• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.2
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• pp.249-255
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• 2012

This paper covers the service quality of packet-based voice service that is provided over a high speed downlink packet access (HSDPA) system. Using the measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over a wireless network[2][3], a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement result, the service quality of the voice service is supposed to be quite good over HSDPA system. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metrics in which mean opinion score (MOS) starts to decrease.

• IEMEK Journal of Embedded Systems and Applications
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• v.5 no.4
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• pp.185-194
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• 2010

In this paper, we propose an application layer-based vertical handover management protocol, called multihoming RTP (mhRTP), for real-time applications with seamless mobility across heterogeneous wireless access networks. The proposed multi-homing RTP provides a soft handover by utilizing multiple available wireless access network interfaces simultaneously. The newly available path is dynamically added to the ongoing session by the mhRTP session manager. Also the decision making of QoS-improving or QoS-guaranteed handover is possible based on the estimation of available bandwidth in each candidate network. The performances of the proposed mhRTP have been analyzed through a series of simulations on OPNET network simulator. From the performance analysis, we confirmed that the proposed mhRTP can provide QoS-guaranteed vertical handover with efficient session managements.

• International Journal of Quality Innovation
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• v.10 no.1
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• pp.115-126
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• 2009

Video transmission effectiveness in the Ad Hoc network is becoming important recently, if different routing protocols are applied. Some researchers conclude that the reactive protocols are better for file transfer protocol (FTP) and constant bit rate (CBR) or hypertext transfer protocol (HTTP) transmission in an Ad Hoc wireless network but the performance report of video transmission is not much. This study adopts Qualnet (Network Simulator) as a simulation tool for environmental designing and performance testing, and employs an experimental design with eight groups. Our experiment shows that: (1) The performance of AODV (reactive) protocol is better than DSDV, ZRP and DSR when the transmission load has only one video stream; (2) Proactive (DSDV) and Hybrid protocols (ZRP) are better for a smaller Ad Hoc network when it transmits a video stream with some applications (VoIP, FTP and CBR). We conclude that packet loss rate is sensitive to the quality of video transmission and it has negative relationship with Peak Signal-to-Noise Ratio (PSNR) value. In addition, our experiment also shows that PSNR is a simple Metric for the performance evaluation of video transmission.

• Journal of Digital Convergence
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• v.10 no.1
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• pp.49-57
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• 2012

The Internet succeeded because of the end-to-end principle which allowed anyone to add functionality to the network. However, as the internet is increasingly becoming the platform for smart IT applications such as VoIP, IPTV, Cloud Computing and Smart Phone, networks are now under increasing strain of traffic congestion and the absence of quality of service insurances. To date, the debate over internet rules has focused on network neutrality rather than network efficiency. This article emphasizes the well-functioning role of market mechanism for the efficient use and further development of the network. To maximize the value of the network, this article proposes a differential treatment to packets based on customer types, and a two-part tariff pricing rule to secure funding to expand and upgrade networks.

• Proceedings of the Korea Information Processing Society Conference
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• 2006.05a
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• pp.1267-1270
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• 2006

최근들어 초고속 통신망(xDSL, FTTH, Cable)을 이용하여 Head-End Server에서 가입자 댁내의 셋탑박스를 TV를 이용하여 멀티미디어(다채널 SD/HD 방송, VOD) 및 양방향 인터넷 연동 서비스(T-Portal, T-Communication) 등을 제공하는 것이 가능해 졌다. 더불어 고객들에게 인터넷을 통한 이 서비스의 품질을 보장하는 망 관리 관점에서의 기능도 필요하게 되었다. 고객들은 예전보다 더 나은 통신품질을 제공받기를 기대하고 있다. 따라서, 기존의 PE-to-PE 통신 품질 뿐만 아니라 CE-to-CE 통신 품질도 제공함으로써 고객체감품질 보증의 만족도를 향상시켜야 한다 본 논문에서는 IP 서비스(IPTV, VoIP 등)의 SLA에서 사용될 품질 지표에 관한 연구 및 PE-to-PE구간에서 부터CE-to-CE 구간 까지의 시스템 망 구성도, SLA 관리시스템에 관해 설명한다. 본 논문의 SLA 시스템에서는 가입자 댁내의 셋탑박스에 내장된 품질측정 Agent를 이용하여 통신품질 데이터를 측정한 후, 가공하여 SLA를 적용함으로써, 보다 정확한 고객체감품질 보증 및 고객에게 SLA의 신뢰성을 줄 수 있다. 또한, 서비스 제공자로 하여금 사전에 통신품질을 감시할 수 있는 기능을 제공함으로써 양질의 통신품질을 보장할 수 있도록 하는데 목적이 있다.

• Journal of Distribution Science
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• v.16 no.6
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• pp.37-53
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• 2018

Purpose - There have been not enough studies on the ways customer Moment of Truth(MOT) activities are structured along with consumption chain and their influences as well as the relative influences of service quality at Moment of Truth on customer performance. Therefore, customer service needs needed at Moment of Truth may differ depending on whether these distribution services are at introduction-growth stages or maturity-decline stage already, but there is no study which illustrates this. Research design, data, and methodology - This study selects VoIP and IPTV as the household telecommunication and distribution services at introduction-growth stages as well as high speed internet and wire telephone as those at maturity-decline stages. Then it identifies which experiences that customers have at Moment of Truth by each service as well as the influences related to what the customers consider as important. Results - As the result of demonstration with the target of 858 respondents, customers' experiences and requests differ at Moment of Truth. For service quality, what takes the positive roles in customer performance includes corporeality and certainty for the services at introduction-growth stages as well as reliability, sympathy, and mutuality for those at maturity-decline stages. Conclusions - Implications of these results as well as further directions for study are suggested.

• The KIPS Transactions:PartC
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• v.13C no.4 s.107
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• pp.427-434
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• 2006

Broadband Convergence Network(BcN) is a critical infrastructure to provide wired-and-wireless high-quality multimedia services by converging communication and broadcasting systems, However, there exist possible danger to spread the damage of an intrusion incident within an individual network to the whole network due to the convergence and newly generated threats according to the advent of various services roaming vertically and horizontally. In order to cope with these new threats, we need to analyze the vulnerabilities of BcN in a system architecture aspect and classify them in a systematic way and to make the results to be utilized in preparing proper countermeasures, In this paper, we propose a new classification of vulnerabilities which has been extended from the ITU-T recommendation X.805, which defines the security related architectural elements. This new classification includes system elements to be protected for each service, possible attack strategies, resulting damage and its criticalness, and effective countermeasures. The new classification method is compared with the existing methods of CVE(Common Vulnerabilities and Exposures) and CERT/CC(Computer Emergency Response Team/Coordination Center), and the result of an application to one of typical services, VoIP(Voice over IP) and the development of vulnerability database and its management software tool are presented in the paper. The consequence of the research presented in the paper is expected to contribute to the integration of security knowledge and to the identification of newly required security techniques.

• Proceedings of the Korea Society for Industrial Systems Conference
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• 2003.05a
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• pp.8-13
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• 2003

Historically, IP-based internets have been able to provide a simple best-effort delivery service to all applications they carry. Best effort treats all packets equally, with no service level, packet loss, and delay. But the needs of users have changed. The want to use the new real-time, multimedia, and multicasting applications. Thus, there is a strong need to be able to support a variety of traffic with a variety of quality-of-service requirements. The DiffServ architecture, proposed by the Internet Engineering Task Force(IETF), has become the most viable solution for provising QoS over IP networks. The DiffServ architecture does not specify any handling method between AF out-profile packets and BE packets. This paper propose a mechanism for supporting inter class fairness in the DiffServ architecture. Ⅰ proposed a modified Weighted Round Robin method to protect the BE traffic from AF out-profile packets in the core routers. The proposed technique is evaluated through simulation. Simulation results indicate that the proposed method provides better protection not only for BE packets from AF out-profile packets, but also for the AF in-profile packets in congested networks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.7B
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• pp.528-535
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• 2012

This paper proposes silence drop first(SDF) active buffer management algorithm to increase the voice capacity when silence suppression is used. This algorithm finds and drops silence packet rather than voice packet in the queue for resolving buffer overflow of queue. Simulations with voice codec of G.729A and G.711 are performed. By using proposed SDF algorithm, the voice capacity is increased by 84.21% with G.729A and 38.46% with G.711. Further more, SDF algorithm reduces the required link capacity and loosens the silence packet inter-arrival time limit to provide target voice quality compared with that of conventional algorithms.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.1
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• pp.23-34
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• 2004

This paper presents architecture and control method of packet scheduler to guarantee QoS of high quality streaming services in high-speed packet-switched networks. Since streaming services need far more stringent QoS requirements than the typical sort of burst data applications, they should be guaranteed minimum bandwidth and end-to-end delay bound to each flow, regardless of the behavior of other flows. To meet these requirements, a packet scheduler isolate a flow from the undesirable effects of other flows and provides end-to-end delay guarantees for individual flow and divides stringently the available link bandwidth among flows sharing the link. Until now, many vendors are developing traffic management chips running at 10Gbps, but most of chips have drawbacks to support high quality streaming services. In this paper, we investigate the drawbacks of commercial TM chips and traffic characteristic of streaming services and present implementation frameworks of the proposed packet scheduler. Finally, we analyze the simulation results of the proposed scheduler.