• Proceedings of the IEEK Conference
• /
• 2003.11c
• /
• pp.153-156
• /
• 2003

In this paper, we propose QoS guarantee system fur multi-functional VoIP end Terminal. This system guarantees low delay of voice data for Internet telephony in VoIP end terminal that has various kinds of Internet dependant application. QoS system we propose support low delay transmission in VoIP terminal interface.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.3B
• /
• pp.398-407
• /
• 2010

Due to increase of user requirement for various traffics and the advance of network technology, each distinct network has converge into FMC(Fixed Mobile Convergence) networks. However, we need to research the performance analysis of VoIP(Voice over Internet Protocol) in the FMC network to provide QoE for the voice user of FMC network. Therefore, this paper introduces the scenario which is the situation of voice quality degradation when a user uses VoIP to communicate with other users in the FMC network. Especially, this paper presents scenario in terms of the component of the network and finds the improvement point of voice quality. In the simulation results, three improvement points of voice quality are found as following: voice quality degradation by packet loss in the physical layer of the HSDPA network, by utilizing GGSN without QoS parameter mapping mechanism which is gateway between 3GPP and IP backbone, and by using non-QoS AP in the WLAN network.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.10 no.8
• /
• pp.3728-3752
• /
• 2016

Wireless mesh networks (WMNs) have been increasingly applied in private and public networks during the last decade. In a different context, voice over IP (VoIP) has emerged as a new technology for making voice calls around the world over IP networks and is replacing traditional telecommunication systems. The popularity of the two technologies motivated the deployment of VoIP over WMNs. However, VoIP over WMNs suffers from inefficient bandwidth utilization because of two reasons: i) attaching 40-byte RTP/UDP/IP header to a small VoIP payload (e.g., 10 bytes) and ii) 841 μs delay overhead of each packet in WMNs. Among several solutions, VoIP packet multiplexing is the most prominent one. This technique combines several VoIP packets in one header. In this study, we will survey all the VoIP multiplexing methods over WMNs. This study provides a clear understanding of the VoIP bandwidth utilization problem over WMNs, discusses the general approaches in which packet multiplexing methods could be performed, provides a detailed study of present multiplexing techniques, shows the aspects that hinder the VoIP multiplexing methods, discusses the factors affected by VoIP multiplexing schemes, shows the merits and demerits of different multiplexing approaches, provides guidelines for designing a new improved multiplexing technique, and provides directions for future research. This study contributes by providing guidance for designing a suitable and robust method to multiplex VoIP packets over WMNs.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2002.11a
• /
• pp.839-843
• /
• 2002

In this paper, we present a next generation VoIP network architecture. Specifically, we present key components such as SoftSwitch and media gateway, and important protocols for VoIP services. We also present basic components and mechanisms for VoIP QoS. The architecture presented in this paper is able to support open interfaces and multimedia traffic, and therefore various multi-services can be supported efficiently using the architecture.

• The Journal of the Acoustical Society of Korea
• /
• v.28 no.2E
• /
• pp.35-42
• /
• 2009

In this paper, we present criteria of objective measurement of speech quality to provide the mobile-VoIP services efficiently over wireless mobile internet. The mobile-VoIP service, which is based on mobility and is error-prone compared to conventional VoIP over wired network, is about to be launched, but there have not been adequate quality indexes and the Quality of Service (QoS) standards for evaluating speech quality of Mobile-VoIP. In addition, there are many factors influencing on the speech quality in packet network of which packet loss contribute directly to the overall voice communication quality. For this reason, we adopt the Gilbert-Elliot Channel Model for modeling packet network based on IP and assess the voice quality through the objective speech method of ITU-T P. 862 PESQ and ITU-T P. 862.1 MOS-LQO under various packet loss rates in the transmission channel environments. Our simulation results address the specific criteria and QoS for the mobile-VoIP services in terms of the various packet loss environments.

• The KIPS Transactions:PartC
• /
• v.10C no.4
• /
• pp.501-508
• /
• 2003

VoIP that conveys voice in internet is getting into the spotlight as means to alternate existing PSTN in corporation as well as users. Current VoIP is furnishing voice efficiently, but it needs to support various services for VoIP acceleration. IETF is developing CPL standard which is call processing language for supporting various services. User has to store script to specific server for the use of CPL, therefore it is required client to support CPL. This paper describes about design and implementation of SP-based CPL client for various services. The CPL client was implemented using LINUX 2.4.x, C, and GTK1.2.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.9 no.2
• /
• pp.73-80
• /
• 2009

Although it is relatively difficult to eavesdrop the commonly used PSTN in that it is connected with direct circuit, it is difficult to ensure the secret of call on Internet because many users can connect to the Internet at the same time. However, it is needed to ensure secret of voice call in a special situation. Due to the fact that many users can connect to the internet at the same time, VoIP can always be in a defenseless state by hackers. Therefore, in this paper, we have developed the increased voice security internet telephone terminal and measured conversation quality by adopting VPN PPTP based on SIP and using tunnel method in transmitting voice data to prevent eavesdrop of internet telephone.

• Smart Media Journal
• /
• v.4 no.4
• /
• pp.47-55
• /
• 2015

Recently, the computing paradigm has been changing and VoIP technology is being revisited to support various services in ICT field. In this paper, we have designed and implemented the systems of software PBX open source Asterisk using light-weighted virtualization Docker technique, hardware platform, and mobile devices to support voice service based on secured mobile VoIP. And we verified the delay test of network traffics and the secured voice communication test in global real network environment.

• Proceedings of the Korea Multimedia Society Conference
• /
• 2001.11a
• /
• pp.41-45
• /
• 2001

본 논문에서는 인터넷 망을 통해 음성 데이터를 전달한 수 있는 VoIP 기술과 인도어(indoor) 환경에서 사용자에게 이동성을 제공하는 블루투스 기술을 연동하여 새로운 VoIP wireless phone을 설계한다. 또한 본 논문에서 제안한 VoIP wireless phone의 호 접속제어를 위하여 새로운 프로토콜과 프로토콜내에서 사용되는 메시지를 설계한다.

• Journal of Digital Convergence
• /
• v.9 no.5
• /
• pp.225-233
• /
• 2011

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 a recording system which records voice data of SIP-based Internet telephone's voice calls. It is developed on the linux system and has some features such as audio mixing of two in/out voice channels, live packet sniffing, and the ability to transfer mixed audio files to the log file server. These functions are implemented using various open source softwares. Afterwards, this VoIP recording system will be applied as a base technology to advanced services like a VoIP-based call center system.