• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.683-690
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• 2003

In this paper we present a theoretical framework for the network design with delay QoS guarantee to a voice at the packet level. Especially, we propose a method for estimating the bandwidth at the ingress edge routers accommodating the voice connections and data sessions in the next-generation If network. First, we describe network architecture for VoIP (Voice over IP) services in the NGN (Next Generation Network). After that, we propose a procedure for dimensioning the bandwidth at the output port of a router that accommodates voice and data traffic using the non-preemptive queuing system with strict priority service scheme. Via numerical experiments we illustrate the implication of the proposition.

• Journal of Intelligence and Information Systems
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• v.17 no.1
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• pp.1-16
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• 2011

The real-time ubiquitous environment is required to be able to process a series of sensor data within limited time. The whole sensor data processing consists of several phases : getting data out of sensor, acquiring context and responding to users. The ubiquitous computing middleware is aware of the context using the input sensor data and a series of data from database or knowledge-base, makes a decision suitable for the context and shows a response according to the decision. When the real-time ubiquitous environment gets a set of sensor data as its input, it needs to be able to estimate the delay-time of the sensor data considering the available resource and the priority of it for scheduling a series of sensor data. Also the sensor data of higher priority can stop the processing of proceeding sensor data. The research field for such a decision making is not yet vibrant. In this paper, we propose a queuing time computation algorithm for sensor data processing in real-time ubiquitous environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.9B
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• pp.1272-1280
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• 2010

WFQ (Weighted Fair Queuing) provides not only fairness among traffic flows in using bandwidth but also guarantees the Quality of Service (QoS) that individual flow requires, which is why it has been applied to the resource reservation protocol (RSVP)-capable router. The RSVP allocates an enough resource to satisfy both the rate and end-to-end delay requirements of the flow in condition of no packet loss, and the WFQ guarantees those QoS requirements with the allocated resource. In a practice, however, most QoS-guaranteed services, specially the Voice of IP, allow a few percent of packet loss, so it is strongly desired that the RSVP and WFQ make the best use of this allowable packet loss. This paper enhances the WFQ to allow packet loss and investigates its performance. The performance evaluation showed that allowing the packet loss of 0.4% can improve the flow admission capability by around 40 percent.

• Proceedings of the KIEE Conference
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• 2005.05a
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• pp.215-217
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• 2005

There are two types of packet loss probabilities used in both the network layer and the physical layer within the wireless transmitter such as a queueing discard probability and transmission loss probability. We analyze these loss performances in order to guarantee Quality of Service (QoS) which is the basic of the future network. The queuing loss probability is caused by a maximum allowable delay time and the transmission loss probability is caused by a wireless channel error. These two types of packet loss probabilities are not easily analyzed due to recursive feedback which, originates as a result at a queueing delay and a number of retransmission attempts. We consider a wireless transmitter to a M/D/1 queueing model. We configurate the model to have a finite-size FIFO buffer in order to analyze the real-time traffic streams. Then we present the approaches used for evaluating the loss probabilities of this M/D/1/K queueing model. To analyze the two types of probabilities which have mutual feedbacks with each other, we drive the solutions recursively. The validity and accuracy of the analysis are confirmed by the computer simulation. From the following solutions, we suggest a minimum of 'a Maximum Allowable Delay Time' for real-time traffic in order to initially guarantee the QoS. Finally, we analyze the required service rate for each type utilizing real-time traffic and we apply our valuable analysis to a N-user's wireless network in order to get the fundamental information (types of supportable real-type traffics, types of supportable QoS, supportable maximum number of users) for network design.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.6B
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• pp.508-516
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• 2006

The explosive growth of wireless network users and the existence of various wireless services have demanded high rate throughput as well as user's QoS guarantees. Towards this, this paper proposes QoS-oriented subcarrier allocation scheme considering the QoS provisioning of multiple users, which is major requirement for wireless network design point of view. This paper introduces joint RR/K&H combined with M-LWDF(Modified Largest Weighted Delay First) scheme throughout observing statistical channel behavior and real time queuing analysis for appropriate resource allocation tightly connected to multiuser scheduling. Accordingly, the system throughput can be enhanced, and the QoS demanded for delay sensitive services can be satisfied. Furthermore, the proposed scheme is applied for OFDMA(Orthogonal Frequency Division Multiple Access) systems to allocate sub-carriers in optimal way. The simulation results verify plausible performances of proposed resource allocation scheme via showing superior effective capacity under time-varying physical-layer channel behaviors.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.2
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• pp.683-701
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• 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 the Institute of Electronics Engineers of Korea CI
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• v.45 no.5
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• pp.117-124
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• 2008

In this paper, the call setup performance of the CSCF(Call Session Control function) that manages the session control in providing the IMS(IP Multimedia subsystem)-based multimedia service in the BcN(Broadband convergence Network) is analyzed. While the performance related with the call/session of the SIP-Based voice service is analysed, the study for the call recessing performance of the IMS-based multimedia service is insufficient. In this paper, as, the processing capacity and subscriber number of CSCF were changed, the call setup delay time according to the session setup procedure of IMS was measured using the M/M/1 queuing model and OPNET simulation. The experimental results show that the sudden increased delay showed up in case the session establishment requirement ratio of total users over 13% of the CSCF processing capacity. Therefore, the user number and capacity of CSCF suitable for the session establishment delay threshold can be calculated or estimated.

• Journal of Korea Multimedia Society
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• v.11 no.7
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• pp.966-976
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• 2008

Providing Quality-of-Service(QoS) guarantee requires the reservation of resource needed to accommodate the traffic flow of service for which resource reservation protocol(RSVP) was introduced. However, if any RSVP-capable router on the path fails to accommodate the traffic flow due to lack of resource, the flow is not allowed no matter how much surplus resources other routers on the path have. In order to solve this problem, this paper proposes a resource reservation method that can adjust adaptively the delay budget of traffic flow allocated to each RSVP-capable router in which the router uses the recently developed general-time fair queuing scheduler. The results of the simulation applying the proposed method to an evaluation network show that it may yield the gain of up to 90% compared to that in the original one in terms of the number of admitted flows.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.2
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• pp.10-16
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• 2016

For higher reliability against a link failure of a control network in railway vehicles, a recovery mechanism is needed. We introduce a problem that, when a link failure occurs in a ring-topology control network, a node may experience a significant increase of transmission delay depending on its relative position within the network. We then propose two mechanisms to solve this problem: (1) differentiating and prioritizing node traffic in forwarding; and (2) switching some nodes to a backup bus-topology network. Our simulation study shows that, while the first mechanism achieves a limited gain by only compensating queuing delay, the second one gets a sufficient gain which is impacted by the number of nodes switched to the bus network.

• Journal of Korean Society of Transportation
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• v.33 no.5
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• pp.478-487
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• 2015

This study proposed a Variable Speed Limits(VSL) algorithm to use traffic information based on Cumulative Demand-Capacity Analysis and evaluated its performance. According to the analysis result, the total of delay consisted of 3 separate parts. There was no change in total travel time although the total of delay decreased. These effects was analysed theoretically and then, evaluated through VISSIM, a microscopic simulator. VISSIM simulation results show almost same as those of theoretical analysis. Furthermore in SSAM analysis with VISSIM simulation log, the number of high risk collisions decreased 36.0 %. However, the total delay decrease effect is not real meaning of decrease effect because the drivers' desired speed is same whether the VSL model is operated or not. Nevertheless this VSL model maintains free flow speed for longer and increases the cycle of traffic speed fluctuation. In other words, this is decrease of delay occurrence and scale. The decrease of speed gap between upstream and downstream stabilizes the traffic flow and leads decrease number of high risk collision. In conclusion, we can expect increase of safety through total delay minimization according to this VSL model.