Improved Handover Through Dual Connectivity in 5G mmWave Mobile Networks

The millimeter wave (mmWave) bands offer the possibility of orders of magnitude greater throughput for fifth generation (5G) cellular systems. However, since mmWave signals are highly susceptible to blockage, channel quality on any one mmWave link can be extremely intermittent. This paper implements a novel dual connectivity protocol that enables mobile user equipment (UE) devices to maintain physical layer connections to 4G and 5G cells simultaneously. A novel uplink control signaling system combined with a local coordinator enables rapid path switching in the event of failures on any one link. This paper provides the first comprehensive end-to-end evaluation of handover mechanisms in mmWave cellular systems. The simulation framework includes detailed measurement-based channel models to realistically capture spatial dynamics of blocking events, as well as the full details of MAC, RLC and transport protocols. Compared to conventional handover mechanisms, the study reveals significant benefits of the proposed method under several metrics.Comment: 16 pages, 13 figures, to appear on the 2017 IEEE JSAC Special Issue on Millimeter Wave Communications for Future Mobile Network

Quality of service optimization of multimedia traffic in mobile networks

Mobile communication systems have continued to evolve beyond the currently deployed Third Generation (3G) systems with the main goal of providing higher capacity. Systems beyond 3G are expected to cater for a wide variety of services such as speech, data, image transmission, video, as well as multimedia services consisting of a combination of these. With the air interface being the bottleneck in mobile networks, recent enhancing technologies such as the High Speed Downlink Packet Access (HSDPA), incorporate major changes to the radio access segment of 3G Universal Mobile Telecommunications System (UMTS). HSDPA introduces new features such as fast link adaptation mechanisms, fast packet scheduling, and physical layer retransmissions in the base stations, necessitating buffering of data at the air interface which presents a bottleneck to end-to-end communication. Hence, in order to provide end-to-end Quality of Service (QoS) guarantees to multimedia services in wireless networks such as HSDPA, efficient buffer management schemes are required at the air interface. The main objective of this thesis is to propose and evaluate solutions that will address the QoS optimization of multimedia traffic at the radio link interface of HSDPA systems. In the thesis, a novel queuing system known as the Time-Space Priority (TSP) scheme is proposed for multimedia traffic QoS control. TSP provides customized preferential treatment to the constituent flows in the multimedia traffic to suit their diverse QoS requirements. With TSP queuing, the real-time component of the multimedia traffic, being delay sensitive and loss tolerant, is given transmission priority; while the non-real-time component, being loss sensitive and delay tolerant, enjoys space priority. Hence, based on the TSP queuing paradigm, new buffer managementalgorithms are designed for joint QoS control of the diverse components in a multimedia session of the same HSDPA user. In the thesis, a TSP based buffer management algorithm known as the Enhanced Time Space Priority (E-TSP) is proposed for HSDPA. E-TSP incorporates flow control mechanisms to mitigate congestion in the air interface buffer of a user with multimedia session comprising real-time and non-real-time flows. Thus, E-TSP is designed to provide efficient network and radio resource utilization to improve end-to-end multimedia traffic performance. In order to allow real-time optimization of the QoS control between the real-time and non-real-time flows of the HSDPA multimedia session, another TSP based buffer management algorithm known as the Dynamic Time Space Priority (D-TSP) is proposed. D-TSP incorporates dynamic priority switching between the real-time and non-real-time flows. D-TSP is designed to allow optimum QoS trade-off between the flows whilst still guaranteeing the stringent real-time component’s QoS requirements. The thesis presents results of extensive performance studies undertaken via analytical modelling and dynamic network-level HSDPA simulations demonstrating the effectiveness of the proposed TSP queuing system and the TSP based buffer management schemes

Some aspects of traffic control and performance evaluation of ATM networks

The emerging high-speed Asynchronous Transfer Mode (ATM) networks are expected to integrate through statistical multiplexing large numbers of traffic sources having a broad range of statistical characteristics and different Quality of Service (QOS) requirements. To achieve high utilisation of network resources while maintaining the QOS, efficient traffic management strategies have to be developed. This thesis considers the problem of traffic control for ATM networks. The thesis studies the application of neural networks to various ATM traffic control issues such as feedback congestion control, traffic characterization, bandwidth estimation, and Call Admission Control (CAC). A novel adaptive congestion control approach based on a neural network that uses reinforcement learning is developed. It is shown that the neural controller is very effective in providing general QOS control. A Finite Impulse Response (FIR) neural network is proposed to adaptively predict the traffic arrival process by learning the relationship between the past and future traffic variations. On the basis of this prediction, a feedback flow control scheme at input access nodes of the network is presented. Simulation results demonstrate significant performance improvement over conventional control mechanisms. In addition, an accurate yet computationally efficient approach to effective bandwidth estimation for multiplexed connections is investigated. In this method, a feed forward neural network is employed to model the nonlinear relationship between the effective bandwidth and the traffic situations and a QOS measure. Applications of this approach to admission control, bandwidth allocation and dynamic routing are also discussed. A detailed investigation has indicated that CAC schemes based on effective bandwidth approximation can be very conservative and prevent optimal use of network resources. A modified effective bandwidth CAC approach is therefore proposed to overcome the drawback of conventional methods. Considering statistical multiplexing between traffic sources, we directly calculate the effective bandwidth of the aggregate traffic which is modelled by a two-state Markov modulated Poisson process via matching four important statistics. We use the theory of large deviations to provide a unified description of effective bandwidths for various traffic sources and the associated ATM multiplexer queueing performance approximations, illustrating their strengths and limitations. In addition, a more accurate estimation method for ATM QOS parameters based on the Bahadur-Rao theorem is proposed, which is a refinement of the original effective bandwidth approximation and can lead to higher link utilisation

Performance analysis of error recovery and congestion control in high-speed networks

In the past few years, Broadband Integrated Services Digital Network (B-ISDN) has received increasing attention as a communication architecture capable of supporting multimedia applications. Among the techniques proposed to implement B-ISDN, Asynchronous Transfer Mode (ATM) is considered to be the most promising transfer technique because of its efficiency and flexibility.In ATM networks, the performance bottleneck of the network, which was once the channel transmission speed, is shifted to the processing speed at the network switching nodes and the propagation delay of the channel. This shift is because the high-speed channel increases the ratio of processing time to packet transmission time and also the ratio of propagation delay to packet transmission time. The increased processing overhead makes it difficult to implement hop-by-hop schemes, which may impose prohibitably high processing at each switching node. The increased propagation delay overhead makes traffic control in ATM a challenge since a large number of packets can be in transit between two ATM switching nodes. Because of these fundamental changes, control schemes developed for traditional networks may not perform efficiently, and thus, new network architectures (congestion control schemes, error control schemes, etc.) are required in ATM networks.In this dissertation, we first present an extensive survey of various traffic control schemes and network protocols for ATM networks. In this survey, possible traffic control schemes are examined, and problems of those schemes and their possible solutions are presented. Next, we investigate two key research issues in ATM networks (and other types of high-speed networks): the effects of protocol-processing overhead and the efficiency of traffic control schemes.We first investigate the effects of protocol-processing overhead on the performance of error recovery schemes. Specifically, we investigate the performance trade-offs between link-by-link and edge-to-edge error recovery schemes. Our results show that for a network with high-speed/low-error-rate channels, an edge-to-edge scheme gives a smaller delay than a link-by-link scheme. We then investigate the effectiveness of a priority packet discarding scheme, a congestion control mechanism suitable for high-speed networks. We derive loss probabilities for each stream and investigate the impact of burstiness of traffic streams on the performance of individual streams

Enhanced adaptive RTCP-based inter-destination multimedia synchronization approach for distributed applications

[EN] Newer social multimedia applications, such as Social TV or networked multi-player games, enable independent groups (or clusters) of users to interact among themselves and share services within the context of simultaneous media content consumption. In such scenarios, concurrently synchronized playout points must be ensured so as not to degrade the user experience on such interaction. We refer to this process as Inter-Destination Multimedia Synchronization (IDMS). This paper presents the design, implementation and evaluation of an evolved version of an RTCP-based IDMS approach, including an Adaptive Media Playout (AMP) scheme that aims to dynamically and smoothly adjust the playout timing of each one of the geographically distributed consumers in a specific cluster if an allowable asynchrony threshold between their playout states is exceeded. For that purpose, we previously had also to develop a full implementation of RTP/RTCP protocols for NS-2, in which we included the IDMS approach as an optional functionality. Simulation results prove the feasibility of such IDMS and AMP proposals, by adopting several dynamic master reference selection policies, to maintain an overall synchronization status (within allowable limits) in each cluster of participants, while minimizing the occurrence of long-term playout discontinuities (such as skips/pauses) which are subjectively more annoying and less tolerable to users than small variations in the media playout rate.This work has been financed, partially, by Universitat Politecnica de Valencia (UPV), under its R&D Support Program in PAID-05-11-002-331 Project and in PAID-01-10. Authors also would like to thank the anonymous reviewers that helped to significantly improve the quality of the paper with their constructive comments.Montagud, M.; Boronat, F. (2012). Enhanced adaptive RTCP-based inter-destination multimedia synchronization approach for distributed applications. Computer Networks. 56(12):2912-2933. https://doi.org/10.1016/j.comnet.2012.05.00329122933561

Traffic and task allocation in networks and the cloud

Communication services such as telephony, broadband and TV are increasingly migrating into Internet Protocol(IP) based networks because of the consolidation of telephone and data networks. Meanwhile, the increasingly wide application of Cloud Computing enables the accommodation of tens of thousands of applications from the general public or enterprise users which make use of Cloud services on-demand through IP networks such as the Internet. Real-Time services over IP (RTIP) have also been increasingly significant due to the convergence of network services, and the real-time needs of the Internet of Things (IoT) will strengthen this trend. Such Real-Time applications have strict Quality of Service (QoS) constraints, posing a major challenge for IP networks. The Cognitive Packet Network (CPN) has been designed as a QoS-driven protocol that addresses user-oriented QoS demands by adaptively routing packets based on online sensing and measurement. Thus in this thesis we first describe our design for a novel ``Real-Time (RT) traffic over CPN'' protocol which uses QoS goals that match the needs of voice packet delivery in the presence of other background traffic under varied traffic conditions; we present its experimental evaluation via measurements of key QoS metrics such as packet delay, delay variation (jitter) and packet loss ratio. Pursuing our investigation of packet routing in the Internet, we then propose a novel Big Data and Machine Learning approach for real-time Internet scale Route Optimisation based on Quality-of-Service using an overlay network, and evaluate is performance. Based on the collection of data sampled each $2$ minutes over a large number of source-destinations pairs, we observe that intercontinental Internet Protocol (IP) paths are far from optimal with respect to metrics such as end-to-end round-trip delay. On the other hand, our machine learning based overlay network routing scheme exploits large scale data collected from communicating node pairs to select overlay paths, while it uses IP between neighbouring overlay nodes. We report measurements over a week long experiment with several million data points shows substantially better end-to-end QoS than is observed with pure IP routing. Pursuing the machine learning approach, we then address the challenging problem of dispatching incoming tasks to servers in Cloud systems so as to offer the best QoS and reliable job execution; an experimental system (the Task Allocation Platform) that we have developed is presented and used to compare several task allocation schemes, including a model driven algorithm, a reinforcement learning based scheme, and a ``sensible’’ allocation algorithm that assigns tasks to sub-systems that are observed to provide lower response time. These schemes are compared via measurements both among themselves and against a standard round-robin scheduler, with two architectures (with homogenous and heterogenous hosts having different processing capacities) and the conditions under which the different schemes offer better QoS are discussed. Since Cloud systems include both locally based servers at user premises and remote servers and multiple Clouds that can be reached over the Internet, we also describe a smart distributed system that combines local and remote Cloud facilities, allocating tasks dynamically to the service that offers the best overall QoS, and it includes a routing overlay which minimizes network delay for data transfer between Clouds. Internet-scale experiments that we report exhibit the effectiveness of our approach in adaptively distributing workload across multiple Clouds.Open Acces

Per-Priority Flow Control (Ppfc) Framework For Enhancing Qos In Metro Ethernet

Day by day Internet communication and services are experiencing an increase in variety and quantity in their capacity and demand. Thus, making traffic management and quality of service (QoS) approaches for optimization of the Internet become a challenging area of research; meanwhile flow control and congestion control will be considered as significant fundamentals for the traffic control especially on the high speed Metro Ethernet. IEEE had standardized a method (IEEE 802.3x standard), which provides Ethernet Flow Control (EFC) using PAUSE frames as MAC control frames in the data link layer, to enable or disable data frame transmission. With the initiation of Metro Carrier Ethernet, the conventional ON/OFF IEEE 802.3x approach may no longer be sufficient. Therefore, a new architecture and mechanism that offer more flexible and efficient flow and congestion control, as well as better QoS provisioning is now necessary

Dynamic bandwidth allocation in ATM networks

Includes bibliographical references.This thesis investigates bandwidth allocation methodologies to transport new emerging bursty traffic types in ATM networks. However, existing ATM traffic management solutions are not readily able to handle the inevitable problem of congestion as result of the bursty traffic from the new emerging services. This research basically addresses bandwidth allocation issues for bursty traffic by proposing and exploring the concept of dynamic bandwidth allocation and comparing it to the traditional static bandwidth allocation schemes

Towards An Optimal Core Optical Network Using Overflow Channels

This dissertation is based on a traditional circuit switched core WDM network that is supplemented by a pool of wavelengths that carry optical burst switched overflow data. These overflow channels function to absorb channel overflows from traditional circuit switched networks and they also provide wavelengths for newer, high bandwidth applications. The channel overflows that appear at the overflow layer as optical bursts are either carried over a permanently configured, primary light path, or over a burst-switched, best-effort path while traversing the core network. At every successive hop along the best effort path, the optical bursts will attempt to enter a primary light path to its destination. Thus, each node in the network is a Hybrid Node that will provide entry for optical bursts to hybrid path that is made of a point to point, pre-provisioned light path or a burst switched path. The dissertation's main outcome is to determine the cost optimality of a Hybrid Route, to analyze cost-effectiveness of a Hybrid Node and compare it to a route and a node performing non-hybrid operation, respectively. Finally, an example network that consists of several Hybrid Routes and Hybrid Nodes is analyzed for its cost-effectiveness. Cost-effectiveness and optimality of a Hybrid Route is tested for its dependency on the mean and variance of channel demands offered to the route, the number of sources sharing the route, and the relative cost of a primary and overflow path called path cost ratio. An optimality condition that relates the effect of traffic statistics to the path cost ratio is analytically derived and tested. Cost-effectiveness of a Hybrid Node is compared among different switching fabric architecture that is used to construct the Hybrid Node. Broadcast-Select, Benes and Clos architectures are each considered with different degrees of chip integration. An example Hybrid Network that consists of several Hybrid Routes and Hybrid Nodes is found to be cost-effective and dependent of the ratio of switching to transport costs