• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.683-690
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• 2003

In this paper we present a theoretical framework for the network design with delay QoS guarantee to a voice at the packet level. Especially, we propose a method for estimating the bandwidth at the ingress edge routers accommodating the voice connections and data sessions in the next-generation If network. First, we describe network architecture for VoIP (Voice over IP) services in the NGN (Next Generation Network). After that, we propose a procedure for dimensioning the bandwidth at the output port of a router that accommodates voice and data traffic using the non-preemptive queuing system with strict priority service scheme. Via numerical experiments we illustrate the implication of the proposition.

• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.388-390
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• 2001

인터넷을 통해 음성 서비스를 가능하도록 해주는 VoIP (Voice over IP) 기술은 다양한 멀티미디어 기술과 결합하여 차세대 이동 통신망에서 핵심적인 서비스로 발전할 것이다. 하지만 차세대 이동 통신망에서는 다양한 단말기와 엑세스망 기술이 지원될 것이기 때문에 서로 다른 통신 능력을 가진 사용자 사이에서 직접적인 세션 설정이 불가능한 다양성의 문제(Diversity Problem)가 발생할 것이다. 이러한 다양성의 문제를 해결하기 위해서는 사전에 미디어 협약 (Media Negotiation)이라고 하는 과정을 거쳐야 한다. 기존의 SIP(Session Initiation Protocol) 기반의 VoIP 시스템에서는 이미 미디어 협약 과정이 정의되어 있지만 많은 세션 설정 시간이 소비되는 단점이 있다. 본 고에서는 이러한 단점을 개선하여 효과적인 미디어 협약이 가능하도록 해주는 SIP 기반 의 새로운 VoIP 아키텍쳐를 제안한다. 개선된 YoIP 아키텍쳐는 기존의 아키텍쳐 상의 지역 도메인 내의 기존 요소를 확장하여 구현 가능하기 때문에 높은 상호 호환성과 개발의 용이성을 지닌다. 이러한 새로운 VoIP 아키텍쳐의 구현을 통한 세션 시간 측정 결과 기존의 아키텍쳐에 비해 50%이상의 세션 설정 시간이 단축됨을 알 수 있었다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.1
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• pp.24-51
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• 2011

When supporting both voice and TCP in a wireless multihop network, there are two conflicting goals: to protect the VoIP traffic, and to completely utilize the remaining capacity for TCP. We investigate the interaction between these two popular categories of traffic and find that conventional solution approaches, such as enhanced TCP variants, priority queues, bandwidth limitation, and traffic shaping do not always achieve the goals. TCP and VoIP traffic do not easily coexist because of TCP aggressiveness and data burstiness, and the (self-) interference nature of multihop traffic. We found that enhanced TCP variants fail to coexist with VoIP in the wireless multihop scenarios. Surprisingly, even priority schemes, including those built into the MAC such as RTS/CTS or 802.11e generally cannot protect voice, as they do not account for the interference outside communication range. We present VAGP (Voice Adaptive Gateway Pacer) - an adaptive bandwidth control algorithm at the access gateway that dynamically paces wired-to-wireless TCP data flows based on VoIP traffic status. VAGP continuously monitors the quality of VoIP flows at the gateway and controls the bandwidth used by TCP flows before entering the wireless multihop. To also maintain utilization and TCP performance, VAGP employs TCP specific mechanisms that suppress certain retransmissions across the wireless multihop. Compared to previous proposals for improving TCP over wireless multihop, we show that VAGP retains the end-to-end semantics of TCP, does not require modifications of endpoints, and works in a variety of conditions: different TCP variants, multiple flows, and internet delays, different patterns of interference, different multihop topologies, and different traffic patterns.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4A
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• pp.310-315
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• 2002

The Bark Coherence Function (BCF) defies a coherence function within perceptual domain as a new cognition module, robust to linear distortions due to the analog interface of digital mobile system. Our previous experiments have shown the superiority of BCF over current measures. In this paper, a new BCF suitable for VoIP is developed. The unproved BCF is based on the wavelet series expansion that provides good frequency resolution while keeping good time locality. The proposed Wavelet based Bark Coherence function (WBCF) is robust to variable delay often observed in packet-based telephony such as Voice over Internet Protocol (VoIP). We also show that the refinement of time synchronization after signal decomposition can improve the performance of the WBCF. The regression analysis was performed with VoIP speech data. The correlation coefficients and the standard error of estimates computed using the WBCF showed noticeable improvement over the Perceptual Speech Quality Measure (PSQM) that is recommended by ITU-T.

• 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).

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.5
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• pp.129-132
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• 2009

Frame erasure is one of the most difficult problems in voice over IP (VoIP) networks and is a major source of speech quality degradation. In this paper, a frame erasure concealment algorithm based on nonlinear regression analysis is presented to minimize speech quality deterioration in code-excited linear prediction (CELP) based coders. We applied the proposed scheme to the ITU-T G.729 standard and obtained improved perceptual evaluation of speech quality (PESQ) scores compared to the conventional methods.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.844-848
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• 2002

In this paper, we present a VoIP service provisioning architecture based on MEGACO/H.248 which is one of the key protocols for VoIP services. MEGACO/H.248 is a media gateway control protocol standardized by both ITU-T and IETF, and many ITSPs, carriers, and vendors currently have a lot of interest in the protocol. MEGACO/H.248 is used by a softswitch a key component of the next generation VoIP network, in order to control various media gateways and provide seamless interworking between PSTN and Yon networks.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.13-15
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• 2003

SIP(Session Initiation Protocol)는 사용자간의 멀티미디어 세션을 처리하기 위한 응용 계층의 시그널링 프로토콜로서 유연성 및 확장이 용이한 장점을 가지고 있다. Caller Preference는 이러한 SIP의 기본적인 프로토콜을 확장한 형태로서 송신자가 Preference를 명시하여 서버가 처리할 응답 기능을 선택하거나 수신자의 수신 능력(Callee Capabilities)에 따라 적절한 호처리를 진행할 수 있는 서비스이다. 본 논문에서는 SIP를 기반으로 하는 VoIP(Voice over IP) 시스템을 구현함에 있어 UA(User Agent)내에 Preference를 선택적으로 명시할 수 있는 기능을 포함시키고 또한 이의 요청에 대한 수용이 가능하고 호처리를 진행할 수 있는 네트워크 서버를 구현하였다.

• IE interfaces
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• v.22 no.1
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• pp.95-103
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• 2009

During the past few years, design of experiments (DOE) has been gaining acceptance in the telecommunications research community as a mean for designing and analyzing experiments economically and efficiently. In addition, the need for introducing a systematic robust design methodology (i.e., one of the most popular DOE methodologies) to network simulations has been increasing. In this paper, we present an architecture of voice over IP (VoIP) application and the E-Model for calculating the perceived quality of service (QoS). Then, we apply the Taguchi robust design methodology to optimize the perceived QoS of VoIP application, and describe the detailed step-by-step procedures. We have used ns-2 simulator to collect experimental data in which the SN ratio, a robustness measure, is analyzed to determine an optimal design condition. The analysis shows that "initial delay time in playout buffer" is a major control factor for ensuring robust behaviors of the perceived QoS of VoIP. Finally, we verify the proposed optimal design condition using a confirmation experiment.