• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.05a
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• pp.233-236
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• 2010

This paper describes the packet traffic management of the Ubiquitous Healthcare System. In this system, ECG signal and accelerometer signal is transmitted from a wearable health shirt (WHS) to the base station. However, with the increment of users in this system, traffic over-load issue occurs. The main aim of this paper is to reduce the traffic over-load issue between sensor nodes by only transmitting the required signals to the base station when irregular activities are observed. In order to achieve this, in-network processing is adapted where the process of observation is conducted inside the sensor node of WHS. Results shows that irregular activities such as fall can be detected on real-time inside the sensor node and thus resolves traffic over-load issue.

• Journal of Internet Computing and Services
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• v.7 no.3
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• pp.53-60
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• 2006

As it has been continuously increased the volume of traffic over Internet, it is hard for a network traffic monitoring system to analysis every packet in a real-time manner. While it is increased usage of applications which are dynamically allocated port number such as peer-to-peer(P2P), steaming media, messengers, users want to analyze traffic data generated from them. This high level analysis of each packet needs more processing time. This paper proposes to introduce load shedder for limiting the number of packets. After it determines what application generates a selected packet, the packet is analyzed with a defined application protocol.

• Management Science and Financial Engineering
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• v.4 no.2
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• pp.43-58
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• 1998

Packet switch is a key component in high speed digital networks. This paper investigates congestion phenomena in the packet switching with input buffers. For large value of switch size N, mathematical models have been developed to analyze asymptotic maximum switch throughput under nonuniform traffic. Simulation study has also been done for small values of finite N. The rapid convergence of the switch performance with finite switch size to asymptotic solutions implies that asymptotic analytical solutions approximate very closely to maximum throughputs for reasonably large but finite N. Numerical examples show that non-uniformity in traffic pattern could result in serious degradation in packet switch performance, while the maximum switch throughput is 0.586 when the traffic load is uniform over the output trunks. Window scheduling policy seems to work only when the traffic is relatively uniformly distributed. As traffic non-uniformity increases, the effect of window size on throughput is getting mediocre.

• Journal of Communications and Networks
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• v.16 no.5
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• pp.548-558
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• 2014

Load distribution is vital to the performance of multipath transport. The task becomes more challenging in real-time multimedia applications (RTMA), which impose stringent delay requirements. Two key issues to be addressed are: 1) How to minimize end-to-end delay and 2) how to alleviate packet reordering that incurs additional recovery time at the receiver. In this paper, we propose sub-packet based multipath load distribution (SPMLD), a new model that splits traffic at the granularity of sub-packet. Our SPMLD model aims to minimize total packet delay by effectively aggregating multiple parallel paths as a single virtual path. First, we formulate the packet splitting over multiple paths as a constrained optimization problem and derive its solution based on progressive approximation method. Second, in the solution, we analyze queuing delay by introducing D/M/1 model and obtain the expression of dynamic packet splitting ratio for each path. Third, in order to describe SPMLD's scheduling policy, we propose two distributed algorithms respectively implemented in the source and destination nodes. We evaluate the performance of SPMLD through extensive simulations in QualNet using real-time H.264 video streaming. Experimental results demonstrate that: SPMLD outperforms previous flow and packet based load distribution models in terms of video peak signal-to-noise ratio, total packet delay, end-to-end delay, and risk of packet reordering. Besides, SPMLD's extra overhead is tiny compared to the input video streaming.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.937-942
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• 2003

In this paper, we measured and examined RTT delays and packet losses according to the changes of stationary loads for two typical stream-type traffics, a DV and a MPGE2 on the R＆D Gigabit Network testbed, JGN. As the result of our actual measurements, we realized that the packet size of stationary load have no effects on a DV and a MPGE2 stream on the very high-speed network(50Mbps, IP over ATM). When its bandwidth and stationary load exceeds 95％ of network bandwidth, packet losses appeared and RTT delay increased rapidly. Also we realized that the number and size of Receive ＆ Transmit buffer on the end systems have no effects on packet losses and RTT delays.

• Journal of Communications and Networks
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• v.17 no.4
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• pp.370-383
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• 2015

Effective link quality estimation is a vital issue for reliable routing in wireless networks. This paper studies the performance of expected transmission count (ETX) under different traffic loads. Although ETX shows good performance under light load, its performance gets significantly worse when the traffic load is high. A broadcast packet storm due to new route discoveries severely affects the link ETX values under high traffic load, which makes it difficult to find a good path. This paper presents the design and implementation of a variation of ETX called high load - ETX (HETX), which reduces the impact of route request broadcast packets to link metric values under high load. We also propose a reliable routing protocol using link quality metrics, which is called link quality distance vector (LQDV). We conducted the evaluation of the performance of three metrics - HETX, ETX and minimum hop-count. The simulation results show that HETX improves the average route throughput by up to 25% over ETX under high traffic load. Minimum hop-count has poor performance compared with both HETX and ETX at all of the different traffic loads. Under light load, HETX and ETX show the same performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2090-2095
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• 1998

In this paper, we measured voice packet loss over the Korean Internet and analyzed percentage articulation by variation of the packet loss. To do this, we reviewed real-time transmission service based on RTP/UDP/IP and test method of the transmission quality. and implemented the real-time speech transmission system using GSM and UDP/IP. Monosyllable list has been chosen for the percentage articulation test, each voice packet has been coded and compressed by GSM and it has sequence number to measured packet loss and to recover out-of-order packets. In transmission results using seven router over the Korean Internet, we have show that loss rates reached 1.6% (unload), 22.5%(load) and loss rates after packet recovery by resequencing and FEC are from 9% to 35%. Finally, we have shown that percentage articulations by variation of the network traffic are Table 4.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.3
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• pp.179-196
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• 1986

In this paper, the peformance of a statistical packet voice/data multiplexer is studied. In ths study we assume that in the packet voice/data multiplexer two separate finite queues are used for voice and data traffics, and that voice traffic gets priority over data. For the performance analysis we divide the output link of the multiplexer into a sequence of time slots. The voice signal is modeled as an (M+1) - state Markov process, M being the packet generation period in slots. As for the data traffic, it is modeled by a simple Poisson process. In our discrete time domain analysis, the queueing behavior of voice traffic is little affected by the data traffic since voice signal has priority over data. Therefore, we first analyze the queueing behavior of voice traffic, and then using the result, we study the queueing behavior of data traffic. For the packet voice multiplexer, both inpur state and voice buffer occupancy are formulated by a two-dimensional Markov chain. For the integrated voice/data multiplexer we use a three-dimensional Markov chain that represents the input voice state and the buffer occupancies of voice and data. With these models, the numerical results for the performance have been obtained by the Gauss-Seidel iteration method. The analytical results have been verified by computer simylation. From the results we have found that there exist tradeoffs among the number of voice users, output link capacity, voic queue size and overflow probability for the voice traffic, and also exist tradeoffs among traffic load, data queue size and oveflow probability for the data traffic. Also, there exists a tradeoff between the performance of voice and data traffics for given inpur traffics and link capacity. In addition, it has been found that the average queueing delay of data traffic is longer than the maximum buffer size, when the gain of time assignment speech interpolation(TASI) is more than two and the number of voice users is small.

• Proceedings of the Korea Society for Simulation Conference
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• 2001.10a
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• pp.440-443
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• 2001

As inter-network traffics grows rapidly, the router systems as a network component becomes to be capable of not only wire-speed packet processing but also plentiful programmability for quality services. A network processor technology is widely used to achieve such capabilities in the high-end router. Although providing two such capabilities, the network processor can't support a deep packet processing at nominal wire-speed. Considering QoS may result in performance degradation of processing packet. In order to achieve foster processing, one chipset of network processor is occasionally not enough. Using more than one urges to consider a problem that is, for instance, an out-of-order delivery of packets. This problem can be serious in some applications such as voice over IP and video services, which assume that packets arrive in order. It is required to develop an effective packet processing mechanism leer using more than one network processors in parallel in one linecard unit of the router system. Simulation analysis is also needed for verifying the mechanism. We propose the packet processing mechanism consisting of more than two NPs in parallel. In this mechanism, we use a load-balancing algorithm that distributes the packet traffic load evenly and keeps the sequence, and then verify the algorithm with simulation analysis. As a simulation tool, we use DEVSim++, which is a DEVS formalism-based hierarchical discrete-event simulation environment developed by KAIST. In this paper, we are going to show not only applicability of the DEVS formalism to hardware modeling and simulation but also predictability of performance of the load balancer when implemented with FPGA.

• ETRI Journal
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• v.23 no.2
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• pp.52-60
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• 2001

In this paper, a medium access control protocol is proposed for integrated voice and data services in wireless local networks. Uplink channels for the proposed protocol are composed of time slots with multiple spreading codes per slot based on slotted code division multiple access (CDMA) systems. The proposed protocol uses spreading code sensing and reservation schemes. This protocol gives higher access priority to delay-sensitive voice traffic than to data traffic. The voice terminal reserves an available spreading code to transmit multiple voice packets during a talkspurt. On the other hand, the data terminal transmits a packet without making a reservation over one of the available spreading codes that are not used by voice terminals. In this protocol, voice packets do not come into collision with data packets. The numerical results show that this protocol can increase the system capacity for voice service by applying the reservation scheme. The performance for data traffic will decrease in the case of high voice traffic load because of its low access priority. But it shows that the data traffic performance can be increased in proportion to the number of spreading codes.