• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.1B
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• pp.7-20
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• 2011

Recently, Proxy Mobile IPv6(PMIPv6) networks have been studied as the mobility management protocol to effectively use limited wireless resources. And the multicasting, which is core technology of the Internet broadcast system such as mobile IPTV, has been discussed mainly based on PMIPv6 network. However, multicasting based on PMIPv6 network causes disconnection of services because it does not solve problems of packet loss during binding and group joining procedure. Hence, we propose a seamless multicast scheme which prevents packet loss in PMIPv6 networks. The proposed scheme achieves lower latency than the existing scheme because it combines binding and group joining procedure, a1so it does not cause about packet losses due to performing buffering. We proved the performance using the simulations.

• The Transactions of the Korea Information Processing Society
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• v.7 no.2S
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• pp.721-732
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• 2000

Multimedia communication systems require not only high-performance computer hardwares and high-speed networks, but also a buffer management mechanism to process many data efficiently. Two buffer handling methods, Push and Pull, are commonly used. In the Push method, a server controls the flow of dat to a client, while in the Pull method, a client controls the flow of data from a server. Those buffering schemes can be applied to the data transfer between the packet receiving buffer, which receives media data from a network server, and media playout devices, which play the recived media data. However, the buffer management mechanism in client-sides mainly support either one of the Push or the Pull method. Consequently, they have some limitations to support various media playout devices. Futhermore, even though some of them support both methods, it is difficult to use since they can't provide a unified structure. To resolved these problems, in this paper, we propose an efficient and flexible Push/Pull buffer management mechanism at client-side. The proposed buffer management scheme supports both Push and Pull method to provide various media playout devices and to support buffering function to absorb network jitter. The proposed scheme can support the various media playback devices using a single buffer space which in consequence, saves memory space compared to the case that a client keeps tow types of buffers. Moreover, it facilitates the single buffer as a mechanism for the absorbing network jitter effectively and efficiently. The proposed scheme has been implemented in an existing multimedia communication system, so called ISSA (Integrated Streaming Service Architecture), and it shows a good performance result compared to the conventional buffering methods in multimedia communication environments.

• The KIPS Transactions:PartC
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• v.13C no.7 s.110
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• pp.821-830
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• 2006

As internet worms are spread out worldwide, the detection and filtering of worms becomes one of hot issues in the internet security. As one of implementation methods to detect worms, the Linux Netfilter kernel module can be used. Its basic operation for worm detection is a string matching where coming packet(s) on the network is/are compared with predefined worm signatures(patterns). A worm can appear in a packet or in two (or more) succeeding packets where some part of worm is in the first packet and its remaining part is in its succeeding packet(s). Assuming that the maximum length of a worm pattern is less than 1024 bytes, we need to perform a string matching up to two succeeding packets of 2048 bytes. To do so, Linux Netfilter keeps the previous packet in buffer and performs matching with a combined 2048 byte string of the buffered packet and current packet. As the number of concurrent connections to be handled in the worm detection system increases, the total size of buffer (memory) increases and string matching speed becomes low In this paper, to reduce the memory buffer size and get higher speed of string matching, we propose a string matching scheme without using buffer. The proposed scheme keeps the partial matching result of the previous packet with signatures and has no buffering for previous packet. The partial matching information is used to detect a worm in the two succeeding packets. We implemented the proposed scheme by modifying the Linux Netfilter. Then we compared the modified Linux Netfilter module with the original Linux Netfilter module. Experimental results show that the proposed scheme has 25% lower memory usage and 54% higher speed compared to the original scheme.

• Journal of the Korean Society for Railway
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• v.10 no.6
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• pp.758-765
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• 2007

In this paper, we propose an efficient seamless handoff scheme to minimize packet loss and unnecessary packets on the Internet using the peculiar mobility characteristics of public vehicles such as trains. MR (Mobile Router) in a train visits each AR (Access Router) in the fixed order. As the MR detects reachability to the NAR (Next Access Router) on the new link, the PAR (Previous Access Router) can directly deliver packets from MR's HA (Home Agent) to the NAR according to the HML (Handoff Mobile router List). Then. the NAR buffers them until the MR finishes L3 (Layer 3) handoff procedure with the NAR. Therefore, our scheme can support a seamless handoff without the packet loss and unnecessary packets on the Internet. The result of our performance evaluation has shown that the proposed scheme could provide excellent performance, compared with the NEMO basic support protocol and the Bi-casting protocol.

• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.

• 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.

• ETRI Journal
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• v.24 no.4
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• pp.311-322
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• 2002

Presently, optical burst switching (OBS) technology is under study as a promising solution for the backbone of the optical Internet in the near future because OBS eliminates the optical buffer problem at the switching node with the help of no optical/electro/optical conversion and guarantees class of service without any buffering. To implement the OBS network, there are a lot of challenging issues to be solved. The edge router, burst offset time management, and burst assembly mechanism are critical issues. In addition, the core router needs data burst and control header packet scheduling, a protection and restoration mechanism, and a contention resolution scheme. In this paper, we focus on the burst assembly mechanism. We present a novel data burst generation algorithm that uses hysteresis characteristics in the queueing model for the ingress edge node in optical burst switching networks. Simulation with Poisson and self-similar traffic models shows that this algorithm adaptively changes the data burst size according to the offered load and offers high average data burst utilization with a lower timer operation. It also reduces the possibility of a continuous blocking problem in the bandwidth reservation request, limits the maximum queueing delay, and minimizes the required burst size by lifting up data burst utilization for bursty input IP traffic.

• Journal of KIISE:Information Networking
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• v.28 no.2
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• pp.242-250
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• 2001

This paper deals with the TCP bandwidth guarantee problem in a differentiated serviccs(Diffserv) network. The Diffserv assured s<:rvice differentiates packet drop probabilities to guarantee the promised bandwidth even under network congestion. However a token buffer marker fails to show adequate performance because TCI' generates packets according to the unique Tel' congestion control mechanism. We propose a marker that uses a data buffer as well as a token buffer. The marker with a data buffer works well with the assured service mechanism because it smooths Tel' traffic. We showed that the marker with a data buffer achieves the target throughput better than a marker with a token buffer only. We also showed that the optimal buffer size is proportional to reserved throughput and HTT.

• International journal of advanced smart convergence
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• v.2 no.1
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• pp.6-11
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• 2013

Network Coding (NC) is a new paradigm for network communication. In network coding, intermediate nodes create new packets by algebraically combining ingress packets and send it to its neighbor node by broadcast manner. NC has rapidly emerged as a major research area in information theory due to its wide applicability to communication through real networks. Network coding is expected to improve throughput and channel efficiency in the wireless multi-hop network. Many researches have been carried out to employ network coding to wireless ad-hoc network. In this paper, we proposed a dynamic queue management to improve coding opportunistic to enhance efficiency of NC. In our design, intermediate nodes are buffering incoming packets to encode queue. We expect that the proposed algorithm shall improve encoding rate of network coded packet and also reduce end to end latency. From the simulation, the proposed algorithm achieved better performance in terms of coding gain and packet delivery rate than static queue management scheme.

• Journal of Internet Computing and Services
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• v.3 no.2
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• pp.27-37
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• 2002

This paper analyzes problems of existing Mobile IP and redesigns component functions, and proposes handover method using LRS and BCS for handover performance improvement. When mobile node moves another domain, the proposed BCS recognizes its mobility. So, BCS manages mobile node's binding information and serves micro mobility. When handover occurs among domains, micro mobility enables using LRS, As BCS and LRS have buffering function, they can reduce packet loss by forwarding the buffered datagram. It's designed component is implemented in LINUX environment. And then, handover performance was evaluated by simulation. The results of proposed handover method are better than existing method in both transmission delay time and packet loss.