• Proceedings of the Korean Information Science Society Conference
• /
• 1999.10c
• /
• pp.423-426
• /
• 1999

In offering a statistical end-to-end bandwidth guarantee service, typically called Assured Service, in Differentiated Serviced (Diff-Serv) framework, the biggest issue is its inconsistency. Larger profile TCP flows fail to achieve the guaranteed rate when competing with many smaller profile flows. This phenomenon, which we call "bandwidth skew", stems from the fact that larger profile flows take longer time to recover from the congestion window size backoff after a packet drop. Proposed solutions to this problem, therefore, are focused on modifying the TCP behavior. However, TCP modification is not practicable, mainly due to its large installation base. We look to other mechanisms in the Diff-Serv framework to find more realistic solutions. In particular, we demonstrate that RIO, the de facto standard packet differentiation mechanism used for Assured Service, also contributes to the bandwidth skew. Based on this new finding, we design a modified RIO mechanism called RI+O. RI+O uses OUT queue length in addition to IN and IN+OUT queue length to calculate OUT packet drop probability. We show through extensive simulation that RI+O significantly alleviates the bandwidth skew, expanding the operating regime for Assured Service.d Service.

• 제어로봇시스템학회:학술대회논문집
• /
• 2000.10a
• /
• pp.133-133
• /
• 2000

This paper implements Message Service of TCN or IEC 61375-1. TCN is divided into two services, Variable and Message Service. Variable Service uses the broadcasting method with Source Address, but Message Service uses peer-to-peer method with Destination Address and has OSI 7 Layer. In TCN, interface between Transport and Network Layer has not been defined and Meaning of Packet Pool has not been defined exactly. Therefore, this paper proposes the Implementation method for both the interface between Transport and Network Layer and the packet pool for Message Service of TCN.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.32 no.3A
• /
• pp.261-270
• /
• 2007

The Delay Threshold-based Priority Queueing (DTPQ) scheme has been shown useful for scheduling both real-time (RT) and non-real-time (NRT) service traffic in mobile broadband wireless access (MBWA) systems. The overall system capacity can be maximized subject to their QoS requirement by the DTPQ scheme, which takes the urgency of the RT service into account only when their head-of-line (HOL) packet delays exceed a given delay threshold. In practice, the optimum delay threshold must be configured under the varying service scenarios and a corresponding traffic load, e.g., the number of RT and NRTusers in the system. In this paper, we propose an adaptive version of DTPQ scheme, which updates the delay threshold by taking the urgency and channel conditions of RT service users into account. By evaluating the proposed approach in an orthogonal frequency division multiple access/time division duplex (OFDM/TDD)-based broadband mobile access system, it has been found that our adaptive scheme significantly improves the system capacity as compared to the existing DTPQ scheme with a fixed delay threshold.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.10 no.1
• /
• pp.79-95
• /
• 2016

In order to improve the Quality-of-Service (QoS) of latency sensitive applications in next-generation cellular networks, multi-path is adopted to transmit packet stream in real-time to achieve high-quality video transmission in heterogeneous wireless networks. However, multi-path also introduces two important challenges: out-of-order issue and reordering delay. In this paper, we propose a new architecture based on Software Defined Network (SDN) for flow aggregation and flow splitting, and then design a Multiple Access Radio Scheduling (MARS) scheme based on relative Round-Trip Time (RTT) measurement. The QoS metrics including end-to-end delay, throughput and the packet out-of-order problem at the receiver have been investigated using the extensive simulation experiments. The performance results show that this SDN architecture coupled with the proposed MARS scheme can reduce the end-to-end delay and the reordering delay time caused by packet out-of-order as well as achieve a better throughput than the existing SMOS and Round-Robin algorithms.

• Journal of Communications and Networks
• /
• v.4 no.4
• /
• pp.266-281
• /
• 2002

The General Packet Radio Service (GPRS) is being deployed by GSM network operators world-wide, and promises to provide users with “always-on” data access at bandwidths comparable to that of conventional fixed-wire telephone modems. However, many users have found the reality to be rather different, experiencing very disappointing performance when, for example, browsing the web over GPRS. In this paper, we examine the causes, and show how unfortunate interactions between the GPRS link characteristics and TCP/IP protocols lead to poor performance. A performance characterization of the GPRS link-layer is presented, determined through extensive measurements taken over production networks. We present measurements of packet loss rates, bandwidth availability, link stability, and round-trip time. The effect these characteristics have on TCP behavior are examined, demonstrating how they can result in poor link utilization, excessive packet queueing, and slow recovery from packet losses. Further, we show that the HTTP protocol can compound these issues, leading to dire WWW performance. We go on to show how the use of a transparent proxy interposed near the wired-wireless border can be used to alleviate many of these performance issues without requiring changes to either client or server end systems.

• Journal of the Institute of Electronics Engineers of Korea SC
• /
• v.43 no.1 s.307
• /
• pp.7-12
• /
• 2006

In this paper we proposed a real-time remote monitoring system for interoperability between local area and wide area for wireless data communication. In local area, we used a miniaturized low-power wireless module and in wide area used CDMA Cellular System's Packet Data Service. The measurement results can be spread via Internet access in real-time

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.14 no.2
• /
• pp.473-494
• /
• 2020

To provide a guaranteed Quality of Service (QoS) to real-time traffic in High-Speed Downlink Packet Access (HSDPA) core network, we proposed an enhanced mechanism. For an enhanced QoS, a Class-Based Low Latency Fair Queueing (CBLLFQ) packet scheduling algorithm is introduced in this work. Packet classification, metering, queuing, and scheduling using differentiated services (DiffServ) environment was the points in focus. To classify different types of real-time voice and multimedia traffic, the QoS provisioning mechanisms use different DiffServ code points (DSCP).The proposed algorithm is based on traffic classes which efficiently require the guarantee of services and specified level of fairness. In CBLLFQ, a mapping criterion and an efficient queuing mechanism for voice, video and other traffic in separate queues are used. It is proved, that the algorithm enhances the throughput and fairness along with a reduction in the delay and packet loss factors for smooth and worst traffic conditions. The results calculated through simulation show that the proposed calculations meet the QoS prerequisites efficiently.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.16 no.6
• /
• pp.1229-1234
• /
• 2012

In this paper, we analyze the problems that the DDS middleware, which is a standard data-centric communication interface, uses the fixed packet transmission method by the pre-defined protocol for exchanging data packets. The packet transmission method selected in a fixed manner cannot handle appropriately the increasing of resource overhead in an environment where the load of the DDS network changes dynamically. If the load on the node and network exceeds the threshold, the performance of the packet transmission may be degraded rapidly. This results in a failure of ensuring the real-time characteristic of DDS middleware. To solve this problem, we propose the scheme of the adaptive packet transmission for adjusting the transmission method in real-time based on the overhead on the DDS network.

• 한국정보통신설비학회:학술대회논문집
• /
• 2009.08a
• /
• pp.275-280
• /
• 2009

Femtocells are viewed as a promising option for mobile operators to improve coverage and provide high-data-rate services in a cost-effective manner Femtocells can be used to serve indoor users, resulting in a powerful solution for ubiquitous indoor and outdoor coverage. TThe frequency accuracy and phase alignment is necessary for ensuring the quality of service (QoS) forapplications such as voice, real-time video, wireless hand-off, and data over a converged access medium at the femtocell. But, the GPS has some problem to be used at the femtocell, because it is difficult to set-up, depends on the satellite condition, and very expensive. The IEEE 1588 specification provides a low-cost means for clock synchronisation over a broadband Internet connection. The Time of Packet (ToP) specified in IEEE 1588 is able to synchronize distributed clocks with an accuracy of less than one microsecond in packet networks. However, the timing synchronization over packet switched networks is a difficult task because packet networks introduce large and highly variable packet delays. This paper proposes an enhanced filter algorithm to reduce ths packet delay variation effects and maintain ToP slave clock synchronization performance. The results are presented to demonstrate in the intra-networks and show the improved performance case when the efficient ToP filter algorithm is applied.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.23 no.8
• /
• pp.2090-2095
• /
• 1998

In this paper, we measured voice packet loss over the Korean Internet and analyzed percentage articulation by variation of the packet loss. To do this, we reviewed real-time transmission service based on RTP/UDP/IP and test method of the transmission quality. and implemented the real-time speech transmission system using GSM and UDP/IP. Monosyllable list has been chosen for the percentage articulation test, each voice packet has been coded and compressed by GSM and it has sequence number to measured packet loss and to recover out-of-order packets. In transmission results using seven router over the Korean Internet, we have show that loss rates reached 1.6% (unload), 22.5%(load) and loss rates after packet recovery by resequencing and FEC are from 9% to 35%. Finally, we have shown that percentage articulations by variation of the network traffic are Table 4.