Biblio
The major challenge of Real Time Protocol is to balance efficiency and fairness over limited bandwidth. MPTCP has proved to be effective for multimedia and real time networks. Ideally, an MPTCP sender should couple the subflows sharing the bottleneck link to provide TCP friendliness. However, existing shared bottleneck detection scheme either utilize end-to-end delay without consideration of multiple bottleneck scenario, or identify subflows on switch at the expense of operation overhead. In this paper, we propose a lightweight yet accurate approach, EMPTCP, to detect shared bottleneck. EMPTCP uses the widely deployed ECN scheme to capture the real congestion state of shared bottleneck, while at the same time can be transparently utilized by various enhanced MPTCP protocols. Through theory analysis, simulation test and real network experiment, we show that EMPTCP achieves higher than 90% accuracy in shared bottleneck detection, thus improving the network efficiency and fairness.
In multi-tenant datacenters, the hardware may be homogeneous but the traffic often is not. For instance, customers who pay an equal amount of money can get an unequal share of the bottleneck capacity when they do not open the same number of TCP connections. To address this problem, several recent proposals try to manipulate the traffic that TCP sends from the VMs. VCC and AC/DC are two new mechanisms that let the hypervisor control traffic by influencing the TCP receiver window (rwnd). This avoids changing the guest OS, but has limitations (it is not possible to make TCP increase its rate faster than it normally would). Seawall, on the other hand, completely rewrites TCP's congestion control, achieving fairness but requiring significant changes to both the hypervisor and the guest OS. There seems to be a need for a middle ground: a method to control TCP's sending rate without requiring a complete redesign of its congestion control. We introduce a minimally-invasive solution that is flexible enough to cater for needs ranging from weighted fairness in multi-tenant datacenters to potentially offering Internet-wide benefits from reduced interflow competition.
Selecting the best path in multi-path heterogeneous networks is challenging. Multi-path TCP uses by default a scheduler that selects the path with the minimum round trip time (minRTT). A well-known problem is head-of-line blocking at the receiver when packets arrive out of order on different paths. We shed light on another issue that occurs if scheduling have to deal with deep queues in the network. First, we highlight the relevance by a real-world experiment in cellular networks that often deploy deep queues. Second, we elaborate on the issues with minRTT scheduling and deep queues in a simplified network to illustrate the root causes; namely the interaction of the minRTT scheduler and loss-based congestion control that causes extensive bufferbloat at network elements and distorts RTT measurement. This results in extraordinary large buffer sizes for full utilization. Finally, we discuss mitigation techniques and show how alternative congestion control algorithms mitigate the effect.
With the development of modern High-Speed Railway (HSR) and mobile communication systems, network operators have a strong demand to provide high-quality on-board Internet services for HSR passengers. Multi-path TCP (MPTCP) provides a potential solution to aggregate available network bandwidth, greatly overcoming throughout degradation and severe jitter using single transmission path during the high-speed train moving. However, the choose of MPTCP algorithms, i.e., Coupled or Uncoupled, has a great impact on the performance. In this paper, we investigate this interesting issue in the practical datasets along multiple HSR lines. Particularly, we collect the first-hand network datasets and analyze the characteristics and category of traffic flows. Based on this statistics, we measure and analyze the transmission performance for both mice flows and elephant ones with different MPTCP congestion control algorithms in HSR scenarios. The simulation results show that, by comparing with the coupled MPTCP algorithms, i.e., Fully Coupled and LIA, the uncoupled EWTCP algorithm provides more stable throughput and balances congestion window distribution, more suitable for the HSR scenario for elephant flows. This work provides significant reference for the development of on-board devices in HSR network systems.
Opportunistic spectrum access is one of the emerging techniques for maximizing throughput in congested bands and is enabled by predicting idle slots in spectrum. We propose a kernel-based reinforcement learning approach coupled with a novel budget-constrained sparsification technique that efficiently captures the environment to find the best channel access actions. This approach allows learning and planning over the intrinsic state-action space and extends well to large state spaces. We apply our methods to evaluate coexistence of a reinforcement learning-based radio with a multi-channel adversarial radio and a single-channel carrier-sense multiple-access with collision avoidance (CSMA-CA) radio. Numerical experiments show the performance gains over carrier-sense systems.
The idea to use multiple paths to transport TCP traffic seems very attractive due to its potential benefits it may offer for both redundancy and better utilization of available resources by load balancing. Fixed and mobile network providers employ frequently load-balancers that use multiple paths on either per-flow or per-destination level, but very seldom on per-packet level. Despite of the benefits of packet-level load balancing mechanisms (e.g., low computational complexity and high bandwidth utilization) network providers can't use them mainly because of TCP packet reorderings that harm TCP performance. Emerging network architectures also support multiple paths, but they face with the same obstacle in balancing their load to multiple paths. Indeed, packet level load balancing research is paralyzed by the reordering vulnerability of TCP.A couple of TCP variants exist that deal with TCP packet reordering problem, but due to lack of end-to-end transparency they were not widely deployed and adopted. In this paper, we revisit TCP's packet reorderings problem and present a transparent and light-weight algorithm, Out-of-Order Robustness for TCP with Transparent Acknowledgment (ACK) Intervention (ORTA), to deal with out-of-order deliveries.ORTA works as a transparent thin layer below TCP and hides harmful side-effects of packet-level load balancing. ORTA monitors all TCP flow packets and uses ACK traffic shaping, without any modifications to either TCP sender or receiver sides. Since it is transparent to TCP end-points, it can be easily deployed on TCP sender end-hosts (EHs), gateway (GW) routers, or access points (APs). ORTA opens a door for network providers to use per-packet load balancing.The proposed ORTA algorithm is implemented and tested in NS-2. The results show that ORTA can prevent TCP performance decrease when per-packet load balancing is used.
Congestion diffusion resulting from the coupling by resource competing is a kind of typical failure propagation in network systems. The existing models of failure propagation mainly focused on the coupling by direct physical connection between nodes, the most efficiency path, or dependence group, while the coupling by resource competing is ignored. In this paper, a model of network congestion diffusion with resource competing is proposed. With the analysis of the similarities to resource competing in biomolecular network, the model describing the dynamic changing process of biomolecule concentration based on titration mechanism provides reference for our model. Then the innovation on titration mechanism is proposed to describe the dynamic changing process of link load in networks, and a novel congestion model is proposed. By this model, the global congestion can be evaluated. Simulations show that network congestion with resource competing can be obtained from our model.
In recent years, integration of Passive Optical Net-work(PON) and WiMAX (Worldwide Interoperability Microwave Access Network) network is attracting huge interest among many researchers. The continuous demand for large bandwidths with wider coverage area are the key drivers to this technology. This integration has led to high speed and cost efficient solution for internet accessibility. This paper investigates the issues related to traffic grooming, routing and resource allocation in the hybrid networks. The Elastic Optical Network forms Backbone and is integrated with WiMAX. In this novel approach, traffic grooming is carried out using light trail technique to minimize the bandwidth blocking ratio and also reduce the network resource consumption. The simulation is performed on different network topologies, where in the traffic is routed through three modes namely the pure Wireless Network, the Wireless-Optical/Optical-Wireless Network, the pure Optical Network keeping the network congestion in mind. The results confirm reduction in bandwidth blocking ratio in all the given networks coupled with minimum network resource utilization.
We have proposed the Media Access Control method based on the Synchronization Phenomena of coupled oscillators (SP-MAC) to improve a total throughput of wireless terminals connected to a Access Point. SP-MAC can avoid the collision of data frames that occur by applying Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) based on IEEE 802.11 in Wireless local area networks (WLAN). Furthermore, a new throughput guarantee control method based on SP-MAC has been proposed. This method enable each terminal not only to avoid the collision of frames but also to obtain the requested throughput by adjusting the parameters of SP-MAC. In this paper, we propose a new throughput control method that realizes the fairness among groups of terminals that use the different TCP versions, by taking the advantage of our method that is able to change acquired throughput by adjusting parameters. Moreover, we confirm the effectiveness of the proposed method by the simulation evaluation.
We present ctrlTCP, a method to combine the congestion controls of multiple TCP connections. In contrast to the previous methods such as the Congestion Manager, ctrlTCP can couple all TCP flows that leave one sender, traverse a common bottleneck (e.g., a home user's thin uplink) and arrive at different destinations. Using ns-2 simulations and an implementation in the FreeBSD kernel, we show that our mechanism reduces queuing delay, packet loss, and short flow completion times while enabling precise allocation of the share of the available bandwidth between the connections according to the needs of the applications.
Communication between two Internet hosts using parallel connections may result in unwanted interference between the connections. In this dissertation, we propose a sender-side solution to address this problem by letting the congestion controllers of the different connections collaborate, correctly taking congestion control logic into account. Real-life experiments and simulations show that our solution works for a wide variety of congestion control mechanisms, provides great flexibility when allocating application traffic to the connections, and results in lower queuing delay and less packet loss.
In wireless sensor networks (WSNs), congestion control is a very essential region of concern. When the packets that are coming get increased than the actual capacity of network or nodes results into congestion in the network. Congestion in network can cause reduction in throughput, increase in network delay, and increase in packet loss and sensor energy waste. For that reason, new complex methods are mandatory to tackle with congestion. So it is necessary to become aware of congestion and manage the congested resources in wireless sensor networks for enhancing the network performance. Diverse methodologies for congestion recognition and prevention have been presented in the previous couple of years. To handle some of the problems, this paper exhibits a new technique for controlling the congestion. An efficient and reliable routing protocol (ERRP) based on bio inspired algorithms is introduced in this paper for solving congestion problem. In the proposed work, a way is calculated to send the packets on the new pathway. The proposed work has used three approaches for finding the path which results into a congestion free path. Our analysis and simulation results shows that our approach provides better performance as compared to previous approaches in terms of throughput, packet loss, delay etc.
Denial of service (DoS) is a process of injecting malicious packets into the network. Intrusion detection system (IDS) is a system used to investigate malicious packets in the network. Software-defined network (SDN) physically separates control plane and data plane. The control plane is moved to a centralized controller, and it makes a decision in the network from a global view. The combination between IDS and SDN allows the prevention of malicious packets to be more efficient due to the advantage of the global view in SDN. IDS needs to communicate with switches to have an access to all end-to-end traffic in the network. The high traffic in the link between switches and IDS results in congestion. The congestion between switches and IDS delays the detection and prevention of malicious traffic. To address this problem, we propose a historical database (Hdb), a scheme to reduce the traffic between switches and IDS, based on the historical information of a sender. The simulation shows that in the average, 54.1% of traffic mirrored to IDS is reduced compared to the conventional schemes.
Originally implemented by Google, QUIC gathers a growing interest by providing, on top of UDP, the same service as the classical TCP/TLS/HTTP/2 stack. The IETF will finalise the QUIC specification in 2019. A key feature of QUIC is that almost all its packets, including most of its headers, are fully encrypted. This prevents eavesdropping and interferences caused by middleboxes. Thanks to this feature and its clean design, QUIC is easier to extend than TCP. In this paper, we revisit the reliable transmission mechanisms that are included in QUIC. More specifically, we design, implement and evaluate Forward Erasure Correction (FEC) extensions to QUIC. These extensions are mainly intended for high-delays and lossy communications such as In-Flight Communications. Our design includes a generic FEC frame and our implementation supports the XOR, Reed-Solomon and Convolutional RLC error-correcting codes. We also conservatively avoid hindering the loss-based congestion signal by distinguishing the packets that have been received from the packets that have been recovered by the FEC. We evaluate its performance by applying an experimental design covering a wide range of delay and packet loss conditions with reproducible experiments. These confirm that our modular design allows the protocol to adapt to the network conditions. For long data transfers or when the loss rate and delay are small, the FEC overhead negatively impacts the download completion time. However, with high packet loss rates and long delays or smaller files, FEC allows drastically reducing the download completion time by avoiding costly retransmission timeouts. These results show that there is a need to use FEC adaptively to the network conditions.
Multi-tag identification technique has been applied widely in the RFID system to increase flexibility of the system. However, it also brings serious tags collision issues, which demands the efficient anti-collision schemes. In this paper, we propose a Multi-target tags assignment slots algorithm based on Hash function (MTSH) for efficient multi-tag identification. The proposed algorithm can estimate the number of tags and dynamically adjust the frame length. Specifically, according to the number of tags, the proposed algorithm is composed of two cases. when the number of tags is small, a hash function is constructed to map the tags into corresponding slots. When the number of tags is large, the tags are grouped and randomly mapped into slots. During the tag identification, tags will be paired with a certain matching rate and then some tags will exit to improve the efficiency of the system. The simulation results indicate that the proposed algorithm outperforms the traditional anti-collision algorithms in terms of the system throughput, stability and identification efficiency.
This paper focuses on optimizing the sigmoid filter for detecting Low-Rate DoS attacks. Though sigmoid filter could help for detecting the attacker, it could severely affect the network efficiency. Unlike high rate attacks, Low-Rate DoS attacks such as ``Shrew'' and ``New Shrew'' are hard to detect. Attackers choose a malicious low-rate bandwidth to exploit the TCP's congestion control window algorithm and the re-transition timeout mechanism. We simulated the attacker traffic by editing using NS3. The Sigmoid filter was used to create a threshold bandwidth filter at the router that allowed a specific bandwidth, so when traffic that exceeded the threshold occurred, it would be dropped, or it would be redirected to a honey-pot server, instead. We simulated the Sigmoid filter using MATLAB and took the attacker's and legitimate user's traffic generated by NS-3 as the input for the Sigmoid filter in the MATLAB. We run the experiment three times with different threshold values correlated to the TCP packet size. We found the probability to detect the attacker traffic as follows: the first was 25%, the second 50% and the third 60%. However, we observed a drop in legitimate user traffic with the following probabilities, respectively: 75%, 50%, and 85%.
With the developing understanding of Information Security and digital assets, IT technology has put on tremendous importance of network admission control (NAC). In NAC architecture, admission decisions and resource reservations are taken at edge devices, rather than resources or individual routers within the network. The NAC architecture enables resilient resource reservation, maintaining reservations even after failures and intra-domain rerouting. Admission Control Networks destiny is based on IP networks through its Security and Quality of Service (QoS) demands for real time multimedia application via advance resource reservation techniques. To achieve Security & QoS demands, in real time performance networks, admission control algorithm decides whether the new traffic flow can be admitted to the network or not. Secure allocation of Peer for multimedia traffic flows with required performance is a great challenge in resource reservation schemes. In this paper, we have proposed our model for VoIP networks in order to achieve security services along with QoS, where admission control decisions are taken place at edge routers. We have analyzed and argued that the measurement based admission control should be done at edge routers which employs on-demand probing parallel from both edge routers to secure the source and destination nodes respectively. In order to achieve Security and QoS for a new call, we choose various probe packet sizes for voice and video calls respectively. Similarly a technique is adopted to attain a security allocation approach for selecting an admission control threshold by proposing our admission control algorithm. All results are tested on NS2 based simulation to evalualate the network performance of edge router based upon network admission control in VoIP traffic.
Traffic normalization, i.e. enforcing a constant stream of fixed-length packets, is a well-known measure to completely prevent attacks based on traffic analysis. In simple configurations, the enforced traffic rate can be statically configured by a human operator, but in large virtual private networks (VPNs) the traffic pattern of many connections may need to be adjusted whenever the overlay topology or the transport capacity of the underlying infrastructure changes. We propose a rate-based congestion control mechanism for automatic adjustment of traffic patterns that does not leak any information about the actual communication. Overly strong rate throttling in response to packet loss is avoided, as the control mechanism does not change the sending rate immediately when a packet loss was detected. Instead, an estimate of the current packet loss rate is obtained and the sending rate is adjusted proportionally. We evaluate our control scheme based on a measurement study in a local network testbed. The results indicate that the proposed approach avoids network congestion, enables protected TCP flows to achieve an increased goodput, and yet ensures appropriate traffic flow confidentiality.
Cloud-based communications system is now widely used in many application fields such as medicine, security, environment protection, etc. Its use is being extended to the most demanding services like multimedia delivery. However, there are a lot of constraints when cloud-based sensor networks use the standard IEEE 802.15.3 or IEEE 802.15.4 technologies. This paper proposes a channel characterization scheme combined to a cross-layer admission control in dynamic cloud-based multimedia sensor networks to share the network resources among any two nodes. The analysis shows the behavior of two nodes using different network access technologies and the channel effects for each technology. Moreover, the existence of optimal node arrival rates in order to improve the usage of dynamic admission control when network resources are used is also shown. An extensive simulation study was performed to evaluate and validate the efficiency of the proposed dynamic admission control for cloud-based multimedia sensor networks.
Wireless mesh networks (WMNs) are attracting more and more real time applications. This kind of applications is constrained in terms of Quality of Service (QoS). Existing works in this area are mostly designed for mobile ad hoc networks, which, unlike WMNs, are mainly sensitive to energy and mobility. However, WMNs have their specific characteristics (e.g. static routers and heavy traffic load), which require dedicated QoS protocols. This paper proposes a novel traffic regulation scheme for multimedia support in WMNs. The proposed scheme aims to regulate the traffic sending rate according to the network state, based on the buffer evolution at mesh routers and on the priority of each traffic type. By monitoring the buffer evolution at mesh routers, our scheme is able to predict possible congestion, or QoS violation, early enough before their occurrence; each flow is then regulated according to its priority and to its QoS requirements. The idea behind the proposed scheme is to maintain lightly loaded buffers in order to minimize the queuing delays, as well as, to avoid congestion. Moreover, the regulation process is made smoothly in order to ensure the continuity of real time and interactive services. We use the interval type-2 fuzzy logic system (IT2 FLS), known by its adequacy to uncertain environments, to make suitable regulation decisions. The performance of our scheme is proved through extensive simulations in different network and traffic load scales.