• 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.

• Journal of information and communication convergence engineering
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• v.7 no.3
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• pp.309-313
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• 2009

In this paper, end-to-end VoIP(Voice over Internet Protocol) performance is evaluated by simulation with NS-2 simulation tool. There are many results studied and published for VoIP performance over TCP/IP networks. But, almost all of them were focused on wired or wireless Internet environments. About MANET (Mobile Ad Hoc Network), VoIP is currently studying several points of research. In this paper, analysis of VoIP performance is done with focusing on the mobility of MANETs. MOS(Mean Opinion Score), network delay, packet loss rates are considered as end-to-end QoS performance parameters.

• Journal of the Korea Society of Computer and Information
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• v.20 no.6
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• pp.29-35
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• 2015

The heterogeneous LTE (Long Term Evolution)-Advanced networks comprising a macrocell and femtocells can provide an efficient solution not only to extend macrocell coverage but also to deal with packet traffics increasing explosively within macrocells. An efficient resource management scheme is necessary to maintain the QoS (Quality of Service) of mobile multimedia services because the LTE-Advanced system should support not only voice but also mobile applications such as data, image and video. This paper proposes a resource management scheme to guarantee QoS continuity of multimedia services and to maximize the resource utilization in OFDMA (Orthogonal Frequency Division Multiple Access) based LTE-Advanced systems. This scheme divides the resources into several service blocks and allocates those resources based on the competition between macrocell and femtocell. Simulation results show that it provides better performances than the conventional one in respect of handover failure rate and blocking rate.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.7-10
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• 2000

본 논문은 이동통신상에서 송신측의 송신 채널 에러정보를 수신측에서 송신측으로 전달하는 음성패킷을 이용하여 송신측에 알려주어 압축과정에 이용하게 할 수 있는 방법을 제안한다. ACELP(Algebraic CELP)구조 방식을 가지고 있는 음성압축기들 중 G.729을 사용하며 음성 패킷정보 중 Pitch Delay와 Fixed Codebook를 이용하여 전송음성 패킷안에 상대방의 송신 채널정보를 싣는다. 수신측에서 받은 패킷이 Erasure로 판단되었을 때 패킷정보들 중 Fixed Codebook Index를 만들게 되는 4개의 Optimal 펄스 중 2개의 펄스만 사용하며 나머지 2개의 펄스는 약속된 임의의 위치에 위치시킨 후 송신측에 전송시킨다. 상대방에서 약속에 맞는 위치의 펄스를 보내왔을 때를 체크함으로써 자신의 송신채널 상태를 알 수 있게 된다 송신채널에$5\%$ Erasure 채널 에러가 발생했을 때 채널정보를 가진 패킷의 음질은 약 0.1dB 정도 떨어지게 된다. 하지만 음성압축 전송 시 송신채널의 정보를 이용하여 무선채널에러에 강하게 할 수 있다.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.53 no.6
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• pp.440-445
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• 2004

The design and implementation of a Network Processor using System-on-a-chip(SoC) technology is presented. The proposed network processor can handle several protocols as well as various types of traffics simultaneously. The proposed SoC consists of ARM processor core, ATM block, AAL processing block, Ethernet block and a scheduler. The scheduler guarantees QoS of the voice traffic and supports multiple AAL2 packet. The SoC is manufactured on the 0.35 micron fabrication line of HYNIX semiconductor, the total number of gates is about 312,000, for a maximum operating frequency of over to 50㎒.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.4B
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• pp.387-393
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• 2009

In this paper, we have considered the network performance of the VoIP service with the measurement by an emulator, and analyzed the major factors to affect its performance. Also, we have used the measured values to investigate the traffic variations, where their values were observed in the commercial operated network after the delay, jitter and packet loss, and loss compensation methods were applied as the dominant elements. It is expected that our presented results will be a good data to provide the high-quality of voice service in the Internet.

• The KIPS Transactions:PartC
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• v.16C no.5
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• pp.661-668
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• 2009

The Extended real-time Polling Service (ErtPS) is proposed tosupport QoS of VoIP service with silence suppression which generates variable size data packets in IEEE 802.16e systems. If the silence is suppressed, VoIP should support Comfort Noise Generation (CNG) which generates comfort noise for receiver's auditory sense to notify the status of connection to the user. CNG mode in silent-period generates a data with lower bit rate at long packet transmission intervals in comparison with talk-spurt. Therefore, if the ErtPS, which is designed to support service flows that generate data packets on a periodic basis, is applied to silent-period, resources of the uplink are used inefficiently. In this paper, we proposed the Improved ErtPS algorithm for efficient resource utilization of the silent-period in VoIP traffic supporting CNG. In the proposed algorithm, the base station allocates bandwidth depending on the status of voice at the appropriate interval by havingthe user inform the changes of voice status. The Improved ErtPS utilizes the Cannel Quality Information Channel (CQICH) which is an uplink subchannel for delivering quality information of channel to the base station on a periodic basis in 802.16e systems. We evaluated the performance of proposed algorithm using OPNET simulator. We validated that proposed algorithm improves the bandwidth utilization of the uplink and packet transmission latency

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.6
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• pp.1313-1319
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• 2011

VoIP service by taking advantage of the current PSTN network and internet over the existing telephone network at an affordable price allows you to make voice calls to the service is being expanded. However, the security of public must be maintained for security vulnerabilities in Smart Phone VoIP case problems arise, and is likely to be attacked by hackers. In this paper, the Internet, using wired and Smart Phone VoIP services may occur during analysis of the type of incident and vulnerability analysis, the eavesdropping should conduct an attack. Smart Phone VoIP with institutional administration to analyze the vulnerability OmniPeek, AirPcap the equipment is installed in a lab environment to conduct eavesdropping attack. Packet according to the analysis and eavesdropping attacks, IP confirmed that the incident as an attack by the eavesdropping as to become the test proves. In this paper, as well as Smart Phone VoIP users, the current administration and the introduction of Smart Phone service and VoIP service as a basis for enhanced security will be provided.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.12B
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• pp.1017-1027
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• 2006

Recently, Internet service providers are offering triple play service which combines voice and video services with the existing high speed Internet service under the common network infrastructure of IP protocol, so that they can create more benefit from operating a single network architecture. In line with this movement in the convergence of network and services, in this work we propose a bandwidth dimensioning method for the subscriber network of the IP network at flow level. To that purpose, let us propose a series of bandwidth dimensioning methods: bandwidth for best effort service only, bandwidth dimensioning for premium services, and bandwidth dimensioning schemes for the premium Internet services as well as the best effort service which comprise the TPS. Our link dimensioning method is based on the flow level that incorporates the flow blocking probability as a measure of grade of services(GoS), and investigates the characteristics of the proposed methods via extensive numerical experiments. After that, let us carry out a simulation experiment concerning the delay and loss performance of the packet scheduling for the premium services (QoS) using the bandwidth designed by our proposed method, via which the packet level quality of service (QoS) for the proposed link dimensioning method can be observed.

• The Transactions of the Korea Information Processing Society
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• v.7 no.8
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• pp.2463-2472
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• 2000

In this paper, we consider pits and falls of V oIP service scheme over the air link environment. It results that V oIP over packet switching is a more attractive approach in several points. We point out the ffilIior requirements for successful V oIP service over the air. Also, we propose V oIP CP concept for efficient wireless channel utilization. Additionally, we analyze and evaluate the performance. According to the results, It shows that the long cycle VolP vocoder CODEC such as ITU-T G.723 is better than short cycle V oIP vocoder CODEC. In this case, the increase of the simultaneous user of system is almost 60% larger than conventional circuit switching.