System and method for deriving an aggregation delay for a packet aggregation in a wireless network comprising a controller for deriving an optimal aggregation delay at a network node depending on information relating to the channel quality related parameters at the network node in the wireless network.

Wireless mesh networks (WMNs) are wireless multi-hop backhaul networks in which mesh routers relay traffic on behalf of clients or other routers. Due to large MAC layer overhead, applications such as Voice over IP, which send many small packets, show poor performance in WMNs. Packet aggregation increases the capacity of IEEE 802.11-based WMNs by aggregating small packets into larger ones and thereby reducing overhead. In order to have enough packets to aggregate, packets need to be delayed and buffered. Current aggregation mechanisms use fixed buffer delays or do not take into account the delay characteristics of the saturated IEEE 802.11 MAC layer. In this work, we present FUZPAG, a novel packet aggregation architecture for IEEE 802.11-based wireless mesh networks. It uses fuzzy control to determine the optimum aggregation buffer delay under the current channel utilization. By cooperation among neighboring nodes FUZPAG distributes the buffer delay in a fair way. We implemented and evaluated the system in a wireless mesh testbed. For different network topologies we show that FUZPAG outperforms standard aggregation in terms of end-to-end latency under a wide range of traffic patterns.

Wireless mesh networks (WMNs) are wireless multihop networks comprised of mesh routers, which relay traffic on behalf of clients and other nodes. Using the standard IEEE 802.11 distributed coordination function (DCF) as MAC layer, a node needs to contend for the medium each time it wants to transmit a packet. This creates high overhead in particular for small packets and leads to poor performance for real-time applications such as Voice over IP (VoIP) or online gaming. Burst packet transmission can increase the efficiency. For example, with the Transmission Opportunity limit (TXOPlimit) in IEEE 802.11e, a station may transfer several packets without contending for the channel in between. Similarly, IP packet aggregation combines several IP packets together and sends them as one MAC Service Data Unit. Originally, both schemes have been developed for singlehop networks only. Thus the impact on WMNs is unclear if the packets need to contend over multiple hops. In this paper, we use measurements from a 9-node WMN testbed to compare TXOPs and IP packet aggregation for VoIP in terms of fairness, network capacity and quality of user experience. We show that for low networks loads, both TXOPs and IP packet aggregation increase the VoIP quality compared to IEEE 802.11 DCF. However, in highly loaded networks IP packet aggregation outperforms the other schemes.

We present a software framework for channel assingment and routing experiments in multi-channel/multi-radio wireless mesh networks.Based on a plug-in architecture we develop a traffic demand-aware channel assignment algorithm. The evaluation in KAUMesh - an experimental multi-radio mesh network - shows around 30% throughput improvement with respect to a traffic un-aware channel assignment scheme

The coverage area of Access Points CAPS) in Wireless Local Area Networks (WLANs) often overlaps considerably. Hence, a station can potentially associate with many APs. In traditional IEEE 802.11 systems, the station associates to the AP with the strongest signal. This strategy may result in load imbalance between APs and thus low overall network throughput. This paper proposes a new mechanism for selecting the "best" AP based on a novel available bandwidth estimation scheme. The available bandwidth provided by an AP depends mainly on the signal quality and the load on the wireless channel. Based on measurements we first analyze how those factors vary stochastically over time and motivate why a frequent estimation of available bandwidth is necessary. We then develop BEST-AP, a system for Bandwidth ESTimation of Access Points, which uses regular data traffic to estimate the available bandwidth from all APs in reach in a non-intrusive way, even if they are on a different channel. Based on OpenFlow, BEST-AP allows the station to be associated with multiple APs simultaneously and to switch between APs with low overhead. Using the available bandwidth estimates, the system exploits the "best" AP for longer duration while probing the less good APs for shorter durations to update the bandwidth estimations. The evaluation in a WLAN testbed shows that with background load created from real WLAN traces, the dynamic selection of APs improves the throughput of a station by around 81%, compared to a static selection. When the station is mobile, the throughput increases by 176% on average.

Optimizing the operation of IEEE 802.11 networks requires to estimate the load of the wireless channel. The channel busy fraction, which is the fraction of time in which the wireless channel is sensed busy due to successful or unsuccessful transmissions, can be used as such indicator. It can be obtained from e.g. the IEEE 802.11k channel load report or hardware-specific interfaces. Previously, the channel busy fraction has been used as a metric for different purposes such as routing and admission control. However, a thorough evaluation of the relationship between the busy fraction and other important characteristics such as the collision probability and throughput is missing. In this paper, we present an analytical model to study the channel busy fraction in non-saturated IEEE 802.11 networks. We validate the model with measurements in a testbed. The predictions from the model match measurements well. Furthermore, we demonstrate how to apply the model to estimate the available link bandwidth. Using measurements obtained from a testbed operated at 6 Mbit/s, we show that the channel busy fraction allows an accurate prediction of the available bandwidth with an error smaller than 70 Kbit/s.

Wireless Mesh networks are multi-hop networks mostly based on IEEE 802.11 technology and are considered as a viable alternative for providing broadband wireless Internet access. As a consequence, they require support for Quality of Service or advanced mechanisms for selecting Internet gateways. One important requiredinformation is the one-way delay between different nodes. In this paper, we have developed, implemented, and evaluated an one-way delay estimation technique for wireless mesh networks which is based on estimating intra node queuing and inter node forwarding delay. An IP-header option field is used to accumulate the per hop delay estimate to provide an end-to-end estimate. We also outline problems with the implementation and compare results with real one-way delays obtained from a 14 node mesh testbed.We show how estimation accuracy depends on network load and provide insights into further improvements

Cognitive Radio (CR) technology constitutes a promising approach to increase the capacity of Wireless Mesh Networks (WMNs). Using this technology, Mesh Routers (MRs) and the attached Mesh Clients (MCs) are allowed to opportunistically transmit on the licensed band, but under the constraint not to interfere with the Primary Users (PUs) of the spectrum. Thus, the effective deployment of CR- WMNs require that each MR must be able to: sense the current spectrum, select an available PU-free channel and perform the spectrum handoff to a new channel in case of PU arrival on the current one. How to coordinate these actions in the optimal way which maximizes the performance of the CR-WMNs while minimizing the interference to the PUs constitutes an open research issue in CR systems. In this paper, we propose an adaptive spectrum scheduling and allocation scheme which allows a MR to identify the best schedule of (i) when to sense the current channel, (ii) when to transmit, (iii) when to perform a spectrum handoff. Due the large number of parameters involved, we propose Reinforcement Learning (RL) techniques to allow a MR to learn by itself the optimal balance between spectrum sensing-exploitation- exploration actions based on network feedbacks coming from the MCs. We perform extensive simulations which confirm the adaptivity and efficiency of our approach in terms of increased throughput when compared with non-learning based schemes for CR-WMNs.

Cognitive radio ad hoc networks (CRAHNs) constitute a viable solution to solve the current problems of inefficiency in the spectrum allocation, and to deploy highly reconfigurable and self-organizing wireless networks. Cognitive radio (CR) devices are envisaged to utilize the spectrum in an opportunistic way by dynamically accessing different licensed portions of the spectrum. To this aim, most of the recent research has mainly focused on devising spectrum sensing and sharing algorithms at the link layer, so that CR devices can operate without interfering with the transmissions of other licensed users, also called primary users (PUs). However, it is also important to consider the impact of such schemes on the higher layers of the protocol stack, in order to provide efficient end-to-end data delivery. At present, routing and transport layer protocols constitute an important yet not deeply investigated area of research over CRAHNs. This paper provides three main contributions on the modeling and performance evaluation of end-to-end protocols (e.g. routing and transport layer protocols) for CRAHNs. First, we describe NS2-CRAHN, an extension of the NS-2 simulator, which is designed to support realistic simulation of CRAHNs. NS2-CRAHN contains an accurate yet flexible modeling of the activities of PUs and of the cognitive cycle implemented by each CR user. Second, we analyze the impact of CRAHNs characteristics over the route formation process, by considering different routing metrics and route discovery algorithms. Finally, we study TCP performance over CRAHNs, by considering the impact of three factors on different TCP variants: (i) spectrum sensing cycle, (ii) interference from PUs and (iii) channel heterogeneity. Simulation results highlight the differences of CRAHNs with traditional ad hoc networks and provide useful directions for the design of novel end-to-end protocols for CRAHNs.

Wireless mesh networks (WMNs) based on the IEEE 802.11 standard are becoming increasingly popular as a viable alternative to wired networks. WMNs can cover large or difficult to reach areas with low deployment and management costs. Several multi-path routing algorithms have been proposed for such kind of networks with the objective of load balancing the traffic across the network and providing robustness against node or link failures. Packet aggregation has also been proposed to reduce the overhead associated with the transmission of frames, which is not negligible in IEEE 802.11 networks. Unfortunately, multi-path routing and packet aggregation do not work well together, as they pursue different objectives. Indeed, while multi-path routing tends to spread packets among several next-hops, packet aggregation works more efficiently when several packets (destined to the same next-hop) are aggregated and sent together in a single MAC frame. In this paper, we propose a technique, called aggregation aware forwarding, that can be applied to existing multi-path routing algorithms to allow them to effectively exploit packet aggregation so as to increase their network performance. In particular, the proposed technique does not modify the path computation phase,but it just influences the forwarding decisions by taking the state of the sending queues into account.We demonstrated our proposed technique by applying it to Layer-2.5, a multi-path routing and forwarding paradigm for WMNs that has been previously proposed.We conducted a thorough performance evaluation by means of the ns-3 network simulator, which showed that our technique allows to increase the performance both in terms of network throughput and end-to-end delay.

When delivering multimedia services over Internet, different media types are impacted by resource limitations in a different way. While an interactive audio service calls for low-latency communication, video streams should be routed over network paths with sufficient capacity. However, in current networks flows towards the same destination follow the same path, which may lead to a suboptimal resource utilization that effectively penalizes end-users' quality of experience (QoE). This paper proposes Q-POINT, a QoE-driven path optimization model to fairly maximize aggregated end-user QoE for competing clients' service flows by calculating the best path for each flow, subject to resource constraints. We formulate the problem as a mixed integer linear program integrating QoE models for audio, video and data transfer. Such an approach can be leveraged within the software-defined networking paradigm, which provides a control plane to orchestrate path set-up. We evaluate our model and illustrate its benefits over shortest path selection.

There is a clear trend towards making multimedia applications context-aware so as to customize them by taking into account any collection of information which may be relevant, such as e.g. user location. However, current multimedia services are dominated by IMS, which is seen as a service platform that uses the SIP protocol to access all services that the internet can provide. In this paper, we describe the Daidalos approach on making IMS based multi-media services context-aware. We also demonstrate, how generic sensor networks can be integrated into the context management system of our platform thus enabling sensor network detected events to influence behavior of context-aware multimedia applications

The globalisation of our society leads to an increasing need for spontaneous communication. However, the development of such applications is a tedious and error-prone process. This results from the fact that in general only basic functionality is available in terms of protocol implementations and encoders/decoders. This leads to inflexible proprietary software systems implementing unavailable functionality on their own. In this work we introduce Instant-X, a novel component-based middleware platform for multimedia applications. Unlike related work, Instant-X provides a generic programming model with an API for essential tasks of multimedia applications with respect to signalling and data transmission. This API abstracts from concrete component implementations and thus allows replacing specific protocol implementations without changing the application code. Furthermore, Instant-X supports dynamic deployment, i.e., unavailable components can be automatically loaded at runtime. To show the feasibility of our approach we evaluated our Instant-X prototype regarding code complexity and performance.

This draft proposes a mode for RFC 3640 to support an MPEG-4 AAC-BSAC audio codec format with an optional attached bitstream description. The bitstream description employs the MPEG-21 generalized Bitstream Syntax Description Language (gBSDL). The description is attached as auxiliary header and can be used to support adaptation

Unmanned aerial vehicles (UAVs) have gained a lot of popularity in diverse wireless communication fields. They can act as high- altitude flying relays to support communications between ground nodes due to their ability to provide line-of- sight links. With the flourishing Internet of Things, several types of new applications are emerging. In this paper, we focus on bandwidth hungry and delay-tolerant applications where multiple pairs of transceivers require the support of UAVs to complete their transmissions. To do so, the UAVs have the possibility to employ two different bands namely the typical microwave and the high-rate millimeter wave bands. In this paper, we develop a generic framework to assign UAVs to supported transceivers and optimize their trajectories such that a weighted function of the total service time is minimized. Taking into account both the communication time needed to relay the message and the flying time of the UAVs, a mixed non-linear programming problem aiming at finding the stops at which the UAVs hover to forward the data to the receivers is formulated. An iterative approach is then developed to solve the problem. First, a mixed linear programming problem is optimally solved to determine the path of each available UAV. Then, a hierarchical iterative search is executed to enhance the UAV stops' locations and reduce the service time. The behavior of the UAVs and the benefits of the proposed framework are showcased for selected scenarios.

Network Function Virtualization (NFV) focuses on decoupling network functions from proprietary hardware (i.e., middleboxes) by leveraging virtualization technology. Combining it with Software Defined Networking (SDN) enables us to chain network services much easier and faster. The main idea of using these technologies is to consolidate several Virtual Network Functions (VNFs) into a fewer number of commodity servers to reduce costs, increase VNFs fluidity and improve resource efficiency. However, the resource allocation and placement of VNFs in the network is a multifaceted decision problem that depends on many factors, including VNFs resource demand characteristics, arrival rate, configuration of underlying infrastructure, available resources and agreed Quality of Services (QoS) in Service Level Agreements (SLAs). This paper presents a bottom-up open-source NFV analysis platform (NFV-Inspector) to (1) systematically profile and classify VNFs based on resource capacities, traffic demand rate, underlying system properties, placement of VNFs in the network, etc. and (2) extract/calculate the correlation among the QoS metrics and resource utilization of VNFs. We evaluated our approach using an emulated virtual Evolved Packet Core platform (Open5GCore) to showcase how complex relation among various NFV service chains can be systematically profiled and analyzed.

Discovering insights about Virtual Network Function (VNFs) resource demand characteristics will enable cloud vendors to optimize their underlying Network Function Virtualization (NFV) system orchestration and dramatically mitigate CapEx and OpEx spendings. However, analyzing large-scale NFV systems, especially in mobile network environments, is a challenging task and requires tailor-made approaches for each particular application. In this demo, we showcase NFV-Inspector, an open source and extensible VNF analysis platform that is capable of systematically benchmark and profile NFV deployments. Based on its pluggable framework, NFV-Inspector classifies VNFs resource demand characteristics and correlate their Key Performance Indicators (KPIs) with system-level Quality of Service (QoS) measurements. 

Future wireless systems will be heterogeneous and highly adaptive. In this environment, it is important to dis-cover, create and adapt (multimedia) services and content and to integrate these processes into a platform so that pervasive systems, application services and other user-centric services can utilise them easily. In this paper, we present a Multimedia Service Provisioning Platform (MMSPP), designed for sys-tems beyond 3G. The MMSPP orchestrates multimedia ses-sion control and content adaptation. Adaptation processes, based on MPEG-21 DIA, are coordinated via SIP/SDPng and guided through user/terminal profiles and network characteris-tics. The platform provides mechanisms for service discovery and interacts with accounting, charging and network QoS mechanisms

The deployment of multimedia services in next-generation networks is a challenge due to the high configuration complexity of the streaming process in different stationary and mobile sub-networks and for various user devices. The Session Initiation Protocol (SIP) has proven to be a suitable mechanism to handle the control of multimedia services in such networks. However, the current standardization and implementation of SIP do not allow the simultaneous coordination of multiple concurrent applications on a single device, as the prescribed realization of the SIP state machines (transactions) does not consider mutual access of applications to a single SIP stack. This paper presents a SIP-based mechanism for synchronized management of services in a shared environment. We have developed a middleware that facilitates the uniform access of multiple applications towards one or multiple SIP stacks to enable prioritization of services and centralized resource coordination of concurrent SIP applications on a single terminal or server

The delivery and adaptation of multimedia content in distributed and heterogeneous environments requires flexible control and management mechanisms in terminals and in control entities inside the network. In the near future, it is important to reach interoperability between the IETF approaches on multimedia session establishment and control and the MPEG-21 efforts for multimedia streaming and adaptation to bring advanced multimedia service provisioning and adaptation services towards the customer. MPEG-21 Digital Item Adaptation (DIA) provides normative descriptions for supporting adaptation of multimedia content, but does not define interactions with transport and control mechanisms. On the other hand, the IETF standardization efforts on multimedia session control provide the necessary transport (e.g. RTP) and control mechanisms (SDP/SDPng). We thus bridge the gap between those approaches by creating a converged XML model that enables the integration of session management and negotiation protocols (e.g. SIP or Megaco) inspired by the XML formats of MPEG-21 DIA and SDPng. We also present preliminary implementation results of the converged model along with concepts and implementation of network-based content adaptation mechanisms through media gateways that enable flexible multimedia management for heterogeneous consumer terminals

The Ethernet VPN (EVPN) technology has emerged as a key solution for the interconnection of geo-distributed Data Centers (DCs) over provider-managed MPLS networks. Such interconnections need to satisfy service-level agreements, which can be achieved by enforcing Traffic Engineering (TE) policies. However, deploying an effective TE policy is challenging and complex. This stems from the fact that network administrators should have a detailed insight into the network status and protocol specifics. Software-Defined Networking (SDN) may facilitate both the policy definition and deployment based on its comprehensive network view and existing integration with DC management systems, such as OpenStack. This paper presents an SDN-based framework for policy-driven DC interconnections that are built around EVPN. The framework is designed to translate routing and other TE policies, which are defined for EVPN instances, into appropriate low-level network actions to meet the policy goals. A generic programming interface allows an SDN controller to load different TE strategies so as to implement the policy, without the need to hard-code it. Moreover, our evaluations illustrate how clients might benefit from specific TE strategies and what is their impact on network performance

Datacenter networks ofer a large degree of multipath in order to provide large bisectional bandwidth. The end-to-end performance is determined by the load-balancing strategy which needs to be designed to efectively manage congestion. Consequently, congestion aware load-balancing strategies such as CONGA or HULA have been designed. Recently, more and more applications that are hosted on cloud servers use multipath transport protocols such as MPTCP. However, in the presence of MPTCP, existing load-balancing schemes including ECMP, HULA or CONGA may lead to suboptimal forwarding decisions where multiple MPTCP subfows of one connection are pinned on the same bottleneck link. In this paper, we present MP-HULA, a transport layer multi-path aware load-balancing scheme using Programmable Data Planes. First, instead of tracking congestion information for the best path towards the destination, each MP-HULA switch tracks congestion information for the best-k paths to a destination through the neighbor switches. Second, we design MP-HULA using Programmable Data Planes, where each leaf switch can identify, using P4, which MPTCP subfow belongs to which connection. MP-HULA then load-balances diferent MPTCP subfows of a MPTCP connection on diferent next hops considering congestion state while aggregating bandwidth. Our evaluation shows that MP-HULA with MPTCP outperforms HULA in average flow completion time (2.1x at 50% load, 1.7x at 80% load).

Predicting the expected throughput of TCP is important for several aspects such as e.g. determining handover criteria for future multihomed mobile nodes or determining the expected throughput of a given MPTCP subflow for load-balancing reasons. However, this is challenging due to time varying behavior of the underlying network characteristics. In this paper, we present a genetic-algorithm-based prediction model for estimating TCP throughput values. Our approach tries to find the best matching combination of mathematical functions that approximate a given time series that accounts for the TCP throughput samples using genetic algorithm. Based on collected historical datapoints about measured TCP throughput samples, our algorithm estimates expected throughput over time. We evaluate the quality of the prediction using different selection and diversity strategies for creating new chromosomes. Also, we explore the use of different fitness functions in order to evaluate the goodness of a chromosome. The goal is to show how different tuning on the genetic algorithm may have an impact on the prediction. Using extensive simulations over several TCP throughput traces, we find that the genetic algorithm successfully finds reasonable matching mathematical functions that allow to describe the TCP sampled throughput values with good fidelity. We also explore the effectiveness of predicting time series throughput samples for a given prediction horizon and estimate the prediction error and confidence. 

OpenStack has been widely acknowledged to be one of the most important open source cloud platforms. In order to perform experimentally driven research in the area of cloud and cloud networking, there is however a big gap, because most researchers do not have access to a large cloud deployment and cannot change networking or compute infrastructure in order to test their algorithms and protocols on a large-scale. We developed OpenStackEmu, which is to the best of our knowledge the first attempt that combines OpenStack infrastructure with a Software Defined Networking (SDN) based controller such as OpenDaylight and a large-scale network emulator CORE (Common Open Research Emulator). The OpenStack compute and control nodes are connected to the CORE emulation server using TUN/TAP interfaces that inject the control (e.g. for VM migration) and data (VM-to-VM traffic) packets into a customizable network topology that is emulated using configurable Open vSwitches using CORE emulator. Experimenters can define e.g. fat-tree or distributed data center topologies and study the behavior of real VMs and services in those VMs under different background loads and SDN routing policies. We integrated the data center traffic generator DCT2Gen that allows to generate realistic background traffic based on traces from real data centers. Experimenters can study the performance impact of different VM migration strategies or different routing and load balancing schemes on real VM and application performance using different emulated topologies. We believe that OpenStackEmu is an important tool for both the SDN and OpenStack community in order to evaluate the performance of novel algorithms and protocols in the area of cloud networking.

In the field of mobile communications, there coexist various access technologies that support wireless connections, such as Bluetooth, IEEE 802.11 and GPRS. These technologies are diverse in terms of bandwidth, delay, coverage area, etc. It is not uncommon that a mobile device is equipped with multiple wireless network interfaces to achieve efficient and ubiquitous communications. In wireless overlay networks, the mobile node can even utilize multiple network interfaces simultaneously to achieve greater QoS performance. This paper presents the design of such a system that supports multi-homed communications. In particular, we design the policy for mapping different network flows to different interfaces, and propose a load-sharing algorithm based on Weighted-Round-Robin channel selection and Jump-Ahead packet scheduling. Simulation results show that the algorithm can efficiently distribute data packets among multiple channels to achieve bandwidth aggregation

The placement of Virtual Network Functions (VNF) in distributed data centers is an important problem to solve for the next generation cloud based telecom architectures. This is because where to place the VNFs and how to route the traffic in the physical network impacts the energy consumption of the cloud infrastructure, the resiliency of the service chains and the SLA with the tenants. For network operators, it is important to minimize the operational costs of their infrastructure, provide robustness of the placement and routing in order to cope with potential hardware failures and imprecise resource demand specifications. In this paper, we develop a new optimization model for the green multi-period VNF placement and traffic routing problem, where different service chain configurations exist over time. The model is formulated as a Mixed Integer Linear Program (MILP), considers latency due to network propagation and VNF processing and provides different protection methods for the NFV traffic routing to cope with link failures. By applying Soyster's robustness principle, the model yields a network configuration that can cope with load that deviates from the expected demand. Because the MILP is complex to solve, we develop a fast variable fixing heuristic. In our numerical evaluation, we use the virtualized Evolved Packet Core and study the energy cost of different robustness levels and protection schemes for VNF service flow routing.

A Mobile Ad Hoc Network (MANET) is a collection of mobile nodes (MN) that communicate using wireless links without support from any pre-existing infrastructure network. Packets are delivered from a source to a destination using packet forwarding capabilities of intermediate nodes. Therefore, MNs act as both end systems and routers. Mobile Ad Hoc networking has been considered as one of the most important and essential technologies that support future Pervasive Computing Scenarios. Recently, the usage of MANETs in the scope of 4G scenarios has attracted much research efforts and MANETs are seen as one way to extend coverage of hotspots in order to provide Internet connectivity to mobile users. However, TCP performance is crucial for user satisfaction but TCP is well known to suffer from low performance in wireless environments. In this paper, we evaluate several alternative TCP protocols on their suitability for Internet connected MANETs. We conclude that TCP- Vegas is a viable option as its performance is not significantly worse than those TCP variants highly specialized for MANETs but it is compatible with standard Internet protocols

In multi-radio wireless mesh networks, a networkdevice simultaneously transmits packets over different channelsby using multiple radios. Such frequency diversity not only increasesthroughput but makes multi-path routing approaches extremelyinteresting. This is because the channel diversity reducesthe risk for intra- and inter-flow interference. A fundamentalproblem to solve is the forwarding strategy which determineswhich packets to be sent over what multi-path segments atany given time. Ideally, the forwarding strategy should scheduleflows according to the capacity constraints imposed by thechannel assignment. However, the possibility to improve MAClayer efficiency by aggregating small packets into larger onesis reduced when packets are forwarded to different next-hops.In this paper, we develop a novel packet forwarding strategyfor multi-radio mesh networks that combines the benefits ofmulti-path routing with packet aggregation. In our cross-layerapproach, we effectively trade-off aggregation opportunities withchannel diversity. Simulation results show that our approach canimprove network throughput and delay by up to 15 percentand 25 percent, respectively, compared with aggregation unawareforwarding strategies.

Tactical networks are used in military and rescue operations to provide timely and accurate information to operating teams.  Tactical networks have traditionally used long distance narrow band radio links. However, although these links provide robust real-time communication the limited bandwidth makes them less suited for high data-rate applications.  To support high-data rate TCP applications such as providing digital images and maps, emerging tactical networks use shorter range but higher data-rate wide band radio links and multi-hop.  Due to the requirement of cheap up-front cost, most MANET research has focused on Carrier Sense Multiple Access (CSMA) networks. However, in tactical networks, where bounded delays are important, Time Division Multiple Access (TDMA) can give better possibility to support the Quality of Service needed for real-time communication.  The purpose of this paper is to assess and compare the throughput of three state-of-the-art TCP versions and two routing protocols over TDMA based MANETs.

Many  networks are multi-path; mobile devices have multiple interfaces, data centers have redundant paths and ISPs forward traffic over disjoint paths to perform load-balancing. Multi-path TCP (MPTCP) is a new mechanism that transparently divides a TCP connection into subflows and distributes them over a host's network interfaces. While this enables multi-homed systems like e.g. smartphones to use several interfaces and thus different, and mostly disjoint, network paths for a single transmission, most end-systems are still single-homed. With one interface, standard MPTCP creates only a single subflow, making single-homed systems unable to benefit from MPTCP's functionality. In this paper we propose PathFinder, an MPTCP extension that tries to estimate the number of subflows required to fully utilize  the network capacity, enabling single-homed hosts to reap the benefits of MPTCP. We evaluate MPTCP with PathFinder and compare its performance to standard MPTCP. The evaluation shows that PathFinder is able to open a limited but sufficient amount of subflows to significantly increase the throughput when compared to using standard MPTCP.

Routing packets over multiple disjoint paths towards a destination can increase network utilization by load-balancing the traffic over the network. The drawback of load-balancing is that different paths might have different delay properties, causing packets to be reordered. This can reduce TCP performance significantly, as reordering is interpreted as a sign of congestion. Packet reordering can be avoided by letting the network layer route strictly on flow-level. This will, however, also limit the ability to achieve optimal network throughput. There are also several proposals that try to mitigate the effects of reordering at the transport layer. In this paper, we perform an initial evaluation of such TCP reordering mitigations in multi-radio multi-channel wireless mesh networks when using multi-path routing. We evaluate two TCP reordering mitigation techniques implemented in the Linux kernel. The transport layer mitigations are compared using different multi-path routing strategies. Our findings show that, in general, flow-level routing gives the best TCP performance and that transport layer reordering mitigations only marginally can improve performance.

Routing packets over multiple disjoint paths to- wards a destination can increase network utilization by load- balancing the traffic over the network. The drawback of load-balancing is that different paths might have different delay properties, causing packets to be reordered. This can reduce TCP performance significantly, as reordering is interpreted as a sign of congestion. Packet reordering can be avoided by letting the network layer route strictly on flow-level. This will, however, also limit the ability to achieve optimal network throughput. There are also several proposals that try to mitigate the effects of reordering at the transport layer. In this paper, we perform an initial evaluation of such TCP reordering mitigations in multi-radio multi-channel wireless mesh networks when using multi-path routing. We evaluate two TCP reordering mitigation techniques implemented in the Linux kernel. The transport layer mitigations are compared using different multi-path routing strategies. Our findings show that, in general, flow-level routing gives the best TCP performance and that transport layer reordering mitigations only marginally can improve performance. 

Routing packets over multiple disjoint paths towards a destination can increase network utilization by load-balancing the traffic over the network. In wireless mesh networks, multi-radio multi-channel nodes are often used to create a larger set of interference-free paths thus increasing the chance of load-balancing. The drawback of load-balancing is that different paths might have different delay properties, causing packets to be reordered. This can reduce TCP performance significantly, as reordering is interpreted as a sign of congestion. Packet reordering can be avoided by letting the network layer forward traffic strictly on flow-level. This would avoid the negative drawbacks of packet reordering, but will also limit the ability to achieve optimal network throughput. On the other hand, there are several proposals that try to mitigate the effects of reordering at the transport layer. In this paper, we perform an in-depth evaluation of such TCP reordering mitigations in multi-radio multi-channel wireless mesh networks when using multi-path forwarding. We evaluate two TCP reordering mitigation techniques implemented in the Linux kernel. The transport layer mitigations are compared using different multi-path forwarding strategies. Our findings show that, in general, flow-level forwarding gives the best TCP performance and that transport layer reordering mitigations only marginally can improve performance

Recently, Wireless Mesh Networks (WMNs) have attracted attention as means to provide alternative internet connectivity to rural areas or communities. In WMNs, wireless access points communicate with each other wirelessly forming a true wireless, mesh based access network of mesh relay nodes (MRN). Mesh gateways (MG) provide internet connectivity and standard mobile clients attach to MRNs, which forward packets via other MRNs to other meshed clients or through MGs to the internet. Therefore, the wireless backbone comprised of MRNs and MGs is similar to static, internet connected Ad Hoc networks. A major problem is however scalability of WMNs as well as MAC and PHY layer overhead for packet transmission. Capacity of WMNs can be increased significantly by aggregating (combining) several smaller packets into larger ones [1][2]. This is in particular beneficial to VoIP flows where packet sizes are small. The overall number of packets is reduced, average packet size increased and contention will take place only once for the larger packets. A relay node will then process fewer large packets instead of many small ones. While such aggregation mechanisms have been proposed for single-hop infrastructure WLAN, designing an aggregation strategy for multi-hop WMNs is a hard problem because in this multi-hop environment, signal quality and congestion for each link is different. When mesh relay nodes aggregate small packets, there is an inherent trade-off regarding packet size. Aggregating more packets leads to larger packets, which reduces the overall number of packets in the mesh and thus reduces multi-hop contention and packet loss due to collisions. However, such larger (aggregated) packets can lead to higher packet loss for a link that operates at low signal quality [3]. To find the optimum frame size for packet aggregation is therefore not trivial and depends on traffic and link quality which can vary over time and might be different for each link in a multi hop path. In this paper, we will evaluate if packet aggregation is beneficial when applied to standard TCP in a WMN environment with low quality links. For integration of Mesh Networks into 4G environment, the transport layer needs to be compatible and interoperable with already deployed transport protocols. We will therefore compare end-to-end performance of standard TCP (with Selective and Delayed ACKs) using different MSS sizes in a mesh environment, both with and without deploying packet aggregation. Delayed ACKs and Selective ACKs has shown to be beneficial in multihop environments as the number of packets is reduced leading to less contention [4]

Wireless Mesh Networks (WMNs) have attracted attention as a way to provide alternative wireless Internet connectivity. In a WMN, access points communicate with each other wirelessly, forming a wireless mesh backbone network. One major problem in WMNs is low performance due to MAC and PHY layer overhead for transmission of small packets. Around 44 percent of the packets in the Internet are smaller than 100 bytes and a majority of these packets are TCP packets. The capacity of WMNs can therefore be increased significantly by packet aggregation (concatenation) of TCP packets. In this paper, we evaluate the impact of packet aggregation on TCP performance in WMNs. Using both synthetic and trace based traffic distributions we demonstrate that packet aggregation can not only increase capacity for TCP in such networks but also reduce round trip times.

Recently, Wireless Mesh Networks (WMNs) have attracted attention as a way to provide alternative Internet connectivity to rural areas or communities. In WMNs, wireless access points communicate with each other wirelessly, forming a true wireless mesh based access network of mesh relay nodes (MRNs). A major problem is, however, scalability of WMNs as well as MAC and PHY layer overhead for packet transmission. Capacity of WMNs can be increased significantly by aggregating (combining) several smaller packets into larger ones. This is in particular beneficial to flows where many packets are of small sizes such as voice flows. TCP could also benefit, as by aggregating several packets together there is a reduced collision risk between TCP DATA and TCP ACKs in addition to the reduced MAC layer contention. In this paper, we investigate the impact of packet aggregation on TCP in Wireless Mesh Networks. Using several different scenarios we demonstrate that packet aggregation can not only increase capacity for TCP in such networks but also improve fairness and reduce end-to-end delay