• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.4
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• pp.678-683
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• 2003

This study presents a result of performance with the design of network topology for voice and data integration under computer network. This network is consisted of FastEthernet, other LANs and ATM WAN(wide area network), and performance evaluation of delay in a PBX+IP network, delay in a VoIP network and delay in a IP+ATM network will be shown. We use parameters including network bandwidth, number of packet, routing protocol(IGRP, OSPF). We simulate integrated of voice and data used PBX. we will study further about the case of integrated of voice and data environments using PBX. and, evaluate IP+ATM WAN average measured network delay and average delay of VoIP network.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.89-92
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• 2001

In this paper, it is proposed that RSVP and WFQ must be a good way of a better service for the better quality for Mobile If Network. For the Performance Analysis of working it was composed of Mobile IP and VoIP Network model, and further more test of the postpone and QoS was implemented. The results of the test is as follows, Before the movement of mobile agent was 2ms, after that, 3ms, And before QoS was adapted the value was 30ms, after being adapted, analyzed as 10ms. This research that the problem of put off was improved by adaping QoS in the mobile IP Network.

• Proceedings of the Korea Electromagnetic Engineering Society Conference
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• 2000.11a
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• pp.20-24
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• 2000

VoIP는 Packet Netwark(ATM, xDSL, Frame Relay, Cable Network)망을 이용하여 음성데이터를 전송 하는 기술로서 PSTN을 통해 음성데이터를 전송하는 것보다 비용절감의 효과가 크다. 본 연구에서는 최적의 VoIP 서비스 제공을 위한 음성/데이터 통합 네트워크 망을 설계하기 위하여 IP와 ATM을 이용한 서로 다른 2개의 망을 설계하여 지연과 Routing 정책, Traffic 추가 후 지연현상에 대하여 살펴보았다. 지연은 순수한 VoIP 망을 구성 할 경우 8-10ms. VoIP+ATM으로 망을 구성 할 경우 2ms로 나타났고, 라우팅 정책(RIP, IGRP, OSPF 적용)에서는 IP 또는 IP+ATM으로 망을 구성 할 경우 RIP는 25ms, IGRP는 22ms로 나타났고, OSPF를 이용할 경우 14ms로 평가되어 OSPF를 이용한 라우팅 정책을 설정하는 것이 바람직하다고 볼 수 있다. 결론적으로 본 연구의 결과 VoIP망을 구성 할 경우 IP+ATM을 기반으로 구축하면 보다 더 효과적인 인터넷 망을 구성할 수 있음을 확인하였다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.6
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• pp.89-94
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• 2015

FMS which has a concept that wireless terminal can replace wire terminal services is a technologies that is can provide service costs same as wire terminal in the special zone. Enterprise type of FMC that is developed making up for the weak point is must have to improve voice quality and FMC performance in the soft phone. This paper measure voice quality based on the one way of the total estimated delay time of FMC to carry out IMS services between IP-PBX and FMC soft-phone to operate it's controller optimally and put forward evidence to be in 120ms and 150ms in the VoIP FMC voice quality. To measure FMC performances in four categories evaluated trials and prove its performances.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.3B
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• pp.304-310
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• 2009

In this paper, we evaluate the performance of VoIP in OFDMA-based relay systems with various capabilities of relays. We classify relays according to capability as "mid-capability (MC)" and "high-capability (HC)" relay. In system with HC relays, not only base station (BS) but also relay station (RS) performs scheduling at its ova whereas only BS performs scheduling in system with MC relays using the information reported by MS (i.e. the received signal-to-interference-plus-noise ratio (SINR) of mobile station (HS), the amount of MS traffic, etc). In system with MC relays, the controling overhead of BS is larger than that of system with HC relays. However, since BS has all MS information, efficient resource allocation and scheduling is possible. We derived the "average packet delay," "good packet ratio," and "cell goodput" in a VoIP environment. The simulation results demonstrate that the system with MC relays has better VoIP performance over that with HC relays.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6A
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• pp.445-454
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• 2005

To measure the performance of call quality, we have built the experiment environment and observed that the delay caused by encapsulation between internet and VoIP telephones is under 5ms at most. The major delay is assumed to be the time required to capsulate the packet for tunnelling of VPN. Because the difference of average delay time is under $4ms{\sim}5ms$, the difference of call quality between VoIP and VoIP telephone adopting VPN is negligible. We have concluded that the capsulation process between PAC and PNS is the major factor influencing the network load by changing the number of fames in a packet during communication Also, we have concluded that the most suitable frame numbers is tow or three by adding the frame numbers in a packet to obtain the suitable frames in a packet and setting up end-to-end delay under 150ms.

• Journal of IKEEE
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• v.5 no.1 s.8
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• pp.8-15
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• 2001

This paper remarks IETF based requirements for IP/UDP/RTP header compression issued in 3GPP2 All IP Ad Hoc Meeting and protocol stacks of the next generation mobile station. All IP Network, for real time application such as Voice over IP (VoIP) multimedia services based on 3GPP2 3G cdma2000. Frames for various protocols expected in the All IP network Mobile Station (MS) are explained with several figures including the bit-for-bit notation of header format based on IETF draft of Robust Header Compression Working Group (ROHC). Especially, this paper includes problems of IS-707 Radio Link Protocol (RLP) for header compression which will be expected to modify in All IP network MS's medium access layer to accommodate real time packet data service[1]. And also, since PPP has also many problems in header compression and mobility aspects in MS protocol stacks for 3G cdma2000 packet data network based on Mobile IP (PN-4286)[2], we introduce the problem of solution for header compression of PPP. Finally. we suggest the guidelines for All IP network MS header compression about expected protocol stacks, radio resource efficiency and performance.

• Journal of the Korea Society for Simulation
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• v.12 no.3
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• pp.55-68
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• 2003

As mobile IP has been suggested only to support mobility of mobile station(MS) by which it dose nothing but guarantee MS's new connection to the network, it is for nothing in Quality of Service(QoS) after handoff of MS. QoS is very important factor in mobile IP network to provide multimedia applications and real-time services in mobile environment, and it is closely related to handoff delay Therefore as a main issue in mobile IP research area, handoff delay problem is actively studied to guarantee and promote QoS. In this paper, in order to resolve such a problem, we suggest Simple Network Management Network(SNMP) information-based routing that adds keyword management method to information-based routing in active network, and then propose QoS controlled handoff by SNMP information-based routing. After setting up routing convergence time, modeling of suggested method and existing handoff method is followed in order to evaluate the simulations that are carried out with NS-2. The result of simulation show the improvement of handoff delay, and consequently it turns off the QoS has been improved considerably.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.687-695
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• 2002

Media-specific FEC techniques, suggested to confront with VoIP speech packet loss, improve speech quality at the expense of generating additional one-frame delay. In this paper, we suggest new media-specific FEC, i.e, LSF FEC technique which is able to improve speech quality with much shortened additional delay. In the proposed technique, the LSF parameters of the future frame are utilized to recover a lost packet. To evaluate performance of the proposed technique, we use ITU-T G.723.1 and G.729 Codec and apply Gilbert packet loss model and estimate MOS per every packet loss rate using PESQ speech quality estimation algorithm. The proposed technique has effect of shortening delay over from 6.5ms to 27ms compared with existing media-specific FEC techniques. Simulation results for comparison of reconstructed speech quality show this novel technique improves the MOS over 0.1 in practical lossy environment of 5 % packet loss rate.

• Journal of the Korea Society of Computer and Information
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• v.13 no.4
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• pp.161-167
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• 2008

A method to compute the size of packet and the optimal way to transmit the packets are proposed in this work for the VoIP communication using the MIL-STD-188-220C, military wireless Ad-hoc protocol which is used for the amicable communications of both speeches and data between several radiotelegraph. The expected time of data transmission is estimated beforehand, and then the size of package and transmission method are decided in the consideration of VoIP speech quality for the users as well as the data transmission quality of radiotelegraph.