서의 Watcher 정보 서비스 구현)

  • 조현규;장춘서
    • Proceedings of the Korean Information Science Society Conference
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    • 2003.04d
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    • pp.214-216
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    • 2003
  • SIP(Session Initiation Protocol)는 사용자간에 멀티미디어 세션을 생성, 수정, 종료를 위한 응용계층의 시그널링 프로토콜로써 확장성이 뛰어나 여러 인터넷 응용분야에 적용이 가능하다. SIP를 이용한 확장 서비스의 하나인 Presence 서비스는 사용자들간에 현재 위치의 변화에 따른 최신 정보를 제공함으로써 VoIP(Voice over IP), 인스턴트 메시징(Instant Messaging) 등 Presence 정보를 기반으로 하는 여러 응용 서비스와 연계하여 활용이 가능하다. Presence 서비스에서의 Watcher 정보는 특정한 자원(resource)에 대한 모든 등록(subscription) 상태를 의미하며 이 상태는 동적(dynamic)으로 변화하게 된다. 따라서 사용자는 이러한 변화되는 상태를 알기 위하여 watcher 정보에 등록할 필요가 생긴다. 따라서 본 논문에서는 SIP 기반의 Presence 시스템에서 동적으로 변화하는 사용자들의 등록 상태를 알려주는 Watcher 정보 서비스를 구현하였다.

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The Interoperability Issue in Broadband Convergence network Implementation (광대역통합망 구축에서 상호운용성 이슈)

  • Lee, Jae-Jeong;Ryu, Han-Yang;Nam, Ki-Dong;Kim, Chang-Bong
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.48 no.2
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    • pp.57-64
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    • 2011
  • The NGN (Next Generation Network) means the kernel infrastructure technology to provide information and communication services which are able to be used at present and future when a ubiquitous computing era has been realized. In other words, NGN can be the frame providing the same information and communication services anytime and anywhere regardless of wire and wireless. The broadband convergence network that has been built in the public institution has established a broadband multimedia communication network supporting voice telephone, task net, internet network, video conference network, voice over IP (VoIP) network and etc. It is possible for a requested bandwidth and services to be served, only if a broadband convergence network provide the interoperability between the various classes which include a transport network layer, network control layer, service control layer and other layers. In this paper, we analyzed the interoperability issues of the present broadband convergence network and propose a guideline for the future one.

Measurement Scheme for One-Way Delay Variation with Detection and Removal of Clock Skew

  • Aoki, Makoto;Oki, Eiji;Rojas-Cessa, Roberto
    • ETRI Journal
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    • v.32 no.6
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    • pp.854-862
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    • 2010
  • One-way delay variation (OWDV) has become increasingly of interest to researchers as a way to evaluate network state and service quality, especially for real-time and streaming services such as voice-over-Internet-protocol (VoIP) and video. Many schemes for OWDV measurement require clock synchronization through the global-positioning system (GPS) or network time protocol. In clock-synchronized approaches, the accuracy of OWDV measurement depends on the accuracy of the clock synchronization. GPS provides highly accurate clock synchronization. However, the deployment of GPS on legacy network equipment might be slow and costly. This paper proposes a method for measuring OWDV that dispenses with clock synchronization. The clock synchronization problem is mainly caused by clock skew. The proposed approach is based on the measurement of inter-packet delay and accumulated OWDV. This paper shows the performance of the proposed scheme via simulations and through experiments in a VoIP network. The presented simulation and measurement results indicate that clock skew can be efficiently measured and removed and that OWDV can be measured without requiring clock synchronization.

A SIP Extension Method for Closed Multiparty Conference with Guarantee of Security (비공개형 다자간 컨퍼런스의 보안성 확보를 위한 SIP 확장 기법)

  • 김현태;김형진;나인호
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.8 no.2
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    • pp.331-337
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    • 2004
  • Multiparty conference service based on SIP supported by VoIP network is gradually increased in use and the continuous development and standardization works on SIP are in the process of advancing. But SIP used in currently does not support identity discovery and distribution of each participant for multipath conference. In this paper, a SIP extension method for guaranteeing security from the multiparty conference based on SIP is proposed. We design a new SIP header and method for discovering and distributing a participant's identity in closed multiparty conference when the call initiation is established. And it can ensure that each participant is notified before a new participant joins.

Transmission Performance of Lattice Structure Ad-Hoc Network under Intrusions (침해가 있는 격자구조 애드-혹 네트워크의 전송성능)

  • Kim, Young-Dong
    • The Journal of the Korea institute of electronic communication sciences
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    • v.9 no.7
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    • pp.767-772
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    • 2014
  • As temporary network, ad-hoc network has been effected by structures and implemented environments of networks. In this paper, transmission performance of lattice structure ad-hoc network, which is expected to use in sensor network and IoT(Internet of Things), is analyzed in point of intrusions and countermeasure for intrusion is suggested. In this paper, computer simulation based on NS-2 is used for performance analysis, VoIP(Voice over Internet Protocol) as a widely used service is chosen for performance measure. MOS(Mean Opinion Score) and call connection rate is used as performance parameter. As results of performance analysis, it is shown that for MOS, random network is better then lattice network at intrusion environments, but for call connection rate, lattice network is better then random network.

A Study on Voice over Internet Protocol Security Response Model for Administrative Agency (행정기관 인터넷전화 보안 대응 모델 개발 연구)

  • Park, Dea-Woo;Yang, Jong-Han
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2011.10a
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    • pp.237-240
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    • 2011
  • Voice over Internet Protocol calls using administrative agency to build a national information and communication service, 'C' group, providers, the KT, SK Broadband, LG U+, Samsung SDS, as there are four operators. To prepare for an attack on Voice over Internet Protocol for administrative agency, security is a need for research to support the model. In this paper, the Internet telephone business of Administrative Agency to investigate and analyze the specific security measures to respond. Should set priorities around confidentiality about five security threats from NIS to Study of Voice over Internet Protocol Security Response Model for Administrative Agency. (1) Illegal wiretapping, (2) call interception, (3) service misuse, (4) denial of service attacks, (5) spam attacks, write about and analyze attack scenarios. In this paper, an analysis of protection by security threats and security breaches through a step-by-step system to address the research study is a step-by-step development of the corresponding model.

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Packet Delay and Loss Analysis of Real-time Traffic in a DBA Scheme of an EPON (EPON의 DBA 방안에서 실시간 트래픽의 패킷 손실률과 지연 성능 분석)

  • Shim, Se-Yong;Park, Chul-Geun
    • Proceedings of the KIEE Conference
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    • 2004.11c
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    • pp.86-88
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    • 2004
  • As the rapid incensement of the number of internet users has occurred recently, many multimedia application services have been emerging. To improve quality of service, traffic can be suggested to be classified with priority in EPON(Ethernet Passive Optical Network), which is supporting the multimedia application services. In this paper, multimedia application services treat bandwidth classifying device in serving both delay sensitive traffic for real-time audio, video and voice data such as VoIP(Voice over Internet Protocol), and nonreal-time traffic such as BE(Best Effort). With looking through existing mechanisms, new mechanism to improve the quality will be suggested. The delay performances and packet losses of traffic achieved by supporting bandwidth allocation of upstream traffic in suggested mechanism will be analyzed with simulation.

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Classification of BcN Vulnerabilities Based on Extended X.805 (X.805를 확장한 BcN 취약성 분류 체계)

  • Yoon Jong-Lim;Song Young-Ho;Min Byoung-Joon;Lee Tai-Jin
    • The KIPS Transactions:PartC
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    • v.13C no.4 s.107
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    • pp.427-434
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    • 2006
  • Broadband Convergence Network(BcN) is a critical infrastructure to provide wired-and-wireless high-quality multimedia services by converging communication and broadcasting systems, However, there exist possible danger to spread the damage of an intrusion incident within an individual network to the whole network due to the convergence and newly generated threats according to the advent of various services roaming vertically and horizontally. In order to cope with these new threats, we need to analyze the vulnerabilities of BcN in a system architecture aspect and classify them in a systematic way and to make the results to be utilized in preparing proper countermeasures, In this paper, we propose a new classification of vulnerabilities which has been extended from the ITU-T recommendation X.805, which defines the security related architectural elements. This new classification includes system elements to be protected for each service, possible attack strategies, resulting damage and its criticalness, and effective countermeasures. The new classification method is compared with the existing methods of CVE(Common Vulnerabilities and Exposures) and CERT/CC(Computer Emergency Response Team/Coordination Center), and the result of an application to one of typical services, VoIP(Voice over IP) and the development of vulnerability database and its management software tool are presented in the paper. The consequence of the research presented in the paper is expected to contribute to the integration of security knowledge and to the identification of newly required security techniques.

Implementation of Analysis System for H.323 Traffic (H.323 트래픽 분석 시스템의 개발)

  • Lee Sun-Hun;Chung Kwang-Sue
    • The KIPS Transactions:PartC
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    • v.13C no.4 s.107
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    • pp.471-480
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    • 2006
  • Recently, multimedia communication services, such as video conferencing and voice over IP, have been rapidly spread. H.323 is an international standard that specifies the components, protocols and procedures that provide multimedia communication services of real-time audio, video, and data communications over packet networks, including IP based networks. H.323 is applied to many commercial services because it supports various network environments and has a good performance. But communication services based on H.323 may have some problem because of current network trouble or mis-implementation of H.323. The understanding of this problem is a critical issue because it improves the quality of service and is easy to service maintenance. In this paper, we implement the analysis system for H.323 protocol wihch includes H.245, H.225.0, RTP, RTCP, and so on. Tills system is able to capture, parse, and present the H.323 protocol in real-time. Through the operation test and performance evaluation, we prove that our system is a useful to analyze and understand the problems for communication services based on H.323.