• The Transactions of the Korea Information Processing Society
• /
• v.7 no.8S
• /
• pp.2608-2619
• /
• 2000

This paper proposes the Handover Scheme for the mobile and describes the result of the performance analysis. In the conventional scheme of handover request, the withdrawal of terminal may occur because handover request is performed based on fixed signal level without considering network load and terminal mobility. The proposed scheme offers the minimization of withdrawal and handover blocking probability by means of the handover request of terminal based on the network load and terminal mobility. Conventional handover scheme has the sequential procedure that network performs resource check and path rerouting on the handover by MT(Mobile Terminal). Proposed handover scheme pre-processes the resource check before the handover request by predicting the handover request timo so that handover latency can be reduced. Moreover, path optimization is executed after the completion of handover in order to reduce handover latency. The rdduction of handover latency prevents the dropping of service by minimizing backward handover blocking. In summary, we propose the prediction of handover request time and decision method based on terminal, validating the performance of proposed scheme considering various cases of simulation.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.5 no.5
• /
• pp.1049-1068
• /
• 2011

In this paper a simple and effective call admission control (CAC) scheme is proposed for the emerging interleave-division multiple-access (IDMA) systems, supporting a variety of traffic types and offering different quality of service (QoS) requirements and priority levels. The proposed scheme is signal-to-interference-plus-noise ratio (SINR) evolution based CAC (SE-CAC). The key idea behind the scheme is to take advantage of the SINR evolution technique in the process of making admission decisions, which is developed from the effective chip-by-chip (CBC) multi-user detection (MUD) process in IDMA systems. By virtue of this semi-analytical technique, the MUD efficiency can be estimated accurately. Additionally, the computational complexity can be considerably reduced. These features make the scheme highly suitable for IDMA systems, which can combat intra-cell interference efficiently with simple CBC MUD. Analysis and simulation results show that compared to the traditional CAC scheme considering MUD efficiency as a constant, the proposed SE-CAC scheme can guarantee high power efficiency and throughput for multimedia traffic even in heavy load conditions, illustrating the high efficiency of CBC MUD. Furthermore, based on the SINR evolution, the SE-CAC can make accurate estimation of available resource considering the effect of MUD, leading to low outage probability as well as low blocking and dropping probability.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.6 no.2
• /
• pp.683-701
• /
• 2012

IEEE 802.11 standard has achieved huge success in the past decade and is still under development to provide higher physical data rate and better quality of service (QoS). An important problem for the development and optimization of IEEE 802.11 networks is the modeling of the MAC layer channel access protocol. Although there are already many theoretic analysis for the 802.11 MAC protocol in the literature, most of the models focus on the saturated traffic and assume infinite buffer at the MAC layer. In this paper we develop a unified analytical model for IEEE 802.11 MAC protocol in ad hoc networks. The impacts of channel access parameters, traffic rate and buffer size at the MAC layer are modeled with the assistance of a generalized Markov chain and an M/G/1/K queue model. The performance of throughput, packet delivery delay and dropping probability can be achieved. Extensive simulations show the analytical model is highly accurate. From the analytical model it is shown that for practical buffer configuration (e.g. buffer size larger than one), we can maximize the total throughput and reduce the packet blocking probability (due to limited buffer size) and the average queuing delay to zero by effectively controlling the offered load. The average MAC layer service delay as well as its standard deviation, is also much lower than that in saturated conditions and has an upper bound. It is also observed that the optimal load is very close to the maximum achievable throughput regardless of the number of stations or buffer size. Moreover, the model is scalable for performance analysis of 802.11e in unsaturated conditions and 802.11 ad hoc networks with heterogenous traffic flows.

• Journal of KIISE:Information Networking
• /
• v.29 no.1
• /
• pp.1-14
• /
• 2002

The most important issue in providing multimedia traffic on a mobile computing environments is to guarantee the mobile host(client) with consistent QoS(Quality of Service). However, the QoS negotiated between the client and network in one cell may not be honored due to client mobility, causing hand-offs between cells. In this paper, a call admission control mechanism is proposed to provide consistent QoS guarantees for multimedia traffics in a mobile computing environment. Each cell can reserve fractional bandwidths for hand-off calls to its adjacent cells. It is important to determine the right amount of reserved bandwidth for hand-off calls because the blocking probability of new calls may increase if the amount of reserved bandwidth is more than necessary. An adaptive bandwidth reservation based on an MPP(Mobility Pattern Profile) and a 2-tier cell structure has been proposed to determine the amount of bandwidth to be reserved in the cell and to control dynamically its amount based on its network condition. We also propose a call admission control based on this bandwidth reservation and "next-cell prediction" scheme using an MPP. In order to evaluate the performance of our call admission control mechanism, we measure the metrics such as the blocking probability of our call admission control mechanism, we measure the metrics such as the blocking probability of new calls, dropping probability of hand-off calls, and bandwidth utilization. The simulation results show that the performance of our mechanism is superior to that of the existing mechanisms such as NR-CAT1, FR-CAT1, and AR-CAT1.

• Proceedings of the IEEK Conference
• /
• 2000.11a
• /
• pp.189-192
• /
• 2000

This paper proposes a fuzzy-based random access controller with a superimposed frame structure (F$^2$RAC) fur voice/data-integrated wireless networks. F$^2$RAC adopts mini-slot technique for reducing contention cost, and these mini-slots of which number may dynamically vary from one frame to the next as a function of the traffic load are further partitioned into two regions for access requests coming from voice and data traffic with their respective QoS requirements. And F$^2$RAC is designed to properly determine the access regions and permission probabilities for enhancing the data packet delay while ensuring the voice packet dropping probability constraint. It mainly consists of the estimator with Pseudo-Bayesian algorithm and fuzzy logic controller with Sugeno-type of fuzzy rules. Simulation results prove that F$^2$RAC can guarantee QoS requirement of voice and provide the highest throughput efficiency and the smallest data packet delay amongst the different alternatives including PRMA［1］, IPRMA［2］, and SIR［3］.

• Journal of information and communication convergence engineering
• /
• v.9 no.5
• /
• pp.530-538
• /
• 2011

Performance of TCP can be severely degraded in Mobile IP-based wireless networks where packet losses not related to network congestion occur frequently during inter-subnetwork handoffs by user mobility. To solve such a problem in the networks using Mobile IP, the packet buffering method at a base station(BS) recovers those packets dropped during handoff by forwarding the buffered packets at the old BS to the mobile users. But, when the mobile user moves to a congested BS in a new foreign subnetwork, those buffered packets forwarded by the old BS are dropped and TCP transmission performance of a mobile user degrades severely. In this paper, we propose a Modified BLUE(MBLUE) buffer required at a BS to increase TCP throughput in Mobile IP-based networks. When a queue length exceed a threshold and congestion grows, MBLUE increases its packet drop probability. But, when a TCP connection is added at new BS by a handoff, the old BS marks the buffered packets. And new BS receives the marked packets without dropping. Simulation results show that MBLUE buffer reduces congestion during handoffs and increases TCP throughputs.

• Journal of IKEEE
• /
• v.7 no.1 s.12
• /
• pp.80-87
• /
• 2003

This paper proposes a routing scheme for multi-hop LEO satellite networks with inter-satellite links aiming for reducing the number of link handovers while keeping the efficient use of network resource. The proposed routing scheme controls the link handovers by taking account of the deterministic LEO satellite system dynamics, geographical location of a ground terminal and statistic information of call duration. The performance of the proposed routing scheme has been evaluated and compared with previous routing schemes in terms of average number of link handovers during a call, the call blocking and dropping probability, and the network utilization.

• Journal of Communications and Networks
• /
• v.5 no.3
• /
• pp.275-281
• /
• 2003

Random Early Detection (RED) [1] is an active queue management scheme which has been deployed extensively to reduce packet loss during congestion. Although RED can improve loss rates, its performance depends severely on the tuning of its operating parameters. The idea of adaptively varying RED parameters to suit the network conditions has been investigated in [2], where the maximum packet dropping probability $max_p$ has been varied. This paper focuses on adaptively varying the queue weight $\omega_q$ in conjunction with $max_p$ to improve the performance. We propose two algorithms viz., $\omega_q$-thresh and $\omega_q$-ewma to adaptively vary $\omega_q$. The performance is measured in terms of the packet loss percentage, link utilization and stability of the instantaneous queue length. We demonstrate that varying $\omega_q$ and $max_p$ together results in an overall improvement in loss percentage and queue stability, while maintaining the same link utilization. We also show that $max_p$ has a greater influence on loss percentage and queue stability as compared to $\omega_q$, and that varying $\omega_q$ has a positive influence on link utilization.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.8B
• /
• pp.780-786
• /
• 2002

In this paper several access schemes for voice and data packet over CPCH with waiting time signalling are proposed. Waiting time signalling is used for reducing the excessive attempt of UE when all the CPCH channel resources is being used by other UEs. The proposed schemes in this paper are better than other schemes in view of the voice/data packets? dropping probability, average access attempt to transmit voice/data packets successfully and the average throughput.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.19 no.5
• /
• pp.842-856
• /
• 1994

The various services that a broadband integrated services digital network (B-ISDN) carries, have a wide range of delay, delay jitter and cell loss probability requirements. Design of appropriate control schemes for B-ISDN is an extremely important and challenging problem. In this paper, we proposed multiplexing algorithm with both delay and loss priorities in order to satisfy the diverse requirements. For the implementation of cell lose priority, we assumed that voice cells are generated as non-discardable(i.e., high priority) and discardable (i.e., low priotity)cells. The low priority voice cell may be discarded inside the network if congestion occurs. The cell dropping scheme is shown to reduce cell losses as well as delays for both voice and data. Such a load shedding scheme is expected to improve significantly utilization of B-ISDN.