• 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.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.93-99
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• 2011

This paper proposes a novel network jitter estimation algorithm for robust VoIP communication system. The proposed method computes the current network environment mode using the differences of arrival time and generation time from sequential received packets. According to the current network environment mode, the jitter variance weights is adjusted to minimize the error for estimating the network jitter. The jitter average and variance are calculated by the autoregressive estimated algorithm, and then the network jitter is estimated by applying the jitter variance weights.

• Proceedings of the Korea Information Processing Society Conference
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• 2007.05a
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• pp.1221-1224
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• 2007

현재 기업들 간의 전화비를 줄이고 각종 VoIP 부가 서비스를 위해 VoIP 시스템의 도입이 시작 되었다. 이에 VoIP 전화기들 간의 각 기능을 최대한 활용하기 위해 현재 VoIP 시장의 90%를 차지하고 있는 H.323 게이트웨이(Gateway)간의 H.323 프로토콜의 구현차이로 인한 문제점이 발생되기 시작되었다. 본 논문은 VoIP Gateway상에 H.323 프로토콜 통신을 하면서 프로토콜 연결 상 구현의 차이로 인해 VoIP 서비스에 비정상적인 작동으로 호가 종료가 되거나 음성이 들리지 않는 현상 및 전화기의 부가서비스를 사용할 수 없는 문제를 해결하기 위해 H.323 프로토콜의 작동을 분석하고 이기종간의 H.323프로토콜 신호가 호환되지 않을 경우 이를 해결하기 위해 H.323 프로토콜상의 H.245 시그널링 (signaling)을 Media gateway 서버를 이용해 구현한 RFC 2833 DTMF-compliant 프로토콜을 사용하여 H.323 프로토콜 처리함으로써 이기종간의 Call transfer, Hold 그리고 Conferenct 기능에 대한 호환성이 개선됨을 보여 준다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.2
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• pp.293-300
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• 2007

As the development of the various IP-based services, it is expected that Internet telephony service will gradually replace the traditional PSTN-based telephony service. But there are many issues resolved to spread the Internet telephony service. One of them is to support the emergency services in the Internet telephony. In the case of USA, it has been regulated that 911 services should be supported in the Internet telephony services using VoIP on the similar performance level to PSTN 911 service. According to the regulation, basic VoIP 911 calls should be routed to the general access line of LEA without the location information or the callback number, but the enhanced VoIP 911 calls with the location information and callback number should be routed on the dedicated 911 network and destined to the local 911 distribution center such as PSAP. But, in the current VoIP-based Internet telephony network, the emergency call service has not been handled as one of the special services as well at has a worse performance in comparison to it on PSTN. Moreover, the service has a critical problem that it can not be destined to the nearest PSAP because of the insufficient information about the location information and the call back number. In this paper, we suggest the SIP-based emergency call service mechanism in order to resolve the problems above mentioned. This suggested mechanism is implemented to show its effectiveness and efficiency.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.2
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• pp.279-287
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• 2001

In this paper, we discuss the scale of subjective speech quality measurement over VoIP(Voice over IP) network which is a component of broadband networks. Objective parameters of multimedia services like PSNR or jitter can easily measured and defined, but these factors are not easily meet the user's perceptual recognition. We suggest the speech quality measurement scale through the subjective measurement for end-to-end speech quality composed of sender-side quality, transmission quality, receiver-side quality, which provide the degree of correctness of representation of speaker, the degree of impairment caused by various factors, the degree of recognition of processed speech, respectively. Also, we examined the proposed method and verify it's availability.

• 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.

• 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 the Korea Society for Simulation
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• v.15 no.3
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• pp.87-91
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• 2006

With the growth of the internet, the number of wireless internet users has increased continuously up to date. However, mobile communications could not support high speed transmission rate with cheap communication fee and wireless LAN has problems in providing terminal mobility and wide area connectivity, respectively. So the WMAN standard has been newly designed to make up for these limits. The initial 802.16 specification effectively offers a solution for providing fixed users with high speed wireless communication but it does not offer terminal mobility. So the 802.16d and 802.16e have been developed as the next generation solution that can support various PHY layer (SC, SCa, OFDM, OFDMA) and offer the terminal mobility. In this paper, we propose an effective uplink scheduling algorithm for VoIP with using UGS, and we show that our proposed algorithm is superior in view of average delay and management of uplink bandwidth to conventional rtPS algorithm and the scheme in reference, with using NS-2 network simulator.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2006.07a
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• pp.609-609
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• 2006

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2006.07a
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• pp.594-594
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• 2006