• The Magazine of the IEIE
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• v.31 no.7
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• pp.58-65
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• 2004

값싼 이용료를 장점으로 내세워 많은 관심과 투자가 집중되었던 VoIP(Voice ever IP) 서비스가 기대와는 달리 그 시장이 매우 침체되어 있는 것이 사실이다. 이는 기존 전화 사업자들의 소극적인 서비스에 대한 투자와 인터넷의 특성상 낮은 서비스 품질 때문에 많은 이용자들이 사용을 멀리했기 때문이다. 그러나 점차 기간 통신 서비스 사업자들이 차세대 네트워크 및 광대역 통합망을 도입하면서 VoIP 서비스를 기본 서비스로 제공하고, VoIP 관련법과 제도가 정비됨으로써 VoIP 서비스의 활성화가 기대되고 있다. 또한 그동안 문제가 되어 왔던 서비스 품질은 다양한 음질 개선 노력으로 인해 이제는 양질의 품질을 제공하고 있어 더 이상 서비스 이용에 문제가 되지 않는다. (중략)

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6B
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• pp.575-584
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• 2009

In this paper, we design and implement a VoIP traffic generator for simulating call processing in IP contact center systems. Creating a VoIP call based on H.323 and SIP and generating RTP traffic which uses G.711 codec, the generator lets many users simulate situations on which they call each other. With this tool, which is named VoIPTG, users can combine H.323 or SIP session control protocol, the number of users, time variation, and voice codecs and then direct various situations for simulation. This traffic generator can be used for testing functions of an IP contact center and especially it is necessary for testing the quality of IP based call recording systems.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.281-284
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• 2011

Smart phone is being used in the job as the ubiquitous society will Without being restricted by the time and place and devices. The rapid increase in the use of smart phones has brought the activation of the mobile job. And government agencies have brought in the transition to a smart society. In this paper, using a Voice over Internet protocol(VoIP) service for your smart phones to enhance security is the study of encryption technologies. External and internal signals, and call encryption and security standards of administrative agencies is the study of VoIP. Smart phone VoIP service is a study that security of equipment certificate, the internal signal and call encryption. This paper will contribute what using smart phone VoIP security and usability In smart generation.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.18 no.3
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• pp.109-120
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• 2008

VoIP utilizes the Internet for the services, and therefore it is vulnerable to intrusions and attacks. Because provided services deal with information related to privacy of users, it requires high level security including authentication and the confidentiality/integrity of signaling messages and media streams. However, when such a protocol is implemented in a VoIP phone, the implementation can have limitations due to the limited resources. The present study purposed to implement VoIP security protocols and to evaluate their performance in terms of connection quality and voice quality by applying them to SIP proxy and UA (User Agent). In the result of performance evaluation, the application of the security protocols did not lower voice quality, but connection quality was high in the DTLS based security protocol. As the protocol was applicable to signaling and media paths based on DTLS, we found that it can be a solution for the limited resources of VoIP phone.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.947-950
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• 2005

Voice of IP(VoIP) technology provides voice service as well as data service via Internet. It has been a promising technology as Internet grows fast and the requirements are increasing. Recently, serveral protocols have been created to allow telephone calls to be made over IP networks, notably, SIP and H.323. Due to introducing SIP and H.323, In this paper, we consideration interoperability of internet telephony service between ITSPs(internet telephony service provider)based on H.323.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.3
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• pp.275-280
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• 2010

In this paper, VoIP(Voice over Internet Protocol) transmission performance for MANET(Mobile Ad-hoc Networks) is improved and analyzed with packet aggregation scheme which is aggregating some of short length packets to one large packet and sending to networks. VoIP simulator based on NS(Network Simulator)-2 is implemented and used to measure performance of VoIP traffic transmission. In this simulation, VoIP traffics are generated with parameters of some codes such as G.711, G.729A, GSM.AMR and iBLC. MOS(Mean Opinion Score), end-to-end network delay, packet loss rate and transmission bandwidth are measured. Performance improvements of 98% for MOS, 6.4times for end-to-end network delay, 32times for packet loss rate is shown as simulation results. On the other hand, transmission bandwidth is increased about maximum 10%. Finally, VoIP implementation guide for the performance with packet aggregation is suggested.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.1
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• pp.81-89
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• 2005

In this paper, we have described the implementation of a practical simulation tool to design and analyze communication networks. Especially, this study is focused on the implementation and application methods of a simulator supporting VoIP The key characteristics of this particular system are its easy and intuitive usage, the real behaviors implementation of equipment and protocols, the actual generation and transmission of traffic for simulation, supporting of VoIP and so forth. Our system is distinguished from the existing tools which define only the nature of voice traffic, process those packets in the same way as general data, and analyze only the quality of packet transmission such as delay. Our tool presented in this paper generates and processes packets in different way according to the types of traffic distinguishing call signal from voice information traffic. Also, we equipped this system with the various devices such as VoIP gateway and gatekeeper, which enabled this system to analyze the performance of devices and the quality of voice traffic transmission between PSTN and Internet. By presenting the implementation methods and application of this system, we managed to propose the utilization scheme of a simulation tool.

• 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.