• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7B
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• pp.660-667
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• 2006

Voice over Wireless LAN (VoWLAN) is defined as a communication service which transmits voice over wireless LAN. Due to its capability of providing high quality service while reducing communication costs, the VoWLAN is expected to be widely introduced to hospitals, hotels, and many more industry sectors. However, despite the increased interest in the new service, there has been limited research on the economic assessment of deploying VoWLAN. This paper provides a systematic and quantitative approach to the return on investment (ROI) analysis of VoWLAN service. We first analyze the domestic and international market trend of VoWLAN. Then, implementing scenarios of VoWLAN and corresponding alternatives are established. For each scenario, cost and benefit analysis is conducted. The proposed framework is implemented by using VBA program. To validate the result of the analysis, a case study is also conducted.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2012.10a
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• pp.642-645
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• 2012

MANET has disadvantage for information intrusion caused from non-infra structure. That is based on mixed problems caused from terminal devices has no capability of various resources using abilities to run intrusion prevention function, and also from difficulty of no easy using infra structural server like as firewall. In this paper, transmission performances, be effected from information intrusion and IDS(Intrusion Detection System), are analyzed for large scale MANET, and weakness from information intrusion of MANET are studied. This study is for large scale MANET which has some large communication area and lots of nodes, voice traffic, based on VoIP protocols, is considered as application services be transmitted over MANETs. Computer simulation using NS-2 is used to measure and show MOS and call connection ratios.

• 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.

• Journal of Communications and Networks
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• v.11 no.1
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• pp.72-83
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• 2009

We have studied the effect of head of the line blocking (HOLB) on session initiation protocol (SIP) session establishment delays. Our results are based on experiments performed in a test bed and on the public Internet. We used the stream control transmission protocol (SCTP) as a transport for SIP because SCTP can be configured to suffer or to avoid HOLB. Our experiments show that the effect of HOLB on session establishment delays generally starts to be significant starting at fairly low packet loss rates. However, there are scenarios where network conditions are good enough to make the effect of HOLB insignificant.

• IEMEK Journal of Embedded Systems and Applications
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• v.5 no.4
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• pp.185-194
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• 2010

In this paper, we propose an application layer-based vertical handover management protocol, called multihoming RTP (mhRTP), for real-time applications with seamless mobility across heterogeneous wireless access networks. The proposed multi-homing RTP provides a soft handover by utilizing multiple available wireless access network interfaces simultaneously. The newly available path is dynamically added to the ongoing session by the mhRTP session manager. Also the decision making of QoS-improving or QoS-guaranteed handover is possible based on the estimation of available bandwidth in each candidate network. The performances of the proposed mhRTP have been analyzed through a series of simulations on OPNET network simulator. From the performance analysis, we confirmed that the proposed mhRTP can provide QoS-guaranteed vertical handover with efficient session managements.

• Annual Conference of KIPS
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• 2003.11b
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• pp.1061-1064
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• 2003

SIP(Session Initiation Protocol) 기반의 VoIP(Voice over IP) 시스템의 구성 요소에서 레지스트라(Registrar)는 사용자들의 Registration 정보들을 저장 및 관리하는 역할을 한다. 네트워크상에서 이 Registration 상태는 관리자의 정책 등에 따라 동적으로 변화하게 되므로 사용자는 변경는 상태 정보를 알 필요성이 생긴다. Registration Event Package는 UA(User Agent)가 Subscriber로서 Registration 상태의 변경에 대한 통지를 요청할 수 있고 알림(Notification)을 통해 얻을 수 있는 매커니즘를 제공한다. 본 논문에서는 SIP 환경에서 이러한 Registration Event Package를 구현하여 사용자가 동적으로 변화되는 Registration 상태 정보를 알림을 통해 받을 수 있도록 하였다.

• Annual Conference of KIPS
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• 2002.11b
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• pp.1419-1422
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• 2002

SIP(Session Initiation Protocol)는 기존의 VoIP(Voice over IP)를 위한 시스템에서 뿐만 아니라 호설정의 처리와 제어가 필요한 여러 인터넷 응용분야에 적용이 가능한 확장성이 뛰어난 프로토콜이다. 또한 HTTP(Hyper Text Transfer Protocol)와 유사한 텍스트 기반의 응용계층의 프로토콜로서 호처리시 절차가 비교적 간결하다. 본 논문에서는 이러한 SIP의 장점을 이용하여 화상회의를 위한 시스템을 구현하였다. 이때. 인증 기능을 부여하여 화상회의시 세션 연결을 위한 호설정을 이루는 과정에서 상대방을 인증할 수 있도록 하였고 인증방법으로는 메시지 인증기능과 함께 리플레이 공격(replay attack)의 방지기능을 가진 SIP 다이제스트 인증(Digest Authentication) 방법을 사용하였다. 본 화상회의 시스템에 사용된 SIP 사양은 최근에 발표된 RFC 3261을 기준으로 하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.875-879
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• 2002

CPL is a technique that serves various additional service in Internet telephony such as call forwarding, call blocking etc. IETF IPTEL working group developed this CPL standard. Users could request various additional services such as call forwarding, call blocking etc. by registering XML scripts to location servers. This paper would describe the design and the implementation skill of SIP proxy server that support these improved functionalities in detail. SIP registrar and SIP proxy server are designed and implemented in Linux platform because this platform serves fast and low cost development environment.

• Annual Conference of KIPS
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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.