Multipath routing for video delivery over bandwidth-limited networks

The delivery of quality video service often requires high bandwidth with low delay or cost in network transmission. Current routing protocols such as those used in the Internet are mainly based on the single-path approach (e.g., the shortest-path routing). This approach cannot meet the end-to-end bandwidth requirement when the video is streamed over bandwidth-limited networks. In order to overcome this limitation, we propose multipath routing, where the video takes multiple paths to reach its destination(s), thereby increasing the aggregate throughput. We consider both unicast (point-to-point) and multicast scenarios. For unicast, we present an efficient multipath heuristic (of complexity O(|V|/sup 3/)), which achieves high bandwidth with low delay. Given a set of path lengths, we then present and prove a simple data scheduling algorithm as implemented at the server, which achieves the theoretical minimum end-to-end delay. For a network with unit-capacity links, the algorithm, when combined with disjoint-path routing, offers an exact and efficient solution to meet a bandwidth requirement with minimum delay. For multicast, we study the construction of multiple trees for layered video to satisfy the user bandwidth requirements. We propose two efficient heuristics on how such trees can be constructed so as to minimize the cost of their aggregation subject to a delay constraint.     

Understanding Internet topology: principles, models, and validation

Building on a recent effort that combines a first-principles approach to modeling router-level connectivity with a more pragmatic use of statistics and graph theory, we show in this paper that for the Internet, an improved understanding of its physical infrastructure is possible by viewing the physical connectivity as an annotated graph that delivers raw connectivity and bandwidth to the upper layers in the TCP/IP protocol stack, subject to practical constraints (e.g., router technology) and economic considerations (e.g., link costs). More importantly, by relying on data from Abilene, a Tier-1 ISP, and the Rocketfuel project, we provide empirical evidence in support of the proposed approach and its consistency with networking reality. To illustrate its utility, we: 1) show that our approach provides insight into the origin of high variability in measured or inferred router-level maps; 2) demonstrate that it easily accommodates the incorporation of additional objectives of network design (e.g., robustness to router failure); and 3) discuss how it complements ongoing community efforts to reverse-engineer the Internet.     

Joint interference coordination approach in femtocell networks for QoS performance optimization

Future heterogeneous networks with dense cell deployment may cause high intercell interference. A number of interference coordination (IC) approaches have been proposed to reduce intercell interference. For dense small‐cell deployment with high intercell interference between cells, traditional forward link IC approaches intended to improve edge user throughput for best effort traffic (ie, file transfer protocol download), may not necessarily improve quality of service performance for delay‐sensitive traffic such as voice over long‐term evolution traffic. This study proposes a dynamic, centralized joint IC approach to improve forward link performance for delay‐sensitive traffic on densely deployed enterprise‐wide long‐term evolution femtocell networks. This approach uses a 2‐level scheme: central and femtocell. At the central level, the algorithm aims to maximize network utility (the utility‐based approach) and minimize network outage (the graphic‐based approach) by partitioning the network into clusters and conducting an exhaustive search for optimized resource allocation solutions among femtocells (femto access points) within each cluster. At the femtocell level, in contrast, the algorithm uses existing static approaches, such as conventional frequency reuse (ReUse3) or soft frequency reuse (SFR) to further improve user equipment quality of service performance. This combined approach uses utility‐ and graphic‐based SFR and ReUse3 (USFR/GSFR and UReUse3/GReUse3, respectively). The cell and edge user throughput of best effort traffic and the packet loss rate of voice over long‐term evolution traffic have been characterized and compared using both the proposed and traditional IC approaches.     

Analysis of adaptive bandwidth allocation in wireless networks with multilevel degradable quality of service

A wireless/mobile network supporting multilevel quality of service (QoS) is considered. In such a network, users or applications can tolerate a certain degree of QoS degradation. Bandwidth allocation to users can, therefore, be adjusted dynamically according to the underlying network condition so as to increase bandwidth utilization and service provider's revenue. However, arbitrary QoS degradation may be unsatisfactory or unacceptable to the users, hence resulting in their subsequent defection. Instead of only focusing on bandwidth utilization or blocking/dropping probability, two new user-perceived QoS metrics, degradation ratio and upgrade/degrade frequency, are proposed. A Markov model is then provided to derive these QoS metrics. Using this model, we evaluate the effects of adaptive bandwidth allocation on user-perceived QoS and show the existence of trade offs between system performance and user-perceived QoS. We also show how to exploit adaptive bandwidth allocation to increase system utilization (for the system administrator) with controlled QoS degradation (for the users). By considering various mobility patterns, the simulation results are shown to match our analytical results, demonstrating the applicability of our analytical model to more general cases.     

Packet Voice on a Local Area Network with Round Robin Service

In this paper, we propose a combined voice/data protocol suitable for multiple access broadcast networks that provide round robin service to the stations. Such networks are well suited to the integration of voice and data since they guarantee bounded delay and provide high utilization even for high bandwidth channels. Using one such network proposal-namely Expressnet-as a representative scheme, we examine the characteristics of the service that voice traffic experiences under the voice/data protocol. We show that the access protocol is able to utilize the channel efficiently to support a large population of voice sources while maintaining low packet delay and guaranteeing some prespecified minimum bandwidth for data traffic. In addition, we show the advantages of silence suppression, i.e., discarding speech that constitutes silent periods, and we examine the cost of overloading the network in terms of the amount of speech discarded.     

Precomputation schemes for QoS routing

Precomputation-based methods have recently been proposed as an instrument to facilitate scalability, improve response time, and reduce computation load on network elements. The key idea is, in effect, to reduce the time needed to handle an event by performing some computation in advance, i.e., prior to the event's arrival. Such computations are performed as background processes, enabling a solution to be provided promptly upon a request, through a simple, fast procedure. We investigate precomputation methods in the context of quality-of-service (QoS) routing. Precomputation is highly desirable for QoS routing schemes due to the high computational complexity of selecting QoS paths, and the need to provide a satisfactory path promptly upon a request. We consider two major settings of QoS routing. The first case is where the QoS constraint is of the "bottleneck" type, e.g., a bandwidth requirement, and network optimization is sought through hop minimization. The second is the more general setting of "additive" QoS constraints (e.g., delay) and general link costs. The paper mainly focuses on the first setting. We show that, by exploiting the typical hierarchical structure of large-scale networks, a substantial improvement can be achieved in terms of computational complexity. We consider networks with topology aggregation. We show that precomputation is a necessary element for any QoS routing scheme and establish a precomputation scheme appropriate for such settings. We consider the case of additive QoS constraints (e.g., delay) and general link costs. As the routing problem becomes NP-hard, we focus on /spl epsiv/-optimal approximations and derive a precomputation scheme that offers a major improvement over the standard approach.     

On minimizing flow monitoring costs in large-scale software-defined network networks

Recent years have witnessed the rise of novel network applications such as telesurgery, telepresence, and holoportation. As such applications have stringent performance requirements, timely and accurate traffic monitoring becomes of paramount importance to be able to react in a timely and efficient manner, and swiftly adjust the network configuration to achieve the sought-after requirements. However, existing monitoring schemes are either incurring high cost (e.g., high bandwidth consumption) due to the large number of monitoring messages or inefficient when they incur high reporting delay (i.e., the time needed for a monitoring message to reach the controller) making the collected statistics obsolete. In this paper, we address this problem and propose monitoring mechanisms for software defined networks that minimize the monitoring cost while satisfying an upper bound on the reporting delay of the statistics. Our solutions allow to carefully select the switch that should report the statistics about each flow crossing the network taking into consideration the available bandwidth and the capacity of the switch (i.e., the maximum number of flows that it can monitor). In particular, we formulate the switch-to-flow selection problem as an integer linear program and propose two heuristic algorithms to cope with large-scale instances of the problem. We consider the scenario where a single controller is collecting statistics and another where statistics are collected by multiple controllers. Simulation results show that the proposed algorithms provide near-optimal solutions with minimal computation time and outperform existing monitoring strategies in terms of monitoring cost and reporting delay.     

Signaling architectures and recovery time scaling for grid applications in IST Project MUPBED

It is now evident to the research community that local computational resources cannot keep up in an economical way with the demands generated by some users/applications. Therefore, distributed computing and the concept of a computational grid are now emerging. Novel transport network concepts are needed to support such visions, and high-speed intelligent optical networking may be the required infrastructure that will enable global grids. Emerging utility grid applications like business continuity and disaster recovery have strong requirements on the dynamic optical networks connecting the distributed grid resources. Supporting grid networking with an intelligent optical network (ION) infrastructure will allow utility grid applications the necessary flexibility with the required QoS (e.g., high bandwidth, reliability, limited delay). Emerging QoS requirements, such as scalable recovery time, highly depend on the ION's signaling architecture. This article gives simple analytical models for the implementation options of optical control plane signaling, shows simulation models for different resilience strategies, and offers some illustrative numerical comparisons to support the aforementioned efforts. This research area is also discussed, among others, in the European research project Multi-Partner European Testbeds for Research Networking (MUPBED).     

Supporting bursty traffic with bandwidth guarantee in WDM distribution networks

This paper presents new research results of the DARPA-funded ONRAMP consortium on the next generation Internet to study efficient WDM-based network architectures and protocols for supporting broadband services in regional access networks. In particular, we present new efficient scheduling algorithms for bandwidth sharing in WDM distribution networks. The current ONRAMP distribution network architecture has a tree topology with each leaf node (e.g., a router or workstation) sharing access to the root node of the tree, which corresponds to an access node in the feeder network. Our model allows a leaf node to use one or more fixed-tuned or tunable transceivers; moreover, different leaf nodes can support different subsets of wavelengths depending on their expected traffic volumes. An important goal of ONRAMP is to support bandwidth-on-demand services with QoS guarantee over WDM. As a first step toward this goal, we have developed several fast scheduling algorithms for flexible bandwidth reservations in a WDM distribution network. The scheduling algorithms can provably guarantee any bandwidth reservations pattern that does not overbook network resources, i.e., bandwidth reservation (throughput) up to 100% network capacity can be supported.     

Sticky CSMA/CA: Implicit synchronization and real-time QoS in mesh networks

We propose a novel approach to QoS for real-time traffic over wireless mesh networks, in which application layer characteristics are exploited or shaped in the design of medium access control. Specifically, we consider the problem of efficiently supporting a mix of Voice over IP (VoIP) and delay-insensitive traffic, assuming a narrowband physical layer with CSMA/CA capabilities. The VoIP call carrying capacity of wireless mesh networks based on classical CSMA/CA (e.g., the IEEE 802.11 standard) is low compared to the raw available bandwidth, due to lack of bandwidth and delay guarantees. Time Division Multiplexing (TDM) could potentially provide such guarantees, but it requires fine-grained network-wide synchronization and scheduling, which are difficult to implement. In this paper, we introduce Sticky CSMA/CA, a new medium access mechanism that provides TDM-like performance to real-time flows without requiring explicit synchronization. We exploit the natural periodicity of VoIP flows to obtain implicit synchronization and multiplexing gains. Nodes monitor the medium using the standard CSMA/CA mechanism, except that they remember the recent history of activity in the medium. A newly arriving VoIP flow uses this information to grab the medium at the first available opportunity, and then sticks to a periodic schedule, providing delay and bandwidth guarantees. Delay-insensitive traffic fills the gaps left by the real-time flows using novel contention mechanisms to ensure efficient use of the leftover bandwidth. Large gains over IEEE 802.11 networks are demonstrated in terms of increased voice call carrying capacity (more than 100% in some cases). We briefly discuss extensions of these ideas to a broader class of real-time applications, in which artificially imposing periodicity (or some other form of regularity) at the application layer can lead to significant enhancements of QoS due to improved medium access.     

Stability and performance analysis of networks supporting elasticservices

We consider the stability and performance of a model for networks supporting services that adapt their transmission to the available bandwidth. Not unlike real networks, in our model, connection arrivals are stochastic, each has a random amount of data to send, and the number of ongoing connections in the system changes over time. Consequently, the bandwidth allocated to, or throughput achieved by, a given connection may change during its lifetime as feedback control mechanisms react to network loads. Ideally, if there were a fixed number of ongoing connections, such feedback mechanisms would reach an equilibrium bandwidth allocation typically characterized in terms of its “fairness” to users, e.g., max-min or proportionally fair. We prove the stability of such networks when the offered load on each link does not exceed its capacity. We use simulation to investigate performance, in terms of average connection delays, for various fairness criteria. Finally, we pose an architectural problem in TCP/IPs decoupling of the transport and network layer from the point of view of guaranteeing connection-level stability, which we claim may explain congestion phenomena on the Internet     

Path diversity for enhanced media streaming

Media streaming over best effort packet networks such as the Internet is quite challenging because of the dynamic and unpredictable available bandwidth, loss rate, and delay. Recently, streaming over multiple paths to provide path diversity has emerged as an approach to help overcome these problems. This article provides an overview of the benefits and use of path diversity for media streaming. The different approaches to media coding and streaming over multiple paths are examined, together with architectures for achieving path diversity between single or multiple senders and a single receiver. Important examples include using the distributed servers in a content delivery network to provide path diversity to a requesting client, using multiple 802.11 wireless access points to provide path diversity to a mobile client, and using relays to provide low-latency media communication. The design, analysis, and operation of media streaming systems that use path diversity are considered, with emphasis on the accurate performance models needed to select the best paths or best servers.     

Capacity and delay of hybrid wireless broadband access networks

An optical network is too costly to act as a broadband access network. On the other hand, a pure wireless ad hoc network with n nodes and total bandwidth of W bits per second cannot provide satisfactory broadband services since the pernode throughput diminishes as the number of users goes large. In this paper, we propose a hybrid wireless network, which is an integrated wireless and optical network, as the broadband access network. Specifically, we assume a hybrid wireless network consisting of n randomly distributed normal nodes, and m regularly placed base stations connected via an optical network. A source node transmits to its destination only with the help of normal nodes, i.e., in the ad hoc mode, if the destination can be reached within L (L /spl geq/ 1) hops from the source. Otherwise, the transmission will be carried out in the infrastructure mode, i.e., with the help of base stations. Two transmission modes share the same bandwidth of W bits/sec. We first study the throughput capacity of such a hybrid wireless network, and observe that the throughput capacity greatly depends on the maximum hop count L and the number of base stations m. We show that the throughput capacity of a hybrid wireless network can scale linearly with n only if m = Ω(n), and when we assign all the bandwidth to the infrastructure mode traffics. We then investigate the delay in hybrid wireless networks. We find that the average packet delay can be maintained as low as Θ(1) even when the per-node throughput capacity is Θ(W).     

Interleaved Polling with Adaptive Cycle Time (IPACT): A Dynamic Bandwidth Distribution Scheme in an Optical Access Network

While in recent years backbone bandwidth has experienced substantial growth, little has changed in the access network. Last mile still remains the bottleneck between a high capacity LAN or home network and the backbone. Passive optical network (PON) is a technology viewed by many as an attractive solution to this problem.In this study, we discuss and evaluate design issues for PON access networks. Specifically, to drive the cost of an access network down, it is very important to have an efficient, scalable solution. We believe that a PON based on polling, with data encapsulated in Ethernet frames, possesses the best qualities, such as dynamic bandwidth distribution, use of a single downstream and a single upstream wavelength, ability to provision a fractional wavelength capacity to each user, and ease of adding a new user.To support dynamic bandwidth distribution, we propose an interleaved polling algorithm. We then suggest a scheme for in-band signaling that allows using a single wavelength for both downstream data and control message transmission.To obtain realistic simulation results, we generated synthetic traffic that exhibits the properties of self-similarity and long-range dependence. We then analyzed the network performance and its effect on various types of traffic, e.g., best-effort data traffic, VBR video traffic and CBR streams.     

End-to-end optimized TCP-friendly rate control for real-time video streaming over wireless multi-hop networks

Rate control is an important issue in video streaming applications. The most popular rate control scheme over wired networks is TCP-Friendly Rate Control (TFRC), which is designed to provide optimal transport service for unicast multimedia delivery based on the TCP Reno’s throughput equation. It assumes perfect link quality, treating network congestion as the only reason for packet losses. Therefore, when used in wireless environment, it suffers significant performance degradation because of packet losses arising from time-varying link quality. Most current research focuses on enhancing the TFRC protocol itself, ignoring the tightly coupled relation between the transport layer and other network layers. In this paper, we propose a new approach to address this problem, integrating TFRC with the application layer and the physical layer to form a holistic design for real-time video streaming over wireless multi-hop networks. The proposed approach can achieve the best user-perceived video quality by jointly optimizing system parameters residing in different network layers, including real-time video coding parameters at the application layer, packet sending rate at the transport layer, and modulation and coding scheme at the physical layer. The problem is formulated and solved as to find the optimal combination of parameters to minimize the end-to-end expected video distortion constrained by a given video playback delay, or to minimize the video playback delay constrained by a given end-to-end video distortion. Experimental results have validated 2–4 dB PSNR performance gain of the proposed approach in wireless multi-hop networks by using H.264/AVC and NS-2.     

Adaptive allocation of resources and call admission control forwireless ATM using genetic algorithms

In wireless ATM-based networks, admission control is required to reserve resources in advance for calls requiring guaranteed services. In the case of a multimedia call, each of its substreams (i.e., video, audio, and data) has its own distinct quality of service (QoS) requirements (e.g., cell loss rate, delay, jitter, etc.). The network attempts to deliver the required QoS by allocating an appropriate amount of resources (e.g., bandwidth, buffers). The negotiated QoS requirements constitute a certain QoS level that remains fixed during the call (static allocation approach). Accordingly, the corresponding allocated resources also remain unchanged. We present and analyze an adaptive allocation of resources algorithm based on genetic algorithms. In contrast to the static approach, each substream declares a preset range of acceptable QoS levels (e.g., high, medium, low) instead of just a single one. As the availability of resources in the wireless network varies, the algorithm selects the best possible QoS level that each substream can obtain. In case of congestion, the algorithm attempts to free up some resources by degrading the QoS levels of the existing calls to lesser ones. This is done, however, under the constraint of achieving maximum utilization of the resources while simultaneously distributing them fairly among the calls. The degradation is limited to a minimum value predefined in a user-defined profile (UDP). Genetic algorithms have been used to solve the optimization problem. From the user perspective, the perception of the QoS degradation is very graceful and happens only during overload periods. The network services, on the other hand, are greatly enhanced due to the fact that the call blocking probability is significantly decreased. Simulation results demonstrate that the proposed algorithm performs well in terms of increasing the number of admitted calls while utilizing the available bandwidth fairly and effectively     

Adaptive algorithms for feedback-based flow control in high-speed,wide-area ATM networks

This paper gives a class of flow control algorithms for the adaptive allocation of bandwidths to virtual connections (VC) in high-speed, wide-area ATM networks. The feedback rate to the source from the network is parsimonious, with each feedback bit indicating whether the buffer at a distant switch is above or below a threshold. The service discipline at the switch is first-come-first-served. The important goal of adaptability aims to make all of the network bandwidth available to the active VCs, even though the number of such VCs is variable over a given range. Each VC has two parameters, one giving its minimum guaranteed bandwidth and the other is the weight for determining its share of the uncommitted bandwidth. Judicious selection of these parameters defines distinctive services, such as best effort and best effort with minimum bandwidth. We derive design rules for selecting the parameters of the algorithms such that the appropriate guarantees and fairness properties are exhibited in the dynamical behavior. The systematic use of “damping” in right proportion with “gain” is shown to be a powerful device for stabilizing behavior and achieving fairness. Our analyses are based on a simple analytic fluid model composed of a system of first-order delay-differential equations, which reflect the propagation delay across the network. Extensive simulations examine the following: (1) fairness, especially to start-up VCs; (2) oscillations; (3) transient behavior, such as the rate of equalization from different initial conditions; (4) disparate bandwidth allocations; (5) multiple paths with diverse propagation delays; (6) adaptability and robustness with respect to parameters; and (7) interoperability of different algorithms     

An optimal solution to resource allocation among soft QoS traffic in wireless network

Optimization theory and nonlinear programming method have successfully been applied into wire‐lined networks (e.g., the Internet) in developing efficient resource allocation and congestion control schemes. The resource (e.g., bandwidth) allocation in a communication network has been modeled into an optimization problem: the objective is to maximize the source aggregate utility subject to the network resource constraint. However, for wireless networks, how to allocate the resource among the soft quality of service (QoS) traffic remains an important design challenge. Mathematically, the most difficult comes from the non‐concave utility function of soft QoS traffic in the network utility maximization (NUM) problem. Previous result on this problem has only been able to find its sub‐optimal solution. Facing this challenge, this paper establishes some key theorems to find the optimal solution and then present a complete algorithm called utility‐based allocation for soft QoS to obtain the desired optimal solution. The proposed theorems and algorithm act as designing guidelines for resource allocation of soft QoS traffic in a wireless network, which take into account the total available resource of network, the users’ traffic characteristics, and the users’ channel qualities. By numerical examples, we illustrate the explicit solution procedures.Copyright © 2013 John Wiley & Sons, Ltd.     

Transport control protocol in optical burst switched networks: issues, solutions, and challenges

Since its advent in 1981, TCP has been subject to a tremendous amount of research effort and enhancements for achieving better performance over various network environments and application scenarios. Due to the transmission characteristics of optical burst switched networks, such as random burst dropping, retro-blocking (i.e., bursts proceeding or delayed from their actual reservation time slot), burstification delay, and burst signaling delay, TCP could be significantly affected if no corresponding countermeasure and enhancement are developed. In this review article we provide a comprehensive survey on reported studies for TCP enhancements over OBS networks in order to mitigate the numerous side effects due to the buffer- less characteristic of burst transmission. Furthermore, we closely analyze TCP behavior over OBS networks with various burst transmission characteristics while highlighting the open challenges that have not yet been extensively tackled or solved.     

Performance Evaluation of Polling Protocols for Data Transmission on Wireless Communication Networks

The growing interest in mobile computing and communication devices leads to the necessity of wireless broadband network. Data transmission over such networks requires suitable error control schemes to guarantee high data reliability as well as efficient bandwidth utilization.In this paper we propose an accurate yet simple analytical approach to evaluate the performance of wireless networks using gated and exhaustive polling protocols combined with the Stop and Wait (SW) or Go Back N (GBN) ARQ schemes [Bertsekas and Gallager, 2]. Moreover, simulation results concerning the performance of polling protocols combined with the Selective Repeat (SR) ARQ scheme are also shown for comparison purposes.Protocol performance is estimated under very general assumptions, such as: AWGN or fading channels, arbitrary value of the round trip delay and arbitrary distribution of the traffic load (i.e., both symmetric or asymmetric system have been considered).