Multiple TCP Connections Research Articles

Adaptive Quality-of-Service Allocation Scheme for Improving Video Quality over a Wireless Network

The need to ensure the quality of video streaming transmitted over wireless networks is growing every day. Video streaming is typically used for applications that are sensitive to poor quality of service (QoS) due to insufficient bandwidth, packet loss, or delay. These challenges hurt video streaming quality since they affect throughput and packet delivery of the transmitted video. To achieve better video streaming quality, throughput must be high, with minimal packet delay and loss ratios. A current study, however, found that the adoption of the adaptive multiple TCP connections (AM-TCP), as a transport layer protocol, improves the quality of video streaming by increasing throughput and lowering packet delay and loss on high QoS wireless networks. The main objective of this study is to develop a Quality of Service (QoS) method for enhancing video streaming quality over wireless networks. For that, this paper proposes a bandwidth aggregation adaptive multiple TCP-connection (BAM-TCP) scheme to enhance the video streaming quality. BAM-TCP combines AM-TCP with a bandwidth aggregation (BAG) method to aggregate bandwidth from numerous connections. Simulation experiments demonstrate that the BAM-TCP method enhances throughput while minimizing packet loss ratio and end-to-end delay.

Stochastic Optimization of Multipath TCP for Energy Minimization and Network Stability over Heterogeneous Wireless Network

Multipath Transport Control Protocol (MPTCP) is a transport layer protocol that enables multiple TCP connections across various paths. Due to path heterogeneity, it incurs more energy in a multipath wireless network. Recent work presents a set of approaches described in the literature to support systems for energy consumption in terms of their performance, objectives and address issues based on their design goals. The existing solutions mainly focused on the primary system model but did not discourse the overall system performance. Therefore, this paper capitalized a novel stochastically multipath scheduling scheme for data and path capacity variations. The scheduling problem formulated over MPTCP as a stochastic optimization, whose objective is to maximize the average throughput, avoid network congestion, and makes the system more stable with greater energy efficiency. To design an online algorithm that solves the formulated problem over the time slots by considering its min-drift-plus penalty form. The proposed solution was examined under extensive simulations to evaluate the anticipated stochastic optimized MPTCP (so-MPTCP) outcome and compared it with the base MPTCP and the energy-efficient MPTCP (eMPTCP) protocols. Simulation results justify the proposed algorithm''s credibility by achieving remarkable improvements, higher throughput, reduced energy costs, and lower-end to end delay.

Ns-2'de Farklı Arka Plan TCP Trafik Koşulları Altında VoIP'nin Kullanıcı Düzeyinde Performans Değerlendirmesi

Voice over IP (VoIP) is gaining more and more importance and displaces the traditional telephony. For example, more than 300 million monthly active users worldwide use the popular VoIP application "Skype". However, a big problem in the VoIP environment is the voice quality. The purpose of this study is to investigate the effects of background TCP traffic on perceived voice quality of a VoIP conversation at the user-level using the G.711 codec. Two different playout buffering policies including static buffering and optimal buffering have been applied by the VoIP server. For comparison purposes, the same experiments have also been repeated when no playout buffering policy has been applied by the VoIP server. A three -hop network topology consisting of a source, a transit, and a destination subnetwork was simulated whereas the end-to-end capacity of the entire network was limited by a 1.5 Mbps Asymmetric Digital Subscriber Line ( ADSL) link. Multiple simultaneous TCP connections with different segment sizes were established to simulate the various conditions of background traffic. By using the ns2voip framework, an enhancement to the popular Network Simulator 2 (ns-2), extensive simulation experiments for analyzing the VoIP user-level performance have been carried out and the voice quality has been evaluated by calculating the Mean Opinion Score (MOS). The results show that the voice quality is strongly negatively affected by background TCP traffic, even in the presence of a single TCP flow with 1500 Byte segments. Also, the size of background TCP segments significantly influences the achievable MOSs of VoIP conversations. However, it has also been observed that aggregating multiple speech frames into a single IP packet can increase the MOS. Particularly, depending on the number and segment size of background TCP flows, aggregation of the optimal number of voice frames into the same IP packet improves the MOSs up to 14.61% over a 1.5 Mbps ADSL link.

PABO: Mitigating congestion via packet bounce in data center networks

In today’s data center, a diverse mix of throughput-sensitive long flows and delay-sensitive short flows are commonly presented. However, commodity switches used in a typical data center network are usually shallow-buffered for the sake of reducing queueing delay and deployment cost. The direct outcome is that the queue occupation by long flows could potentially block the transmission of delay-sensitive short flows, leading to degraded performance. Congestion can also be caused by the synchronization of multiple TCP connections for short flows, as typically seen in the partition/aggregate traffic pattern. The congestion is usually transient and any end-device intervention through the timeout-based pathway would result in suboptimal performance. While multiple end-to-end transport-layer solutions have been proposed, none of them has tackled the real challenge: reliable transmission in the network. In this paper, we fill this gap by presenting PABO — a novel link-layer design that can mitigate congestion by temporarily bouncing packets to upstream switches. PABO’s design fulfills the following goals: (i) providing per-flow based flow control on the link layer, (ii) handling transient congestion without the intervention of end devices, and (iii) gradually back propagating the congestion signal to the source when the network is not capable to handle the congestion. We present the detailed design of PABO and complete a proof-of-concept implementation. We discuss the impact of system parameters on packet out-of-order delivery and conduct extensive experiments to prove the effectiveness of PABO. We examine the basic properties of PABO using a tree-based topology, and further evaluate the overall performance of PABO using a realistic Fattree topology for data center networks. Experiment results show that PABO can provide prominent advantage of mitigating transient congestions and can achieve significant gain on flow completion time.

Game of protocols: Is QUIC ready for prime time streaming?

SummaryQuick User Datagram Protocol (UDP) Internet Connections (QUIC) is an experimental and low‐latency transport protocol proposed by Google, which is still being improved and specified in the Internet Engineering Task Force (IETF). The viewer's quality of experience (QoE) in HTTP adaptive streaming (HAS) applications may be improved with the help of QUIC's low‐latency, improved congestion control, and multiplexing features. We measured the streaming performance of QUIC on wireless and cellular networks in order to understand whether the problems that occur when running HTTP over TCP can be reduced by using HTTP over QUIC. The performance of QUIC was tested in the presence of network interface changes caused by the mobility of the viewer. We observed that QUIC resulted in quicker start of media streams, better streaming, and seeking experience, especially during the higher levels of congestion in the network and had a better performance than TCP when the viewer was mobile and switched between the wireless networks. Furthermore, we measured QUIC's performance in an emulated network that had a various amount of losses and delays to evaluate how QUIC's multiplexing feature would be beneficial for HAS applications. We compared the performance of HAS applications using multiplexing video streams with HTTP/1.1 over multiple TCP connections to HTTP/2 over one TCP connection and to QUIC over one UDP connection. To that effect, we observed that QUIC provided better performance than TCP on a network that had large delays. However, QUIC did not provide a significant improvement when the loss rate was large. Finally, we analyzed the performance of the congestion control mechanisms implemented by QUIC and TCP, and tested their ability to provide fairness among streaming clients. We found that QUIC always provided fairness among QUIC flows, but was not always fair to TCP.

Utilizing Connection Usage Characteristics for Faster Web Data Transport

The quest for faster data transport to improve web user experience is ongoing and attempts are conducted from various fronts to realize it. On top of improving user experience, the implications of improving web data transport are also on the energy efficiency of wireless devices as well as user retention rates of service providers. HTTP/1.x allow the opening of multiple TCP connections per server and then using those connections for fetching multiple web objects through the use of HTTP pipelining. With the advent of HTTP/2.0, multiplexing is done inside a single connection to fetch multiple objects. In this paper, we analyze the TCP connections between the browser and the servers and examine their characteristics. We describe how an enhanced TCP variant can take advantage of data transport connection patterns. We show the benefits that enhanced TCP system can bring with the understanding of connection usage patterns. We find that such transport protocol can have effect in the page idle times as well as the connection concurrency during web page transfer. The results show significant improvement of page load times for both encryption heavy and unencrypted pages. We discuss the effect of the transport protocol on object transfer, connection duration, idle times during the page load, connections, and concurrency of flows that cumulate into page load times.

TCP Performance over Wi-Fi: Joint Impact of Buffer and Channel Losses

We propose an analytical model for a Wi-Fi network acting as a last-mile Internet access with multiple long-lived TCP connections on both the up and down links. Our model considers the joint impact of buffer losses at the access point, contention at the medium access control layer, and packet losses due to the wireless channel being erroneous. We show that the model accurately quantifies the probability of an arbitrary TCP packet being discarded, and the total throughput obtained on the up and down links. Furthermore, quantitative insights can be gained into the throughput that long-lived TCP flows achieve under the joint impact of all aforementioned types of losses. In particular, we find that the wireless channel errors and buffer overflows both lead to throughput unfairness, but that they do so in the opposite direction on the up and down links, respectively. We demonstrate that this insight can be exploited so as to significantly mitigate the throughput unfairness without compromising the total obtainable network throughput.

Server Load – Balancing using Resources of Clustering Technique

Load Balancing is one of the most important issues for clustered servers. Load Balancing is done on the basis of server traffic. The proposed algorithm uses user-session history as the other constraint which is very useful for resource management over the server and easy access for the client. Collection of the data of a particular client from the last user-session, groups the applications that are likely to be opened in a specific time frame. This will provide easy access to the client and will reduce resource wastage on the server by using cluster technology. The primary clusters will be based on the bandwidth of the client’s internet bandwidth, which will map to different servers. Whereas secondary clusters will be based on the similarity of application-usage in the past user-session. This approach can reduce the unnecessary searching time. Cluster management will provide better approachability towards the server also the bandwidth based load balancing will lead to the minimum bandwidth wastage. Keywords Clustering, Load Balancing, Scalability, Client-Server, Resource Management, Cluster Creation, Cluster allocation, Cluster Reallocation, Cluster Management and Cloud Computing 1. INTRODUCTION The inspiration behind any Client-Server technology, arguably is its capacity to handle n no. of transactions in a given span of Time. Today, in the much credited dynamic environment “Scalability” is the concern for modern day Computer Scientist and Algorithm Pundits. You have only few ways left to deal with performance issues. 1. Either you are forced to upgrade hardware resources or 2. A load balancing [1] algorithm to find smart costefficient ways to fetch the user their desired resources without compromising Quality of Service. Coming directly to Load balancing on the Server, various methods have been used in past to balance the same, yet some haven’t got desired outcome. Clustering technique had been used in the near past for better fetch and efficient server loadmanagement Clustering Technique [2] is exploratory technique used in many areas to analyze large data sets. Given a proper notion of similarity, they find groups of similar variables/objects by partitioning the data set in “similar subsets”. Typically, several metrics over which a distance measure can be defined are associated with points (named samples) in the data set. There usually are various Clustering techniques available. Most used amongst them are I) Hierarchical and II) Partitioning clustering techniques. User Session Deduction is one of most important criteria for Load-Balancing, given the technique to be used is Server-Clusters. User-Session Identification has been non-trivial task to be performed on-the-go real time. Applications such as telnet or ssh typically generate a single TCP connection per single user-session, whereas application layer protocols such as HTTP, IMAP/SMTP and X11 usually generate multiple TCP connections per user-session. User-session characterization genuinely allows researchers to build realistic scenarios when assessing the performance of a complex network via simulation. Earlier Problems included I) Missed user Sessions, as threshold value had to be statically stated to tell the server that it is running over capacity. Vague, as it may sound, any usersession could be missed once there is bottleneck of connections asking permission for a connection establishment. II) Possibility of a huge data lose once there is a system failure since no backup image available, only to add to the chaos, a main server did not have any backup server available to route traffic towards that server. III) Every returning User had to start-over every time he/she terminates the session, it led to huge delay in page requested and page Fetched. As every user was forced to visit the prespecified pages and not the user-desired page. Our Approach will ensure a proper user session deduction takes place. The Algorithms will make sure to time stamp every usersession so that in case of any missing sessions the timestamp will auto-correct it’s reading. Apart from these a smart Artificial Intelligence Engine will predict user behavior with respect to their usage at different point of times. This will help user to directly visit his/ her required page rather than logging in again and getting diverted once again to Homepage. Also a smart cluster allocation Engine and cluster management algorithm will work hand in hand and deliver realtime cluster allocation to users without any hassles whatsoever. Even Cluster Allocation depends upon various parameters viz. Bandwidth, Predicted userbehavior. International Journal of Computer Applications (0975 – 8887) Volume 125 – No.5, September 2015 26 Fig 1: Architecture Diagram

A performance enhanced transmission control protocol for inter-planetary networks

Activities in deep space have prompted researchers to develop new protocols for inter-planetary network. In this paper, we propose a performance enhanced inter-planetary transmission control protocol (IP-TCP) for deep space communication. IP-TCP opens multiple parallel TCP connections to transfer a number of TCP segments simultaneously over a single physical link. We have developed theoretical models to understand the performance of IP-TCP with regard to long RTT, asymmetric bandwidth and high bit error rate comparable to deep space environment. Through simulations we evaluate the proposed protocol considering RTT values typically seen in Mars exploration networks under different conditions of packet error rates. We observe that the IP-TCP outperforms few popular existing protocols which are also extensions of terrestrial TCP.

A novel fairness-aware parallel download scheme

Parallel download protocols that establish multiple TCP connections to distributed replica servers have the potential to reduce file download times and to achieve server-side load balancing. Unfortunately, parallel download protocols are inherently unfair towards single-flow downloads and may even starve them. This paper presents a fairness-aware parallel download protocol that allows the dynamic mitigation of throughput and fairness in distributed networks. The key novelty is a receiver-based framework that allows a dynamic adjustment of the congestion and rate control of each subconnection. The proposed Fairness-aware Control Coordinator (FCC) adjusts the priority of the subconnection between the TCP-fair rate (high priority) and the TCP-LP fair rate (low priority) according to the dynamics on the network without adding to the unfairness towards other downloads. Simulation results show the flexibility of the proposed protocol in balancing performance and fairness.

Harnessing the High Bandwidth of Multiradio Multichannel 802.11n Mesh Networks

There has been an increasing interest in deploying wireless mesh networks (WMNs) for communication and video surveillance purposes thanks to its low cost and ease of deployment. It is well known that a major drawback of WMN is multihop bandwidth degradation, which is primarily caused by contention and radio interference. The use of mesh nodes with multiple radios and channels has been regarded as a straightforward solution to the problem in the research community. However, we demonstrate in this paper through real-world experiments that such an approach cannot resolve the multihop TCP throughput degradation problem in IEEE 802.11n mesh networks. With extensive experimentation, we verify that the degradation is principally caused by the increase in TCP Round-Trip Time (RTT) when the number of hops increases. TCP throughput is fundamentally limited inversely by the RTT. We find that the multihop TCP throughput (up to five hops) when using 802.11n is no better than when using 802.11a, despite the much higher data rate 802.11n. We attempt to use multiple parallel TCP connections as a remedy to the problem, and it turns out that the wireless bandwidth can be fully utilized with a sufficient number of parallel streams. In general, our results give a key message that TCP tuning (e.g., setting the correct TCP buffers and use of parallel streams) is of paramount importance in high-bandwidth multihop wireless mesh networks that employ the latest wireless standards. These tuning techniques have to be implemented into commercial products to fully leverage the ever advancing wireless technologies to support the growing demand of multihop communications in wireless mesh networks.

Resource-aware and quality-fair video-streaming using multiple adaptive TCP connections

In this paper, we present a resource-aware and quality-fair video content sharing system. When a video sharing server has insufficient uplink bandwidth and needs to serve multiple video content sharing services via streaming or downloading to other client peers using TCP transport, each service shares the limited uplink bandwidth equitably, due to the fair sharing characteristics inherent in TCP. However this bandwidth fair sharing cannot always guarantee quality fairness among the services, due to the specific requirements for video-streaming services, such as the playout rate and the size of the playout buffer. In our system, the server uses multiple TCP connections adaptively, depending on the anticipated status of each client playout buffer, to guarantee the bandwidth of each video-streaming session. By guaranteeing the quality of each video-streaming session, without the quality loss of other service sessions, the proposed system can successfully achieve service quality fairness. Simulation results show that our proposed algorithm can dramatically enhance the quality of each streaming session and thus provide service quality fairness among simultaneous multiple heterogeneous video-streaming services and content download services.

GridFTP-APT: Automatic Parallelism Tuning Mechanism for GridFTP in Long-Fat Networks

In this paper, we propose an extension to GridFTP that optimizes its performance by dynamically adjusting the number of parallel TCP connections. GridFTP has been used as a data transfer protocol to effectively transfer a large volume of data in Grid computing. GridFTP supports a feature called parallel data transfer that improves throughput by establishing multiple TCP connections in parallel. However, for achieving high GridFTP throughput, the number of TCP connections should be optimized based on the network status. In this paper, we propose an automatic parallelism tuning mechanism called GridFTP-APT (GridFTP with Automatic Parallelism Tuning) that adjusts the number of parallel TCP connections according to information available to the Grid middleware. Through simulations, we demonstrate that GridFTP-APT significantly improves the performance of GridFTP in various network environments.

Multimedia streaming using multiple TCP connections

In recent years, multimedia applications over the Internet become increasingly popular. However, packet loss, delay, and time-varying bandwidth of the Internet have remained the major problems for multimedia streaming applications. As such, a number of approaches, including network infrastructure and protocol, source and channel coding, have been proposed to either overcome or alleviate these drawbacks of the Internet. In this article, we propose the MultiTCP system, a receiver-driven, TCP-based system for multimedia streaming over the Internet. Our proposed algorithm aims at providing resilience against short term insufficient bandwidth by using multiple TCP connections for the same application. Our proposed system enables the application to achieve and control the desired sending rate during congested periods, which cannot be achieved using traditional TCP. Finally, our proposed system is implemented at the application layer, and hence, no kernel modification to TCP is necessary. We analyze the proposed system, and present simulation and experimental results to demonstrate its advantages over the traditional single-TCP-based approach.

Probe-Aided MulTCP

An aggregate congestion control mechanism, namely Probe-Aided MulTCP (PA-MulTCP), is proposed in this paper. It is based on MulTCP,a proposal for enabling an aggregate to emulate the behavior of multiple concurrent TCP connections. The objective of PA-MulTCP is to ensure the fair sharing of the bottleneck bandwidth between the aggregate and other TCP or TCP-friendly flows while keeping lightweightness and responsiveness. Unlike MulTCP, there are two congestion window loops in PA-MulTCP, namely the probe window loop and the adjusting window loop.The probe window loop constantly probes the congestion situation and the adjusting window loop dynamically adjusts the congestion window size for the arriving and departing flows within the aggregate Our simulations demonstrate that PA-MulTCP is more stable and fairer than MulTCP over a wide range of the weight N in steady conditions as well as in varying congestion conditions. PA-MulTCP is also responsive to flow arrival/departure and thus reduces the latency of short-lived transfers. Furthermore, PA-MulTCP is lightweight, since it enjoys above advantages at the cost of only an extra probe window loop, which has a marginal influence on the implementation complexity. Finally, the design of PA-MulTCP decouples the congestion management from the other functionalities in the aggregate flow management. As a result, PA-MulTCP could be potentially applied to a wider range of scenarios, e.g. wireless TCP proxies, edge-to-edge overlays, QoS provisioning and mass data transport. TCP, Fairness, Congestion Control, Aggregate Flow Management

Exploring TCP Parallelisation for performance improvement in heterogeneous networks

TCP Parallelisation uses a set of parallel (modified or standard) TCP connections to transfer data for an application process. With standard TCP connections, TCP Parallelisation has been used to effectively utilise bandwidth for data intensive applications over high bandwidth-delay product (BDP) networks. On the other hand, it has been argued that a single TCP connection with proper modification can emulate and capture the robustness of TCP Parallelisation and thus can well replace TCP Parallelisation. A typical example is HSTCP, which can emulate the behaviour of a set of multiple standard TCP connections. However, through the analysis and the proposed use of TCP Parallelisation in this paper, we found that TCP Parallelisation has its unique merits, which cannot be emulated by a single-connection based approach. As a result, the single-connection based approach (such as HSTCP) may not be able to achieve the same effects as TCP Parallelisation, especially in heterogeneous networks and highly dynamic environments (e.g., high BDP networks combined with wireless access links). In this paper, we demonstrate some of these unique merits of TCP Parallelisation (namely, localisation of disturbances such as Fast Recovery, loss synchronisation between parallel connections and self- healing) over single-connection based approach in the heterogeneous environment.

Comparison of FEC types with regard to the efficiency of TCP connections over AWGN satellite channels

Optimizing the end-to-end throughput of a TCP connection (goodput) over geostationary satellite links is a challenging research topic. This is because the high delay-bandwidth product, together with a non-negligible random loss of packets, is a condition that considerably differs from the environments TCP was originally designed for. As a result, TCP performance is significantly impaired by the channel bit error rate. The literature is full of suggestions for improving TCP goodput, most based on modifications of the protocol itself. We only investigated the application of different FEC (forward error correction) types for TCP goodput optimization, leaving the end-to-end protocol unaltered. Using a method midway between analysis and simulation to evaluate the goodput of TCP long-lived connections, we first studied the influence of packet loss rate, introduced by errors on the channel, on the TCP goodput. We showed that, in some cases, the packet loss rate does not need to be negligible with respect to that caused by congestion, as it is widely-held opinion. We then applied physical-level FEC techniques, such as convolutional encoding/Viterbi decoding, Reed Solomon, link-level erasure codes and their combinations, over a wide field of signal to noise conditions of the satellite channel. For each FEC type, we found the FEC rate that maximizes the TCP goodput, in each channel condition. We finally compared the results of all FECs used between them, and presented the case of multiple TCP connections sharing the same link as well

Mean field convergence of a model of multiple TCP connections through a buffer implementing RED

RED (Random Early Detection) has been suggested when multiple TCP sessions are multiplexed through a bottleneck buffer. The idea is to detect congestion before the buffer overflows by dropping or marking packets with a probability that increases with the queue length. The objectives are reduced packet loss, higher throughput, reduced delay and reduced delay variation achieved through an equitable distribution of packet loss and reduced synchronization. Baccelli, McDonald and Reynier [Performance Evaluation 11 (2002) 77–97] have proposed a fluid model for multiple TCP connections in the congestion avoidance regime multiplexed through a bottleneck buffer implementing RED. The window sizes of each TCP session evolve like independent dynamical systems coupled by the queue length at the buffer. The key idea in [Performance Evaluation 11 (2002) 77–97] is to consider the histogram of window sizes as a random measure coupled with the queue. Here we prove the conjecture made in [Performance Evaluation 11 (2002) 77–97] that, as the number of connections tends to infinity, this system converges to a deterministic mean-field limit comprising the window size density coupled with a deterministic queue.

De-layered grid storage server

Networks have become faster and disks have become fatter at a pace that, despite Moore's law, CPU developments have simply not been been able to keep up with. We present a Grid Storage Server which is capable of scaling up to meet the "terabit-terabyte" demands of very large scale grid computation applications with large data sets.The Grid Storage Server is implemented almost completely in silicon, whether FPGA or ASIC; the fast-path of this server does not use a CPU or von Neumann style (instruction/data) machine. Instead, multiple layers of a protocol stack are compiled into a hardware engine that processes all layers concurrently on-chip. This "serverless" design allows it to scale up to match "terabit" network speeds and "terabyte" disk capacities enabling large scale grid applications.At a price-point a small percent of that of a server-based design, the Grid Server incorporates a standards compliant high-performance TCP stack that can saturate 40Gbps using a single or multiple TCP connections. The current design directly attaches to a storage array of 48TB capacity. The storage array is organized with a fault-tolerant RAID for performance and reliability; the RAID configuration is adaptive and can be tuned to conflicting application needs. As a bonus, because the control-plane in the silicon-based TCP engine has very low jitter, the protocol engine has mechanisms for nanosecond precision clock synchronisation across very large distances, thus, for the first time, enabling transcontinental real-time and temporal grid computation and database applications.

Video Data Transmission Protocol "SVFTP" Using Multiple TCP Connections and Its Application

This paper presents a novel data transmission protocol SVFTP, which enables high-speed and error-free video data transmission over IP networks. A video transmission system based on SVFTP is also presented. While conventional protocols are designed for file transmission, SVFTP focuses on video data as a continuous media. In order to fit a flexible video transmission system, SVFTP achieves higher throughput on the long distance link as well as transmission interruption/resumption and progressive download and play back. In addition, a rate shaping mechanism for SVFTP is introduced in order to control greediness and burst traffic of multiple-TCP sessions. Laboratory and field transmission experiments show that SVFTP achieves high performance and functionality.