Transmission Control Protocol Packet Research Articles

Performance and Scalability Analysis of SDN-Based Large-Scale Wi-Fi Networks

The Software-Defined Networking (SDN) paradigm is one that is utilized frequently in data centers. Software-Defined Wireless Networking, often known as SDWN, refers to an environment in which concepts from SDN are implemented in wireless networks. The SDWN is struggling with challenges of scalability and performance as a result of the growing number of wireless networks in its coverage area. It is thought that SDN techniques, such as Mininet-Wi-Fi and Ryu Controller for wireless networks, can overcome the problems with scalability and performance. Existing Wi-Fi systems do not provide SDN execution to end clients, which is one reason why the capability of Wi-Fi is restricted on SDN architecture. Within the scope of this study, we analyzed Wi-Fi networks operating on SDN using the Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). By utilizing a testbed consisting of Ryu Controller and Mininet-Wi-Fi, we were able to test Wi-Fi over SDN and evaluate its performance in addition to its scalability. When evaluating the performance of a network, we take into account a number of different metrics, such as bandwidth, round-trip time, and jitter. In order to assess the level of performance, the SDN-based Wi-Fi controller Ryu is linked to an increasing number of access points (1, 2, 3, and 4) as well as stations (10, 30, 50, and 100). The experimental findings obtained using Mininet-Wi-Fi indicate the scalability and dependability of the network performance provided by the SDN Wi-Fi network controller Ryu in an SDN environment. In addition, the round-trip time for TCP packets grows proportionally with the number of hops involved. A single access point is capable of simultaneously supporting up to fifty people at once.

Detection of Crime Patterns in Digital Forensic Investigation to Trace the Adversaries

The use of the internet has increased significantly over the past couple of years. Access to the internet has become so common that a person without computer knowledge can also use this facility easily. This ease of availability has provided a lot of benefits to society but on the other hand misuse of the internet for personal or corporate benefits is also increasing. To prosecute cybercriminals and make some lawful checks on everyone's digital activities, digital forensic science comes into the light. In this context, we developed a new framework that improves the digital forensic investigation process. This research paper proposes a method in which we can identify the illegal activities and trace the adversaries. We capture the TCP (Transmission Control Protocol) packets from the servers and workstations. This data collected from the TCP log is stored in the database and preprocessed to eliminate redundant data. Furthermore, the database also contains past data. The proposed framework has three major processes collection of TCP packets, storing and preprocessing of collected data in a database, and mining of the pattern through a digital forensic anomaly collection algorithm. For the evaluation of our proposed framework, we have developed a java based application. The results are shown in the form of reports and tables.

Low-High Burst: A Double Potency Varying-RTT Based Full-Buffer Shrew Attack Model

The full-buffer Shrew (FB-Shrew) denial of service (DoS) attack is a variant of the classic Shrew attack that exploits the congestion control mechanism of transmission control protocol (TCP). Here, an attacker sends a high-rate burst of attack packets only after the router buffer is filled with TCP packets, causing the router to drop legitimate packets, and forcing the retransmission of TCP packets. As such, an FB-Shrew attack can cause maximum damage with minimum resources. In this paper, we challenge an assumption of constant round trip time adopted in the original FB-Shrew model. As a result, this model fails to achieve its expected attack effect. In response, we analyze the TCP congestion window and queue behaviors to develop two low-high burst models for maximizing the potency of the FB-Shrew attack. Model 1 is designed to achieve the attack effect expected of the original model. Then, the attack potency of Model 1 is enhanced by simply adjusting the starting time of the attack burst to form Model 2. Mode 1 only exploits the retransmission timeout (RTO) mechanism. Model 2 takes advantage of both the RTO mechanism and the fast retransmission mechanism. In this way, Model 2 further slows down the growth of the congestion window and extends the attack period. A combination of theoretical analyses and simulations are adopted to first validate the proper functioning and effectiveness of the two models for a standard network configuration, and then we assess their attack performances with variations in different network parameters. Our performance assessment demonstrates that one attack unit of Model 2 damages almost twice the number of TCP units as one attack unit of Model 1, which represents an increase in attack potency of nearly 200 percent. The present study provides an expanded basis to explore FB-Shrew attack patterns that may be utilized by attackers. Moreover, the damage that could be inflicted by such attack and the extent to which defense strategies are capable of mitigating the attack's impact could be assessed more precisely by defenders.

Green TCP Transmission Over Cognitive Radio Networks

The last decade has witnessed an unprecedented growth of mobile data traffic and energy consumption in wireless networks. Energy consumption can incur greenhouse gas emission and high expenditure for maintaining wireless networks. To reduce the energy consumption, some previous works have so far considered improving the energy efficiency of various wireless networks. However, most of them largely ignore the energy efficiency of delivering transmission control protocol (TCP) applications, one of the most popular transmission protocols used for many Internet applications. Even though a couple of previous works have studied the problem of energy-efficient TCP transmission, they do not optimize its energy efficiency by taking into account the interweaving of parameters deployed at different layers of the protocol stack. Hence, they may not indeed lead to energy-efficient TCP transmission in wireless networks. In this paper, we consider a unified way of reducing energy consumption for transmitting TCP applications in cognitive radio networks (CRNs). In particular, we design an energy-efficient TCP transmission scheme for CRNs, which takes into account spectrum sensing, spectrum access, modulation and coding scheme, transmission power, and TCP packet length to enhance the energy efficiency. To validate our proposed scheme, we offer simulation results to show the significant energy efficiency improvement.

Exploiting Caching for Millimeter-Wave TCP Networks: Gain Analysis and Practical Design

Millimeter-wave (mmWave) communications are a key enabler of gigabit access using large spectrum resources. Most relevant works focus on the design of radio access networks (RANs), but a transport layer design tends to be overlooked despite its importance as a data supply to the RAN. Recent studies reported that transmission control protocol (TCP)—the de facto standard for the transport layer—is problematic in mm-Wave communications because the sending rate cannot be controlled due to the drastic channel status change between line-of-sight (LoS) and non-LoS (NLoS); this is known as TCP performance collapse problem (TPCP). The TPCP can be alleviated if a cache is deployed in a base station, such that TCP packets from servers are stored in the cache during NLoS and forwarded to the RAN immediately when the channel becomes LoS. This paper overcomes the TPCP by making full use of the mmWave spectrum via efficient cache utilization. Here, a caching gain is analyzed and defined as the end-to-end rate ratio between TCPs with and without cache. This results show that the gain is maximized by forwarding packets to the RAN whenever the wireless channel is available. Based on this, we propose a novel cache-enabled TCP framework called mmWave TCP (mmTCP) with two key functionalities. The first is a batch retransmission, where packets likely to be lost during NLoS are simultaneously retransmitted once the channel becomes LoS. The second is an online cache management, where the cache is being reallocated to different users given the current channel and cache status. The performance of the mmTCP is evaluated by a realistic network simulator–version 3 under different environments, such that 48% and 10% end-to-end rate improvements are made in single- and multi-user cases, respectively.

VoIP transmission mechanism based on TCP

As the widespread employment of firewalls on the Internet, user datagram protocol (UDP) based voice over Internet protocol (VoIP) system will be unable to transmit voice data. This paper proposed a novel method to transmit voice data based on transmission control protocol (TCP). The method adopts a disorder TCP transmission strategy, which allows discontinuous data packets in TCP queues read by application layer directly without waiting for the retransmission of lost data packets. A byte stream data boundary identification algorithm based on consistent overhead byte stuffing algorithm is designed to efficiently identify complete voice data packets from disordered TCP packets arrived so as to transmit the data to the audio processing module timely. Then, by implementing the prototype system and testing, we verified that the proposed algorithm can solve the high time delay, jitter and discontinuity problems in standard TCP protocol when transmitting voice data packets, which caused by its error control and retransmission mechanism. We proved that the method proposed in this paper is effective and practical.

Refining TCP's RTT dependent mechanism by utilizing link retransmission delay measurement in Wireless LAN

SummaryNetwork utilization by legacy transmission control protocol (TCP) is determined by its round trip time (RTT) dependent mechanism for the growth of its sending rate. The RTT does not always reflect the actual network conditions, especially in the case of wireless local area network (WLAN). Consequently, it influences the RTT‐dependent mechanism falsely. This paper proposes a novel cross‐layer scheme between TCP and the Institute of Electrical and Electronics Engineers (IEEE) 802.11 medium access control (MAC) that compensates for any diminished growth of TCP's sending rate because of the inclusion of non‐congestion delay component in RTT, if any. The proposed scheme has two refinements. The first refinement is at the MAC layer that notifies the additional propagation delay on account of link retransmission of a TCP packet. The second refinement is at the TCP layer in which the sender adapts the sending rate by relating the aforementioned additional propagation delay to the RTT estimation. To examine the benefits, the proposed scheme is evaluated after implementing it on top of representative TCP variants; Newreno and Westwood+ (TCPW). Simulation results showed that with the proposed scheme, network utilization by native TCP is improved by more than 35% in a dedicated WLAN environment, while the performance remained protected in a shared WLAN environment. Copyright © 2015 John Wiley & Sons, Ltd.

A packet-reordering solution to wireless losses in transmission control protocol

The wireless medium may cause substantial packet losses, rendering Transmission Control Protocol (TCP) inefficient. We propose a cross-layer solution by combining link-layer retransmission techniques and a solution for TCP packet reordering. It is costly to conduct link-layer retransmission with the constraint of orderly packet delivery. We require the link layer to provide reliable packet delivery, but without orderly delivery guarantee, thus transforming the problem of high packet error rates to the problem of packet reordering. The latter is dealt with by enhancing TCP with a solution for packet reordering. We justify our design by analyzing both general scenarios and the case of IEEE 802.11n. Our simulation results demonstrate that the proposed method is effective in improving TCP connection goodput in wireless networks.

Analytical modeling of TCP flow in wireless LANs

TCP (Transmission Control Protocol) is one of the protocols which are widely used in Internet environments. This paper presents an analytical model developed using the stochastic reward net (SRN) modeling technique for the TCP flow behavior in WLANs. The purpose of developing an analytical model is the evaluation of stationary TCP flow behavior. In this paper, we focus the attention on the TCP variant called TCP Reno which is implemented by most operating systems. The performance of TCP Reno is investigated through the SRN model. The model captures aspects such as congestion window (cwd) evolution, slow start and congestion avoidance phases, TCP packet transmissions, management of packet losses due to time-out, and due to generation of triple duplicate acknowledgements. The performance metrics such as throughput and delay of the TCP traffic are obtained using the SRN model. On the basis of these performance measures, the existing unfairness between the downstream and upstream flow at the AP in WLANs is also shown. The proposed analytical model is validated via simulation.

Lighter and faster simulations of high-speed IP networks

Two important factors for the realistic simulation of web-traffic are the heavy-tailed distributed file-sizes at the session level, and the transmission control protocol (TCP) at the transport level. The operation of a source that integrates both factors faces two major scalability problems: the computing resources required by each source put a limit on the total number of sources that can be simulated, and the number of discrete events in the context of high capacity networks leads to unacceptably long simulation times. We introduce a light weight traffic source (LWTS). It generates traffic statistically similar to the traffic produced by a realistic source. Similar to a realistic source, it generates heavy-tailed distributed files at the session layer. However, its transport operation is based on a pseudo-TCP (P-TCP) protocol that approximates TCP. The thin code of P-TCP makes LWTS ∼ 50 times less resource hungry as compared to a realistic traffic source. To solve the second scalability problem, we introduce a novel abstraction technique at the transport level: we send a whole window of TCP packets as one big packet. This abstraction results in reducing the number of discrete events leading to ∼ 28 times faster simulations.

Analyzing and Improving the TCP Flow Fairness in Wireless Ad Hoc Networks

The TCP(transmission control protocol) flow fairness problem in wireless multi-hop ad hoc networks is investigated. It is identified that the IEEE 802.11 DCF protocol can lead to severe unfairness, i.e., some nodes seize the whole channel capacity while others are starved. The TCP flow unfairness problem is first analyzed by simulation and it is found that the main reason lies in the unfairness of MAC(media access and control) protocol, while the TCP timeout mechanism makes the unfairness more severe. Then a probability model is used to quantitatively analyze the relation between the TCP unfairness and MAC parameters, which shows that the TCP flow fairness heavily correlates with the TCP packet length, and increasing the initial contention window of the MAC protocol can improve the fairness effectively. Based on these observations, a novel adaptive backoff algorithm is proposed, which dynamically adjusts the initial contention window according to the TCP packet length. Both theoretic analysis and simulation results show that the proposed algorithm can relieve the fairness problem to a

Wireless video streaming with TCP and simultaneous MAC packet transmission (SMPT)

AbstractVideo streaming is expected to account for a large portion of the traffic in future networks, including wireless networks. It is widely accepted that the user datagram protocol (UDP) is the preferred transport protocol for video streaming and that the transmission control protocol (TCP) is unsuitable for streaming. The widespread use of UDP, however, has a number of drawbacks, such as unfairness and possible congestion collapse, which are avoided by TCP. In this paper we investigate the use of TCP as the transport layer protocol for streaming video in a multi‐code CDMA cellular wireless system. Our approach is to stabilize the TCP throughput over the wireless links by employing a recently developed simultaneous MAC packet transmission (SMPT) approach at the link layer. We study the capacity, i.e. the number of customers per cell, and the quality of service for streaming video in the uplink direction. Our extensive simulations indicate that streaming over TCP in conjunction with SMPT gives good performance for video encoded in a closed loop, i.e. with rate control. We have also found that TCP is unsuitable (even in conjunction with SMPT) for streaming the more variable open‐loop encoded video. Copyright © 2004 John Wiley & Sons, Ltd.

Analytical Expression and PerformanceEvaluation of TCP Packet Loss Probability Over Geostationary Satellite

The letter contains the formulation of an analytical expression of the Transmission Control Protocol (TCP) packet loss probability over geostationary satellite channels on the basis of the state-of-the-art in the field. The expression obtained is function of the bandwidth available and is suited to be used in control mechanisms. It is compared with the results obtained through a satellite network emulator already validated in the literature.

Delay analysis for CBR traffic under static-priority scheduling

We examine the delay performance of packets from constant-bit-rate (CBR) traffic whose delay is affected by non-real-time traffic. The delay performance is analyzed by solving the nD/D/1 queue with vacations. We obtain an exact and closed form solution, hence obviating the need of any approximations or numerical Laplace inversions. We then provide various numerical results for low-bit-rate transmission links, in which packets can experience large delay. From our quantitative evaluation, we conclude that there exists an optimum packet size for a given delay bound. In extremely slow links, such as modem links, transmission control protocol (TCP) packets should be segmented to reduce the CBR delay. We therefore investigate the delay impact of TCP packet sizes as well.

Efficient demultiplexing of incoming TCP packets

When a transport protocol segment arrives at a receiving system, the receiving system must determine which application is to receive the protocol segment. This decision is typically made by looking up a protocol control block (PCB) for the segment, based on information in the segment's header. PCB lookup (a form of demultiplexing) is typically one of the more expensive operations in handling inbound protocol segment [Fe190]. Many recent protocol optimizations for the Transmission Control Protocol (TCP) [Jac88] assume that a large component of TCP traffic is bulk-data transfers, which result in packet trains [JR86]. If packet trains are prevalent, there is a high likelihood that the next TCP segment is en route to the same application (i.e. uses the same PCB) as the previous TCP segment. In these environments a very simple one-PCB cache like those used in BSD systems yields very high cache hit rates. However, there are classes of applications that do not form packet trains, and these applications do not perform well with a one-PCB cache. Examples of such applications are quite common in the area of heads-down data entry into on-line transaction-processing (OLTP) systems. OLTP systems make heavy use of computer communications networks and have large aggregate-packet-rates but are also characterized by large numbers of connections, low per-connection packet rates, and rather small packets. This combination of characteristics results in a very low incidence of packet trains. This paper uses a simple analytic approach to examine how different PCB lookup schemes perform with OLTP traffic. One scheme is shown to work an order of magnitude better for OLTP traffic than the one-PCB cache approach while still maintaining good performance for packet-train traffic.