• Korean Journal of Human Ecology
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• v.15 no.6
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• pp.961-971
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• 2006

The purpose of this study was firstly to provide fundamental data on the activation for the IP-based video phone for the special consumer related to the physically handicapped; secondly to inform empirical data for the consumer public policy in the information technology market, specially for the deaf people. The results of study showed that consumer needs extend to not only simple voice communication for general consumers but also special demands for both the handicapped and the elderly. This study also indicated that VoIP's characteristics of technology would be easily applied to the TRS or VRS which can be adapted to the special consumer market so that VoIP service would be optimal technology for the special consumers like the deaf. In order to successfully implement TRS & VRS business, the paper proposed as follows; 1) the provision of VoIP service enable to satisfying consumers in special market such as the deaf market and the elderly market, 2) the necessity of supporting policy by the related law, and 3) the construction of the system inducing interests from the market participants.

• 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.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.2
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• pp.245-252
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• 2011

VoIP transmits voices over IP-based networks and it is the abbreviation of Voice over Internet Protocol. Recently, VoIP provides various services in addition to voices. Since VoIP services' provision is extending, VoIP service quality management is becoming an important issue. Therefore, in this paper, we study VoIP service quality management. We examine VoIP technology, service types, and network architecture. Then, we investigate key quality indicators(KQIs)/key performance indicators(KPIs) in terms of customers, not network service providers. Toward this, we also study the concept of general service quality management as well as the concept of telecommunication related service quality management. Moreover, we apply $\bar{x}$ and R charts to show how to use statistical quality control techniques in real telecommunication companies with one KQI.

• TTA Journal
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• s.71
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• pp.93-104
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• 2000

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.6
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• pp.719-726
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• 2014

This paper addresses quality of experience(QoE) and how to measure and evaluate QoE including its subjective aspects. Adopting the real measurements on the field, a various quality metric have been measured for VoIP(voice over IP) service provided through a wireless interface of WiBro(Wireless Broadband). By analyzing the measured values and correlation between the metrics, we attempt to find a method to evaluate QoE of the VoIP service in a objective way. As a result, it has been shown that QoE of the VoIP service through WiBro network has close relation to the packet-level end-to-end delay, and the delay has close relation to received signal strength indicator(RSSI).

• Proceedings of the KIEE Conference
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• 2002.07d
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• pp.2316-2318
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• 2002

지금까지 디지털 통신에서는 반향이 통신품질의 관점에서 별다른 문제가 되지 않았다. 그러나 인터넷의 발달로 인하여 음성 데이터 통합(VoIP:Voice over Internet Protocol)을 이용한 인터넷폰의 사용이 요구되고 있으며, 시외 또는 국제 통화의 경우에 음성신호를 서킷에서 패킷으로 전송하는 과정에서 전송 지연 증가에 따른 반향에 대한 문제가 발생되고 있다. 본 논문에서는 VoIP 기반의 음성통신에서 발생하는 반향을 적응 반향제어기를 통해 제거하는 방법에 대해 연구하였다. 모의 실험을 통해 ECLMS 알고리즘을 적용한 반향제거기가 우수한 반향제거 성능을 보여줌을 확인하였다.

• Journal of Communications and Networks
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• v.14 no.5
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• pp.510-517
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• 2012

This paper analyzes the performance of voice-over-Internet protocol (VoIP) services in terms of the system throughput, the packet delay, and the signaling overhead in a mobile WiMAX system with a hybrid automatic repeat request (HARQ) mechanism. Furthermore, a queueing analytical model is developed with due consideration of adaptive modulation and coding, the signaling overhead, and the retransmissions of erroneous packets. The arrival process is modeled as the sum of the arrival rate at the initial transmission queue and the retransmission queue, respectively. The service rate is calculated by taking the HARQ retransmissions into consideration. This paper also evaluates the performance of VoIP services in a mobile WiMAX system with and without persistent allocation; persistent allocation is a technique used to reduce the signaling overhead for connections with a periodic traffic pattern and a relatively fixed payload. As shown in the simulation results, the HARQ mechanism increases the system throughput as well as the signaling overhead and the packet delay.

• Proceedings of the KIEE Conference
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• 2003.07d
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• pp.2127-2129
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• 2003

지금까지 디지털 통신에서는 반향이 통신 품질의 관점에서 별다른 문제가 되지 않았다. 그러나 인터넷의 발달로 인하여 음성 데이터 통합(VoIP:Voice over Internet Protocol)을 이용한 인터넷폰의 사용이 요구되고 있으며, 시외 또는 국제 통화의 경우에 음성 신호를 서킷에서 패킷으로 전송하는 과정에서 전송 지연 증가에 따른 반향에 대한 문제가 발생되고 있다. 현재는 DSP chip의 급속한 발달로 반향의 제거가 실시간으로 처리할수 있게 되었다. 본 논문에서는 VoIP기반의 음성 통신에서 발생하는 반향을 적응 반향제어기를 통해 제거하는 방법에 대해 연구하였다. DSP processor를 사용한 실험을 통해 알고리즘을 적용한 반향제거기의 성능이 우수함을 확인하였다.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.2 no.3
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• pp.9-16
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• 2003

In this paper, we propose the burst error reduction (BER) algorithm for VoIP service in the wireless LAN network. In end point device, this BER algorithm can be achieved packet loss bounded QoS provisioning using interleaving in buffering and FEC (Forward Error Correction) through transmitting voice packet. BER algorithm can reduced the voice packet loss rate 5.5%-60% in VoIP network using wireless LAN.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.292-294
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• 2001

VoIP(Voice over IP) system은 현재 크게 2가지 형태로 진행되어가고 있다. 첫 번째는 H.323을 이용한 방법이고, 두 번째는 SIP(Session Initiation Protocol)를 이용한 방법이다. H.323은 실제 데이터를 전송하기전 호처리에 많은 signaling이 이루어지는 관계로 SIP보다 많은 RTT(Round Trip Time)를 소모하게 된다. 따라서 매우 복잡하고, LAN환경을 바탕으로 만들어서 확장성면에서도 여러 문제점을 가지고 있다. 그래서 본 논문은 호처리는 SIP를 이용하고, 실제 음성전송은 RTP(Real-Time Transport Protocol)와 RTCP(RTP Control Protocol)를 이용하는 시스템 구현을 제시한다. RTP는 실시간 특성을 가지는 데이터에 대해서 종단간 전송 서비스를 제공해주는 프로토콜로, 어떠한 인코딩에도 적합한 프레임워크를 제공한다. 그런데, RTP는 완전한 하나의 프로토콜이 되기 위해서는 RTP와 페이로드 포맷이 함께 제공되어야 하므로, 구현시스템은 음성신호를 PCM(Pulse Code Modulation), ADPCM(Adaptive Differential PCM)등의 여러 압축기술을 이용하여 파일을 생성하여 실시간으로 RTP와 RTCP를 이용하여 전송하는 방법을 제시한다.