• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.3B
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• pp.255-263
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• 2009

In this paper, we propose an adaptive beacon scheduling algorithm to control a reception period of actual data according to variation of amount of traffic in IEEE 802.15.4 LR-WPAN(Low Rate-Wireless Personal Area Network) with the cluster-tree topology. If a beacon interval is shortened, the amount of the traffic a node receives can be increased while consumption of the energy can be also increased. In this sense, we can achieve optimal on orgy consumption by controlling the beacon interval when the amount of data to be received is being decreased. The result of simulation using NS-2 shows that the proposed algorithm improves performances in terms of packet loss rate and end-to-end delay compared with algorithm using a fixed beacon interval. For a design of cluster-tree based LR-WPAN managing delay-sensitive services, the proposed algorithm and the associated results can be applied usefully.

• The KIPS Transactions:PartC
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• v.9C no.3
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• pp.375-384
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• 2002

In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.12
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• pp.2635-2643
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• 1997

In this paepr, a routing algorithm is proposed in order to reduce average end-to-end delay and congesting in the high speed network. The algorithm proposed here uses the existing one which adaptively modifies routes and the amount of traffic allocated to each link as user traffic partterns flutuate. This algorithm is ELB(Equitable Load Balancing). Also, the new algorithm considers the proportional to the distance between source and destination. It reduces congestion from the ELB and average end-to-end delay from the propagation dealy concepts respectively. Through the simulation, it shows that the algorithm proposed here reduces average end-to-end delay over low load to high load, and it also guarantees the congestion control.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.5
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• pp.7-15
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• 2007

In wireless communication networks, single-radio single-channel architecture degrades throughput and end-to-end delay due to half-duplex transmission of wireless node and route intra interference. Also, In contention-based MAC (Medium Access Control) architecture, channel contention reduces throughput and packet collision enlarges end-to-end delay. In this paper, we use multi-radio multi-channel architecture which will make wireless node to operate in full duplex mode, and exclude route intra interference. Based on this architecture, we propose a new link metric, ccf which reflects the characteristics of a contention-based wireless link, and propose a routing path metric MCCR considering channel switching delay and route intra interference. MCCR is compared with MCR by simulation, the performance of a route established by MCCR outperforms the performance of a route by MCR in terms of throughput and end-to-end delay.

• KIPS Transactions on Computer and Communication Systems
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• v.7 no.9
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• pp.227-234
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• 2018

Guaranteeing the end-to-end delay deadline is an important issue for quality of service (QoS) of delay sensitive systems, such as real-time system, networked control system (NCS), and cyber-physical system (CPS). Most routing algorithms typically use the mean end-to-end delay as a performance metric and select a routing path that minimizes it to improve average performance. However, minimum mean delay is an insufficient routing metric to reflect the characteristics of the unpredictable wireless channel condition because it only represents average value. In this paper, we proposes a deadline-aware routing algorithm that maximizes the probability of packet arrival within a pre-specified deadline for CPS by considering the delay distribution rather than the mean delay. The proposed routing algorithm constructs the end-to-end delay distribution in a given network topology under the assumption of the single hop delay follows an exponential distribution. The simulation results show that the proposed routing algorithm can enhance QoS and improve networked control performance in CPS by providing a routing path which maximizes the probability of meeting the deadline.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10B
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• pp.1852-1860
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• 1999

This paper studied performance of the AAL2 for variable rate real time services in ATM network with discrete-time simulation model. In this simulation, input parameters are packet fill delay for AAL2 PDU generation, guard time for ATM cell generation, burstness and number of channels. Though variation of the above mentioned parameters, we obtained end-to end delay variations and throughput, analyzed performance effect of the each parameter for voice packet service.

• Journal of the Korea Society of Computer and Information
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• v.7 no.2
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• pp.101-114
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• 2002

The real multimedia application in wide area communication needs the guaranteed performance of communication service. Therefore, the resource is reserved at the moment of traffic burst and the region of connection admission possibility is widened at the basis of maximum cell rate. This of study shows that the end-to-end traffic delay to the traffic of burst state is guaranteed when the total of maximum transmission rate is higher than link speed by using the region of deterministic delay. The network load rate of connection admission can be improved by the inducement of delay bounds consideration each traffic characteristic to guarantee the end-to-end delay of network from single switch. This suggested buffering system using deterministic service do not give any influence to service quality and can guarantee the bounds of end-to-end delay. And it can also reduce the load of network even if the delay is increased according to the burst traffic characteristic. The above suggested system can be applied effectively to the various kinds of general network specification which admit both real time trafnc service and non-real time traffic service.

• Proceedings of the Korean Operations and Management Science Society Conference
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• 2006.05a
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• pp.1609-1615
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• 2006

Overlay multicast technique is an effective way as an alternative to IP multicast. Traditional IP multicast is not widely deployed because of the complexity of IP multicast technology and lack of application. But overlay multicast can be easily deployed by effectively reducing complexity of network routers. Because overlay multicast resides on top of densely connected IP network, In case of multimedia streaming service over overlay multicast tree, real-time data is sensitive to end-to-end delay. Therefore, moderate algorithm's development to this network environment is very important. In this paper, we are interested in minimizing maximum end-to-end delay in overlay multicast tree. The problem is formulated as a degree-bounded minimum delay spanning tree, which is a problem well-known as NP-hard. We develop tabu search heuristic with intensification and diversification strategies. Robust experimental results show that is comparable to the optimal solution and applicable in real time

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.5B
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• pp.253-263
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• 2005

This paper proposed a mobile multicast technique to satisfy end-to-end QoS for various user requirements in mobile network environment. In order to provide seamless mobility, fast handoff technique was applied. By using L2 mobile trigger, it was possible to minimize remarkable amount of packet loss by delay occurred during handoff. To provide efficient multicast, concept of hierarchy was introduced to Xcast++, which results in a creation of HXcast++. HXcast++ optimized transfer path of multicast and reduced expensive multicast maintenance costs caused by frequent handoff. Suggestion of GMA (Group Management Agent) mechanism allows joining to group immediately without waiting IGMP Membership query during handoff. GMA mechanism will minimize the delay for group registration process and the resource usage due to delay of withdrawal process. And also use of buffering & forwarding technique minimized packet loss during generation of multicast tree. IntServ/RSVP was used to provide End-to-End QoS in local domain and DiffServ was used in global domain. To minimize reestablishment of RSVP session delay, extended HXcast++ control messages ware designed to require PATH message. HXcast++ proposed in this thesis is defined as multicast technique to provide end-to-end QoS and also to satisfy various user requirements in mobile network environment.

• Journal of the Korea Society for Simulation
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• v.10 no.2
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• pp.15-24
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• 2001

In this paper we study the delay patterns of voice traffic for end-to-end users Caused by serving the whole bunch of applications traffic at the same time on the Voice over Internet Protocol (VoIP) network. Given the current situation that voice traffic is served along with other application services on the VoIP network, it is quite necessary to figure out how and by what the voice traffic requiring high QoS is delayed. We compare the delay performance of voice traffic on the VoIP network under FIFO with the one under Weighted Fair Queuing(WFQ), and discover the differences of the delay performance resulting from the use of different voice codec algorithms. The results of our study show that using the voice codec algorithm with a higher coding rate nd the queuing algorithm of WEQ can provide users with high-quality voice traffic.