• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.44 no.9
• /
• pp.38-45
• /
• 2007

With the emergence of new applications, packet classification is essential for supporting advanced internet applications, such as network security and QoS provisioning. As the packet classification on multiple-fields is a difficult and time consuming problem, internet routers need to classify incoming packet quickly into flows. In this paper, we present multi-phase packet classification architecture using an internal buffer for fast packet processing. Using internal buffer between address pair searching phase and remained fields searching phases, we can hide latency from the characteristic that search times of source and destination header fields are different. Moreover we guarantee the improvement by using single entry caching. The proposed architecture is easy to apply to different needs owing to its simplicity and generality.

• The KIPS Transactions:PartC
• /
• v.10C no.3
• /
• pp.345-354
• /
• 2003

IP Multicast is required of more little network resources than one in unicast. Furthermore, reliable multicast has been researched for supporting reliability at IP Multicast mechanism. Although these studies are carried out, they only have focused on general data. In other words, in case that realtime packet, they can not support reliability since they do not consider realtime properties such as dependency of interframe and playback in time. Besides, we also request to support scalability because we are based on Mobile IP network together with internet. Thus, we need a mechanism to guarantee reliability and scalability of realtime stream data. In this paper, we propose PER (Preceding Error Recovery) that reflect characteristics of the realtime data, especially for H.323. PER provides scalable reliability because it is based on tree-based multicast basically and helps to support scalable relibility as reducing control packet and recovers stream buffer space from underflow status as soon as possible. PER shows much better scalable and reliable than existing works.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.8A
• /
• pp.918-927
• /
• 2004

WCDMA system is 3rd generation wireless mobile system specified by 3GPP. In WCDMA downlink, two power control schemes are operated. One is inner loop power control operated in every slot. Another is outer loop power control based on one frame time. Base station (BS) can estimate proper transmission power by these two power control schemes. However, because each MS's transmission power makes a severe effect on BS's performance, BS cannot give excessive transmission power to the specific user. 3GPP defined Power Control Dynamic Range (PCDR) to guarantee proper BS's performance. In this paper, we propose Adaptive PCDR algorithm. By APCDR algorithm, Radio Network Controller (RNC) can estimate each MS's current state using received signal to interference ratio (SIR). APCDR algorithm changes MS's maximum code channel power based on frame. By proposed scheme, each MS can reduce wireless channel effect and endure outages in cell edge. Therefore, each MS can obtain better QoS. Simulation result indicate that APCDR algorithm show more attractive output than fixed PCDR algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.9A
• /
• pp.1048-1057
• /
• 2004

WCDMA system is 3rd generation wireless mobile system specified by 3GPP. In WCDMA downlink, two power control schemes are operated. One is inner loop power control operated m every slot Another is outer loop power control based on one frame time. Base staion (BS) can estimate proper transmission power by these two power control schemes. However, because each MS's transmission power makes a severe effect on BS's performance, BS cannot give excessive transmission power to the speclfic user 3GPP defined Power Control Dynamic Range (PCDR) to guarantee proper BS's performance. In this paper, we propose Adaptive PCDR algorithm. By APCDR algorithm, Radio Network Controller (RNC) can estimate each MS's current state using received signal to interference ratio (SIR) APCDR algorithm changes MS's maximum code channel power based on frame. By proposed scheme, each MS can reduce wireless channel effect and endure outages in cell edge. Therefore, each MS can obtain better QoS. Simulation result indicate that APCDR algorithm show more attractive output than fixed PCDR algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.28 no.3A
• /
• pp.129-139
• /
• 2003

CAC is a very important issue in CDMA system in order to protect the required QoS(quality of service) and increase the system's capacity. In this paper, we proposed and analyzed a call admission control scheme using multiple thresholds, which can provide quick processing time and better performance. There are two effective thresholds used to decide call admission. One is the number of active users, and the other is the signal to interference ratio(SIR). If the threshold of active users are lower than the low number of users threshold, we accept the new call without any other conditions. Otherwise, we check the current SIR to guarantee the quality of our service. System then accepts the new call when the SIR satisfies system requirement. Otherwise, the call will be rejected. Multiple threshold schemes are investigated and their performance is compared with the number of user and power based CAC's. simulation results are provided to evaluate the performance.

• Journal of KIISE:Computer Systems and Theory
• /
• v.34 no.10
• /
• pp.529-538
• /
• 2007

Fair Queueing algorithms based on Generalized Processor Sharing (GPS) not only guarantee sessions with service rate and delay, but also provide sessions with instantaneous fair sharing. This fair sharing distributes server capacity to currently backlogged sessions in proportion to their weights without regard to the amount of service that the sessions received in the past. From a long-term perspective, the instantaneous fair sharing leads to a different quality of service in terms of delay and bandwidth to sessions with the same weight depending on their traffic pattern. To minimize such long-term unfairness, we propose a delay-bandwidth normalization model that defines the concept of value of service (VoS) from the aspect of both delay and bandwidth. A model and a packet-by-packet scheduling algorithm are proposed to realize the VoS concept. Performance comparisons between the proposed algorithm and algorithms based on fair queueing and service curve show that the proposed algorithm provides better long-term fairness among sessions and that is more adaptive to dynamic traffic characteristics without compromising its service rate and delay guarantees.

• Journal of KIISE:Software and Applications
• /
• v.35 no.4
• /
• pp.207-218
• /
• 2008

With the growth of the Internet, multimedia streaming becomes an important means to deliver video contents over the Internet and the amount of the streaming multimedia contents is also getting increased. However, it becomes difficult to guarantee the quality of service in real-time over the IP network environment with instantaneously varying bandwidth. In this paper, we propose an optimal adaptation framework for streaming contents over the Internet in the sense that the perceptual quality of the multi-angie content with multiple visual objects is maximized given the constraints such as available bandwidth and transcoding cost. In the multi-angle video service framework, the user can select his/her preferred alternate views among the given multiple video streams captured at different view angles for a same event. This enhanced experience often entails streaming problems in real-time over the network, such as instantaneous bandwidth changes in the Internet. In order to cope with this problem, we assume that multi-angle video contents are encoded at different bitrates and the appropriate video streams are then selected or transcoded for delivery to meet such bandwidth constraints. For the user selective consumption of the various bitstreams in the multi-angle video service, the bitstream in each angle can be encoded in various bitrate, and the user can select a sub-bitrstream in the given bitrstreams or transcode the corresponding content in order to deliver the optimally adapted video contents to the instantaneously changing network condition. Therefore, we define the transcoding cost which means the time taken for transcoding the video stream and formulate a unified optimization framework which maximizes the perceptual quality of the multiple video objects in the given constraints such as the transcoding cost and the network bandwidth. Finally, we present plenty of the experimental results to show the effectiveness of the proposed method.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.8A
• /
• pp.745-751
• /
• 2006

The transmitter requires knowledge of the channel status information in order to adopt the adaptive modulation and coding scheme(AMC) for OFDM system. But in the outdoor environment which the users have high mobility, the channel status information from the users is outdated, so that it induces the degradation of system throughput and packet error rate(PER) performance. To solve this problem, researches about applying channel prediction technique to the AMC scheme have been proceeded. Most channel prediction techniques assume that there is no channel variation in the predefined time duration, e.g., a slot. As a result, those techniques cannot compensate the degradation of PER performance resulting from the rapid variation of channel during the slot duration. This paper introduces a novel channel prediction technique for OFDM/FDD system to support adaptive modulation and coding scheme over rapidly time-varying multipath fading channel. The proposed channel prediction technique considers the time-varying nature of channel during the slot duration. Simulation results show that the AMC scheme of OFDM/FDD system utilizing the proposed channel prediction technique can guarantee the target PER of 1% without any loss of system throughput compared with the case supported by the conventional channel prediction under ITU-R Veh A 30km/h.

• Journal of Broadcast Engineering
• /
• v.16 no.5
• /
• pp.834-845
• /
• 2011

Massive broadcasting contents such as UHD(Ultra High Definition) TV which requires multi-channel capacity for transmission has been introduced in recent years. A transmission scheme with channel bonding has been considered for transmission of massive broadcasting contents. In HFC(Hybrid Fiber Coaxial) networks, DOCSIS 3.0(Data Over Cable Service Interface Specification 3.0) has already applied channel bonding schemes for up/downstream of data service. A method unlike DOCSIS 3.0 is required to introduce a channel bonding scheme in the broadcasting service having unidirectional transmission with a downstream. Since a massive broadcasting content requires several channels for transmission, VBR(Variable Bit Rate) transmission has been emerging for the bandwidth efficiency. In addition, research on channel allocation and resource scheduling is required to guarantee QoS(Quality of Service) for the broadcasting service based on VBR. In this paper, we propose a transmission method for mass broadcasting service in HFC network and show the UHD transmission simulator developed to evaluate the performance. In order to evaluate the performance, we define various scenarios. Using the simulator, we assess the possibility of channel bonding and VBR transmission for UHD broadcasting system to provide mass broadcasting service efficiently. The developed simulator is expected to contribute to the efficient transmission system development of mass broadcasting service.

• 한국정보통신설비학회:학술대회논문집
• /
• 2008.08a
• /
• pp.223-228
• /
• 2008

Recently, various types of wireless access networks, such as WLAN, WiBro and HSDPA, etc, have been successfully deployed by commercial service providers (i.e., KT, KTF). In this situation, there are many efforts to provide high quality of services to guarantee seamless mobility between heterogeneous networks. The IP layer mobility protocols are efficient mechanisms to provide seamless mobility between IP based heterogeneous networks as well as homogeneous networks. However, to apply IP mobility protocols in real heterogeneous networks (i.e., WiBro and HSDPA), we must consider not only the basic features of techniques of wireless access networks (i.e., Data rate, Coverage, Quality of Service) but also the problem of real environment of service provider (i.e., Expanse cost to change the access network). Due to this reason, it is difficult to satisfy required conditions by using only one IP mobility protocol in real heterogeneous networks. Therefore, in this paper, we propose an efficient mobility protocol to solve the complex problems that are occurred in real heterogeneous networks. The proposed protocol, so-called, "Hybrid Mobile IP" tries to provide a synergy effect by integrating Client Mobile IPv4 (CMIPv4) and Proxy Mobile IPv4 (PMIPv4), and using the two mobility protocols selectively according to the situation of real heterogeneous networks.