• Journal of KIISE:Information Networking
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• v.36 no.6
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• pp.492-504
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• 2009

The delivery of multimedia that efficiently adapts its bit-rate to changing network characteristics and conditions is one of the important challenging tasks in the design of today's real-time multimedia streaming systems such as IPTV, Mobile IPTV and so on. In these work, the primary focus is on network congestion, to improve network stability and inter-protocol fairness. However, these existing works have problems which do not support QoE (Quality of Experience), because they did not consider essential characteristics of contents playback such as the media continuity. In this paper, we propose a novel rate control scheme for improving the QoE of multimedia streaming service in the Internet congestion, called NCAR (Network and Client-Aware Rate control), which is based on network-aware congestion control and client-aware flow control scheme. Network-aware congestion control of the NCAR offers an improving reliability and fairness of multimedia streaming, and reduces the rate oscillation as well as keeping high link utilization. Client-aware flow control of NCAR offers a removing the media discontinuity and a suitable receiver buffer allocation, and provides a good combination of low playback delay. Simulation results demonstrate the effectiveness of our proposed schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.4
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• pp.618-636
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• 1997

DQDB(Distributed Queue Dual Bus) protocol, the IEEE 802.6 standard protocol for metropolitan area networks, does not fully take advantage of the capabilities of dual bus architecture. Although fairness in bandwidth distribution among nodes is improved when using so called the bandwidth balancing mechanism, the protocol requires a considerable amount of time to adjust to changes in the network load. Additionally, the bandwidth balancing mechanism leaves a portion of the available bandwidth unused. In a high-speed backbone network, each node may act as a bridge/ router which connects several LANs as well as hosts. However, Because the existence of high speed LANs becomes commonplace, the congestionmay occur on a node because of the limitation on access rate to the backbone network and on available buffer spaces. to release the congestion, it is desirable to install some congestion control algorithm in the node. In this paper, we propose an efficient congestion control mechanism and fair and waster-free MAC protocol for dual bus network. In this protocol, all the buffers in the network can be shared in such a way that the transmission rate of each node can be set proportional to its load. In other words, a heavily loaded node obtains a larger bandwidth to send the sements so tht the congestion can be avoided while the uncongested nodes slow down their transmission rate and store the incoming segments into thier buffers. this implies that the buffers on the network can be shared dynamically. Simulation results show that the proposed probotol significantly reduces the segment queueing delay of a heavily loaded node and segment loss rate when compared with original DQDB. And it enables an attractive high throughput in the backbone network. Because in the proposed protocol, each node does not send a requet by the segment but send a request one time in the meaning of having segments, the frequency of sending requests is very low in the proposed protocol. so the proposed protocol signigificantly reduces the segment queuing dely. and In the proposed protocol, each node uses bandwidth in proportion to its load. so In case of limitation of available buffer spaces, the proposed protocol reduces segment loss rate of a heavily loaded node. Bandwidth balancing DQDB requires the wastage of bandwidth to be fair bandwidth allocation. But the proposed DQDB MAC protocol enables fair bandwidth without wasting bandwidth by using bandwidth one after another among active nodes.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.8
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• pp.2796-2813
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• 2014

This paper presents a new algorithm called the adaptive logarithmic increase and adaptive decrease algorithm (A-LIAD), which mainly addresses the Round-Trip Time (RTT) fairness problem in satellite networks with a very high propagation delay as an alternative to the current TCP congestion control algorithm. We defined a new increasing function in the fashion of a logarithm depending on the increasing factor ${\alpha}$, which is different from the other logarithmic increase algorithm adopting a fixed value of ${\alpha}$ = 2 leading to a binary increase. In A-LIAD, the ${\alpha}$ value is derived in the RTT function through the analysis. With the modification of the increasing function applied for the congestion avoidance phase, a hybrid scheme is also presented for the slow start phase. From this hybrid scheme, we can avoid an overshooting problem during a slow start phase even without a SACK option. To verify the feasibility of the algorithm for deployment in a high-speed and long-distance network, several aspects are evaluated through an NS-2 simulation. We performed simulations for intra- and interfairness as well as utilization in different conditions of varying RTT, bandwidth, and PER. From these simulations, we showed that although A-LIAD is not the best in all aspects, it provides a competitive performance in almost all aspects, especially in the start-up and packet loss impact, and thus can be an alternative TCP congestion control algorithm for high BDP networks including a satellite network.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.65-75
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• 2005

Measurement of network traffic have shown that the self-similarity is a ubiquitous phenomenon spanning across diverse network environments. In previous work, we have explored the feasibility of exploiting the long-range correlation structure in a self-similar traffic for the congestion control. We have advanced the framework of the multiple time scale congestion control and showed its effectiveness at enhancing performance for the rate-based feedback control. Our contribution is threefold. First, we define a modular extension of the TCP-a function called with a simple interface-that applies to various flavours of the TCP-e.g., Tahoe, Reno, Vegas and show that it significantly improves performance. Second, we show that a multiple time scale TCP endows the underlying feedback control with proactivity by bridging the uncertainty gap associated with reactive controls which is exacerbated by the high delay-bandwidth product in broadband wide area networks. Third, we investigate the influence of the three traffic control dimensions-tracking ability, connection duration, and fairness-on performance.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.9 no.1
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• pp.75-80
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• 2008

Measurement of network traffic have shown that the self-similarity is a ubiquitous phenomenon spanning across diverse network environments. In previous work, we have explored the feasibility of exploiting the long-range correlation structure in a self-similar traffic for the congestion control. We have advanced the framework of the multiple time scale congestion control and showed its effectiveness at enhancing performance for the rate-based feedback control. Our contribution is threefold. First, we define a modular extension of the TCP-a function called with a simple interface-that applies to various flavours of the TCP-e.g., Tahoe, Reno, Vegas and show that it significantly improves performance. Second, we show that a multiple time scale TCP endows the underlying feedback control with proactivity by bridging the uncertainty gap associated with reactive controls which is exacerbated by the high delay-bandwidth product in broadband wide area networks. Third, we investigate the influence of the three traffic control dimensions-tracking ability, connection duration, and fairness-on performance.

• THE INTERNATIONAL COMMERCE & LAW REVIEW
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• v.36
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• pp.111-140
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• 2007

This paper intends to review the qualifications of arbitrator, the disclosure of disqualifications by arbitrator, the challenge grounds of arbitrator, and the challenge procedure of arbitrator under the arbitration laws and rules. There are no provisions for the qualification of arbitrator in the UNCITRAL Model Law on International Commercial Arbitration. Under the UNCITRAL Model Law on person shall be precluded by reason of his nationality from acting as an arbitrators. Under the UNCITRAL Model Law when a person is approached in connection with his possible appointment as an arbitrator, he shall disclose any circumstances likely to give rise to justifiable doubts as to his impartiality or independence. An arbitrator, from the time of his appointment and throughout the arbitral proceedings, shall without delay disclose any such circumstances to the parties. Under the UNCITRAL Model Law an arbitrator may be challenged only if circumstances exist that give rise to justifiable doubts as to his impartiality or independence, or if he does not possess qualifications agreed to by the parties. Under the UNCITRAL Model Law the parties are free to agree on a procedure for challenge an arbitrator. Failing such agreement, a party who intends to challenge an arbitrator shall send a written statement of the reasons for the challenge to the arbitral tribunal within 15 days after becoming aware of the constitution of the arbitral tribunal or any circumstance that give rise to justifiable doubts as to his impartiality or independence. Unless the challenged arbitrator withdraws from his office or the other party agrees to the challenge, the arbitral tribunal shall decide on the challenge. In conclusion, an arbitrator has a responsibility not only to the parties but also to the process of arbitration, and must observe high standards of conduct so that the integrity and must observe high standards of conduct so that the integrity and fairness of the process will be preserved.

• Journal of KIISE:Information Networking
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• v.27 no.1
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• pp.98-107
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• 2000

This paper compares the performance of feedback consolidation algorithms with or without a fast overload indication function which has been recently proposed in the ATM Forum for multicast ABR (Available Bit Rate) services in ATM networks. We use the performance criteria such as the ACR (Allowed Cell Rate) of a source, queue length at a switch, link utilization, fairness, and BRM (Backward Resource Management) cell overhead. Simulation results show that performance of the feedback consolidation algorithms with a fast overload indication function can be improved sign.ificantly than that of algorithms without this function in terms of ramp-down delay, particularly in initial start-up period or under a severe overload situation. The fast overload indication function can be combined with any feedback consolidation algorithm, whereas its performance is highly dependent on an underlying basic feedback consolidation algorithm.

• Journal of KIISE:Computing Practices and Letters
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• v.16 no.7
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• pp.794-798
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• 2010

Assuming that a wireless link is mostly used at the network edge and the wireless NIC driver keeps monitoring the error rate of its link, this paper proposes an enhanced TCP congestion control, TCP-L (TCP Link-Aware). TCP-L predicts true congestion losses occurred inside the wired link area by utilizing the wireless link error rate. As a result, it mitigates performance degradation caused from TCP congestion control actions when segments losses occur in a wireless link. Experimental results show that TCP-L provides better performance and fairness in lossy wireless links than existing TCP congestion control schemes. Our approach utilizing the characteristic of the link at TCP could be well adapted to new wireless environments such as Cognitive Radio and ACK-less IEEE 802.11, where a frame may be delivered with a very long delay or lost in the link.

• The KIPS Transactions:PartC
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• v.13C no.2 s.105
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• pp.203-210
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• 2006

Fair packet scheduling algorithms supporting quality-of-services of real-time multimedia applications can be classified into the following two design schemes in terms of the reference time used in calculating the timestamp of arriving packet: Finish-time Design (FD) and Start-time Design (SD) schemes. Since the former can adjust the latency of a flow with raising the flow's reserved rate, it has been applied to a router for the guaranteed service of the IETF (Internet Engineering Task Force) IntServ model. However, the FD scheme may incur severe bandwidth loss for traffic flows requiring low-rate but strong delay bound such as internet phone. In order to verify the usefulness of the SD scheme based router for the IETF guaranteed service, this paper analyzes and compares two design schemes in terms of bandwidth and payload utilizations. It is analytically proved that the SD scheme is better bandwidth utilization than the FD one, and the simulation result shows that the SD scheme gives better payload utilization by up to 20%.

• Journal of Digital Contents Society
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• v.3 no.1
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• pp.47-57
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• 2002

Recently, wireless networks have been pursuing multimedia data service as voice, data, image, video and various form of data according to development of information communication technology. It guarantees cell delivery delay of real time data in efficient real time multimedia data transfer. Also, it minimizes cell loss rate of non-real time multimedia data. In the wireless ATM, there are based on Asynchronous Transfer Mode(ATM). It implies that there are various service with difficult transmission rates and qualities in the wireless communication network. As a result, it is important to find out the ways to guarantee the Quality of Service(QoS) for each kind of traffic in wireless network. In this thesis, we propose an improved TCRM scheduling algorithms for transmission real-time multimedia data service in wireless ATM Networks. We appear real time multimedia scheduling policy that apply each different method to uplink and downlik to wireless ATM network. It can guarantee QoS requirements for each real time data and non-real time data. It also deals the fairness problem for sharing the scarce wireless resources. We solve fault of TCRM as inefficient problem of non-real data by using arbitrary transmission speed and RB(Reservation Buffer) through VC(Virtual Control) and BS(Base Station).