• Journal of Navigation and Port Research
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• v.27 no.3
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• pp.283-288
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• 2003

Recently, the growth of applications and services over high-speed Internet increases, ATM networks as wide area back-bone has been a major solution. As the conventional TCP/IP suite is still the standard protocol used to support upper application on current. Internet, the issues regarding whether TCP/IP will operate efficiently on top of an ATM infrastructure and how to control its QoS still remain for studies. TCP uses a window-based protocol for flow control in the transport layer. When TCP uses the UBR service in ATM layer, the control method is only buffer management. If a cell is discarded in ATM layer, one whole packet of TCP will be lost; this fact occur the most TCP performance degradation. Several dropping strategies, such as Tail Drop, EPD, PPD, SPD, FBA, have been proposed to improve the TCP performance over ATM. In this paper, to improve the TCP performance, we propose a packet dropping scheme that is based on comparison with EPD, SPD and FBA. Our proposed scheme is applied to schemes discussed in the previous technology. Our proposed scheme does not need to know each connection's mean packet size. When the buffer exceeds the given threshold, it is based on comparison between the number of dropped packet and the approved packet. Our results are reported and discussed for comparing these discarding schemes under similar conditions. Although the number of virtual channel (VC) is increased, the simulation results showed that the proposed scheme can allocate more fairly each VC than other scheme.

• Journal of Internet Computing and Services
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• v.5 no.5
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• pp.39-48
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• 2004

For transporting real-time multimedia traffic, RTP is considered as one of the most promising protocols operating at application layer. In order to manage and control the real-time multimedia traffic within networks, network managers need to monitor and analyze the traffic delivering through their networks. However, conventional traffic analyzing tools can not exactly classify and analyze the real-time multimedia traffic using RTP on the basis of real-time as well as non-real-time operations. In this paper, we propose an efficient online classification method of RTP packets, which can be used on high-speed network links. The accuracy and efficiency of the proposed methodhave been tested using captured data from a KIX node with 100 Mbps links, which interconnects between domestic and overseas Internet networks and is operated by NCA.

• Journal of Korea Multimedia Society
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• v.12 no.5
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• pp.653-663
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• 2009

Synchronization between media is an important aspect in the design of multimedia communication-system. This paper proposes a precise media synchronization mechanism for video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio. In the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.1
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• pp.414-435
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• 2017

As network adaptive streaming technology becomes increasingly common, transport protocol also becomes important in guaranteeing the quality of multimedia streaming. At the same time, because of the appearance of high-quality video such as Ultra High Definition (UHD), preventing buffering as well as preserving high quality while deploying a streaming service becomes important. The Internet Engineering Task Force recently published Multipath TCP (MPTCP). MPTCP improves the maximum transmission rate by simultaneously transmitting data over different paths with multiple TCP subflows. However, MPTCP cannot preserve high quality, because the MPTCP subflows slowly increase the transmission rate, and upon detecting a packet loss, drastically halve the transmission rate. In this paper, we propose a new multipath congestion control scheme for high-quality multimedia streaming. The proposed scheme preserves high quality of video by adaptively adjusting the increasing parameter of subflows according to the network status. The proposed scheme also increases network efficiency by providing load balancing and stability, and by supporting fairness with single-flow congestion control schemes.

• Journal of Institute of Control, Robotics and Systems
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• v.18 no.8
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• pp.774-779
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• 2012

We design and implement a RTT (Round Trip Time) based TCP (Transmission Control Protocol) for USN (Ubiquitous Sensor Network). We adopt a basic update algorithm for window size from FAST TCP that uses the queuing delay at link as the congestion measure. The designed TCP estimates the queuing delay at link from the measured RTT in the network layer, and updates the window size based on the estimated queuing delay. The designed TCP allows to utilize the full capacity of USN links and avoids the waste of the given link capacity that is common without the flow control in the transport layer. The experiment results show that the window size of the sender converges within a small range of variations without any packet loss, and verify the stability and performance of the designed TCP.

• 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.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.2
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• pp.114-124
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• 2009

In this paper, we propose a practical TCP friendly rate control scheme that considers the minimum channel bandwidth of the network when transporting SVC (scalable video coding) video over IP netowrks such as Internet. RTP and RTCP is mainly designed for use with UDP (User Datagram Protocol) for real-time video transport over the Internet. TCP-friendly rate control was proposed to satisfy the demands of multimedia applications while being reasonably fair when competing for bandwidth with conventional TCP applications. However the rate control model of the conventional TCP-friendly rate control scheme does not consider the minimum channel bandwidth of the network. Thus the estimated channel bandwidth by the conventional rate control model might be quite different from the real channel bandwidth when the packet loss ratio of the network is very large. In this paper, we propose a modified TCP-friendly rate control scheme that considers the minimum channel bandwidth of the network. Based on the modified TCP-friendly rate control, we assign the minimum channel bandwidth to the base layer bitstream of SVC video, and remaining available bandwidth is allocated to the enhancement layer of SVC video for the TCP friendly scalable video transmission. It is shown by simulations that the modified TCP-friendly rate control scheme can be effectively used for a wider range of controlled bit rates depending on the packet loss ratio than the conventional TCP-friendly control scheme. Furthermore, the effectiveness of the proposed scheme in terms of objective video quality is proved by comparing PSNR performance with the conventional scheme.

• Convergence Security Journal
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• v.13 no.3
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• pp.25-31
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• 2013

Looking at the recent value change of users and the usage pattern of network users, it is changing from simple web information, one-way information acquisition and data transmission to increase of usage of multimedia, increasing demand for security and customization, and increasing demands for free mobility. Due to this change of demand, the services which were provided individually, developed into a form which is merged, the network also seems to develop into the combined network from the individual network for individual service, and the communication network control technology which is the core technology is also rapidly developing. To reflect the users' demands, the next generation network created the multi-layer network which is based on the WDM/IP transmission system and added the to make it easy to restructure. P-OTS (Packet-Optical Transport System) can be defined as a platform that combines SONET/SDH, Ethernet, DWDM, optical transport network (OTN) switching and reconfigurable optical add-drop multiplexers (ROADMs). In this paper, we suggested that the optimum path choice be performed through diversification of the PCE-based path selection using the information of various layers altogether in the multi-layer environment, compared with the established path selection method when the path was selected using the information of each layers.

• Journal of Korea Multimedia Society
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• v.12 no.11
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• pp.1593-1608
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• 2009

If a certain relay node in multi-hop wireless networks might become a malicious node that does not cooperate with other nodes or a selfish node, network throughput will be dramatically decreased. Most of existing ad hoc routing protocols assuming that the nodes will fully cooperate with other nodes do not resolve the problem of network performance degradation due to malicious and selfish nodes. This paper presents the CARE (Cooperative Ad hoc routing protocol based REputation) scheme incorporating the reputation management that can achieve a multi-hop wireless network with high throughput performance. The proposed scheme provides the horizontal cross-layer approach which can identify misbehaving malicious, selfish nodes dropped out of the hop-by-hop based packet processing in the network and then set up an optimal packet routing path that will detour misbehaving nodes. And the vertical cross-layer approach contained in the CARE scheme attempts to improve the quality of routing paths by exploiting the quality of link information received from the MAC layer. Besides, it provides high TCP throughput by exploiting the reputation values of nodes acquired from the network layer into the transport layer. A case study on experiments and simulations shows that the CARE scheme incorporating vertical and horizontal cross-layer approaches yields better performance in terms of the low rate of packet loss, fast average packet delivery time, and high TCP throughput between end-to-end nodes.

• The KIPS Transactions:PartC
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• v.12C no.6 s.102
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• pp.781-792
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• 2005

This study proposes an Architecture for user level End-to-end Quality of Service(QoS) using overlay In Next Generation Network(NGN). Inexisting NGNs, the IMS of a control plane provides user QoS through direct traffic control and resource-reservation over the IP packet transport network of a user plane. Further, a set of torrent studies are ongoing not only to maximize the QoS for users, but also to minimize the quality deterioration for supporting the user End-to-end QoS. Along with that, an extended QoS in user level must be considered, for Instance, differentiating service quality to support users' expectation, providing optimized contents by users' equipments, and so forth. Accordingly, the Overlay Service Network Architecture proposed by this study provides protocol adaptation for maximum throughput on transport layer by using the most efficient transport layer protocol to various network circumstances. Also, the Overlay Service Network Architecture on application layer distributes processing delay from the data transformation process of the user equipment to the network, and it is capable of intermediate processing depending on user service level. application service feature, and equipment circumstance as well. Thus, this study mainly proposes the Overlay Service Network Architecture for user level end-to-end QoS in NGN with the quality control features both on the transport layer and the application layer, an internal component feature, and a service scenario providing the QoS linking with 3GPP.