• Journal of Information Processing Systems
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• v.7 no.2
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• pp.241-260
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• 2011

In earlier days, most of the data carried on communication networks was textual data requiring limited bandwidth. With the rise of multimedia and network technologies, the bandwidth requirements of data have increased considerably. If a network link at any time is not able to meet the minimum bandwidth requirement of data, data transmission at that path becomes difficult, which leads to network congestion. This causes delay in data transmission and might also lead to packet drops in the network. The retransmission of these lost packets would aggravate the situation and jam the network. In this paper, we aim at providing a solution to the problem of network congestion in mobile ad hoc networks [1, 2] by designing a protocol that performs routing intelligently and minimizes the delay in data transmission. Our Objective is to move the traffic away from the shortest path obtained by a suitable shortest path calculation algorithm to a less congested path so as to minimize the number of packet drops during data transmission and to avoid unnecessary delay. For this we have proposed a protocol named as Congestion Aware Selection Of Path With Efficient Routing (CASPER). Here, a router runs the shortest path algorithm after pruning those links that violate a given set of constraints. The proposed protocol has been compared with two link state protocols namely, OSPF [3, 4] and OLSR [5, 6, 7, 8].The results achieved show that our protocol performs better in terms of network throughput and transmission delay in case of bulky data transmission.

• Journal of the Korea Society of Computer and Information
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• v.18 no.10
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• pp.47-54
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• 2013

In this paper, the structural design of optical Internet is analyzed and by using UNI (User Network Interface) signaling protocol an optical transmission experiment was performed. The hierarchical structure of the basic optical Internet consists of the backbone network, the service network and the access network. The necessary functions for each layer were described as follows: Control structure of the optical transport layer, network operation and management structure, internetworking technology of sub networks, routing and signaling technology. By using UNI signaling protocol from OIF (Optical Internetworking Forum), the optical transmission in the proposed structure of the optical Internet network was experimented. By the traffic generation of LSP (Label Switched Path) data packets along the route-configuration was delivered to UNI. Finally, by showing the value of TCP (Transmission Control Protocol) packets the optical transmission was completely and successfully demonstrated.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.12
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• pp.539-546
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• 2002

In this paper, we consider a wireless multimedia environment to service both real-time video traffic and non-real-time WWW-application traffic In our suggested new packet scheduling algorithm, we consider both the accumulation counter and SIR to reduce delay in real-time traffic. In addition, our packet scheduling algorithm gives priority first to real-time video traffic service and then to non-real-time internet Packet service when real-time traffic service is absent. From the simulation results, we find that the AC (Accumulation Counter) scheme has much smaller delay than the conversional M-LWDF scheme for real-time video data users, which has a special quality sensitive to its own delay. We also consider the transmission structure of using both the frame period in the time-axis and the OVSF codes in the code-axis at the same time, which is similar to the structure of HSDPA system.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.1
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• pp.69-76
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• 2010

We describe an all-optical circuit which recognizes the header information of packet-by-packet in the access networks based on FTTH. The circuit's operation is confirmed by an experiment in the recognition of 3 and 4 header bits. The output from the header recognition circuit appears in a signal assigned in the time axis according to the header information. The recognition circuit of header for self-routing has a very simple structure using only delay lines and switches. The circuit is expected that it can be constructed of the high reliability and the low cost. Also, the circuit can solve the problems of the power loss and private security which is the weak point of the TDM-PON method by being established a unique transmission line to each subscriber.

• Proceedings of the KIEE Conference
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• 2008.11a
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• pp.379-381
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• 2008

TETRA(Terrestrial Trunked Radio) is a digital trunked radio standard developed by the ETSI(European Telecommunications Standards Institute). Currently, TETRA was set Digital TRS in electric power If wireless backbone network. In this time, we use many company's TETRA modem. So, TETRA modem performance evaluation is very important. TETRA modem use two type of Data transfer mode. One is Packet Data using UDP/IP. and the other is SDS(Short Data Service). In this paper, We generate Packet Data using Traffic Generator module. Packet Data transfer 1000 times each 10 bytes to 400 bytes. We analyze transmission delay time, success rate and standard deviation.

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

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4B
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• pp.255-269
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• 2006

The differentiated services(DiffServ) architecture provides packet level service differentiation through the simple and predefined Per-Hop Behaviors(PHBs). The Assured Forwarding(AF) PHB proposed as the assured services uses the RED-in/out(RIO) approach to ensusre the expected capacity specified by the service profile. However, the AF PHB fails to give good QoS and fairness to the TCP flows. This is because OUT(out- of-profile) packet droppings at the RIO buffer are unfair and sporadic during only network congestion while the TCP's congestion control algorithm works with a different round trip time(RTT). In this paper, we propose an Adaptive Regulating Drop(ARD) marker, as a novel dropping strategy at the ingressive edge router, to improve TCP fairness in assured services without a decrease in the link utilization. To drop packets pertinently, the ARD marker adaptively changes a Temporary Permitted Rate(TPR) for aggregate TCP flows. To reduce the excessive use of greedy TCP flows by notifying droppings of their IN packets constantly to them without a decrease in the link utilization, according to the TPR, the ARD marker performs random early fair remarking and dropping of their excessive IN packets at the aggregate flow level. Thus, the throughput of a TCP flow no more depends on only the sporadic and unfair OUT packet droppings at the RIO buffer in the core router. Then, the ARD marker regulates the packet transmission rate of each TCP flow to the contract rate by increasing TCP fairness, without a decrease in the link utilization.

• Journal of Broadcast Engineering
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• v.4 no.1
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• pp.1-11
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• 1999

This paper introduces orthogonally multiplexed modulation and demodulation methods based on Wavelet Packet Bases and particularly describes Wavelet Packet Modulation (WPM) techniques that provide the designer of transmission signal set in time-frequency domain with tree structural information which can be adapted to given channel characterristics. Multi-dimensional signaling methods are also contrasted to common and different characteristics of conventional QAM. multi-tone modulation methods. The paper addresses the mothod how to find a best tree structure that has more adaptivity to impulse and narrowband tone pulse noises using a tunning algorithm which arbitrarily partitions the time-frequency space and makes a suitable orthogonal signaling waveforms. Simulation results exhibits a favorable performance over existing mod/demod methods specially for narrowband tone pulse and impulse interferences.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.561-564
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• 2005

This paper proposes an access control algorithm for guaranteeing fair packet transmissions in CDMA-based slotted ALOHA systems. In the proposed algorithm, the base station calculates the packet transmission and retransmission probabilities based on the offered loads and then broadcasts these probabilities to all mobile stations. Mobile stations, which have a packet to transmit, attempt to transmit a packet with the received probabilities. Simulation results show that the proposed algorithm can offer better system throughput and average delay than the conventional algorithm. Results also show that the proposed algorithm can guarantee a good fairness among all mobile stations regardless of the offered loads.