• The Journal of Korean Institute of Communications and Information Sciences
• /
• 제30권8B호
• /
• pp.507-513
• /
• 2005

As the rapid increasement of the number of internet users has occured recently, many multimedia application services have been emerging. To improve quality of service, traffic can be suggested to be classified with priority in EPON(Ethernet Passive Optical Network), which is supporting the multimedia application services. In this paper, multimedia application services treat bandwidth classifying device in serving both delay sensitive traffic for real-time audio, video and voice data such as Von(Voice over Internet Protocol), and for real-time traffic such as BE(Best Effort). With looking through existing mechanisms, new mechanism to improve the quality will be suggested. The delay performances and packet losses of traffic achieved by supporting bandwidth allocation of upstream traffic in suggested mechanism will be analized with simulations.

• Journal of information and communication convergence engineering
• /
• 제6권1호
• /
• pp.68-72
• /
• 2008

The VoIP(Voice over IP) has been worldwide used and already put to practical use in many fields. However, it is needed to ensure secret of VoIP call in a special situation. It is relatively difficult to eaves-drop the commonly used PSTN in that it is connected with 1:1 circuit. However, it is difficult to ensure the secret of call on Internet because many users can connect to the Internet at the same time. Therefore, this paper suggests a new model of Internet telephone for eavesdrop prevention enabling VoIP(using SIP protocol) to use the VPN protocol and establish the probability of practical use comparing it with Internet telephone.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• 제11권2호
• /
• pp.245-252
• /
• 2011

VoIP transmits voices over IP-based networks and it is the abbreviation of Voice over Internet Protocol. Recently, VoIP provides various services in addition to voices. Since VoIP services' provision is extending, VoIP service quality management is becoming an important issue. Therefore, in this paper, we study VoIP service quality management. We examine VoIP technology, service types, and network architecture. Then, we investigate key quality indicators(KQIs)/key performance indicators(KPIs) in terms of customers, not network service providers. Toward this, we also study the concept of general service quality management as well as the concept of telecommunication related service quality management. Moreover, we apply $\bar{x}$ and R charts to show how to use statistical quality control techniques in real telecommunication companies with one KQI.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• 제40권12호
• /
• pp.18-26
• /
• 2003

In this paper, we design and implement a QoS control function in the SIP-based VoIP system. As a network infrastructure for VoIP service, we select the Intserv over Diffserv architecture where the network resources are managed by a call admission control mechanism. The SIP protocol extended to support QoS signaling procedure is modulized to operate independently with the infrastructure. The performance of the QoS-enabled VoIP system is verified by experiments.

• Proceedings of the Korean Information Science Society Conference
• /
• 한국정보과학회 2002년도 가을 학술발표논문집 Vol.29 No.2 (3)
• /
• pp.124-126
• /
• 2002

Voice over Internet Protocol(VoIP) is expected to be a major application on the Internet in the future This paper propose an approach to VoIP that uses Differentiated Service and Multi-protocol Label Switching(MPLS) to provide quantitative QoS guarantees over an IP network. An algorithm that determines QoS-constrained routes is proposed and a framework that uses such an algorithm for traffic engineering is outlined. the key component of this framework is a Centralize Resource Manager(CRM) responsible for monitoring and managing resources within the network and making all decisions to route／reroute traffic according to QoS requirement

• Journal of the Korea Institute of Information and Communication Engineering
• /
• 제15권11호
• /
• pp.2391-2398
• /
• 2011

National Information and Communication Services of Public Administration and Security organized by the 'A 'group (Line service network), 'B' group (IP service network), 'C' Group (Voice over Internet protocol(VoIP) service, IP application services) are provided by constructing the infrastructure. National Information and Communications Services 'C' group, providers are providing VoIP services. In this paper, national information and communications service 'C' group, providers of domestic calls, international calls, including calls to move we will study the basic telephone service. And text messaging, video telephony, IP-Centrex services, etc. we will study the seven value-added services. In addition, national information and communication service providers on the status of the project based on the analysis of national information and communication Internet telephone network using Internet telephony is the type of analysis. In this study, national information and communications services industry, will serve as the basis for the development.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 한국해양정보통신학회 2011년도 추계학술대회
• /
• pp.249-252
• /
• 2011

National Information and Communication Services of Public Administration and Security organized by the 'A' group (Line service network), 'B' group (IP service network), 'C'Group (Voice over Internet protocol(VoIP) service, IP application services) are provided by constructing the infrastructure. National Information and Communications Services 'C' group, providers are providing VoIP services. In this paper, national information and communications service 'C' group, providers of domestic calls, international calls, including calls to move we will study the basic telephone service. And text messaging, video telephony, IP-Centrex services, etc. we will study the seven value-added services. In addition, national information and communication service providers on the status of the project based on the analysis of national information and communication Internet telephone network using Internet telephony is the type of analysis. In this study, national information and communications services industry, will serve as the basis for the development.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• 제10권5호
• /
• pp.821-831
• /
• 2006

In this paper we researched on the development methods and design of the B-ISDN protocol embedded Internet-Phone PBX System for the previous telephone exchange and communication. We designed the structure of the user terminal md exchange server of the Internet-Phone PBX System, defined message and data structure for call control, and designed call control message flows between sender and receiver terminal for user's communication.

• The Journal of the Korea institute of electronic communication sciences
• /
• 제9권7호
• /
• pp.767-772
• /
• 2014

As temporary network, ad-hoc network has been effected by structures and implemented environments of networks. In this paper, transmission performance of lattice structure ad-hoc network, which is expected to use in sensor network and IoT(Internet of Things), is analyzed in point of intrusions and countermeasure for intrusion is suggested. In this paper, computer simulation based on NS-2 is used for performance analysis, VoIP(Voice over Internet Protocol) as a widely used service is chosen for performance measure. MOS(Mean Opinion Score) and call connection rate is used as performance parameter. As results of performance analysis, it is shown that for MOS, random network is better then lattice network at intrusion environments, but for call connection rate, lattice network is better then random network.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• 제12권12호
• /
• pp.5917-5935
• /
• 2018

Adaptive Multi-Rate Codec is one of the codecs which is used for making voice calls. It helps to connect people who are scattered in various geographical areas. It adjusts its bit-rate according to the user's channel conditions. It plays a vital role in providing an improved speech quality of voice connection in Long Term Evolution (LTE). There are some constraints which need to be addressed in providing this service profitably. Quality of Service (QoS) is the dominant mechanism which determines the quality of the speech in communication. On several occasions, number of users are trying to access the same channel simultaneously by standing in a particular region for a longer period of time. It refers to Multi-user channel sharing problem which leads to resource loss very often. The main aim of this paper is to develop a novel RSA - QoS scheduling algorithm for reducing the Resource Loss Ratio. Eventually, it increases the throughput.The simulation result shows that the RSA - QoS increases the number of users for accessing the resources better than the existing algorithms in terms of resource loss and throughput. Ultimately, it enhances the QoS in Mobile Networks.