• Title, Summary, Keyword: Voice over IP

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Robust speech quality enhancement method against background noise and packet loss at voice-over-IP receiver (배경잡음 및 패킷손실에 강인한 voice-over-IP 수신단 기반 음질향상 기법)

  • Kim, Gee Yeun;Kim, Hyoung-Gook
    • The Journal of the Acoustical Society of Korea
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    • v.37 no.6
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    • pp.512-517
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    • 2018
  • Improving voice quality is a major concern in telecommunications. In this paper, we propose a robust speech quality enhancement against background noise and packet loss at VoIP (Voice-over-IP) receiver. The proposed method combines network jitter estimation based on hybrid Markov chain, adaptive playout scheduling using the estimated jitter, and speech enhancement based on restoration of amplitude and phase to enhance the quality of the speech signal arriving at the VoIP receiver over IP network. The experimental results show that the proposed method removes the background noise added to the speech signal before encoding at the sender side and provides the enhanced speech quality in an unstable network environment.

Implementation of QoS-Measuring System for Voice over IP (VoIP(Voice over Internet Protocol) 품질 측정을 위한 UA(User Agent) 및 서버 기능 연구)

  • Kang, Hyun-Joong;Nam, Heung-Woo
    • Journal of the Korea Society of Computer and Information
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    • v.12 no.1
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    • pp.137-144
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    • 2007
  • Advances in networking technology digital media, and codecs have made it possible for the Internet evolves into a Broadband convergence Network (BcN) and provides various services including Voice over Internet Protocol (VoIP) and IPTV over their high-speed IP networks. In order for the Internet to make a profit as traditional Public Switched Telephone Network (PSTN), it must provide high qualify VoIP services. Therefore, real time qualify measurement framework is the most important requisite to provide VoIP service. For this, IETF (Internet Engineering Task Force) defined RTCP-Extended Reports (RTCP-XR) that extend RTCP (Real-Time Transport Protocol Control Protocol). However, procedure and method tot actually VoIP qualify measurement did not recommended nothing but defined item to measure voice quality. Our objective in this paper is to describes a practical measuring framework for end-to-end QoS of switched voice packet in an IP environment. It includes concepts as well as step-by-step procedures for measuring packetized voice streams. It also proposes new formats that extend RTCP-XR's concept.

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Policy and Managerial Issues of Voice over Internet Protocol(VoIP) (인터넷전화의 정책 및 경영이슈측면에서의 이용자분석)

  • Kim, Ji-Hee;Sung, Yoon-Young;Kweon, O-Sang;Kim, Jin-Ki
    • Journal of Information Technology Applications and Management
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    • v.14 no.4
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    • pp.221-233
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    • 2007
  • Which factors should influence consumer consideration to subscribe to Voice over Internet Protocol (VoIP)? Policy issues, managerial concerns, and demographic variables are possible factors. This paper discusses policy and managerial issues regarding VoIP adoption. A model that explains VoIP adoption is proposed and tested. This study analyzes a survey of 750 prospective VoIP users in Korea. The testing is accompanied by logistic regression and discriminant analysis. The results show that trust in VoIP, relative comparison of Quality to fixed service, numbering plan, satisfactions of call Quality and customer services on both fixed and mobile services have impacts on the adoption of VoIP. Implications for VoIP providers and policy makers are presented.

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Multipoint VoIP of End-point Mixing in Various Environments (다양한 환경에서 단말혼합 방법의 다자간 VoIP 운용)

  • Kim, Do-Yun;Park, Eun-Sung;Lee, Sung-Min;Seong, Dong-Su;Lee, Keon-Bae
    • Proceedings of the IEEK Conference
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    • pp.16-18
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    • 2009
  • VoIP(Voice over IP) is the technology to transport voice and video over IP networks such as Internet. Today, VoIP technology is viewed as the right choice for provide voice, video, and data communication over next generation network. We are sure that the multipoint VoIP will help enhancing the various application services in ubiquitous environment. The paper shows multipoint VoIP system implemented with end-point mixing model and introduces various embedded systems such as UFC(Ubiquitous Fashionable Computer), tourist guide terminal and industrial terminal which use the multipoint VoIP.

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The VoIP Capacity Analysis of 802.11 WLANS with Propagation Errors (전파 오류가 빈번한 802.11 무선 랜에서의 VoIP 용량 분석)

  • Jung, Nak-Cheon;Ahn, Jong-Suk
    • Journal of KIISE:Computing Practices and Letters
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    • v.14 no.1
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    • pp.101-105
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    • 2008
  • This paper proposes an analytical model to calculate VoIP (Voice of IP) capacity over wireless LANs with frequent bit errors. Since the traditional analytical models for VoIP capacity have not included the effect of bit errors, simulations ould only evaluate VoIP capacity over erroneous channels. For analytically accurate estimation of VoIP capacity over noisy channels, we extend the conventional model to include the effect of propagation errors, end-to-end delay, voice quality, the waiting time in AP(Access Point). The experiments show that our model predicts the VoIP capacity of a given network within the range from 3% to 9% difference comparing with the simulation results.

Study on Fraud and SIM Box Fraud Detection Method in VoIP Networks (VoIP 네트워크 내의 Fraud와 SIM Box Fraud 검출 방법에 대한 연구)

  • Lee, Jung-won;Eom, Jong-hoon;Park, Ta-hum;Kim, Sung-ho
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.40 no.10
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    • pp.1994-2005
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    • 2015
  • Voice over IP (VoIP) is a technology for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks. Instead of being transmitted over a circuit-switched network, however, the digital information is packetized, and transmission occurs in the form of IP packets over a packet-switched network which consist of several layers of computers. VoIP Service that used the various techniques has many advantages such as a voice Service, multimedia and additional service with cheap cost and so on. But the various frauds arises using VoIP because VoIP has the existing vulnerabilities at the Internet and based on complex technologies, which in turn, involve different components, protocols, and interfaces. According to research results, during in 2012, 46 % of fraud calls being made in VoIP. The revenue loss is considerable by fraud call. Among we will analyze for Toll Bypass Fraud by the SIM Box that occurs mainly on the international call, and propose the measures that can detect. Typically, proposed solutions to detect Toll Bypass fraud used DPI(Deep Packet Inspection) based on a variety of detection methods that using the Signature or statistical information, but Fraudster has used a number of countermeasures to avoid it as well. Particularly a Fraudster used countermeasure that encrypt VoIP Call Setup/Termination of SIP Signal or voice and both. This paper proposes the solution that is identifying equipment of Toll Bypass fraud using those countermeasures. Through feature of Voice traffic analysis, to detect involved equipment, and those behavior analysis to identifying SIM Box or Service Sever of VoIP Service Providers.

Implementation of VoIP Service in Hybrid Fiber Coaxial Network (Hybrid Fiber Coaxial망에서 VoIP 서비스 구현)

  • Ju, Jae-han
    • Journal of Advanced Navigation Technology
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    • v.21 no.1
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    • pp.113-118
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    • 2017
  • As interest in mobile devices and networks has increased recently, voice over internet protocol (VoIP) service, which is a technology for transmitting voice data using an existing internet protocol (IP) network, has rapidly spread, Cheap voice call service has become possible. As the digital broadcasting service becomes popular, hybrid fiber coaxial (HFC) network technology, which uses broadband cable network through fusion of broadcasting and communication, utilizes existing communication system and network equipment to provide various new services such as interactive broadcasting service. Therefore, if UGS-AD is applied to VoCM and RTPS is applied to MTA in order to guarantee the quality of voice data in actual HFC Internet service network, it is possible to smoothly perform voice data transmission in narrow upstream band which is a problem in actual commercial HFC network We also proposed a method to improve VoIP service by improving QoS of voice data in HFC Internet service network.

A Study of the delay pattern of voice traffic for end-to-end users on the voice IP (VoIP 상에서 다양한 응용 서비스 트래픽에 따른 종단간 사용자의 음성 트래픽 지연 변화 연구)

  • 윤상윤;정진욱
    • Journal of the Korea Society for Simulation
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    • v.10 no.2
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    • pp.15-24
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    • 2001
  • In this paper we study the delay patterns of voice traffic for end-to-end users Caused by serving the whole bunch of applications traffic at the same time on the Voice over Internet Protocol (VoIP) network. Given the current situation that voice traffic is served along with other application services on the VoIP network, it is quite necessary to figure out how and by what the voice traffic requiring high QoS is delayed. We compare the delay performance of voice traffic on the VoIP network under FIFO with the one under Weighted Fair Queuing(WFQ), and discover the differences of the delay performance resulting from the use of different voice codec algorithms. The results of our study show that using the voice codec algorithm with a higher coding rate nd the queuing algorithm of WEQ can provide users with high-quality voice traffic.

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