• Journal of KIISE:Information Networking
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• v.32 no.1
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• pp.80-88
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• 2005

In ubiquitous spaces, a user wants to be provided with a VOD service while moving one space to another freely. Since the traditional VOD system is dependent on user location, the VOD server transmits video data to a single client during a session. If the user moves to another space, he/she should close the old session and make a new request for the same video. However, the location-aware VOD system supports user mobility by closing and opening the session automatically. That is, the VOD system automatically perceives the movement of a user and allows the server to change the data flow so that a client near the user can receive the video data. This paper proposes a location-aware VOD service architecture and a session handoff scheme to provide a mobile user with continuous video delivery and implements a location-aware VOD prototype system based on Jini and Java. In experiment, we show that the proposed handoff scheme has a smaller handoff delay than the other handoff scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.7B
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• pp.745-759
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• 2011

In this paper, we propose the global handover mechanism that is able to provide the unlimited range of next-generation multimedia mobile services in an integrated environment. This mechanism consists of a multicast source based forwarding scheme and a global session management scheme. Global session management scheme provides LMA session information management, global mobility and route optimization. Multicast source based forwarding scheme delivers data between previously attached LMA and newly attached LMA without packet loss. In addition, this scheme removes the redundancy of buffered data. We present a performance evaluation and features analysis by the simulations using the ns-2. Global session management scheme has a less handover latency, propagation delay and signaling cost than the conventional methods. Multicast source based forwarding scheme can deliver buffer data without loss and it has less buffer size than conventional method.

• Journal of KIISE:Information Networking
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• v.31 no.6
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• pp.608-619
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• 2004

Application layer multicast(ALM) has been developed as an alternative to support the multi-receiver applications due to the slow deployment of IP multicast. Unlike routers in IP multicast, end systems participating in a ALM session are not optimized for relaying data and have various processing and network capacity. Therefore, with the increased role of end systems, the delay incurred at the end systems occupy large portion of the end-to-end delay. In this paper, we model the end system delay of ALM sessions and analyze its impact on the performance of the existing ALM schemes. Furthermore, we propose an enhanced scheme, based on the existing mesh-based scheme, which reflects the end system delay We also evaluate the performance of the proposed scheme by simulation and show that it reduces the end-to-end delay compared with the existing scheme.

• Journal of KIISE:Information Networking
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• v.29 no.5
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• pp.543-552
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• 2002

Wireless Session Protocol(WSP) which was updated and supplemented based on HyperText Transfer Protocol(HTTP) was designed by Wireless Application Protocol(WAP) forum regarding the characteristics of wireless environment. WSP improved the performance in wireless network, and introduced various facilities considering wireless environment. In this paper, we more improve the performance of WSP adding protocol message compression capability; we cail improved WSP protocol as WSP+. And, we analysis the performance of each protocol with WSP and WSP+ implementation. As a result of experiment, the capability which proposed in this paper reduced a response traffic about 45%. In $10^{-4}$ bit error rate, we also found the packet loss rate and time delay per transaction of WSP+ was improved over 40%. Finally, we found that the protocol message compression capability reduces message retransmission count in transaction layer and shorten the delay time per transaction by reducing a message size.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.2
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• pp.95-102
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• 2010

This paper proposes a SPIT(Spam over IP Telephony) prevention framework which is using expanded white-list in real-time VoIP environment. The existing schemes are vulnerable to attack from spammers since they can continue to transfer SPIT upon changing their ID. And existing frameworks have experienced the time delay and overload as session initiates due to real-time operation. To solve these problems, the proposed scheme expands the scope of white lists by forming social networks using the white list, but it is to decide quickly whether pass a caller ID without searching the entire database. The proposed framework takes the three-stage architecture and the fast scoring system. The proposed framework minimize user's inconvenience and time delay for initiation of session, therefore, it is proper for real-time VoIP environment.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.6
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• pp.49-57
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• 2010

In the near future IPTV services will be emerged the various types through Internet, but IMS based IPTV service is one of the very attractive IPTV services. This paper describes the interconnecting architectures of Web based IPTV contents provider to IMS(IP Multimedia System) network and describes the three difference architectures as method to find its IP address. One is the architecture using DNS or HSS to find IP address of Web based IPTV contents provider and connecting gateway function to I-CSCF in IMS. The other is the architecture connecting gateway AS to ISC interface of S-CSCF in IMS. This paper describes the characteristics of traffic generating due to interconnect the Web contents provider, and the traffic model of each architectures. The proposed each architecture is emulated the session establishment delay characteristics in CoD service of IPTV by the simulation. This paper shows that the architecture connecting gateway AS to ISC interface of S-CSCF is the excellent method compare to other two methods in view of the session establishment delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.1A
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• pp.57-65
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• 2007

The overlay multicast that has been recently presented as IP alternative for the IP multicast has been getting much persuasion by the computing power of the hardware and the advancement of the network techniques to enforce Routing in application-level. In an overlay multicast, the system resource and the network bandwidth must be utilized efficiently to service real-time HDTV images. Specifically, the system must consider the delay and the jitter that can be incurred at the application-level. In this paper, we implement a server and a client to broadcast HDTV, in the session composed by the existing overlay multicast protocol. The broadcasting server performs the service using a TV tuner, An HDTV camcorder, and files, clients constituting a multicast group relay the received data to other clients. At this time, the information that the clients report periodically, including their delay and the network state, to the server is used as an important information to maintain an overlay session. The implementation is based on the DirectX and its performance is evaluated by the LAN test bed that has been set.

• The KIPS Transactions:PartC
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• v.13C no.4 s.107
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• pp.461-470
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• 2006

Recently mobile Stream Control Transmission Protocol (mSCTP) has been proposed as a transport layer approach for supporting mobility. When a mobile terminal (MT) is not located in the home network. a terminal that wishes to communicate with the MT is not able to establish mSCTP association to the MT, since mSCTP does not include the location management mechanism. In order to solve this problem. an interworking approach using the Session Initiation Protocol (SIP) INVITE method has been proposed. However, this approach has shown subsequent delay in acquiring the current location information of the MT when initiating mSCTP association establishment. In this paper, we propose new SIP methods and an approach that minimizes the address acquisition delay (AAD) by utilizing those SIP methods. Mathematical analysis and simulation results show that the proposed approach is more efficient than the previous approach in terms of AAD in all kinds of SIP environments.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.17 no.2
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• pp.103-114
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• 2007

The existing handover authentication schemes have authentication delay and overhead of the authentication server since they have been separately studied handover authentication at the link layer and the network layer. This paper proposes a handover authentication scheme initiated by Mobile Node on FMIPv6 based mobile access networks. The main idea of the paper is to generate a session key at the mobile node side, and transfer it to the next Access Router through the authentication server. Also, the scheme has a hierarchical key management at access router. There are two advantages of the scheme. First, the generated session key can be utilized for protecting the binding update messages and also for access authentication. Second, hierarchical key management at the access router reduced the handover delay time. The security aspects on the against PFS, PBS, and DoS attack of proposed scheme are discussed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.8A
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• pp.1134-1143
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• 2000

We propose a scheduling algorithm, Bandwidth Allocation Scheme (BAS), that guarantees bounded delay in a switching node. It is based on the notion of the GPS (Generalized Processor Sharing) mechanism, which has clarified the concept of fair queueing with a fluid-flow hypothesis of traffic modeling. The main objective of this paper is to determine the session-level weights that define the GPS sewer. The way of introducing and derivation of the so-called system equation' implies the approach we take. With multiple classes of traffic, we define a set of service curves:one for each class. Constrained to the required profiles of individual service curves for delay satisfaction, the sets of weights are determined as a function of both the delay requirements and the traffic parameters. The schedulability test conditions, which are necessary to implement the call admission control, are also derived to ensure the proposed bandwidth allocation scheme' be able to support delay guarantees for all accepted classes of traffic. It is noticeable that the values of weights are tunable rather than fixed in accordance with the varying system status. This feature of adaptability is beneficial towards the enhanced efficiency of bandwidth sharing.