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

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2015.05a
• /
• pp.108-110
• /
• 2015

In this paper, we proposed the relay node selection scheme using property of node such as velocity, direction in Delay Tolerant Networks. the existing selection scheme is caused the problem increasing the transmission delay and packet loss, if select the relay node for different mobile with destination. To overcome this problem, we proposed the relay node selection scheme using the property of mobile node. the proposed scheme represents and shares the property of mobile node. The proposed algorithm assumed the sketchy position of node from mobile node delivering property of destination. In addition, the propose algorithm recognizes and analyzes the context of mobile node to provide the relay node transferring the data efficiently. The simulation result provides the better result in terms of transmission delay and packet delivery ratio by selecting transmission by relay node according property of node.

• Journal of Korea Society of Digital Industry and Information Management
• /
• v.5 no.2
• /
• pp.141-156
• /
• 2009

As the TCP is the protocol designed for the wired network that packet loss probability is very low, because TCP transmitter takes it for granted that the packet loss by the wireless network characteristics is occurred by the network congestion and lowers the transmitter's transmission rate, the performance is degraded. The Snoop Protocol was designed for the wired network by putting the Snoop agent module on the BS(Base Station) that connect the wire network to the wireless network to complement the TCP problem. The Snoop agent cash the packets being transferred to the wireless terminal and recover the loss by resending locally for the error occurred in the wireless link. The Snoop agent blocks the unnecessary congestion control by preventing the dupack (duplicate acknowledgement)for the retransmitted packet from sending to the sender and hiding the loss in the wireless link from the sender. We evaluated the performance in the wired/wireless network and in various TCP versions using the TCP designed for the wired network and the Snoop designed for the wireless network and evaluated the performance of the wired/wireless hybrid network in the wireless link environment that the continuous packet loss occur.

• Journal of Communications and Networks
• /
• v.4 no.2
• /
• pp.108-117
• /
• 2002

TCP, which was developed on the basis of wired links, supposes that packet losses are caused by network congestion. In a wireless network, however, packet losses due to data corruption occur frequently. Since TCP does not distinguish loss types, it applies its congestion control mechanism to non-congestion losses as well as congestion losses. As a result, the throughput of TCP is degraded. To solve this problem of TCP over wireless links, previous researches, such as split-connection and end-to-end schemes, tried to distinguish the loss types and applied the congestion control to only congestion losses; yet they do nothing for non-congestion losses. We propose a novel transport protocol for wireless networks. The protocol called VS-TCP (Variable Segment size Transmission Control Protocol) has a reaction mechanism for a non-congestion loss. VS-TCP varies a segment size according to a non-congestion loss rate, and therefore enhances the performance. If packet losses due to data corruption occur frequently, VS-TCP decreases a segment size in order to reduce both the retransmission overhead and packet corruption probability. If packets are rarely lost, it increases the size so as to lower the header overhead. Via simulations, we compared VS-TCP and other schemes. Our results show that the segment-size variation mechanism of VS-TCP achieves a substantial performance enhancement.

• The Journal of the Korea Contents Association
• /
• v.10 no.1
• /
• pp.46-58
• /
• 2010

The user valence of VoIP services with SIP protocol is increasing rapidly because of cheap communication cost and its conveniency. But attacker can easily modify the packet contents of SIP protocol as SIP header is transmitted by using UDP methods in text form. The reason is that SIP protocols does not provide an authentication function on the transmission session. Therefore, existing SIP protocol is very weak on SIP Packet Flooding attack etc. In order to solve like this kinds of SIP vulnerabilities, we used SIP status codes under the monitoring module for detecting SIP Flooding attacks and additionally proposed an advanced protocol where the authentication and security function is strengthened about SIP packet. We managed SIP session spontaneously in order to strengthen security with SIP authentication function and to solve the vulnerability of SIP protocol. The proposed mechanism can securely send SIP packet to solves the security vulnerability with minimum traffic transmission. Also service delay in SIP proxy servers will be minimized to solve the overload problem on SIP proxy server.

• Journal of KIISE:Information Networking
• /
• v.32 no.4
• /
• pp.495-507
• /
• 2005

Measuring the end-to-end available bandwidth in the Internet is a useful tool for distributedapplication services or QoS (Quality-of-Service) guarantee. To measure the end-to-end available bandwidth, Single-hop Gap model-based packet train measurement techniques are well-known. However, the error of packet train output gap can happen by network topologies. This error of the output gap causes the inaccuracy of the available bandwidth measurement. In this paper, we propose a new end-to-end available bandwidth measurement technique with accurate capacity measurement and fast convergence methods. To solve the erroneous capacity measurement problem of the back-to-back packet train transmission, we propose a new available bandwidth measurement method by decoupling the capacity measurement with the initial gap of the packet train. Also, we propose a new technique to predict the proper initial gap of the packet train for faster convergence. We evaluate our proposed method by the simulation in various topologies comparing with previous methods.

• Proceedings of the IEEK Conference
• /
• 2000.06a
• /
• pp.9-12
• /
• 2000

A forward error correction (FEC) is usually used to correct the errors of the real-time data occurred at the reciever side which require a real-time transmission. The data transmission is peformed after being encapsulating by RTP and UDP. In the ITU-T study group 16, four FEC schemes using the XORing are presented. In the paper, a new supplementary scheme is proposed. In the delay problem the new scheme performs better than the scheme 3 but in the recovery ability for successive packet loss is worse than scheme 3. The proposed scheme which supplements the present schemes can be adapted easily to the current network environment.

• The KIPS Transactions:PartB
• /
• v.12B no.2 s.98
• /
• pp.167-172
• /
• 2005

A rapid growth in the number of Internet users and in massive media contents tends to cause server overload and high network traffic. As a consequence, it can be safely assumed that the quality of service is on the brink of deterioration. A solution of this problem is the development of the Content Distribution Network. Therefore, in this paper, we propose a transmission model using CDN. This model uses regional media server which closes to client, and uses MDC coding and multi-channel to reduce packet loss and delay. In conclusion, this study improved performance of multimedia service by reducing packet loss and delay.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.15 no.3
• /
• pp.874-890
• /
• 2021

With the widespread deployment of the fifth-generation (5G) communication networks, various real-time applications are rapidly increasing and generating massive traffic on backhaul network environments. In this scenario, network congestion will occur when the communication and computation resources exceed the maximum available capacity, which severely degrades the network performance. To alleviate this problem, this paper proposed an intelligent resource allocation (IRA) to integrate with the extant resource adjustment (ERA) approach mainly based on the convergence of support vector machine (SVM) algorithm, software-defined networking (SDN), and mobile edge computing (MEC) paradigms. The proposed scheme acquires predictable schedules to adapt the downlink (DL) transmission towards off-peak hour intervals as a predominant priority. Accordingly, the peak hour bandwidth resources for serving real-time uplink (UL) transmission enlarge its capacity for a variety of mission-critical applications. Furthermore, to advance and boost gateway computation resources, MEC servers are implemented and integrated with the proposed scheme in this study. In the conclusive simulation results, the performance evaluation analyzes and compares the proposed scheme with the conventional approach over a variety of QoS metrics including network delay, jitter, packet drop ratio, packet delivery ratio, and throughput.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 1995.06a
• /
• pp.97-100
• /
• 1995

In transmission of image sequences over ATM networks, the packet loss problem and channel sharing efficiency are important. As a possible solution two-layer video coding methods have been proposed. These methods transmit video information over the network with different levels of protection with respect to packets loss. In this paper, a two-layer coding method using pyramid structure is proposed and several realizations of two-layer video coding methods are presented and their performances are compared.