• 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 Digital Contents Society
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• v.7 no.3
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• pp.207-210
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• 2006

A synchronized multimedia presentation consists of a collection of objects, with each object having one or more rendering intervals within the presentation timeline. These intervals specify the objects' start times and end times relative to the presentation timeline. In this paper we consider the problem of streaming a multimedia presentation from a server to a client over a bandwith-limited communication network. We suppose that each of the static objects is layered-encoded. For a given maximum delay, we consider the problem of finding the optimal number of quality of the presentation. We devise efficient algorithms for determining an optimal policy for several criteria. We also consider the problem of gradual rendering of objects after their start times.

• Journal of KIISE
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• v.44 no.1
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• pp.107-113
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• 2017

Real-time multicasting is a packet transmission scheme ensuring that multiple destinations receive a packet within the desired time line. In wireless sensor networks, a packet can be delivered to a limited distance under a given deadline, since the end-to-end delay tends to be proportional to the end-to-end physical distance. Existing real-time multicasting protocols select the distance between the source and the furthest destination as the distance limitation and construct a multicasting tree guaranteeing delivery paths to each destination within the distance limitation. However, the protocols might lead to real-time delivery failures and energy efficiency degradation due to the fixed distance limitation. In this study, we proposed a real-time multicasting protocol using time deadline. The proposed protocol obtains the maximum transmittable distance with a given time deadline and subsequently constructs a multicasting tree using the maximum transmittable distance. The form of the multicasting tree varies according to the given time deadline to trade off the energy efficiency against the real-time delivery success ratio. The simulation results showed that the proposed scheme is superior to the existing protocols in terms of energy efficiency and real-time delivery success ratio under various time deadlines.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.10
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• pp.1-9
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• 2004

User-Perceived application-level performance is a key to the adoption and success of CDMA 2000. To predict this performance in advance, a detailed end-to-end simulation model of a CDMA network was built to include application traffic characteristics, network architecture, network element details, and protocol features. We assess the user application performance when a Radio Access Network (RAN) and a Core Network (CN) adopt different transport architectures such as ATM and If. For voice Performance, we found that the vocoder bypass scenario shows 8% performance improvement over the others. For data packet performance, we found that HTTP v.1.1 shows better performance than that of HTTP v.1.0 due to the pipelining and TCP persistent connection. We also found that If transport technology is better solution for higher FER environment since the IP packet overhead is smaller than that of ATM for web browsing data traffic, while it shows opposite effect to small size voice packet in RAN architecture. Though simulation results we showed that the 3G-lX EV system gives much better packet delay performance than 3G-lX RTT, the main conclusion is that end-to-end application-level performance is affected by various elements and layers of the network and thus it must be considered in all phases of the technology evolution process.

• The KIPS Transactions:PartC
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• v.11C no.3
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• pp.379-386
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• 2004

The WAP 2.0 system is a newly proposed wireless communication system by the WAP Forum for interoperability across Internet environment and the system takes charge of communication between WAP terminals and existing origin Web servers. The purpose of this paper is 1) to construct a WAP 2.0 proxy proposed by the WAP Forum and 2) to improve the WAP Proxy in order to increase communication efficiency between wired and wireless communication objects. The Improved WAP proxy constructed in this study provides links between wired and wireless communication environments using the split-TCP concept. However, unlike the split-TCP connection, The improved WAP proxy maintains TCP's end-to-end semantics and reduces overhead by avoiding operations as much as possible on the upper protocol layer. In addition, The improved WAP proxy supports SACK(Selective Acknowledgement ) option and Timestamp option for speedy re-transmission which leads to reduction of performance degradation. After constructing the improved WAP proxy under Linux environment, experiments have been taken. The experimental results show that, compared with the experiments when a WAP proxy proposed by the WAP Forum is used, both data transmission delay time and data transmission size decrease to show that communication efficiency is increased. In particular, as packet missing ratio Increases, data transmission size decreases, which demonstrates that the improved WAP proxy is very effective for performance improvement in wireless communication environment.

• The Journal of the Korea Contents Association
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• v.8 no.6
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• pp.43-53
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• 2008

Hyper text transfer protocol (HTTP) over transmission control protocol (TCP) is currently used to transfer objects in the Internet. Stream control transmission protocol (SCTP), an alternative to TCP, which allows for independent delivery among streams, and can thus reduce the mean response delay of web object. We present an analytical model to find the mean response delay for HTTP over SCTP, therefore, estimate the effectiveness of SCTP over TCP. Typical TCP delay models assume the wired environment. On the contrary, the proposed model in this paper assumes the multiple packet losses and wireless environment where fast retransmission is not possible due to small window. The estimated mean response time can be used the benchmark to meet quality of service (QoS) at end-user. We validate the accuracy of our model using experiments. It is shown that the differences between the results from model and those from experimental are very small below 6 % on average. We also find that the mean response delay for HTTP over SCTP is less than that for HTTP over TCP.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.11
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• pp.1-12
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• 2005

In wireless personal area networks(PANs), the successful design of channel time allocation algorithm is a key factor in guaranteeing the various quality of service(QoS) requirements for the stringent real-time constraints of multimedia services. In this paper we propose a dynamic channel time allocation algerian for providing delay guarantees to multimedia traffics such as MPEG streams in the IEEE 802.15.3 high-rate WPANs. The dynamic algorithm exploits the characteristics of MPEG stream, wherein the devices (DEVs) send their channel time requests only at the end of superframe boundaries. The algerian uses mini packets for feedback control in order to deliver dynamic parameters for channel time requests from the DEVs to the piconet coordinator (PNC). In this scheme, the duration of channel time allocated to a DEV during a superframe is changed dynamically depending on the MPEG frame type, traffic load and delay bound of the frame, etc. Performance of the proposed scheme is investigated by simulation. Our results show that compared to traditional scheme, the proposed scheme is very effective and has high performance while guaranteeing the delay bound.

• The Transactions of the Korea Information Processing Society
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• v.4 no.1
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• pp.237-245
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• 1997

In this paper, a flow control mechanism is proposed which is based on the priority control between communication path of a node. In this scheme, demanding length of a data queue for any pre-defined, then each node in that path is forced to maintains buffer size under the limit by controlling priority level of the path. The communication path which requires higher bandwidth sets its demanding queue length smaller. By providing relationship between the priority of a path and length of its queue, the high bandwidth requesting path has a better chance to get high bandwidth by defining the smaller demanding queue size. And also, by forcing a path which has high flow rate to maintain small queue size in the path of the communication, the scheme keep the transmission delay of the path small. The size of the demanding queue of a path is regularly adjusted to meet the applications requirement, and the load status of the network during the life time of the communication. The priority control based on the demanding queue size is also provided in the intermediate nodes as well as the end nodes. By that the flow control can provide a quicker result than end to-end flow control, it provides better performance advantage especially for the high speed network.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.15 no.5
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• pp.71-82
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• 2016

Information System for rear-end collision avoidance is a unit service of C-ITS pilot project. Road environment that the number of vehicles at the same driving high-speed has a possibility that the communication delay or failure caused by heavy load of vehicle to vehicle communication. In this study, effects of the road network about a communication failure rate of information system for rear-end collision avoidance was analyzed quantitatively with micro traffic simulation. The simulation was carried out in situation that crash of two vehicles are occurred at merging area with speed limit 80km/h and information of collision is prvoided to the rear vehicle. From simulation results, it can confirm the trend of the increasing 14% of potential conflict according to 10% increasing of the communication failure rate. C-ITS service has a goal of increasing safety. The coommunication failure rate increases due to heavy load of vehicle causes a fatal result in road safety administrator position. For the success of C-ITS project, a communication system developers side should perform the effort to reduce the communication failure rate.

• The KIPS Transactions:PartC
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• v.10C no.4
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• pp.447-454
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• 2003

Optical burst switching (OBS) has been proposed to reduce the use of fiber delay lines (FDLs) and to realize the optical switching paradigm of the next-generation ail optical networks. The OBS can provide improvements over wavelength routing in terms of bandwidth efficiency and core network scalability via statistical multiplexing of bursts. Recently, another challenging issue is how to upport quality of service (QoS) in the optical burst switching networks. In this paper, we propose a deflection routing scheme to guarantee the QoS for the OBS networks to detour lower priority burst forward to the deflection routing path when congested. A big advantage of the proposed scheme is the simplicity of QoS provision, that comes from the simple QoS provisioning algorithm. Also, the QoS provisioning scheme be able to make efficient networks by fairly traffic distributing with the reduce of the use of FDLs at core routers. The QoS provisioning scheme has been verified to reliably guarantee the QoS of priority 0, 1, 2 burst and to efficiently utilize network resources by computer simulations using OPNET As results, the end-to-end delay of high priority burst is improved, and the network efficiency is also improved.