• Electronics and Telecommunications Trends
• /
• v.16 no.1 s.67
• /
• pp.18-23
• /
• 2001

본 논문에서는 인터넷망에서 MPLS 기술을 이용해서 고품질 음성 서비스를 지원하기 위한 Voice over MPLS 기술의 표준화 동향에 대해 조사 분석하고, 국내 기술 개발시의 고려 사항에 대해 고찰한다. Voice over MPLS 기술에 대한 표준화는 IETF와 MPLS Forum에서 적극적으로 추진하고 있으며, 기술적으로는 Voice directly over MPLS 구조와 VoIP over MPLS의 두 가지 구조가 연구되고 있다.

• Proceedings of the Korean Institute of Intelligent Systems Conference
• /
• 2000.11a
• /
• pp.471-475
• /
• 2000

Voice and data are transmitted over a single packetized data communications network which is designed for data communications. The public switched telephone network for voice and the packet data network for data are merging into a single data network to get efficiency and to reduce operational cost. However, integrating voice and data transmission over a single data network is not easy because voice should be transmitted without delay but data should be transmitted without error. Advances in technology begin to overcome basic differences. Several integration methods in voice and data will be examined and reviewed here. Moreover, trends and problems on integration will be also discussed.

• The Journal of the Acoustical Society of Korea
• /
• v.37 no.6
• /
• pp.512-517
• /
• 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.

• Broadcasting and Media Magazine
• /
• v.19 no.1
• /
• pp.48-55
• /
• 2014

One of the recent noticeable evolutions in mobile communication systems is that wideband-codec is deployed rapidly in VoLTE (Voice over Long Term Evolution) service or HD voice. This paper is concerned with next generation HD voice or VoLTE service that is coined to describe high quality communication with super-wideband voice codec. 3GPP EVS (Enhanced Voice Service) Codec is being standardized to develop the super-wideband voice codec. This paper deals with the codec design constraints, performance requirements, the status of standardization, and finally perspective on VoLTE service in future.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2018.10a
• /
• pp.568-570
• /
• 2018

Currently, with rapid progress and supply of mobile communication technology, LTE(Long Term Evolution) technology is expanded and widely used to industrial and emergency communications beyond earlier smart-phone based service. In this paper, transmission performance of voice traffic, one of railway communication service based on LTE-R as an application field of LTE technology, is analyzed. This study is performed performance analysis with level of application service and consider effects of satisfaction level for users. Computer Simulation based on ns(Network Simulation)-3 is used for analysis and VoIP(Voice over Internet Protocol) specification is used for voice traffics. Results of this paper is used to implement LTE-R networks and develope application services over LTE-R network.

• Proceedings of the Korean Information Science Society Conference
• /
• 2002.04a
• /
• pp.226-228
• /
• 2002

이동통신망이 발달하면서 기존의 인프라를 생성하였던 ATM망을 음성서비스에 활용하기 위한 Voice over ATM기술이 많은 발전을 하고 있다. 본 논문에서는 ATM 망에서 Low-bit-rate voice 트래픽을 처리하기 위해 1020개의 커넥션을 지원하는 AAL2 Voice gateway를 구현하였다. 본 논문에서 구현된 AAL2 Voice gateway는 ITU-T Recommendations 1.363.2와 1.366.2에 근거하여 설계하였다. 또한, SSCS(Service Specific Convergence Sublayer)와 CPS(Common Part Sublayer) 부계층을 모두 하드웨어로 구현하였다. 본 AAL2 Voice gateway는 VHDL로 구현되었으며, 이동통신망과 VoDSL의 Voice gateway에서 그 효용 가치가 높을 것으로 사료된다.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.425-428
• /
• 2000

• International journal of advanced smart convergence
• /
• v.1 no.2
• /
• pp.34-38
• /
• 2012

The recent increase in use of the IEEE 802.15.4 standard for wireless connectivity in personal area networks makes of it an important technology for low-cost low-power wireless personal area networks. Studies showed that voice communications over IEEE 802.15.4 networks is feasible by Guaranteed Time Slot (GTS) allocation; but there are some constraints to accommodate voice transmission beyond two hops due to the excessive transmission delay. In this paper, we propose a GTS allocation scheme for bidirectional voice traffic in IEEE 802.15.4 multihop networks with the goal of achieving fairness and optimization of resource allocation. The proposed scheme uses a greedy algorithm to allocate GTSs to devices for successful completion of voice transmission with efficient use of bandwidth while considering closest devices with another factor for starvation avoidance. We analyze and validate the proposed scheme in terms of fairness and resource optimization through numeral analysis.

• The KIPS Transactions:PartD
• /
• v.14D no.5
• /
• pp.565-572
• /
• 2007

In this paper, we developed a VoWLAN(Voice over WLAN) system based on an open source software. The system aims to provide VoIP service over wireless LAN with an IP-PBX server. The features of system presented in this paper are as follows. First, the initial cost for the development is reduced since the system is developed based on open source software. Second, the system provides various additional services such as Voice Mail, Conference Call, and Interactive Voice Response with a software IP-PBX server. Third, the VoWLAN terminal provides high-level user applications with minimal system resources using lightweight open software solutions. Finally, it is highly scalable since it is based on the open source software.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.16 no.11
• /
• pp.2405-2410
• /
• 2012

The 4th generation mobile communication system, LTE, does not support an additional core network to provide voice service, and it is merged into a packet network based on all IP. Although Voice service over LTE can be supported by VoIP, it will be provided by the existing 3G networks because of the discontinuity of LTE coverage. However, it is inevitable to adopt VoIP over LTE to provide high quality voice service. In this paper, we investigate the capacity of VoIP over LTE. Our results indicate that spectral efficiency can be significantly improved as channel bandwidth increases in terms of VoLTE capacity. In addition, we can achieve higher VoLTE capacity without decreasing control channel capacity.