• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.12B
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• pp.1112-1122
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• 2008

This paper reviews the Parlay X and proposes SIP multimedia session control mechanism which is able to be implemented over IMS using Extended Call Control APIs of Parlay X. Parlay X is one of the industrial standards of Open API in the telecommunication to open network resources and capabilities to third party service providers. In this paper, SIP session control mechanism is described by depicting call flows of basic and essential session handling methods including session initiation, transfer, restoration, and termination. We also show how easy Parlay X Extended Call Control APIs can be used for the purpose of complicated calls handling in the IT applications. To verify feasibility of the blended services based on convergence of telecommunication and internet with regard to performance, we accomplish experimental performance of Extended Call Control APIs from IT application through open service gateway. We show Open API could be applied to next generation network based on IMS without serious degradation of the network performance.

• The Journal of the Korea Contents Association
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• v.3 no.4
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• pp.55-62
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• 2003

In this paper, a call admission contro(cac) technique is proposed to reduce the dropping probabilities of handoff calls in wireless networks while guaranteeing QoS to the users. The proposed technique is based on the estimated effective load for the target eel if a call is accepted. When the estimated effective load is higher than a predetermined threshold, a nu call is blocked and a handoff call is queued irrespective of the availability channels. The SRN, an extended Stochastic Petri Net, modes are constructed to compare the performance of the techniques. The SRN uses rewards concepts instead of the complicate numerical analysis required for the Markov chain modes. As a result, the SRN modeling techniques provide an easier way to carry out performance analysis for call admission control and channel allocation.

• ETRI Journal
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• v.20 no.2
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• pp.115-132
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• 1998

This paper proposes an extended service filtering technique to prevent overload in service control point (SCP) due to televoting (VOT) or mass calling (MAS) services with the heavy traffic characteristics. Also, this paper compares this extended technique with the existing overload control techniques, and calculates steady state call blocking probabilities in intelligent network (IN) under overload conditions. The proposed technique considers SCP overload and IN Capability Set (CS)-1 services (such as VOT or MAS service) that have to use the specialized resources of intelligent peripheral (IP). This technique uses first an activating step in which SCP requests service filtering to service switching point (SSP). Then, in the filtering step, SSP sends filtering results to SCP periodically or each N-calls. Also, when filtering time-out expires, SSP stops service filtering, and sends service filtering response to SCP in the deactivating step. This paper applies this technique to VOT/MAS service, and calculates SCP and SSP-IP (circuit) call blocking probabilities by using an analytical VOT/MAS service model. With the modeling and analyzing of this new technique, it shows that this technique reduces the traffic flow into SCP from SSP and IP prominently.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.27-36
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• 2002

VoiceXML is a standard dialog mark-up language for the neat generation voice applications. The current VoiceXML 1.0 specification is silent on who place outbound calls for push-based voice applications. The push-barred voice applications become very important in modern information systems such as CRM. In this paper, we design and implement an extended VoiceXML platform that supports both inbound and outbound voice information services. We also extend the VoiceXML DTD so as to be able to inbound/outbound fax based on Call Control Requirements of W3C.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.12
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• pp.529-538
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• 2002

Conventional method to reduce the handoff call blocking probability(PBH) in mobile communication system is to reserve a predetermined number of channels only for handoff calls. To determine the number of reserved channels, an optimization problem, which is generally computationally heavily involved, must be solved. In this Paper, we propose a call admission control (CAC) scheme that can be used to reduce the PBH without reserving channels in advance. For this, we define a new measure, gain, which depends on the state of the system upon the arrival of a new call. The proposed CAC decision rule relies on the gain computed when a new call arrives. SRN, an extended stochastic Petri nets, provides compact modeling facilities for system analysis can be calculated performance index by appropriate reward to the model. In this Paper, we develop SRN models which can perform the CAC with gain. The SRN models are 2 level hierarchical models. The upper layer models are the structure state model representing the CAC and channel allocation methods considering QoS with multimedia traffic The lower layer model Is to compute the gain under the state of the upper layer models.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.10a
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• pp.355-358
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• 2007

In wireless mobile communication systems, priority of voice service through high speed data and multimedia transmission requires increased service diversification. Research is being carried out in this environment, on the call admission control techniques to guarantee the diversified service's QoS. SRN (Stochastic Reward Net) is an extended version of Petri nets, well know modeling and analysis tool. In this paper, we develop SRN call admission control model considering the 4 classes of services in the 4th generation IEE 802.16e mobile communication Technology.

• The Transactions of the Korea Information Processing Society
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• v.3 no.7
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• pp.1727-1738
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• 1996

This paper proposes a extended service filtering technique to solve overload control problems inService Con-trol Poin(SCP) due to Televoting (VOT) services with heavy traffic characteristics. Also, this paper compares this new technique with existing IN overload control lechniques, and calculate steady state call blocking probabilities in In under overload conditions. The proposed technique considers SCP overload and IN CS-1services such as VOT service that need to use the specialized resources of IP. This technique uses first an activating step which SCP sends service filtering request to SSP and IP for the start of service filtering. Then, in the filtering step SSP sendas filtering results to SCP periodically or each N-calls.Also,when filtering time-out timer expired or call counter value exceeded maximum number of calls,SSP stops service filtering,and sends ser-vice filteing response to SCP in the deactivating step. The application of this techique to VOT service defined in IN CS-1 service is investigated and analyzed by using an analytical VOT service model.In order to apply this technique to VOT service, this paper also discusses VOT service and the corresponding N architecture including Service Switching Point(SSP),SCP and IP. Then this paper also constructs an analytical VOT service model,and calculates SCP and SSP/IP(circuit)call blocking probablities.With the application of this new technique,this paper shows good performance enhancements in SCP call blocking probability.

• The Journal of the Korea Contents Association
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• v.7 no.3
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• pp.34-41
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• 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.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.7 s.361
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• pp.70-80
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• 2007

For supporting various mobile networks, this paper proposes a framework of distributed connection admission control, named SmartCAC. Especially, intelligent CAC operations are adopted in terms of interoperation between mobile nodes and mobile networks. This scheme does not need to correct information between networks. Basically vertical handover call can use guard channel that was reserved for handoff, because SmartCAC addresses the identification between vertical handover call and new call, delay and reliability as requirement of QoS for efficient connection control. The scheme also uses mobile terminal speed for network filtering. Especially an extended protocol is proposed to give different network states information to mobile nodes because there have been no ways for mobile nodes to compare the states of different networks. Sophisticated simulation study is performed in order to evaluate SmartCAC in terms of signaling cost. As a result, signaling cost of ours is up to 96% better than that of the existing scheme.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.12
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• pp.18-26
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• 2003

In this paper, we design and implement a QoS control function in the SIP-based VoIP system. As a network infrastructure for VoIP service, we select the Intserv over Diffserv architecture where the network resources are managed by a call admission control mechanism. The SIP protocol extended to support QoS signaling procedure is modulized to operate independently with the infrastructure. The performance of the QoS-enabled VoIP system is verified by experiments.