• MALSORI
• /
• no.64
• /
• pp.105-119
• /
• 2007

Noise reduction technologies are indispensable to achieve acceptable speech quality in VoIP systems. This paper proposes a Wiener filter optimized to the estimated SNR of noisy speech for the noise reduction in VoIP environments. The proposed noise canceller is applied as a pre-processor before speech encoding. The performance of the proposed method is evaluated by the PESQ in various noisy conditions. In this paper, the proposed algorithm is applied to G.711, G.723.1, and G.729A which are all VoIP speech codecs. The PESQ results show that the performance of our proposed noise reduction scheme outperforms those of the noise suppression in the IS-127 EVRC and the ETSI standard for the advanced distributed speech recognition front-end.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.11 no.6
• /
• pp.73-80
• /
• 2011

In this paper, we design and implementation of effective remote management system for VoIP(Voice over IP) terminal. Ordinary VoIP terminal is connected inside of NAT(Network Address Translation) assigned private IP address. NAT provides address mapping between outside public IP address and inside private IP address, for outside system telephone call is connected to inside VoIP terminal. In this paper, we also design and implementation of UDP hole punching function that the outside calls pass through the NAT, for outside management system call is connected to internal VoIP terminal, and gathered to management information of the VoIP terminal. Also, we evaluated and analysed of developed system in the test environment. As a result of test, It showed that it was well performed without any data error and data loss in the connectivity, and It showed that it was well gathered management information of the VoIP terminal.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.15 no.5
• /
• pp.857-864
• /
• 2020

The multi-functional VoIP phone supports the Ethernet protocol in the TIPS(: Tactical IP Switch), which is one of the sub-systems of the tactical information and communication system (TICN). It provides secured voice / video calls in conjunction with VoIP exchanges and supports differential services such as multi-party calls and command functions. In this paper, improving methods have been proposed to reduce power supply defects in the field of multi-functional VoIP phones. The power supply part was improved by applying TVS of the output voltage inlet of the dedicated adapter of the multi-functional VoIP phone, TVS of the PoE module input, adding blocking diodes, and adding DC / DC converters behind the poly-switch. Also, functional and environmental tests were performed to verify the validity of the proposed methods.

• Proceedings of the Korean Information Science Society Conference
• /
• 2004.10c
• /
• pp.325-327
• /
• 2004

VoIP는 음성 및 데이터 통합 뿐만 아니라 차세대 네트웍 등의 기반이 되는 기술이며, 인터넷전화 / IP Telephony, 화상회의, 메신저 서비스 등 여러 서비스에 활용되고 있다. 이러한 VoIP 서비스 제공시에 가장 중요시되는 부분이 음성품질이며 이를 측정 및 관리하는 기술이 필수적으로 필요하다. 지금까지는 품질측정장비를 가지고 직접 측정하는 것이 전부였으나 본 연구는 IETF의 VoIP 표준 프로토콜인 MGCP중 파라미터 값을 이용하여 ITU-T의 음성품질 기준인 R factor(G.107)를 계산해 내고 중앙에서 모든 단말 및 사용자들의 실제 발생한 통화에 대한 음성품질을 관리할 수 있는 시스템을 설계 및 구현한다.

• Annual Conference of KIPS
• /
• 2014.11a
• /
• pp.218-220
• /
• 2014

VoIP(Voice over Internet Protocol)는 음성 및 화상과 같은 멀티미디어 세션을 인터넷과 같은 IP 기반 네트워크를 통해 통신하는 기술이다. 최근에는 기존의 PC 시스템 이외에 이동통신기기와 다양한 무선네트워크 기반 휴대용 기기들의 보급으로 VoIP 의 사용량은 크게 증가하고 있다. 하지만, 무선네트워크는 그 특성과 환경적 요인으로 NAT 에서의 차단, 지연, 유실등과 같이 통신의 연속성을 보장해 주지 못하는 문제가 발생할 수 있다. 본 논문에서는 무선네트워크에서 통신할 때 발생할 수 있는 이런 문제들에 대응하는 해결 방안을 제시하고 RTP 미디어 재생의 연속성을 보장하는 ICE 기반 다자간 VoIP 시스템 설계와 구현모델에 대해서 기술하고자 한다.

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

• Journal of Digital Convergence
• /
• v.10 no.6
• /
• pp.259-267
• /
• 2012

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. Recently, the introduction of smart phones has led to a growth in social networking services and thus, the research and development of Internet telephony has been actively progressed and has the potential to become a replacement for the telephone service that is currently being used. In this paper we designed and implemented an Internet telephony network system which is developed by using Asterisk and open source softwares. It is developed on the linux system and has some features such as VoIP telephony service between SIP phones, voice mail, and call recording. It also supports web-based functions such as SIP users and server system management that is implemented by Apache web server and PHP programs. Afterwards, this system will be applied as VoIP network base technology for small sized companies and organizations. It will paly a role for encouraging companies to use open source softwares.

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

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.30 no.9C
• /
• pp.877-886
• /
• 2005

Various frame recovery algorithms have been suggested to overcome the communication quality degradation problem due to Internet-typical impairments on Voice over IP(VoIP) communications. In this paper, we propose a new receiver-based recovery method which is able to enhance recovered speech quality with almost free computational cost and without an additional increment of delay and bandwidth consumption. Most conventional recovery algorithms try to recover the lost or erroneous speech frames by reconstructing missing coefficients or speech signal during speech decoding process. Thus they eventually need to modify the decoder software. The proposed frame recovery algorithm tries to reconstruct the missing frame itself, and does not require the computational burden of modifying the decoder. In the proposed scheme, the Vector Quantization(VQ) codebook indices of the erased frame are directly estimated by referring the pre-computed VQ Codebook Index Interpolation Tables(VCIIT) using the VQ indices from the adjacent(previous and next) frames. We applied the proposed scheme to the ITU-T G.723.1 speech coder and found that it improved reconstructed speech quality and outperforms conventional G.723.1 loss recovery algorithm. Moreover, the suggested simple scheme can be easily applicable to practical VoIP systems because it requires a very small amount of additional computational cost and memory space.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.51 no.12
• /
• pp.20-29
• /
• 2014

Recently, LTE is preparing to make a new leap forward LTE-A all over the world. As LTE privides high speed service, the role of mobile phones seems to change from voice to data service. According to Cisco, global mobile data traffic will increase nearly 11-fold between 2013 and 2018. Mobile video traffic will reach 75% by 2018 from 66% in 2013 in Korea. However, voice service is still the most important role of mobile phones. Thus, controllability of throughput and low BLER is indispensable for high-quality VoIP service among various type of traffic. Although the maximum AMR-WB, 23.85 Kbps is sufficient to a VoIP call, it is difficult for the LTE which can provide tens to hundreds of MB/s may not keep the certain level VoIP QoS especially in the cell-edge area. This paper proposes a new scheduling algorithm in order to improve VoIP performance after analyzing various scheduling algorithms. The proposal is the technology which applies more priority processing for VoIP than other applications in cell-edge area based on two-tier scheduling algorithm. The simulation result shows the improvement of VoIP performance in the view point of throughput and BLER.