• Journal of KIISE:Information Networking
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• v.32 no.1
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• pp.51-59
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• 2005

In this paper, we propose and implement a hybrid-type video conference system using SIP protocol proposed in IEFT. This conference system, all signaling is controled by the server, while the media communication can be managed either by UAs or by the server. as a result, the complexity of its routing and user state management is reduced and it is applicable to the mobile environment of the ad-hoc models. The implemented server is scale enough to be used in a large conference because it only takes charge of signaling part. And also we see that UAs are only suitable in the small conference where less than ten people can join.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.8
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• pp.27-36
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• 1997

In this paper, we propose a locating strategy using a cache based on the distributed call processing stucture. In our stategy, each originating call server, withc is responsible for call setup, has a cache to memorize the service ratio of call server which serves the called terminal.We present performance comparison of the proposed strategy with the existing strategy via simulation. We can reduce the cost for call setup because the originating call server can simpley establish a call without tracing the location of the called terminal. The proposed strategy also reduceds network loads caused by signaling at call setup time and lessens the frequency of queries for the database systems which can be a bottleneck when it serves many subscriber.

• The Journal of the Korea Contents Association
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• v.7 no.3
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• pp.17-24
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• 2007

SIP-based tightly coupled conference, which has a centrally located conference server for controlling and management, can be classified several models according to location of focus and mixer. These are centralized server model, endpoint server model, media server component model and distributed mixing model. However each model has its strength and weakness. In this paper, we propose and implement a SIP-based effective hybrid-type conference model which decreases amount of SIP signaling messages, lowers load of server media mixer, and can be easily expandable to large scale conference. In this model, when the number of participants exceeds a pre-defined limit, the conference server selects some participants which posses specific functions and let them share functions of notifications of conference state event package and media mixing. When each participant subscribes conference state event package to the server, it can indicates its possession of such functions by a specific header message. The server stores the indication to the conference information database, and later uses it to select participants for load sharing. The performance of our proposed model is evaluated by experiments.

• The Journal of the Korea Contents Association
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• v.10 no.6
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• pp.80-90
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• 2010

In this paper, we introduce some issues to implement an agent system to support service mobility in home network environment, and describe detailed design method in terminal as well as server agent. Specifically, we describe user recognition module, signaling message receiving/parsing module of terminal agent and signaling message receiving/parsing module, multimedia switching module, memory management module of server agent. We define several parameters managed in IP sharing device and design binding table structure to support mobility. And we utilize M/M/1/K queueing theory to obtain relations between memory size, blocking probability and memory utilization. From the obtained results, we show that memory size can be predicted in server agent mounted on IP sharing device.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.1
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• pp.75-82
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• 2003

We can foresee the rapidly growing of users by development of VoIP service what can transmit a audio traffic with low cost among a lots application using internet. VoIP needs a standardized protocol that is able to do signaling for offering high quality of services such as mobility, universal number, multiparty conference, voice mail, automatic call distribution. At the present time, a base composition elements of SIP(Session Intiation Protocol) are developing for offering VoIP based SIP in the inside and outside of the country, because SIP of IETF which has a strength from 'fast connection', 'parsing' & 'easy to compile' points of view. This paper suggests a type of Hybrid SIP Server for providing some services as 'a reducing load of SIP server that process a request method from users', 'efficiency of managing networks', 'offering services to many users'.

• Journal of the Korea Society of Computer and Information
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• v.8 no.4
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• pp.117-125
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• 2003

The proposed mechanism has an group management server, which manages the information about both the receivers and the senders. Any receiver can dynamically join/leave the multicast VC. The signaling overload due to group membership changes is not concentrated on the sender, but it is distributed to many receivers for the scalability improvement. The associated signaling messages propagates from the receivers to the ATM switch dedicated to the multicast VC, and hence no signaling overload exists in the shared links there is no latency for the receiver to wait. Our proposed scheme is superior in the view of scalability, the efficiency and the latency to other schemes.

• KIPS Transactions on Computer and Communication Systems
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• v.1 no.2
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• pp.109-118
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• 2012

As a result of the prevalence of smartphone, various mobile services became faster by LTE networks. Because many mobile devices are used more wireless services, heavy signaling traffic for authentication could be generated. Authentication is an important factor in wireless networks to identify devices; it is the start of wireless networks. This paper analyzes previous patterns for more effective authentication in accessing of another external networks. We propose a fast authentication scheme for minimizing of signaling cost between the authentication server and external networks. And we calculate the rate of authentication occurrence in LTE networks using mathematical modeling as well as the change of signaling cost for authentication in various network environments. Finally, we calculate the optimized number of authentication data and show the effectiveness for authentication signaling costs.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.7
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• pp.1437-1447
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• 1997

This paper proposes a cluster-based PCS(Personal Communication Services) network architecture to support increased signaling and control traffic. The procedures are presented to serve the mjobility management and the efficient procesing of the call-control functional using distributed servers. Especially when call-setups are made within a cluster, this distributed servers. Especially when call-setup time and the amount of control traffic of H-HLS (High level Home Location Server) in the network as L-HLS(Low lever Home Location Server) serves the call on behalf of H-HLS. Performance and reliability of the architecture have been also cairried out.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.429-430
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• 2006

In this paper, we describe the architecture of interactive data broadcasting server that can transmit the various data-service contents of the Terrestrial DMB(digital multimedia broadcasting). In the broadcasting environment of the Terrestrial DMB, we enjoys the PADS(program associated data service), or the PIDS(program independent data service) to be executed on various T-DMB terminals as well as the basic video or audio services. This server provides the function such as the data contests management, data channel management, service information configure, return channel connection. Etc. Particularly, this system provides the method to create and transfer the application signaling information for the T- DMB middle application based the java language.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.101-108
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• 2004

SIP(Session Initiation Protocol) is a suitable protocol for supporting Internet telephony services and currently market requirements fur developing supplementary telephony services such as unconditional call forwarding, call forwarding on busy or no answer, call filtering services have recently grown. CPL(Call Processing Language) is a standard technology that can be used to describe and control internet telephony services. In this paper, we describe the CPL system for supplementary Internet telephony services using SIP as an application level call signaling protocol. Those supplementary services are composed of CPL client which is a SIP UA, SIP Proxy server, Registrar and CPL server In this paper, we describe the design and implementation of the CPL server system in detail which is developed in Linux 7.2 using C and C++ programming languages.