• Journal of Korea Society of Industrial Information Systems
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• v.7 no.5
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• pp.11-20
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• 2002

The tendency of the technology that is aggregated voice and data circuit is VoP, especially VoIP. But if we try VoP, we have to throw away well structured and well known TDM resources. In this case CLEC and ILEC will encounter lots of troubles about standardization of progress of network evolution, heavy cost of network construction and so on. Therefore in this paper, I suggest designing VoDSL terminal and its operation environment (such as DSLAM) based on TDM. This designed VoDSL terminal can be installed with the DSLAM between PSTN and ISP without any additional facilities. This is more convenient low cost, operation and easier to construct than VoP.

• Journal of Communications and Networks
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• v.11 no.1
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• pp.36-41
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• 2009

The IEEE 802.16 wireless metropolitan area network (WMAN) standard is designed to correct expensive communication costs in CDMA-based mobile communication systems and limited coverage problems in wireless LAN systems. Thus, the IEEE 802.16e standard can provide mobile high-speed packet access between mobile stations and the Internet service provider through the base station with cheap communication fees. To efficiently accommodate voice over IP (VoIP) services in IEEE 802.16 systems, an uplink quality of service packet-scheduling scheme is proposed, and its performance is evaluated with an NS-2 network simulator in this paper. Numerical results show that this proposed scheme can increase the system capacity by 100% more than in the unsolicited rand service (UGS) scheme and 30% more than the extended real-time polling service (ertPS) scheme, respectively.

• Annual Conference of KIPS
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• 2011.04a
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• pp.611-614
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• 2011

본 논문에서 우리는 이종사업자망이 연동된 통합망 환경에서 네트워크 성능 저하 구간을 탐색하기 위해 RTCP(Real-time Transport Control Protocol)의 타임스탬프 정보를 이용한 네트워크 구간별 지연 시간을 측정할 수 있는 트래픽 모니터링 방안을 제안한다. 실시간 멀티미디어 서비스(IPTV, VoIP)의 이용이 증가함에 따라 이종망간 연동 환경에서 실시간 서비스에 대한 QoS 관리 방안이 반드시 필요하다. 영상회의, VoIP(Voice over IP) 및 IPTV 서비스와 같은 멀티미디어 서비스는 네트워크 성능(지연, 지연변이 및 패킷 손실)에 매우 민감하기 때문에 연동망 환경에서 서비스 품질이 저하될 경우 어느 네트워크 구간에서 성능 저하가 발생하였는지 탐색하는 것은 매우 중요한 문제이다. 이에 우리는 RTCP 패킷을 이용한 구간별 지연시간 측정 방안을 제안하며 이 방안을 통해 네트워크 성능 저하가 발생한 구간을 탐색하고 정의할 수 있다.

• Journal of Communications and Networks
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• v.9 no.4
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• pp.446-456
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• 2007

Increased access to broadband networks has led to a fast-growing demand for voice and video over IP(VVoIP) applications such as Internet telephony(VoIP), videoconferencing, and IP television(IPTV). For pro-active troubleshooting of VVoIP performance bottlenecks that manifest to end-users as performance impairments such as video frame freezing and voice dropouts, network operators cannot rely on actual end-users to report their subjective quality of experience(QoE). Hence, automated and objective techniques that provide real-time or online VVoIP QoE estimates are vital. Objective techniques developed to-date estimate VVoIP QoE by performing frame-to-frame peak-signal-to-noise ratio(PSNR) comparisons of the original video sequence and the reconstructed video sequence obtained from the sender-side and receiver-side, respectively. Since processing such video sequences is time consuming and computationally intensive, existing objective techniques cannot provide online VVoIP QoE. In this paper, we present a novel framework that can provide online estimates of VVoIP QoE on network paths without end-user involvement and without requiring any video sequences. The framework features the "GAP-model", which is an offline model of QoE expressed as a function of measurable network factors such as bandwidth, delay, jitter, and loss. Using the GAP-model, our online framework can produce VVoIP QoE estimates in terms of "Good", "Acceptable", or "Poor"(GAP) grades of perceptual quality solely from the online measured network conditions.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2006.07a
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• pp.662-662
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• 2006

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.4
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• pp.83-89
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• 2011

Nowadays, Internet Telephony Services based on SIP are gaining an explosive increase for general user. Thus a various demands of the user about convenience of IP-PBX are growing. The most important requirements of these is hybrid IP telephony system which is connected not only internet telephone but also general PSTN telephone. Therefore, in this paper, we have developed Hybrid IP-PBX connected all type of telephony terminals for small office. We have developed FXS module for connecting all type and FXO for using PSTN telephone number also. This Hybrid IP-PBX that is can be connected Soft-phone provide various optional services. We measured voice quality and test simultaneous calling for proof of validity.

• ETRI Journal
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• v.38 no.5
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• pp.922-933
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• 2016

IP header compression schemes offer a valuable measure for bandwidth preservation. Such schemes have been practically implemented in infrastructure-based IP networks for point-to-point links. However, minimal research and practical implementation efforts have been conducted in the direction of an IP header compression strategy that can meet the peculiar requirements of multi-hop ad hoc wireless networks. In this paper, we present a practically implemented multi-hop IP header compression scheme using the Robust Header Compression (ROHC) protocol suite. The scheme runs on a novel identifier (ID) based networking architecture, known as an ID-based ad hoc network (IDHOCNET). IDHOCNET additionally solves a number of bottlenecks of pure IP-based ad hoc networks that have emerged owing to IP address auto-configuration service, distributed naming and name resolution, and the role of an IP address as an identifier at the application layer. The proposed scheme was tested on a multi-hop test bed. The results show that the implemented scheme has better gain and requires only O (1) ROHC contexts.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.12B
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• pp.1135-1144
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• 2003

In this paper, we proposed a semi-soft handoff mechanism to provide link mobility in IEEE 802.11 wireless LAN environment. Buffers and routing tables in APs and portals are provided in order to reroute frames, which have not been received during handoff time and have been buffered in an old AP, to a new AP after handoff is performed. For the re -routing operation, the MAC routing table should be updated by exchanging information of a mobile terminal between neighbor APs. With our proposed scheme. a wireless LAN node can perform semi soft handoff while changing its attached AP and provide mobile IP and/or real time service like voice over IP. Also, we have done simulation for evaluation of the performance of the proposed scheme. We show that our semi soft handoff mechanism can be applied for real-time service with no frame loss in mobile environment.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.1094-1097
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• 2002

Realtime applications like VoIP(Voice over IP) in All-IP networks need smooth handoffs in order to minimize or eliminate packet loss as a Mobile Host(MH) transitions between network links. In this paper, we design a new dynamic buffering(DB) mechanism for IPv6 by which an MH can request that the router on its current subnet buffers packets on its behalf while the MH completes registration procedures with the router of a new subnet. Performance results show that our proposed buffering scheme with a dynamic buffer space allocation is quite appropriate for mobile Internet, or the All-IP environment in terms of the datagram loss rate.

• Journal of the Korea Computer Industry Society
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• v.5 no.9
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• pp.1017-1024
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• 2004

In this paper, we propose the progressed interworking method between H.323 and SlP, then explain the improved property. The VolP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easil)· accept the various of multimedia services from internet. Previous connectionmethod of VoIP used H.323 protocol, but it is very complex to connection establishment. so, the SIP(Session Initiation Protocol) protocol that propose in SIP-Working Group is in use recently. Therefore, we need new interworking methodology between H.323 and SIP Products. In this thesis, the progress interworking method between H.323 and SIP are Propose, then interpret unnecessary packet delay for call setup and improved feature of message exchange.