Dimensioning of the LTE access network

This paper proposes efficient analytical models to dimension the necessary transport bandwidths for the Long Term Evolution (LTE) access network satisfying the QoS targets required by different services. In this paper, we consider two major traffic types: elastic traffic and real time traffic. For each type of traffic, individual dimensioning models are proposed for both the S1 interface and the X2 interface. For elastic traffic the dimensioning models are based on the Processor Sharing models; while for real time traffic the dimensioning models are based on the fundamental queuing models. For validating these analytical dimensioning models, a developed LTE system simulation model is used. Extensive simulations are performed for various traffic and network scenarios. The analytical results derived from the proposed dimensioning models are compared with the simulation results. The presented results demonstrate that the proposed analytical models can appropriately estimate the required performances for different service classes and priorities. Hence they are suitable to be used for dimensioning of the LTE access network with different traffic and network conditions.     

Reduction of Outage Probability due to Handover by Mitigating Inter‐cell Interference in Long‐Term Evolution Networks

The burgeoning growth of real‐time applications, such as interactive video and VoIP, places a heavy demand for a high data rate and guarantee of QoS from a network. This is being addressed by fourth generation networks such as Long‐Term Evolution (LTE). But, the mobility of user equipment that needs to be handed over to a new evolved node base‐station (eNB) while maintaining connectivity with high data rates poses a significant challenge that needs to be addressed. Handover (HO) normally takes place at cell borders, which normally suffers high interference. This inter‐cell interference (ICI) can affect HO procedures, as well as reduce throughput. In this paper, soft frequency reuse (SFR) and multiple preparations (MP), so‐called SFRAMP, are proposed to provide a seamless and fast handover with high throughput by keeping the ICI low. Simulation results using LTE‐Sim show that the outage probability and delay are reduced by 24.4% and 11.9%, respectively, over the hard handover method — quite a significant result.     

Optimal rate and power allocation under quality‐of‐service requirements for wireless multihop networks

In wireless networks, real‐time applications have strict QoS requirements for packet delay, packet loss, and reliability. However, most existing work has not considered these QoS metrics when allocating wireless resources so that the QoS requirements of real‐time applications may not be satisfied. To overcome this shortcoming, a rate and power allocation framework incorporating these QoS metrics is first proposed for slow‐fading systems. Second, two distributed algorithms are developed to solve this optimization framework although it is nonconvex and nonseparable. Third, an improved framework is proposed to deal with the rate and power allocation with QoS requirements for fast‐fading systems. It is shown that the fast‐fading state of the network does not need to be considered in this improved framework, and it can be solved using algorithms that are similar to those for the framework of slow‐fading systems. In the end, simulations show that our algorithms converge closely to the globally optimal solution. By comparison with an existing model, simulations also verify the validity of our frameworks on dealing with the rate and power allocation with QoS requirements. Copyright © 2012 John Wiley & Sons, Ltd.     

Modelling multi‐dimensional QoS: some fundamental constraints

In this paper, we model multi‐dimensional QoS in a unified framework, and study some fundamental constraints from the network and the traffic on realizing multiple QoS goals. Multi‐dimensional QoS requirements are quantitatively represented using a QoS region. Based on the theory of effective bandwidths, the framework connects the throughput, the delay, and the loss rate in a uniform formula. Important traffic and network factors, namely, the burst size and the link speed, are involved. With this framework, it is found that the burst size sets hard limit on the QoS region that can be achieved, and that the matching between the link speed and the node processing power can greatly improve the limit. It is also made clear that while pure load imbalance among links does not affect the QoS region, the heterogeneities of burst size or link speed may severely degrade the QoS performance. Applying the theory to real‐time services in differentiated services architecture, we show it provides a useful tool for QoS prediction and network dimensioning. Copyright © 2004 John Wiley & Sons, Ltd.     

Improvement in DRX Power Saving for Non‐real‐time Traffic in LTE

A discontinuous reception (DRX) operation is included in the Long Term Evolution (LTE) system to achieve power saving and prolonged battery life of the user equipment. An improvement in DRX power saving usually leads to a potential increase in the packet delay. An optimum DRX configuration depends on the current traffic, which is not easy to estimate accurately, particularly for non‐real‐time applications. In this paper, we propose a novel way to vary the DRX cycle length, avoiding a continuous estimation of the data traffic when only non‐real‐time applications are running with no active real‐time applications. Because a small delay in non‐real‐time traffic does not essentially impact the user's experience adversely, we deliberately allow a limited amount of delay in our proposal to attain a significant improvement in power saving. Our proposal also improves the delay in service resumption after a long period of inactivity. We use a stochastic analysis assuming an M/G/1 queue to validate this improvement.     

Cross‐layer bandwidth and power allocation for a two‐hop link in wireless mesh network

In this paper, a cross‐layer analytical framework is proposed to analyze the throughput and packet delay of a two‐hop wireless link in wireless mesh network (WMN). It considers the adaptive modulation and coding (AMC) process in physical layer and the traffic queuing process in upper layers, taking into account the traffic distribution changes at the output node of each link due to the AMC process therein. Firstly, we model the wireless fading channel and the corresponding AMC process as a finite state Markov chain (FSMC) serving system. Then, a method is proposed to calculate the steady‐state output traffic of each node. Based on this, we derive a modified queuing FSMC model for the relay to gateway link, which consists of a relayed non‐Poisson traffic and an originated Poisson traffic, thus to evaluate the throughput at the mesh gateway. This analytical framework is verified by numerical simulations, and is easy to extend to multi‐hop links. Furthermore, based on the above proposed cross‐layer framework, we consider the problem of optimal power and bandwidth allocation for QoS‐guaranteed services in a two‐hop wireless link, where the total power and bandwidth resources are both sum‐constrained. Secondly, the practical optimal power allocation algorithm and optimal bandwidth allocation algorithm are presented separately. Then, the problem of joint power and bandwidth allocation is analyzed and an iterative algorithm is proposed to solve the problem in a simple way. Finally, numerical simulations are given to evaluate their performances. Copyright © 2008 John Wiley & Sons, Ltd.     

Energy-efficient and QoS-aware discontinuous reception using a multi-cycle mechanism in 3GPP LTE/LTE-advanced

In the paper we present a novel energy-efficient and Quality-of-Service (QoS)-aware Discontinuous Reception (DRX) scheme by using a multi-cycle mechanism for the Long Term Evolution (LTE) networks. The proposed scheme is capable of improving the power-saving efficiency while simultaneously meeting the specific QoS requirements by dynamically tuning the DRX parameters according to the traffic intensity. It is a table-driven method which off-line pre-establishes the optimal DRX parameters corresponding to the traffic intensities and different QoS requirements such as the packet delays and packet loss rates based on a theoretical analysis approach. At runtime, the DRX parameters of each User Equipment (UE) can be determined optimally with a simple table lookup without spending much computational time and resource. The analytical model is validated against simulation experiments. Based on the analysis results, the power-saving efficiency of our proposed multi-cycle DRX scheme can be effectively boosted by about 37 % comparing to that of the typical DRX scheme while simultaneously meeting the specific QoS requirements like the packet delays and packet loss rates under the scenarios with different traffic intensities.     

QoS‐aware LTE‐A downlink scheduling algorithm: A case study on edge users

4G/LTE‐A (Long‐Term Evolution—Advanced) is the state of the art wireless mobile broadband technology. It allows users to take advantage of high Internet speeds. It makes use of the OFDM technology to offer high speed and provides the system resources both in time and frequency domain. A scheduling algorithm running on the base station holds the allocation of these resources. In this paper, we investigate the performance of existing downlink scheduling algorithms in two ways. First, we look at the performance of the algorithms in terms of throughput and fairness metrics. Second, we suggest a new QoS‐aware fairness criterion, which accepts that the system is fair if it can provide the users with the network traffic speeds that they demand and evaluate the performance of the algorithms according to this metric. We also propose a new QoS‐aware downlink scheduling algorithm (QuAS) according to these two metrics, which increases the QoS‐fairness and overall throughput of the edge users without causing a significant degradation in overall system throughput when compared with other schedulers in the literature.     

Resource allocation algorithm for LTE networks using fuzzy based adaptive priority and effective bandwidth estimation

In this paper, we propose a scheme to allocate resource blocks for the Long Term Evolution (LTE) downlink based on the estimation of the effective bandwidths of traffic flows, where users’ priorities are adaptively computed using fuzzy logic. The effective bandwidth of each user traffic flow that is estimated through the parameters of the adaptive β-Multifractal Wavelet Mode modeling, is used to attain their quality of service (QoS) parameters. The proposed allocation scheme aims to guarantee the QoS parameters of users respecting the constraints of modulation and code schemes (modulation and coding scheme) of the LTE downlink transmission. The proposed algorithm considers the average channel quality and the adaptive estimation of effective bandwidth to decide about the scheduling of available radio resources. The efficiency of the proposed scheme is verified through simulations and compared to other algorithms in the literature in terms of parameters such as: system throughput, required data rate not provided, fairness index, data loss rate and network delay.     

Enabling novel premium service classes in DiffServ over MPLS‐enabled network

Network resources dimensioning and traffic engineering influence the quality in provisioned services required by the Expedited Forwarding (EF) traffic in production networks established through DiffServ over MPLS‐enabled network. By modeling EF traffic flows and the excess of network resources reserved for it, we derive the range of delay values which are required to support these flows at DiffServ nodes. This enables us to develop an end‐to‐end (e2e) delay budget‐partitioning mechanism and traffic‐engineering techniques within a framework for supporting new premium QoS levels, which are differentiated based on e2e delay, jitter and loss. This framework enables ingress routers to control EF traffic flow admission and select appropriate routing paths, with the goal of EF traffic balancing, avoiding traffic congestion and getting the most use out of the available network resources through traffic engineering. As a result, this framework should enable Internet service providers to provide three performance levels of EF service class to their customers provided that their network is DiffServ MPLS TE aware. Copyright © 2008 John Wiley & Sons, Ltd.     

A Multi-link Mechanism for Heterogeneous Radio Networks

Nowadays, smart mobile devices drive the mobile traffic growth rapidly. Most smart mobile devices are equipped with multiple radio network interfaces, such as High Speed Packet Access (HSPA), Long Term Evolution (LTE), and Wi-Fi. Therefore, integration of multiple networks is a viable solution to fulfill traffic offloading and the Quality-of-Service (QoS) requirement of data usage for mobile users. In this paper, we propose a multi-link mechanism to handle the radio network selection and switching between LTE and Wi-Fi networks. A Multi-Link Adaptor (MLA) and a Multi-Connection Manager (MCM) are proposed for the User Equipment (UE) and the core network, respectively, to handle the multi-link mechanism. The applications executed in the UEs do not need to be modified under the proposed approach. The MLA maintains a QoS class table and a routing table for the network selection procedure and uses the GPRS Tunneling Protocol-Control plane (GTP-C) control messages to execute network switching. In the future, we will measure the throughput of the multi-link network and the switch delay between the heterogeneous radio networks.     

Hybrid offset‐time and burst assembly algorithm (H‐OTBA) for delay sensitive applications over optical burst switching networks

Optical burst switching (OBS) is the most favourable switching paradigm for future all‐optical networks. Burst assembly is the first process in OBS and it consists of aggregating clients packets into bursts. Assembled bursts wait for an offset time before being transmitted to their intended destinations. Offset time is used to allow burst control packet reserve required resources prior to burst arrival. Burst assembly process and offset‐time create extra delay in OBS networks. To make OBS suitable for real time applications, this extra latency needs to be controlled. This paper proposes and evaluates a novel offset time and burst assembly scheme to address this issue. Constant bit rate (CBR) traffic that has stringent end‐to‐end delay QoS requirements is used in this study. The proposed scheme is called hybrid offset‐time and burst assembly algorithm (H‐OTBA). The objective of the paper is achieved by controlling maximum burst transfer delay parameters of CBR. The proposed scheme was evaluated via network simulation. Simulation results demonstrate that, H‐OTBA has effectively reduced end‐to‐end delay for CBR traffic compared with current solutions. Copyright © 2014 John Wiley & Sons, Ltd.     

Performance analysis of handover delay and buffer capacity in mobile OpenFlow‐based networks

Software‐defined networking (SDN) is a network concept that brings significant benefits for the mobile cellular operators. In an SDN‐based core network, the average service time of an OpenFlow switch is highly influenced by the total capacity and type of the output buffer, which is used for temporary storage of the incoming packets. In this work, the main goal is to model the handover delay due to the exchange of OpenFlow‐related messages in mobile SDN networks. The handover delay is defined as the overall delay experienced by the mobile node within the handover procedure, when reestablishing an ongoing session from the switch in the source eNodeB to the switch in the destination eNodeB. We propose a new analytical model, and we compare two systems with different SDN switch designs that model a continuous time Markov process by using quasi‐birth–death processes: (1) single shared buffer without priority (model SFB), used for all output ports for both control and user traffic, and (2) two isolated buffers with priority (model priority finite buffering [PFB]), one for control and the other for user plane traffic, where the control traffic is always prioritized. The two proposed systems are compared in terms of total handover delay and minimal buffer capacity needed to satisfy a certain packet error ratio imposed by the link. The mathematical modeling is verified via extensive simulations. In terms of handover delay, the results show that the model PFB outperforms the model SFB, especially for networks with high number of users and high probability of packet‐in messages. As for the buffer dimensioning analysis, for lower arrival rates, low number of users, and low probability of packet‐in messages, the model SFB has the advantage of requiring a smaller buffer size.     

Managing QoS requirements for video streaming: from intra‐node to inter‐node

Streaming video over IP networks has become increasingly popular; however, compared to traditional data traffic, video streaming places different demands on quality of service (QoS) in a network, particularly in terms of delay, delay variation, and data loss. In response to the QoS demands of video applications, network techniques have been proposed to provide QoS within a network. Unfortunately, while efficient from a network perspective, most existing solutions have not provided end‐to‐end QoS that is satisfactory to users. In this paper, packet scheduling and end‐to‐end QoS distribution schemes are proposed to address this issue. The design and implementation of the two schemes are based on the active networking paradigm. In active networks, routers can perform user‐driven computation when forwarding packets, rather than just simple storing and forwarding packets, as in traditional networks. Both schemes thus take advantage of the capability of active networks enabling routers to adapt to the content of transmitted data and the QoS requirements of video users. In other words, packet scheduling at routers considers the correlation between video characteristics, available local resources and the resulting visual quality. The proposed QoS distribution scheme performs inter‐node adaptation, dynamically adjusting local loss constraints in response to network conditions in order to satisfy the end‐to‐end loss requirements. An active network‐based simulation shows that using QoS distribution and packet scheduling together increases the probability of meeting end‐to‐end QoS requirements of networked video. Copyright © 2005 John Wiley & Sons, Ltd.     

Performance of delay‐sensitive traffic in multi‐layered satellite IP networks with on‐board processing capability

In this article, performance of delay‐sensitive traffic in multi‐layered satellite Internet Protocol (IP) networks with on‐board processing (OBP) capability is investigated. With OBP, a satellite can process the received data, and according to the nature of application, it can decide on the transmission properties. First, we present a concise overview of relevant aspects of satellite networks to delay‐sensitive traffic and routing. Then, in order to improve the system performance for delay‐sensitive traffic, specifically Voice over Internet Protocol (VoIP), a novel adaptive routing mechanism in two‐layered satellite network considering the network's real‐time information is introduced and evaluated. Adaptive Routing Protocol for Quality of Service (ARPQ) utilizes OBP and avoids congestion by distributing traffic load between medium‐Earth orbit and low‐Earth orbit layers. We utilize a prioritized queueing policy to satisfy quality‐of‐service (QoS) requirements of delay‐sensitive applications while evading non‐real‐time traffic suffer low performance level. The simulation results verify that multi‐layered satellite networks with OBP capabilities and QoS mechanisms are essential for feasibility of packet‐based high‐quality delay‐sensitive services which are expected to be the vital components of next‐generation communications networks. Copyright © 2007 John Wiley & Sons, Ltd.     

Hub‐polling‐based IEEE 802.11 PCF with integrated QoS differentiation

IEEE 802.11 is one of the most influential wireless LAN (WLAN) standards. Point coordination function (PCF) is its medium access control (MAC) protocol with real‐time traffic (rt‐traffic) quality‐of‐service (QoS) guarantees. In PCF, it is very likely that non‐real‐time traffic (nrt‐traffic) will use the contention free period (CFP) that should be dedicated to traffic having higher priority such as rt‐traffic. Therefore, a modified PCF protocol called MPCF, which is based on hub polling and an integrated QoS differentiation, is presented in this paper. With the integrated QoS differentiation, MPCF can prioritize bandwidth requests according to service classes and QoS requirements. With hub polling, MPCF can reduce the bandwidth for control frames and improve the network throughput. A simple and accurate analytical model is derived and presented in this paper to calculate the system throughput of MPCF. Simulation results show that MPCF protocol is much better than PCF in terms of system capacity and rt‐traffic QoS guarantees. Copyright © 2008 John Wiley & Sons, Ltd.     

A congestion reduction mechanism using D2D cooperative relay for M2M communication in the LTE‐A cellular network

The Long Term Evolution‐advanced cellular network is designed for human‐to‐human communication. When a large number of machine‐to‐machine (M2M) devices are trying to access the network simultaneously, it leads to a low random access (RA) successful rate and high congestion problem, which may cause the waste of radio resources, packet loss, latency, extra power consumption, and the worst, M2M service error. There is an urge to propose an efficient method for M2M communication on the LTE‐A network to resolve the congestion problem. In this paper, we propose a congestion reduction mechanism, which can analyze and model the RA procedure on the Long Term Evolution‐advanced network, to find out the collapse point in the RA procedure and then design a scheme named device‐to‐device cooperative relay scheme to relieve the congestion problem. Meanwhile, this work also adds a relay access barring algorithm to improve performance and an RA resource separation mechanism for human‐to‐human communication. The proposed method can effectively reduce the network congestion problem. Simulation results show that the network throughput and the congestion can be significantly improved using the proposed mechanism. Copyright © 2016 John Wiley & Sons, Ltd.     

An efficient resource allocation to improve QoS of 5G slicing networks using general processor sharing‐based scheduling algorithm

Fifth generation (5G) slicing is an emerging technology for software‐defined networking/network function virtualization–enabled mobile networks. Improving the utilization and throughput to meet the quality of service (QoS) requirements of 5G slicing is very important for the operators of mobile networks. With growing data traffic from different applications of numerous smart mobile devices having several QoS requirements, we expect networks to face problems of congestion and overload that prevent the effective functioning of a radio access network (RAN). This paper proposes a more effective packet‐based scheduling scheme for data traffic by 5G slicing with two operation modes for improving the resource utilization of 5G cloud RAN and providing an efficient isolation of the 5G slices. These two operation modes are referred to as static sharing resource (SSR) scheme and dynamic sharing resources (DSR) scheme. The SSR scheme is a modified version of an existing method. The goal of this approach is to reallocate the shared available resources of 5G network fairly and maximize the utilization of bandwidth while protecting a 5G slice from overwhelming other 5G slices. Throughput and delays of the system model are also discussed to show its performance limits. On the basis of the simulation outcomes, we observed that the proposed DSR scheme outperforms the SSR scheme in terms of provided delay and throughput. In addition, the token bucket parameters together with the assigned capacity weight for each slice can be selected and configured based on the required QoS. Finally, a good estimate for the maximum delay bounds of the slices is provided by the derived theoretical delay bound.     

Network dimensioning at the call level for the always‐on network

It is becoming common for the network to provide always‐on access services, where subscribers are guaranteed that their call requests will never be blocked. This paper studies the call‐level link dimensioning for the always‐on network with single‐class traffic. The call‐level QoS requirement is expressed in terms of the probability of a poor‐quality call, which is the probability that a call experiences packet‐level QoS violation at any time during its duration, as opposed to the probability of blocking in the network with call admission control (CAC). The system is modelled as the M/M/infinite system with finite population and an analytic expression for the probability of a poor‐quality call is derived based on performability analysis. The effects of the call‐level traffic characteristics on the required link resources are studied. It is also shown that the call‐level link dimensioning for the always‐on network needs more link resources than the network with CAC, and the call‐level link dimensioning based on the analytic expression can be used to conservatively dimension the always‐on network with arbitrarily distributed call holding time and inter‐call time. The paper also studies the problem of estimating the call‐level traffic characteristics when the knowledge of call boundaries is not available. Copyright © 2004 John Wiley & Sons, Ltd.