• Journal of the Korean Geophysical Society
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• v.4 no.4
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• pp.239-249
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• 2001

NAT (Network Address Translation) is an IP address modification protocol that translates private IP addresses into authentic Internet addresses. The main features of NAT are to improve network security and to save IP addresses. Generally speaking, in order to perform its functionality. NAT uses the address informaiton in the packet header. Certain application protocols, however, use the information in the packet data as well as the imformation in the packet header to perform end-to-end communication. Therefor, to support these types of application protocols, NAT should be able to perform appropriate translation of protocol information in the packet data. In this thesis, we design and implement a method which translates virtual IP information in the packet data into real IP information by using port proxy server.

• The KIPS Transactions:PartC
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• v.10C no.5
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• pp.641-646
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• 2003

Packet processing speed of routers as well as transmission speed of physical links gives a great effect on IP packet transfer rate in Internet. The router forwards a packet after determining the next hop to the packet's destination. IP address lookup is a main design issue for high performance routers. In this paper, we propose a partitioned compressed-trie for speeding-up IP address lookup algorithms based on tie data structure by exploiting path compression. In the ,proposed scheme, IP prefixes are divided into several compressed-tries and lookup is performed on only one partitioned compressed-trie. Memory access time for IP address lookup is lessen due to compression technique and memory required for maintaining partition does not increased.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.3
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• pp.275-280
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• 2010

In this paper, VoIP(Voice over Internet Protocol) transmission performance for MANET(Mobile Ad-hoc Networks) is improved and analyzed with packet aggregation scheme which is aggregating some of short length packets to one large packet and sending to networks. VoIP simulator based on NS(Network Simulator)-2 is implemented and used to measure performance of VoIP traffic transmission. In this simulation, VoIP traffics are generated with parameters of some codes such as G.711, G.729A, GSM.AMR and iBLC. MOS(Mean Opinion Score), end-to-end network delay, packet loss rate and transmission bandwidth are measured. Performance improvements of 98% for MOS, 6.4times for end-to-end network delay, 32times for packet loss rate is shown as simulation results. On the other hand, transmission bandwidth is increased about maximum 10%. Finally, VoIP implementation guide for the performance with packet aggregation is suggested.

• The Journal of the Acoustical Society of Korea
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• v.33 no.1
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• pp.75-80
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• 2014

The quality of real-time Voice over Internet Protocol (VoIP) network is affected by network impariments such as delays, jitters, and packet loss. This paper proposes a packet loss concealment algorithm based on voice classification for enhancing VoIP speech quality. In the proposed method, arriving packets are classified by an adaptive thresholding approach based on the analysis of multiple features of short signal segments. The excellent classification results are used in the packet loss concealment. Additionally, linear prediction-based packet loss concealment delivers high voice quality by alleviating the metallic artifacts due to concealing consecutive packet loss or recovering lost packet.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.57 no.11
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• pp.2124-2126
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• 2008

In the wireless communication systems, the demand of multimedia services is also increased. Unlike typical data packets, realtime service such as VoIP packets have delay bound and low loss rate requirement. In this paper we propose a new scheduling algorithm that be able to allocate resources to the different kinds of services such as VoIP and data packet. The proposed algorithm considers both time delay and channel condition toe determine the priority. Simulation results show that the proposed algorithm works more efficiently than the conventional algorithms.

• Journal of Korea Society of Industrial Information Systems
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• v.10 no.2
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• pp.67-75
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• 2005

In this paper, we design and implement real time IP security packet analyzer on IPv4 and IPv6 network. This packet analyzer sniffs and analyzes the packets generated by the protocols that are used by IPsec, IKE, IPv4 and IPv6 such as AH, ESP, ISAKMP, IP, ICMP and so on. The purpose of this analyzer is to check current security status of the network automatically. In this paper we provide implementation details and the examples of security evaluation by using our security packet analyzer system.

• The KIPS Transactions:PartC
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• v.9C no.1
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• pp.89-96
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• 2002

Handoff is one of the most important factors that ma? degrade the performance of TCP connections in wireless data networks. In this paper, we present a lossless and duplication free handoff scheme called LPM (Last Packet Marking) for improving Cellular If semisoft handoff. LPM signals the safe handoff cue by sending a specially marked packet to mobile hosts. SPM (Semisoft rePly Message) is the only newly introduced control packet. Our performance study shows that LPM achieves lossless packet delivery without duplication and increases TCP throughput significantly.

• Speech Sciences
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• v.11 no.4
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• pp.67-73
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• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3E
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• pp.99-108
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• 2009

In this paper, we suggest criteria for objective measurement of speech quality in mobile VoIP (Voice over Internet Protocol) services over wireless mobile internet such as mobile WiMAX networks. This is the case that voice communication service is available under other networks. When mobile VoIP service users in the mobile internet network based on packet call up PSTN and mobile network users, but there have not been relevant quality indexes and quality standards for evaluating speech quality of mobile VoIP. In addition, there are many factors influencing on the speech quality in packet network. Especially, if the degraded speech with packet loss transfers to the other network users through the handover, voice communication quality is significantly deteriorated by the transformation of speech codecs. In this paper, we eventually adopt the Gilbert-Elliot channel model to characterize packet network and assess the voice quality through the objective speech quality method of ITU-T P. 862. 1 MOS-LQO for the various call scenario from mobile VoIP service user to PSTN and mobile network users under various packet loss rates in the transmission channel environments. Our simulation results show that transformation of speech codecs results in the degraded speech quality for different transmission channel environments when mobile VoIP service users call up PSTN and mobile network users.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1B
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• pp.1-8
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• 2006

Currently, there are many efforts underway to provide Internet service on integrated wireless and wired networks. Supporting IP mobility is one of the major issues to construct IP based wireless network. Mobile IP has been proposed to solve the IP Mobility problem. But, in processing frequent handoffs in cellular based wireless access network, Micro mobility protocols have been proposed to solve these problems. Micro mobility protocols proposed the Cellular IP, HAWII, and Hierarchical Mobile IP. Cellular IP attracts special attention for it's seamless mobility support in limited geographical areas. New BS must be known to occur begging of handoff in Cellular IP indirect handoff. Therefore during perceiving of hanoff, packet loss or packet duplication still can occur in Cellular IP indirect handoff, which results in the degradation of UDP and TCP performance. In this paper, we propose a enhanced indirect handoff scheme for Cellular IP. Proposed handoff scheme is using a crossover node to minimize the signalling procedure and using a buffering to minimize the packet loss or packet duplication.