High-speed Packet Research Articles

Adaptive Modulation and Coding for LTE Wireless Communication

Long Term Evolution (LTE) is the new upgrade path for carrier with both GSM/UMTS networks and CDMA2000 networks. The LTE is targeting to become the first global mobile phone standard regardless of the different LTE frequencies and bands use in other countries barrier. Adaptive Modulation and Coding (AMC) is used to increase the network capacity or downlink data rates. Various modulation types are discussed such as Quadrature Phase Shift Keying (QPSK), Quadrature Amplitude Modulation (QAM). Spatial multiplexing techniques for 4×4 MIMO antenna configuration is studied. With channel station information feedback from the mobile receiver to the base station transmitter, adaptive modulation and coding can be applied to adapt to the mobile wireless channels condition to increase spectral efficiencies without increasing bit error rate in noisy channels. In High-Speed Downlink Packet Access (HSDPA) in Universal Mobile Telecommunications System (UMTS), AMC can be used to choose modulation types and forward error correction (FEC) coding rate.

Network delay modeling for assisted GPS

Currently, conventional Global Positioning System (GPS) receivers work well in open-sky environments. However, location finding in weak-signal conditions, such as urban canyons and indoors, is challenging and has been the subject of extensive research. Assisted GPS (A-GPS) is one of the concepts that helps receivers acquire weak signals by receiving assistance data from wireless networks, such as orbital parameters, and coarse time and location references. A-GPS also improves time-to-first-fix (TTFF) through faster delivery of information that is recovered from navigation data broadcast by GPS satellites. Receiver technology developers rely on signal simulators for testing real-world scenarios. While A-GPS support has been integrated in many simulators, probabilistic models of delays occurring during assistance data delivery have not been studied properly. This paper provides a methodology of A-GPS network delay modeling that is applicable to various simulation environments. Particularly, a testbed is designed to collect delay data, model them statistically and integrate the model in a simulator. The testbed employs secure user plane location (SUPL) architecture in two modes, mobile station (MS) based (MS-based) and MS-assisted, where data channels are used to communicate A-GPS assistance data.Measurement campaigns are conducted and network delay models are derived for various representative distances between assistance servers and receivers, and for various networks that users connect to, such as LAN, WLAN, third-generation mobile telecommunication (3G), high-speed downlink packet access (HSDPA), and fourth-generation long term evolution (4G LTE), with the Transmission Control Protocol and the Internet Protocol (TCP/IP) connection between a server and a receiver.

On the Utilization of Multi-Mode User Equipment in Multi-Radio Access Technology Cellular Communication Systems

Multi-radio access technology (RAT) cellular communication systems limit connected users to utilizing a single RAT even when employing multi-mode user equipment (UE) capable of utilizing multi-RATs. Single-mode access, combined with static spectrum partitioning between co-deployed RATs and independent resource allocation for employed RATs, results in suboptimal spectrum utilization in multi-RAT systems. This paper models user access in multi-RAT systems and proposes enabling multi-mode UE to simultaneously utilize multiple RATs, using multi-RAT carrier aggregation, to improve the performance and spectrum utilization of multi-RAT systems. Several realizations of multi-mode access with varying implementation requirements are presented and discussed. Detailed system-level simulations, for a system co-deploying High Speed Packet Access (HSPA) and Long-Term Evolution (LTE), are performed to investigate the gains and limitations of different user access configurations in multi-RAT systems.

Optimization of Learning Time for Learning-Assisted Rendezvous Channel in Cognitive Radio System

SUMMARY This paper derives the optimal learning time for the learning-assisted rendezvous channel. One problem with the dynamic spectrum access system of cognitive radio is access channel mismatch between two wireless terminals. In the learning-assisted rendezvous channel, before exchanging packets for link connection, the rate of channel occupancy by the other system is estimated within the learning time; it is referred to as the channel occupancy rate (COR). High speed packet exchange is made possible by selecting a low COR channel. However, the optimal learning time and the impact of COR estimation errors have not been clarified yet. This paper analyzes the time to rendezvous channel (TTR), where TTR is the time needed to complete the rendezvous with a certain probability. The results indicate that the learning time and TTR have a concave relationship which means that the optimal learning time can be determined.

Packet Switch Scheduler for Increasing Sending Packet

Recently, the need of the high speed packet switch is increased. The Re-2DRR scheduling algorithm based on 2DRR scheduling algorithm provides high throughput communication on a packet switch. However, computer network is using many cases, that huge data communication, complex, and other. This paper proposes a new method to increase choices in algorithm variation for three specific systems and more easily implementation than Re-2DRR. The effectiveness of the proposed algorithm is shown through simulation studies.

MODELLING THE PERFORMANCE OF CLASS-BASED WEIGHTED FAIR QUEUE USING OPNET

Congestion control and internet network resources management are complex and critical issues in a high-speed packet switch data network, due to the emergence growth of modern multimedia streaming services. The increasing number of computer users in various organizations and institutions of higher learning have spurred a great deal of research on network traffic control. Network administrators are facing the challenges of providing efficient services that can satisfy user requirements. This research study proposed queuing algorithm based on classbased weighted fair queue scheme to complement congestion. Network simulation environment are designed and modeled using OPNET simulation software in-order to overcome the limitation of the traditional queuing approach. Various simulations scenarios are conducted. Analysis comparison with first-in-first-out and priority queue is recorded. And also, various network traffics such as: HTTP, video conferencing and voice applications among others are considered. From the graphical results obtained clearly shows that the overall applications services performances optimize significantly. In terms of the throughput, packet loss and queuing delay, the algorithms performs excellently compared with the FIFO and priority queue. This paper examines the implication of queuing scheduling algorithms on an IP router. It also outlines the effectiveness of the proposed algorithm in managing network resources during the period of congestion.

Exploiting Channel Fragmentation and Aggregation/Bonding to Create Security Vulnerabilities

We address a unique security vulnerability due to spectrum fragmentation, aggregation, and bonding in IEEE 802.22-based dynamic spectrum access (DSA) networks and in Long-Term Evolution (LTE) and Evolved High-Speed Packet Access (HSPA+) networks. Typically, channel fragmentation, aggregation, and bonding have been perceived as a means of enhancing the bandwidth and throughput for the users. However, this could also result in losing orthogonality between the bonded or fragmented spectrum bands. We show this leads to a security vulnerability that can be exploited by an attacker to cause service disruptions. We present an analysis of two types of attacks, i.e., the MAXimum IMPact (MAXIMP) attack, wherein the attackers try to cause maximum service disruptions by transmitting at maximum power, and the MINPOW attack, wherein the attackers transmit at minimum power just to create a targeted level of service disruption. Results indicate that, although the MAXIMP attack can cause up to about 16% loss in the capacity of the system, the MINimum POWer (MINPOW) attack, which is more difficult to detect than the MAXIMP attack, can cause 11%-15% loss in throughput.

Design and Implementation of Low Latency Weighted Round Robin (LL-WRR) Scheduling for High Speed Networks

Today’s wireless broadband networks are required to provide QoS guarantee as well as fairness to different kinds of traffic. Recent wireless standards (such as LTE and WiMAX) have special provisions at MAC layer for differentiating and scheduling data traffic for achieving QoS. The main focus of this paper is concerned with high speed packet queuing/scheduling at central node such as base station (BS) or router to handle network traffic. This paper proposes novel packet queuing scheme termed as Low Latency Weighted Round Robin (LL-WRR) which is simple and effective amendment to weighted round robin (WRR) for achieving low latency and improved fairness. Proposed LL-WRR queue scheduling scheme is implemented in NS-2 considering IEEE 802.16 network [1] with real time video and Constant Bit Rate (CBR) audio traffic connections. Simulation results show improvement obtained in latency and fairness using LL-WRR. The proposed scheme introduces extra complexity of computing coefficient but its overall impact is very small.

A multi-pipeline architecture for high-speed packet classification

In typical algorithmic packet classification methods, the data structure is tailored for the given ruleset. It is common among published algorithmic methods that the worst case number of memory accesses per classification depends on the properties of the ruleset, such as the distribution of the address prefixes and port ranges. As a result, existing methods cannot assure constant classification rate. A novel multi-pipeline architecture for packet classification is presented in this paper. Our method has outstanding performance in both space and time. We incorporate the prefix inclusion coding scheme to achieve outstanding memory efficiency. For rulesets with 10 thousand rules, the storage cost of our method is between 16 and 24.5bytes per rule. The hardware uses fixed-length linear pipelines. Hence, the classification rate is constant regardless of the ruleset properties. To demonstrate the feasibility of our method, the proposed architecture is implemented on a Virtex-6 FPGA and the device can achieve a classification rate of 340million packets per second. Power dissipation of the device is about 1.43W.

Measurement and Modeling of Network Delays for MS-Based A-GPS Assistance Delivery

While conventional global positioning system (GPS) receivers have gained popularity for applications in open-sky environments, their performance degrades in challenging conditions, such as urban canyons and indoors. The assisted GPS (A-GPS) concept enhanced receiver capability to operate in weak signal conditions using terrestrial links to communicate helpful information such as satellite orbital parameters, coarse estimates of time and location, and so forth. The A-GPS mode has been standardized for many wireless networks and is integrated in high-end simulators to support development and testing efforts. As the terrestrial-assistance data channel becomes an important link for the receiver, the statistical modeling of information delivery delays should be addressed to simulate more adequate operation scenarios. This paper addresses this aspect by presenting a delay measurement and modeling approach for Mobile Station Based assistance data. Even though the approach can be applied to other measurement setups, this paper is presented for standardized secure user plane location architecture that exploits data channels of wireless networks to deliver assistance data. Measurement data are collected and modeled for various scenarios of relative deployment of assistance servers and receivers. Delays are modeled for LAN, WLAN, third-generation mobile telecommunication, high-speed downlink packet access, and fourth-generation long-term evolution networks, with the transmission control protocol and the Internet protocol connection between a server and receiver.

Asynchronous vs synchronous input-queued switches

Input-queued (IQ) switches are one of the reference architectures for the design of high-speed packet switches. Classical results in this field refer to the scenario in which the whole switch transfers the packets in a synchronous fashion, in phase with a sequence of fixed-size timeslots, tailored to transport a minimum-size packet. However, for switches with large number of ports and high bandwidth, maintaining an accurate global synchronization and transferring all the packets in a synchronous fashion is becoming more and more challenging. Furthermore, variable size packets (as in the traffic present in the Internet) require rather complex segmentation and reassembly processes and some switching capacity is wasted due to partial filling of timeslots.Thus, we consider a switch able to natively transfer packets in an asynchronous fashion thanks to a simple and distributed packet scheduler. We investigate the performance of asynchronous IQ switches with different queueing architectures (one queue per input and one queue for input–output pairs) and show that, despite their simplicity, their performance is comparable or even better than those of synchronous switches. We highlight the peculiar role of the variation coefficient of the packet length. Finally, for synchronous switches we evaluate the actual bandwidth overhead due to packet segmentation, by considering a large set of traffic traces covering the period 2008–2013. We show that an impressive amount of bandwidth (up to 30%) can be lost due to segmentation, even if the internal cell size is optimally chosen.These results demonstrate the potential interest of the asynchronous approach in the design of high-performance switches.

A Scalable and Modular Architecture for High-Performance Packet Classification

Packet classification is widely used as a core function for various applications in network infrastructure. With increasing demands in throughput, performing wire-speed packet classification has become challenging. Also the performance of today's packet classification solutions depends on the characteristics of rulesets. In this work, we propose a novel modular Bit-Vector (BV) based architecture to perform high-speed packet classification on Field Programmable Gate Array (FPGA). We introduce an algorithm named StrideBV and modularize the BV architecture to achieve better scalability than traditional BV methods. Further, we incorporate range search in our architecture to eliminate ruleset expansion caused by range-to-prefix conversion. The post place-and-route results of our implementation on a state-of-the-art FPGA show that the proposed architecture is able to operate at 100+ Gbps for minimum size packets while supporting large rulesets up to 28 K rules using only the on-chip memory resources. Our solution is ruleset-feature independent , i.e. the above performance can be guaranteed for any ruleset regardless the composition of the ruleset.

통계적 분석을 통한 무선 채널 품질이 사용자 체감 품질에 미치는 영향 분석

무선 접속을 통한 인터넷 서비스가 보편화된 최근 사람이 서비스를 이용하는 과정에서 실제로 느끼는 품질인 사용자 체감 품질(QoE; Quality of Experience)의 중요성이 더욱 강조되고 있는데 사용자 체감 품질은 서비스 품질 (QoS; Quality of Service)와 같이 객관적인 수치화가 불가능하다는 특징이 있다. 유선과는 달리 무선 접속을 통해서 제공되는 IP 서비스는 매우 많은 요인에 의해서 사용자 체감 품질이 영향을 받을 수 있다. 따라서 본 논문에서는 대표적인 무선 접속 서비스인 HSPA (High Speed Packet Access)를 통해서 음성 서비스가 제공될 때 측정 가능한 품질지표를 선정하고 이들에 대한 실측값을 통계적으로 분석하여 서비스 품질과 사용자 체감 품질 지표와의 상관관계를 밝히고자 한다. 분석 결과 RSSI (Received Signal Strength Indicator)와 전송 지연의 상관관계가 매우 높고 그에 이어 전송 지연과 MOS (Mean Opinion Score)와 매우 높은 상관관계를 가짐을 알 수 있었다. As internet services are being provided through a wireless access, the importance of quality of experience (QoE) is stressed that is defined as the quality that indicates user's actual feeling when a service is provided. Unlike quality of service (QoS) that can be expressed as a numerical value, it is difficult to represent QoE in an objective way. If an internet service is serviced over a wireless channel, its QoE can be affected by a number of factors such as fading, mobility and so on. This paper, therefore, attempts to specify the relationship between QoE and QoS by conducting practical measurements for the voice service through 3G high speed packet access (HSPA) access network. Analysing the measured results, it has been shown that received signal strength indicator (RSSI) has a great influence on mean opinion score (MOS) through transmission delay.

Research and Simulation on Wireless Packet Scheduling Scheme Based on High-Speed Packet Access Network

The efficiency of high speed packet access (HSPA) systems is mainly based on Packet Scheduling Scheme which will be the key to influencing on the system performance and ensuring the service quality for the users. In this paper, we introduce a three step packet scheduling scheme to provide quality of services (QoS) guaranteed for high speed uplink packet access. Then, we proposed an improved packet scheduling algorithm based on QoS parameter for mixed service applications in order to meet the actual QoS needs of the different systems. This algorithm calculated the scheduling priority based on the comprehensive factors, including the queuing status of each user, QoS requirements and the sum of speeds on each channel. The simulation results show that the performance efficiency of HSPA systems employed proposed algorithm is superior compared to the conventional PF and M-LWDF algorithm in average packet drop and average cell throughput.

A New High-speed Packet Parsing Architecture

随着新协议的不断涌现和网络速率的迅猛增长，报文解析结构在解析灵活度和解析速率上面临挑战。该文结合流水线设计和二叉trie树查表思想，提出一种应用于路由转发的报文协议解析结构(Parsing Pipeline Architecture for Forwarding, PPAF)，通过构建协议二叉trie树来支持报文协议解析的灵活度，利用硬件多级流水查表提升报文协议解析处理速率，采用节点映射算法解决协议二叉trie树节点到流水线映射过程中存储资源不均衡的问题。基于NetFPGA平台的仿真结果表明，相对于现有的高速解析结构，PPAF在处理速率和资源占用上取得较好的均衡的同时，能够提供基于接口的独立灵活解析能力。

A Probabilistic Framework for Estimating Call Holding Time Distributions

Call holding time (CHT) is a statistical indicator in a cellular network, directly affecting network performance and providing critical insight for the network service provider. CHT distribution estimation literature relies on the classical estimation theory that targets to determine parameters of a function. Hence, related work can be considered as making use of parametric approaches. However, the required assumptions for these approaches may not be correct for obtaining an accurate model. In this paper, we introduce a probabilistic framework for CHT distribution estimation, which makes use of Dirichlet process mixture of lognormal distributions. The purpose of this work is to provide a practical Bayesian inference framework to enable the extraction and identification of user behaviors, which are not available through traditional estimation techniques. The performance of the proposed framework is tested on a large data set that is obtained from a mobile switching center of a cellular network service provider composed of calls from Global System for Mobile Communications (GSM) and High-Speed Packet Access (HSPA) networks. Accuracy of the obtained CHT distributions is verified through several performance tests, showing that all distribution estimates have significance levels of 0.99.

Quasi-distributed Interference Coordination for HSPA HetNet

The heterogeneous network (HetNet) has been discussed in detail in the Long-Term Evolution (LTE) and LTE Advanced standards. However, the standardization of High-Speed Packet Access HetNet (HSPA HetNet) launched by 3GPP is pushing at full steam. Interference coordination (IC), which is responsible for dealing with the interference in the system, remains a subject worthy of investigation in regard to HSPA HetNet. In this paper, considering the network framework of HSPA HetNet, we propose a quasi-distributed IC (QDIC) scheme to lower the interference level in the co-channel HSPA HetNet. Our QDIC scheme is constructed as slightly different energy-efficient non-cooperative games in the downlink (DL) and uplink (UL) scenarios, respectively. The existence and uniqueness of the equilibrium for these games are first revealed. Then, we derive the closed-form best responses of these games. A feasible implementation is finally developed to achieve our QDIC scheme in the practical DL and UL. Simulation results show the notable benefits of our scheme, which can indeed control the interference level and enhance the system performance.

ARQ Optimization for UMTS Networks

Initially developed for use in fixed networks, the ARQ (Automatic Repeat Request) technique was adopted in UMTS systems aiming the QoS improvement for data transmission. However, real time services like VoIP cannot benefits of ARQ, because of the related delay problems in WCDMA systems. With the advent of HSPA (High Speed Packet Access) and the LTE (Long Term Evolution) networks, there was a consequent redefinition of the ARQ functionalities, aiming to become possible to use it for real time services. Delay problems were minimized as a consequence of the facilities introduced for the new methods of ARQ generation. The main objective of this work is the definition of a methodology toward ARQ optimization relatively to the mean power required to transfer the information from the origin to the destination, accordingly to a given error probability.To reduce the amount of the power transfer in the uplink direction is of paramount importance for the minimization of interference levels as well as to economize battery of cellular equipment.Though the treatment here presented is focused to E-DCH (Enhanced Uplink Dedicated Channel) of HSPA systems, the results obtained are applicable to other related networks using ARQ techniques.

Performance Gain Evaluation from High Speed Packet Access Evolution (HSPA+)

This paper presents a study on the HSPA+ technology and the impact it have on today's mobile communication networks. To this end we study how to use features such as MIMO (Multiple Input Multiple Output) and HOM (Higher Order Modulation) and how they can increase the data rate transfer per user. The extracted results show that we can increase the data transmission rate by varying the configuration of antennas and the modulation type, obtaining the maximum theoretical order of 43.8 Mbps and 23 Mbps in Downlink (DL) and Uplink (UL), respectively using 4x4 MIMO and 64-QAM (Quadrature Amplitude Modulation) modulation. These results are obtained based on the direct application or manipulation of existing models and integrated in a simulator implemented for this purpose.

Online log‐likelihood ratio scaling for robust turbo decoding

Optimal iterative log-MAP decoding of turbo codes requires accurate knowledge of the operating signal-to-noise ratio (SNR). However, the SNR information, available at practical decoders for bit-interleaved coded modulation systems, such as the third generation partnership project high-speed packet access and long-term evolution wireless cellular systems, may be inaccurate. In this study, two decoder architectures for improved turbo decoding in the presence of SNR mismatch are proposed. The SNR-mismatch aware turbo decoder selects the decoder which is estimated to have the best performance at the current mismatch, according to the test criterion. The SNR-mismatch compensated turbo decoder provides a more accurate estimation of the noise variance and concurrently scales the channel and the decoder log-likelihood ratios (LLRs) to continue decoding. Two different methods are proposed to find the optimal scaling factors online, one on the symbol level and the other on the bit level. This study shows that online LLR scaling, without prior knowledge about the noise mismatch statistics, can result in near-optimal turbo decoding regardless of the initial SNR mismatch.