• 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 the Korea institute of electronic communication sciences
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• v.11 no.3
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• pp.309-316
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• 2016

Recently, the voice service over a mobile communication system tends to be provided based on the packet-based technology. Even though the sufficient transmission rate is supported by LTE mobile communication system, the quality of VoIP service that is experienced by the user can be degraded by the change in the transmission conditions and the terminal mobility. This paper has established an environment on which experiments are conducted for the different values of the major parameters that represent the transmission conditions. The result can contribute to the decision of the requirement that the mobile system should meet for maintaining the quality of VoIP service.

• Journal of KIISE:Information Networking
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• v.35 no.5
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• pp.415-424
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• 2008

In this paper, we compare the QoS performance enhancement schemes according to the network architecture of mobile WiMAX system in order to improve the end-to-end QoS performance and propose QoS parameter mapping method and IP packet scheduling algorithm. To evaluate the end-to-end QoS performance, we implemented an end-to-end simulator of mobile WiMAX system using OPNET. Simulation results show that the proposed QoS parameter mapping scheme reduces the average delay of VoIP packet and improves uplink resource efficiency. And also, when the proposed IP packet scheduling algorithm is applied to the system, the end-to-end packet transmission delay of VoIP service can be reduced by 44-67 percent compared to FIFO and WRR scheduler.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.398-407
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• 2010

Due to increase of user requirement for various traffics and the advance of network technology, each distinct network has converge into FMC(Fixed Mobile Convergence) networks. However, we need to research the performance analysis of VoIP(Voice over Internet Protocol) in the FMC network to provide QoE for the voice user of FMC network. Therefore, this paper introduces the scenario which is the situation of voice quality degradation when a user uses VoIP to communicate with other users in the FMC network. Especially, this paper presents scenario in terms of the component of the network and finds the improvement point of voice quality. In the simulation results, three improvement points of voice quality are found as following: voice quality degradation by packet loss in the physical layer of the HSDPA network, by utilizing GGSN without QoS parameter mapping mechanism which is gateway between 3GPP and IP backbone, and by using non-QoS AP in the WLAN network.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2E
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• pp.35-42
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• 2009

In this paper, we present criteria of objective measurement of speech quality to provide the mobile-VoIP services efficiently over wireless mobile internet. The mobile-VoIP service, which is based on mobility and is error-prone compared to conventional VoIP over wired network, is about to be launched, but there have not been adequate quality indexes and the Quality of Service (QoS) standards for evaluating speech quality of Mobile-VoIP. In addition, there are many factors influencing on the speech quality in packet network of which packet loss contribute directly to the overall voice communication quality. For this reason, we adopt the Gilbert-Elliot Channel Model for modeling packet network based on IP and assess the voice quality through the objective speech method of ITU-T P. 862 PESQ and ITU-T P. 862.1 MOS-LQO under various packet loss rates in the transmission channel environments. Our simulation results address the specific criteria and QoS for the mobile-VoIP services in terms of the various packet loss environments.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.795-797
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• 2009

Voice over Internet Protocol(VoIP) service becomes more and more popular nowadays. As such, it is developed over many kinds of network models, especially wireless networks. Mean Opinion Score(MOS) computes the QoS of VoIP service which should be supported by robust network environment. However, MANET is not stable enough to supply high MOS values for VoIP service. In this paper, VoIP service over MANET is simulated using ns-2(Network Simulation 2). In oder to get different MOS values in the results, we differentiate between network environments by adjusting the parameters of MANET.Through comparing the results we can know how to improve the QoS.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.4 s.346
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• pp.1-12
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• 2006

In this paper we propose an efficient mobile multimedia communication protocol, mobile terminal software platform and mobile VoIP application for IP-based convergence networks. The Proposed mobile multimedia communication protocol is called as ST-MRSVP (Split tunnel based Mobile Resource reServation Protocol) which integrates split tunnel based Mobile IP and RSVP in order to support hish speed mobility. Since mobile terminal platform supports QoS (Qualify of Service) with keeping seamless mobility, mobile QoS supporting modules are developed and interworked together by means of shared memory mechanism. Testbed is composed of a core-network embedding the proposed protocols and wireless LAN-based access networks. We verify functionality and performance of the proposed techniques by using various mobility test over the testbed. As a result, the proposed architecture can reduce the handover delay time with QoS support under 30% comparing with the standard mechanisms and support voice quality as good as CDMA phone.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.1
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• pp.145-152
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• 2009

IEEE 802.16e standard, a kind of WMAN standard, was established to support data services with cheaper cost to mobile users than traditional mobile communications system and wireless LAN system can do. In this paper, we propose an uplink QoS packet-scheduler for VoIP service which can be installed in IEEE 802.16 system and evaluate its performance with NS-2 network simulator. The proposed QoS packet-scheduler consists of three procedures: determining scheduler interval, determining the amount of resource assignment, and deciding which mobile station the base station should serve first among multiple mobile stations. According to numerical results, the proposed QoS packet-scheduler can provide more increased system capacity by 220% than UGS service scheme does and by 25 % than ertPS service scheme does.

• 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.

• Proceedings of the Korean Information Science Society Conference
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• 2000.10c
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• pp.529-531
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• 2000

인터넷이 급속도로 발전해 감에 따라 그 활용 분야가 확장되고 보편화되어 가고 있다. 그 중 최근 대두되고 있는 분야가 바로 통신 서비스 분야이다. 이를 위해 등장한 것이 VoIP인데 IP 위로 음성 데이터를 전송하는 것을 목적으로 한다. VoIP를 지원하기 위한 프로토콜로는 크게 ITU-T에서 제안한 H.323과 IETF에서 제안한 SIP를 들 수 있다. 인터넷을 이용한 통신 서비스를 위해서는 음성 데이터를 패킷 망에서 전송하는 기능 외에 터미널의 이동성을 지원하고, QoS를 보장할 수 있어야 한다. 이러한 측면에서 볼 때, 상기한 두 프로토콜은 음성 데이터를 패킷 망에서 전송하기 위한 다양한 기능과 서비스를 제공하고 있는 반면 호스트의 이동성 및 QoS 보장을 위한 방안에 있어서는 미비한 것이 현재 실정이다. 본 논문에서는 이러한 문제점의 해결을 위해, Mobile IP 위에서 동작하도록 확장된 자원 예약 프로토콜을 기반으로 하여 VoIP를 구현함으로써 호스트의 이동성과 QoS를 보장하도록 하는 방안에 대해 제안하고자 한다.