• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2013.05a
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• pp.544-547
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• 2013

In this paper defined UID management, profile management, and they are also implemented and tested for interoperability in testbed including heterogeneous wireless networks environment. Also Architecture and roles of UID, Streaming Service Provider, information flows, unified profiles are studied, designed and implemented to be test. The study result include the functional architectures, requirements and main information flows of UID and Streaming Service Provider during roaming to heterogeneous wireless networks.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.9 no.1
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• pp.24-28
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• 2010

The Quality of Experience (QoE) of next Generation wired/wireless network services based upon IP networking is becoming a popular issue in recent years. The user experience of Internet services such as IPTV, online game, web surfing and etc, are becoming the most desirable factors to service providers to improve service performance and customer's satisfaction. However, collecting user experience from customers and obtaining the QoE parameters from the Quality of Service (QoS) parameters such as bandwidth, delay, jitter or admission control algorithm, are difficult subjects because of the various service types and user characteristics. In this paper, we propose a framework which contains service classification, QoE analysis and service enhancement steps for a suitable QoE measurement and management protocol. We define the user satisfaction indicators of the Internet services, classify the categories of each type of services, and analyse the Key Performance Indicator (KPI) in each type of services to perform the QoS parameters and improving the service qualities.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.3B
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• pp.208-213
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• 2011

Since the wireless technologies based on the IP have been developed recently, a variety of mobile services has been proposed. For broadcasting services to utilize network resources efficiently, there is a need to support handover of multicast traffic in mobile networks. In this paper, we add a network function block to solve the tunnel convergence problem, and propose a new fast handover scheme to reduce handover latency for PMIPv6 networks with multicast function. With the simulation result, we show that our fast handover scheme has better performance than the draft in the IETF in the viewpoint of handover latency and multicast service disruption.

• Journal of information and communication convergence engineering
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• v.7 no.2
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• pp.135-140
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• 2009

The wide deployment of wireless access technologies and the integration of various access network interfaces into a single terminal, allows mobile end-users to be always connected to the IP network, and to use those interface simultaneously. In this paper the CTE provides various access network interfaces capabilities, allowing reception of data over multiple service providers with different characteristics. Considerations for multiple network and service provider environments are regarded as essential for the successful deployment of convergence services in Next Generation Network (NGN). Event Driven Service (EDS) is regarded as a typical convergence service converging different functions of multiple service providers. This paper first describes the deployment model of NGN convergence services for multiple service provider environments. It also covers the service scenario of EDS, a convergence service for multiple service provider environments in NGN. Multiple provider environments stimulates the unified identifier management, namely ubiquitous identification (U-ID), to enable users to be provided convergence services without awareness of multiple provides. Then the designed structure and procedures of U-ID based EDS are given. Finally, the implementation results tested on Korea Advanced Research Network (KOREN) are described.

• Journal of KIISE:Information Networking
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• v.28 no.3
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• pp.389-405
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• 2001

In these days, we have increasing demands on the real-time services, especially for the multimedia data transmission in both of wired and wireless environments and thus efficient and stable ways of transmitting realtime data are needs. Although RTP is widely used for internet-based realtime applications, it cannot avoid packet losses, due to the use of UDP stack and its underlying layers. In the case of mobile computing applications, the packet losses are more frequent and consecutive because of the limited bandwidth. In this paper, we first statistically analyze the characteristics of packet losses in the wired and wireless communications, based on Gilbert model, and a new packet recovery scheme for realtime data transmission is presented. To reflect the transmission characteristics of the present network environment, our scheme makes the sender to dynamically adjust the amount of redundant information, using the current packet loss characteristic parameters reported by the receiver. Additionally, we use relatively large and discontinuous offset values, which enables us to recover from both of the random and consecutive packet losses. Due to these characteristics, our scheme is suitable for the mobile computing environment where packet loss rates are relatively high and varies rapidly in a wide range. Since our scheme is based on the analytic model form statistics, it can also be used for other network environments. We have implemented the scheme with Mobile IP and RTP/RTCP protocols to experimentally verify its efficiency.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.3 no.3
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• pp.266-284
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• 2009

The challenges of rapidly growing numbers of mobile nodes in IPv6-based networks are being faced by mobile computing researchers worldwide. Recently, IETF has standardized Mobile IPv6 and Fast Handover for Mobile IPv6(FMIPv6) for supporting IPv6 mobility. Even though existing literatures have asserted that FMIPv6 generally improves MIPv6 in terms of handover speed, they did not carefully consider the details of the whole handover procedures. Therefore, in conventional protocols, the handover process reveals numerous problems manifested by a time-consuming network layer based movement detection and latency in configuring a new care of address with confirmation. In this article, we study the impact of the address configuration and confirmation procedure on the IP handover latency. To mitigate such effects, we propose a new scheme which can reduce the latency taken by the movement detection, address configuration and confirmation from the whole handover latency. Furthermore, a mathematical analysis is provided to show the benefits of our scheme. In the analysis, various parameters are used to compare our scheme with the current procedures, while our approach is focused on the reduction of handover latency. Finally, we demonstrate total handover scenarios for the proposed techniques and discussed the major factors which contribute to the handover latency.

• Management & Information Systems Review
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• v.28 no.3
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• pp.191-210
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• 2009

Internet was first designed only for sending data, but as the time passed, internet started to evolve into a broadband multi-media web that is capable of transmitting sound, video, high-capacity data and more due to the demands of internet users and the rapid changing internet-communication technology. Domestically, in January, 2000 Saerom C&T, launched a free VoIP, but due to limited ways of conversation(PC to PC) and absence of a revenue model, and bad speech quality, it had hit it's growth limit. This research studied VoIP based on technological enhancement in super-speed internet. According to IDC, domestic internet market's size was 80,800 million in 2008, and it formed a percentage of 12.5% out of the whole sound-communication market. in case of VoIP, it is able to maximize it's profit by connecting cable and wireless network, also it has a chance of becoming firm-concentrated monopoly market by fusing with IPTV. Considering the fact that our country is insignificant in MVNO revitalization, regulating organizations will play a significant roll on regulating profit between large and small businesses. Further research should be done to give VoIP a secure footing to prosper and become popularized.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.223-228
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• 2008

Recently, various types of wireless access networks, such as WLAN, WiBro and HSDPA, etc, have been successfully deployed by commercial service providers (i.e., KT, KTF). In this situation, there are many efforts to provide high quality of services to guarantee seamless mobility between heterogeneous networks. The IP layer mobility protocols are efficient mechanisms to provide seamless mobility between IP based heterogeneous networks as well as homogeneous networks. However, to apply IP mobility protocols in real heterogeneous networks (i.e., WiBro and HSDPA), we must consider not only the basic features of techniques of wireless access networks (i.e., Data rate, Coverage, Quality of Service) but also the problem of real environment of service provider (i.e., Expanse cost to change the access network). Due to this reason, it is difficult to satisfy required conditions by using only one IP mobility protocol in real heterogeneous networks. Therefore, in this paper, we propose an efficient mobility protocol to solve the complex problems that are occurred in real heterogeneous networks. The proposed protocol, so-called, "Hybrid Mobile IP" tries to provide a synergy effect by integrating Client Mobile IPv4 (CMIPv4) and Proxy Mobile IPv4 (PMIPv4), and using the two mobility protocols selectively according to the situation of real heterogeneous networks.

• Journal of Korea Multimedia Society
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• v.11 no.5
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• pp.673-684
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• 2008

In this paper, we propose a signaling protocol which can support QoS for multimedia applications continuously after handover of a mobile node in the network environment where IP-based wireless access networks co-exist. Since existing RSVP-based signaling protocols can cause the waste of network resources due to double reservation and do not provide fast resource reservation/release, they are not suitable for mobile networks. The proposed QoS signaling protocol can support QoS for multimedia applications using the fast reservation of necessary resources after handover and avoid the waste of resources by resolving the double reservation problem. This paper presents design and implementation architectures of the proposed signaling protocol and analyzes its performance via a physical testbed. Based on the analysis results, it was shown that the proposed protocol is able to support seamless QoS for multimedia applications.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.7
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• pp.10-18
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• 2013

With advancing of mobile communication services and commercializing of VoLTE (Voice of LTE), it is getting to pay attention on QoS of VoLTE. This paper proposes an integrated E-model in which some factors influenced to service quality of VoIP and VoLTE based voice communication system are considered in calculating the voice quality of Wideband Codec. The model aims to calculate R value which reflects the situations of access network, network characteristics, terminals' usage and mobility. We mainly deal with the integrated E-model's structure, related algorithms and optimal parameters for VoLTE. Some experiments show that the voice quality difference between VoIP and VoiceChecker, and VoLTE and POLQA, is below 10%. With the proposed model, we can calculate the voice quality by making use of the factors directly affected to service quality and the environment of VoLTE terminal and network. As a result, we can estimate the service quality in advance, without measuring it in real wireless environment.