• Journal of KIISE:Information Networking
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• v.32 no.4
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• pp.483-494
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• 2005

The standard streaming delivery Is mostly based on UDP with no end-to-end congestion control. For this reason, wide usage of multimedia applications in Internet might lead to congested networks. To avoid such a situation, studies on the congestion controlled streaming delivery has been increasingly done after the 1990s. However, by considering only the stability aspect of network, these works ignore the characteristics of multimedia streaming applications. Moreover, most of previous works have no consideration on the network delay which produces an effect on streaming service. In this thesis, in order to overcome limitations of the previous transmission schemes for streaming, we propose a new transmission scheme called 'BEST(Buffer-driven Efficient STreaming)'. The BEST takes a hybrid approach that considers both user-level requirements and network-level requriements. Therefore, the BEST improves the stability of networks by adjusting the sending rate suitable for network status and it also provides the smoothed playback by preventing buffer underflow or overflow. The BEST is designed to consider high-delay networks. Through the simulation, we prove that the BEST satisfies both user-level and network-level requirements in a high-delay network environments.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.5
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• pp.16-24
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• 2011

The IEEE 802.11 wireless LAN (Local Area Network) is widely used for wireless access due to its easy deployment and low cost. Multicast in wireless LANs is very useful for transmitting data to multiple receivers compared to unicast to each receiver. In the IEEE 802.11 wireless LAN, multicast transmissions are unreliable since multicast data packets are transmitted without any feedback from receivers. Recently, various protocols have been proposed to enhance the reliability of multicast transmissions. They still have serious problems in reliability and efficiency due to the excessive control overhead by the use of a large number of control packets in the error recovery process, and due to a large number of retransmissions to satisfy all receivers. In this paper, we propose an effective scheme called PTRM(Proactive Transmission based Reliable Multicast). The proposed scheme uses a block erasure code to generate parity packets and to reduce the impact of independent packet error among receivers. After generating parity packets, the PTRM transmits data packets as many as receivers need to recover error, and then requests feedback from them. The simulation results show that the proposed scheme provides reliable multicast while minimizing the feedback overhead.

• The KIPS Transactions:PartC
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• v.16C no.4
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• pp.527-534
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• 2009

Almost every mobile phone supports MMS to send and receive multimedia messages among users. MMS is suitable for different types of mobile phones and effectively operates in wireless environments. So the mobile content delivery method using MMS can be efficient. However, there would be some problems if a mobile service provider tries to deliver mobile content to subscribers by using the existing MMS. In this paper, we propose an improved data transmission and control scheme for MMS. Our scheme assists mobile service providers to deliver mobile contents easily by modifying the MMS system software. By using our scheme, mobile service providers are able to manage mobile contents stored in mobile devices. Therefore, they can provide various services such as mobile daily news service, automatic software upgrade service, mobile advertising service, etc. We verify the possibility of the proposed scheme through simulation.

• Journal of KIISE:Information Networking
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• v.32 no.1
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• pp.12-19
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• 2005

Reliable transport protocols such as TCP are tuned to Perform well in traditional networks where packet losses occur mainly because of congestion. In a wireless network, however, packet losses will occur more often due to reasons such as the high bit error rate and the handoff rather than due to congestion. When using TCP over wireless network, TCP responds to losses due to the high bit error rate and the handoff by invoking congestion control and avoidance algorithms, resulting in the degraded end-to-end performance in the wireless network. There have been several schemes for improving TCP performance over wireless links. Among them, SNOOP Is a very promising scheme because of the localized retransmission. In this thesis, an efficient scheme is proposed by modifying SNOOP scheme. The invocation of congestion control mechanism is now minimized by knowing the cause of packet loss.

• Journal of Korea Multimedia Society
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• v.7 no.5
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• pp.680-688
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• 2004

In this paper, we propose a new handoff scheme which is efficient in hybrid mobile data network consists of cellular mobile network and wireless LAN. In this scheme, handoff is delayed until connections with wireless LAN and data rates are smoothly decreased according to becon signal strength of wireless LAN. By doing so, data transfer capacity is increased and required data buffer in handoff for mobile and network system can be decreased. We analyze new handoff scheme by computer simulation. The results show that 180Mbytes data can be transferred additionally in handoff processing and required buffer size can be decreased 1/2 with the conditions that mobile speed is 1Km/hr and the data rate of a original call is 2,048Kbps.

• The KIPS Transactions:PartC
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• v.12C no.6 s.102
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• pp.917-926
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• 2005

In this paper, we propose a transport layer mobility support protocol for multimedia streaming services in mobile/wireless environments by extending PR-SCTP to support mobility. We named the proposed scheme to be uPR-mSCTP. The rules related to provide unordered and unreliable data transfer, as well as an approach to minimize handover latency we proposed. Through the simulation results, it is shown that, since uPR-mSCTP reduces transmission rates during handover, not only it reduces the waste of network resources, but it also has the packet loss rate similar to FMIPv6, which is a lot smaller than UDP over MIPv6 or HMIPv6 It also incurs smaller control packet overhead compared to the network layer mobility support approaches.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.9B
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• pp.777-784
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• 2006

In this paper, we propose the PON-based access network based on conventional TDM-PON architecture, which utilizes WDM wavelength channel and bandwidth dynamically. It is also described a dynamic MAC protocol in order to increase the number of subscribers and efficiency of resource utilization. Of particular importance in the proposed approach for MAC protocol is that the wavelength channel and time slot for up/downlink is dynamically allocated according to the required QoS level and the amount of data in data transmission, through the dedicated control channel between OLT and ONU. We evaluate the performance of average packet end-to-end delay in a statistical analysis and numerical analysis. In addition, through simulations with various traffic models, we verified the superior performance of the proposed approach by comparing with the results of other E-PONs.

• Journal of KIISE:Computer Systems and Theory
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• v.32 no.9
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• pp.465-474
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• 2005

For high density SoC design, on-chip communication based on bus interconnection encounters bandwidth limitation while an NoC(Network-on-Chip) approach suffers from unacceptable complexity in its Implementation. This paper introduces a new on-chip communication protocol, SNP (SoC Network Protocol) to overcome these problems. In SNP, conventional on-chip bus signals are categorized into three groups, control, address, and data and only one set of wires is used to transmit all three groups of signals, resulting in the dramatic decrease of the number of wires. SNP efficiently supports master-master communication as well as master-slave communication with symmetric channels. A sequencing rule of signal groups is defined as a part of SNP specification and a phase-restoration feature is proposed to avoid redundant signals transmitted repeatedly over back-to-back transactions. Simulation results show that SNP provides about the same bandwidth with only $54\%$ of wires when compared with AMBA AHB.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.9
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• pp.55-65
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• 2004

ATM was recommended by the ITU and ATM Forum as a means of transportation for B-ISDN. At this time, due to the comprehensive mature of ATM protocol, ATM has been adapted as the backbone system for carrying Internet traffi $c^{[1,2,3,4]}$. But major conceptsregarding the ATN protocol will be used on future technology. This paper presents preventive congestion control mechanisms for detecting HTR(Hard-To Reach) code in ATM systems, in particular for an improved HTR call registration method using network signaling information will discussed. In high speed circuit switching system environments, a fast HTR control mechanism is necessary. We present research results for improving HTR call registration and control methods using network signaling information and fuzzy control mechanisms. We concluded that it showed fast congestion avoidance mechanisms with a fewer system load maximized the efficiency of network resources by restricting ineffective machine attempts.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.4
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• pp.741-746
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• 2012

Transmission control protocol(TCP) widely used as a transport protocol in the Internet includes a loss recovery function that detects and recovers packet losses by retransmissions. The loss recovery function consists of the two algorithms; fast retransmit and fast recovery. There have been researches to avoid nonnecessary retransmission timeouts (RTOs), which leads to selective acknowledgement (SACK) option and limited transmit scheme that are standardized by IETF (Internet Engineering Task Force). Recently, a method that covers the case in which a retransmitted packet is lost again has been propsed. The method, however, is not proved in terms of the additive increase multiplicative decrease (AIMD) principle of TCP congestion control. In this paper, therefore, we analyzed the method in terms of the principle by ns-simulations.