• Journal of Communications and Networks
• /
• v.15 no.1
• /
• pp.15-24
• /
• 2013

Provisioning of quality of service (QoS) is a key issue in any multi-media system. However, in wireless systems, supporting QoS requirements of different traffic types is a more challenging problem due to the need to simultaneously minimize two performance metrics - the probability of dropping a handover call and the probability of blocking a new call. Since QoS requirements are not as stringent for non-real-time traffic, as opposed to real-time traffic, more calls can be accommodated by releasing some bandwidth from the already admitted non-real-time traffic calls. If the released bandwidth that is used to handle handover calls is larger than the released bandwidth that is used for new calls, then the resulting probability of dropping a handover call is smaller than the probability of blocking a new call. In this paper, we propose an efficient call admission control algorithm that relies on adaptive multi-level bandwidth-allocation scheme for non-realtime calls. The scheme allows reduction of the call dropping probability, along with an increase in the bandwidth utilization. The numerical results show that the proposed scheme is capable of attaining negligible handover call dropping probability without sacrificing bandwidth utilization.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.26 no.12B
• /
• pp.1677-1685
• /
• 2001

In this paper, we propose a call admission control scheme that can be performed within guaranteeing of required QoS(Quality of Services) in DS-CDMA(Direct Sequence-Code Division Multiple Access) reverse link. It has been performed rely on a physical channel numberonly and based on quality of received signal from MODEM(modulator/demodulator) part in established study. In other methods, the standard for services would have been set from statistical analysis of users\` location and using received power level in BTS(Base Transceiver Station). These ways bring about not only system loads but time delay or great differences from real environment. To solve these problems, we propose a call drop rate estimation algorithm for the purpose of call admission control based on measured value at LNA(Low Noise Amplifier) ports of BTS(Base Transceiver Station) in real time. This method proposed in this paper estimates a quality of offered service in real time, reduce system loads and shorten time delay which is needed to determine the standard for call admission control. But it requires a additional 17W complexity which can measure received signal power in BTS and estimate call drop rate.

• Journal of Communications and Networks
• /
• v.11 no.2
• /
• pp.157-165
• /
• 2009

In this paper, we investigate a dynamic spectrum sharing problem for the centralized uplink cognitive radio networks using orthogonal frequency division multiple access. We formulate a large-scale joint rate and power allocation as an optimization problem under quality of service constraint for secondary users and interference constraint for primary users. We also suggest admission control to nd a feasible solution to the optimization problem. To implement the resource allocation on a large-scale, we introduce a notion of using the conservative factors $\alpha$ and $\beta$ depending on the outage and violation probabilities. Since estimating instantaneous channel gains is costly and requires high complexity, the proposed algorithm pursues a practical and implementation-friendly resource allocation. Simulation results demonstrate that the large-scale joint rate and power allocation incurs a slight loss in system throughput over the instantaneous one, but it achieves lower complexity with less sensitivity to variations in shadowing statistics.

• Journal of Advanced Navigation Technology
• /
• v.12 no.1
• /
• pp.22-27
• /
• 2008

This paper presents adaptive-MMOSPRED method for prediction of resource demands requested by multimedia calls, and shows the performance of the call admission control based on proposed resource prediction method in multimedia wireless networks. The proposed method determines (I-CDP) random variables of the standard normal distribution by using LMS algorithm that minimize errors of prediction in resource demands, while parameters in an existing method are constant all through the prediction time. Our simulation results show that prediction error in adaptive-MMOSPRED method is much smaller than in fixed-MMOSPRED method. Also we can see via simulation the CAC performance based on the proposed method improves the new call blocking performance compared with the existing method under the desired handoff dropping probability.

• 제어로봇시스템학회:학술대회논문집
• /
• 2000.10a
• /
• pp.65-65
• /
• 2000

For Video on Demand(VoD) servers, a design of an efficient scheduler is important to the support a large number of clients having various playback speeds and receiving rates. In this paper, we propose the scheduling algorithm to handle establishing deadlines and selection using the earliest deadline first. To establish deadlines and selections, the period of the receiving rates for each client is located between the over-max receiving rate and the over-playback rate. To avoid video starvation and the buffer overflow of each client, the proposed algorithm guarantees providing the admission control. Because of establishing deadlines and selection, period of each client receiving is between one over max receiving rate and one over play back rate. Using Virtual Buffer in server, scheduling load is reduced. The efficiency of the proposed algorithm is verified using a Petri Net_Based simulation tool, Exspect.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.30 no.5A
• /
• pp.417-425
• /
• 2005

KOINONIA is high-rate Wireless Personal Area Network (WPAN) technology, and is developed for multimedia traffic transmission in personal area. A KOINONIA piconet is a collection of one or more associated slaves under a single master. Efficient scheduling of a master for the traffic of slaves is essential to use channel effectively and to guarantee QoS of multimedia traffic. We propose a new scheduling algorithm to allocate channel time at desired intervals regardless of superframe length, and a Connection Admission Control(CAC) algorithm to regulate the number of traffics in a piconet. Our proposed algorithms have been shown to save channel time and to meet QoS requirements compared to the conventional weighted round-robin algorithm.

• Journal of Advanced Navigation Technology
• /
• v.13 no.3
• /
• pp.372-377
• /
• 2009

This paper evaluates the performance of three methods for predicting resources requested by mobile's calls and a call admission algorithm based on these predicting methods respectively in wireless networks. The first method is based on Wiener prediction model and the second method is based on the MMOSPRED algorithm and the third method is based on the neural network. The proposed call admission algorithm is based on prioritized handoff call in resource allocation. The resources for future handoff calls are therefore predicted and reserved in advance, and then new calls are admitted as long as the remaining resources are sufficient. We compare their performances in terms of prediction error, new call blocking and handoff dropping probabilities by simulation. Results show that the CAC based on Wiener prediction model performs favorably due to exact resources prediction.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.24 no.8B
• /
• pp.1421-1425
• /
• 1999

A restoration scheme is presented when a failure occurs at nodes or links on the ATM network. Each node on the network contains the same topology information for resources on the network by Exchanging NSE(Network State Element) which has the network state information through RCC(Routing Control Channel). We also propose a new establishing algorithm that set up a new alternate path reflecting the state of the network changed dynamically at the source node. In the result of applying our scheme, the rates of restoration can be improved and the number of messages decreased so that fast recovering could be possible by reducing traffics on the network by lowering the rates of the call-setup failure at the stage of CAC(Connection Admission Control).

• Journal of Advanced Navigation Technology
• /
• v.18 no.4
• /
• pp.347-352
• /
• 2014

In this paper, we propose a Markovian queueing model(M/M/1)-based call admission control to reduce forced terminating rate of non-real secondary user's call for Multi-services Cognitive Radio Networks. A existing control has a problem that the forced terminating rate increases because of adopting a policy of spectrum priority allocation to real calls. In our scheme the rate can be reduced as the call that has no useful spectrum waits in a queue until getting an available spectrum. Our scheme use a neural-net based prediction of primary user's reappearance. Through the simulation, we analysis the call forced terminating rate, access delay and spectrum utilization efficiency, and then show that our scheme can more reduce the forced terminating rate of the call, compared to that of the existing algorithm.

• Journal of KIISE:Information Networking
• /
• v.30 no.4
• /
• pp.461-473
• /
• 2003

Multimedia applications are much more sensitive to QoS(Quality of Service) than text based ones due to their data continuity. In order to provide a fast moving MH(Mobil Host) using multimedia application with a consistent QoS,an efficient call admission mechanism is in need. This paper proposes the 2SCA(2-Step Call Admission) scheme based on cal admission scheme using pripority to guarantee the consistent QoS for mobile multimedia applications. A calls of MH are classified new calls, hand-off calls, and QoS upgrading calls. The 2SCA is composed of the basic call admission and advanced call admission; the former determines the call admission based on bandwidth available in each cell and the latter determines the call admission by applying DTT(Delay Tolerance Time), PQeueu(Priority Queue), and UpQueue(Upgrade Queue) algorithm according to the type of each call blocked at the basic call admission stage. In order to evaluate the performance of our mechanism, we measure the metrics such as the dropping probability of new calls, dropping probability of hand-off calls, and bandwidth utilization. The result shows that the performance of our mechanism is superior to that of existing mechanisms such as CSP(Complete Sharing Policy), GCP(Guard Channel Policy) and AGCP(Adaptive Guard Channel Policy).