• Journal of KIISE:Information Networking
• /
• 제30권5호
• /
• pp.621-630
• /
• 2003

In this paper, we implemented SIP protocol that supports IPv6 and differentiated call processing scheme for NGN(Next Regeneration Network). In NGN, SIP processes call signaling among various application services. A softswitch and SIP server must give priority to sensitive services such as Fax, network management and home networking that require a fast call setup time. Also, the support of IPv6 is needed under consideration of All-IP. We proposed differentiated call processing scheme. The differentiated call processing scheme supports differentiated call processing as priority of service class on call processing in SW server We defined three service classes and use the Flow Label field of the IPv6 header for setting service class. Through the performance analysis, we proved that it improves throughput for call message with the high priority. The result of performance analysis demonstrates that differentiated call processing scheme gives better performance for the service requiring a fast session establishment in NGN.

• Proceedings of the Korean Institute of Communication Sciences Conference
• /
• 한국통신학회 1987년도 춘계학술발표회 논문집
• /
• pp.67-71
• /
• 1987

No 1A ESS must operate in real time. Call processing capacity is the maximm number of originating plus incoming calls which the processor can process in some fixed interval of real time shile all service criteria have been satisfied. In this paper. we discuss the postcutower method of determining call processing capacity of No 1A ESS which can be applicable to an in service office and explain the usuage mithod of estimated call processing capacity of No 1A ESS in service

• IE interfaces
• /
• 제11권3호
• /
• pp.77-87
• /
• 1998

Intelligent Peripheral(IP) system is to provide specialized resource functions (SRF) such as playing announcement, collecting user information, and receiving messages in the Advanced Intelligent Network (AIN). We analyze the call processing capacity of an AIN IP system being developed in ETRI through an extensive simulation using SLAM II under a variety of AIN service scenarios. We consider televoting (VOT) and universal personal telecommunication (UPT) services which are to be provided at the fit implementation of the AIN in Korea. As the performance criteria to determine the call processing capacity, processor utilization, delay and call loss probability are considered. It turns out that the major processor called SAMP is the bottleneck processor, the service response delay dominates the delay performance, and the call loss probability becomes the primary criterion in determining the call processing capacity of the AIN IP system. It is also shown that the call processing capacity of the AIN IP system is determined by the utilization of the processor and the delay performance when the VOT ratio is below 70 percent but it is determined by the call loss probability due to the lack of service channels for providing the SRF operations.

• Journal of the Korean Institute of Telematics and Electronics S
• /
• 제34S권8호
• /
• pp.27-36
• /
• 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
• /
• 제7권3호
• /
• pp.34-41
• /
• 2007

The presence service which provides user's presence information by subscription and notification is one of SIP(session initiation protocol) extension services, and it is used importantly in VoIP(Voice over IP) and Instant Messaging service. In this paper, we propose a new method in which users can combine and control presence service and call processing services in various ways by extending call processing language, and only changed parts of the presence information are published instead of full presence information document. Each user registers full presence information document with his own call processing script during the first publication to a presence server. The presence server executes these call processing scripts, so it can provide various services with combination of presence service and call processing services during the presence subscriptions and notifications. Afterwards, users publish only changed parts of the presence information and the presence server notify only these changed parts to watchers. Therefore the efficiency of the overall system can be improved. The performance of our proposed model is evaluated by experiments.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• 제21권10호
• /
• pp.2605-2619
• /
• 1996

In this paper, we evaluated performance of mobile exchange control part. Queueing network model is used for modeling of mobile exchange control part. We developed a call control processing and location registration scenartio which has a message exchange function between processors in mobile exchange control part. A network symbol are used the simulation models that are composed of the initialization module, message generation module, message routing module, message processing module, message generation module, HIPC network processing module, output analysis module. as a result of computer simulation, we obtain the processor utilization, the mean queue length, the mean waiting time of control part based on call processing and location registration capacity. The call processing and location registration capacity is referred by thenumber of call attempts in the mobile exchange and must be satisfied with the quality of service(delay time).

• The Transactions of the Korea Information Processing Society
• /
• 제3권5호
• /
• pp.1253-1261
• /
• 1996

The base station (BS) of CDMA mobile system (CMS) provides mobile call services by controlling call processing functions. The call Processing part of the BS is divided into two components; the base station transceiver subsystem (BTS) and the base station controller (BSC). In this paper, we present the call capacity estimation of the BS about control signals and traffic data of calls by the simulation, and find the bottleneck points and problems which may occur in the BS. In order to estimate the call capacity, first we extract the major parameters for the modeling the BS. Second, we suggest the simulation model for the BS. Third, we estimate the simulation results by finding major objective factors such that the call blocking probability, the utilization ratio and the delay time in the traffic channel elements (TCEs), the BTS interconnection network (BIN)-COMA interconnection network (CIN) trunks, the transcoding channels and the CIN packet router.

• The Transactions of the Korea Information Processing Society
• /
• 제7권4호
• /
• pp.1217-1226
• /
• 2000

Traditionally, Mobile Teletraffic model consists of two sub-models, i.e. the network traffic model and the traffic source model. In this paper, we present the traffic source model by developing MobCall (Mobile Call Simulator) which analyses various mobile wireless environments based on regional characteristics that the base stations are located. User mobility is presented by regional average vehicle speeds and the transportation share rate. Moreover, the user mobility on subway, which is increasing in urban area, is considered in MobCall. And also, user's movements on highway are considered in MobCall. The object-oriented simulation platform, C++SIM, is used to implement MobCall. Using MobCall, the accumulated number of calls in residential and commercial regions, the handoff rate with respect to traffic sources of Seoul, the handoff rate on highway, and the handoff rate according to the call duration are presented. MobCall enables the simulation of dynamic handoff buffering and functional entity control of one base station according to the changes in user's calling pattern at the design phase. Also, when a new town is under construction by a detailed plan, MobCall is used to design the mobile network with regional characteristics and user mobility considered.

• Proceedings of the IEEK Conference
• /
• 대한전자공학회 1998년도 추계종합학술대회 논문집
• /
• pp.229-232
• /
• 1998

Modern telecommunications are increasingly demanding the addition of complex switching features to switching system software. These improvements also should satisfy customer demands for quick provisioning of new service features. But whenever we have developed the switching system in various types, we met problems that are lack of system engineers and short developments terms. We need new guidelines of software design to meet the demands and overcome our problems. So we propose new software platform, scenario driven call control engine (SDCCE) for ATM switching system software and implement an effective call control software using scenarios for ATM Switching System. A scenario is a set of procedures and supplementary scenarios written in data structure and gives benefits to easy adapt new or modified function by adding a scenario. The goal of SDCCE is to increase reusability and readability of software. Especially, it gives a good software extensibility and maintainability. And when we developed ptmp call processing control software, we could get high productivity by reusing scenarios and procedures of ptp call processing control software. We applied the scenario driven call control software to ATM switching system.

• The Transactions of the Korea Information Processing Society
• /
• 제2권4호
• /
• pp.453-465
• /
• 1995

RPC(Remote Procedure Call) has been studied for programmer to easily write distributed program of little bit higher efficiency and reliability. In this study, fault-tolerant remote procedure call for hardware failures is proposed. Fault-tolerance is supplied by replicated procedures with node group, so called chain, and copies along chains are linearly ordered. Calls for procedure are sent to primary copy along chains, and other copies are propagated internally. If failures happen, first copy in faultless chain returns the result to the caller, Especially, in this study processing of redundant call message and result message, while using limited ack message, are avoided. This method supplies efficient and reliable fault-tolerance compared with existing remote procedure call.