A comparative study of aggregate TCP retransmission rates

Segment retransmissions are an essential tool in assuring reliable end-to-end communication in the Internet. Their crucial role in TCP design and operation has been studied extensively, in particular with respect to identifying non-conformant, buggy, or underperforming behaviour. However, TCP segment retransmissions are often overlooked when examining and analyzing large traffic traces. In fact, some have come to believe that retransmissions are a rare oddity, characteristically associated with faulty network paths, which, typically, tend to disappear as networking technology advances and link capacities grow. We find that this may be far from the reality experienced by TCP flows. We quantify aggregate TCP segment retransmission rates using publicly available network traces from six passive monitoring points attached to the egress gateways at large sites. In virtually half of the traces examined we observed aggregate TCP retransmission rates exceeding 1%, and of these, about half again had retransmission rates exceeding 2%. Even for sites with low utilization and high capacity gateway links, retransmission rates of 1%, and sometimes higher, were not uncommon. Our results complement, extend and bring up to date partial and incomplete results in previous work, and show that TCP retransmissions continue to constitute a non-negligible percentage of the overall traffic, despite significant advances across the board in telecommunications technologies and network protocols. The results presented are pertinent to end-to-end protocol designers and evaluators as they provide a range of "realistic" scenarios under which, and a "marker" against which, simulation studies can be configured and calibrated, and future protocols evaluated

Performance of Bursty World Wide Web (WWW) Sources over ABR

We model World Wide Web (WWW) servers and clients running over an ATM network using the ABR (available bit rate) service. The WWW servers are modeled using a variant of the SPECweb96 benchmark, while the WWW clients are based on a model by Mah. The traffic generated by this application is typically bursty, i.e., it has active and idle periods in transmission. A timeout occurs after given amount of idle period. During idle period the underlying TCP congestion windows remain open until a timeout expires. These open windows may be used to send data in a burst when the application becomes active again. This raises the possibility of large switch queues if the source rates are not controlled by ABR. We study this problem and show that ABR scales well with a large number of bursty TCP sources in the system.Comment: Submitted to WebNet `97, Toronto, November 9

Performance Evaluation of Bonding Techniques at Wireless 802.11n

Demands for high throughput bandwidth, encourage Point to Point wireless to serve more bandwidth for many kind application such as real-time multimedia services. We conduct research with testbed experimental at Point to Point topology use wireless 802.11n in LAB environment. The aim is to studying the performance that would be achieved by Interface Bonding and Channel Bonding techniques. We proposed experiment process and design to evaluate the performance of those techniques. Several parameters such as delay, jitter, data loss rate and throughput applied on TCP/UDP protocols with different Packet Sizes and Directional Traffic Flows. The results experiment showed that Channel Bonding has significant throughput improvement. However, the Interface Bonding results are far from expectation, we found that the performance is least than single normal link. As our finding we analyze it caused by Media Independent Interface (MII), and Scheduling Algorithm unable to work properly at wireless 802.11n using Point to Point connection

On the effectiveness of an optimization method for the traffic of TCP-based multiplayer online games

This paper studies the feasibility of using an optimization method, based on multiplexing and header compression, for the traffic of Massively Multiplayer Online Role Playing Games (MMORPGs) using TCP at the Transport Layer. Different scenarios where a number of flows share a common network path are identified. The adaptation of the multiplexing method is explained, and a formula of the savings is devised. The header compression ratio is obtained using real traces of a popular game and a statistical model of its traffic is used to obtain the bandwidth saving as a function of the number of players and the multiplexing period. The obtained savings can be up to 60 % for IPv4 and 70 % for IPv6. A Mean Opinion Score model from the literature is employed to calculate the limits of the multiplexing period that can be used without harming the user experience. The interactions between multiplexed and non-multiplexed flows, sharing a bottleneck with different kinds of background traffic, are studied through simulations. As a result of the tests, some limits for the multiplexing period are recommended: the unfairness between players can be low if the value of the multiplexing period is kept under 10 or 20 ms. TCP background flows using SACK (Selective Acknowledgment) and Reno yield better results, in terms of fairness, than Tahoe and New Reno. When UDP is used for background traffic, high values of the multiplexing period may stress the unfairness between flows if network congestion is severe

Overcoming TCP Degradation in the Presence of Multiple Intermittent Link Failures Utilizing Intermediate Buffering

It is well documented that assumptions made in the popular Transmission Control Protocol\u27s (TCP) development, while essential in the highly reliable wired environment, are incompatible with today\u27s wireless network realities in what we refer to as a challenged environment. Challenged environments severely degrade the capability of TCP to establish and maintain a communication connection with reasonable throughput. This thesis proposes and implements an intermediate buffering scheme, implemented at the transport layer, which serves as a TCP helper protocol for use in network routing equipment to overcome short and bursty, but regular, link failures. Moreover, the implementation requires no modifications to existing TCP implementations at communicating nodes and integrates well with existing routing equipment. In a simulated six-hop network with five modified routers supporting four challenged links, each with only 60% availability, TCP connections are reliably established and maintained, despite the poor link availability, whereas 94% fail using standard routing equipment, i.e., without the TCP helper protocol

Improving Large-Scale Network Traffic Simulation with Multi-Resolution Models

Simulating a large-scale network like the Internet is a challenging undertaking because of the sheer volume of its traffic. Packet-oriented representation provides high-fidelity details but is computationally expensive; fluid-oriented representation offers high simulation efficiency at the price of losing packet-level details. Multi-resolution modeling techniques exploit the advantages of both representations by integrating them in the same simulation framework. This dissertation presents solutions to the problems regarding the efficiency, accuracy, and scalability of the traffic simulation models in this framework. The ``ripple effect\u27\u27 is a well-known problem inherent in event-driven fluid-oriented traffic simulation, causing explosion of fluid rate changes. Integrating multi-resolution traffic representations requires estimating arrival rates of packet-oriented traffic, calculating the queueing delay upon a packet arrival, and computing packet loss rate under buffer overflow. Real time simulation of a large or ultra-large network demands efficient background traffic simulation. The dissertation includes a rate smoothing technique that provably mitigates the ``ripple effect\u27\u27, an accurate and efficient approach that integrates traffic models at multiple abstraction levels, a sequential algorithm that achieves real time simulation of the coarse-grained traffic in a network with 3 tier-1 ISP (Internet Service Provider) backbones using an ordinary PC, and a highly scalable parallel algorithm that simulates network traffic at coarse time scales

TCP with Adaptive Pacing for Multihop Wireless Networks

In this paper, we introduce a novel congestion control algorithm for TCP over multihop IEEE 802.11 wireless networks implementing rate-based scheduling of transmissions within the TCP congestion window. We show how a TCP sender can adapt its transmission rate close to the optimum using an estimate of the current 4-hop propagation delay and the coefficient of variation of recently measured round-trip times. The novel TCP variant is denoted as TCP with Adaptive Pacing (TCP-AP). Opposed to previous proposals for improving TCP over multihop IEEE 802.11 networks, TCP-AP retains the end-to-end semantics of TCP and does neither rely on modifications on the routing or the link layer nor requires cross-layer information from intermediate nodes along the path. A comprehensive simulation study using ns-2 shows that TCP-AP achieves up to 84% more goodput than TCP NewReno, provides excellent fairness in almost all scenarios, and is highly responsive to changing traffic conditions

MANETs: Internet Connectivity and Transport Protocols

A Mobile Ad hoc Network (MANET) is a collection of mobile nodes connected together over a wireless medium, which self-organize into an autonomous multi-hop wireless network. This kind of networks allows people and devices to seamlessly internetwork in areas with no pre-existing communication infrastructure, e.g., disaster recovery environments. Ad hoc networking is not a new concept, having been around in various forms for over 20 years. However, in the past only tactical networks followed the ad hoc networking paradigm. Recently, the introduction of new technologies such as IEEE 802.11, are moved the application field of MANETs to a more commercial field. These evolutions have been generating a renewed and growing interest in the research and development of MANETs. It is widely recognized that a prerequisite for the commercial penetration of the ad hoc networking technologies is the integration with existing wired/wireless infrastructure-based networks to provide an easy and transparent access to the Internet and its services. However, most of the existing solutions for enabling the interconnection between MANETs and the Internet are based on complex and inefficient mechanisms, as Mobile-IP and IP tunnelling. This thesis describes an alternative approach to build multi-hop and heterogeneous proactive ad hoc networks, which can be used as flexible and low-cost extensions of traditional wired LANs. The proposed architecture provides transparent global Internet connectivity and address autocofiguration capabilities to mobile nodes without requiring configuration changes in the pre-existing wired LAN, and relying on basic layer-2 functionalities. This thesis also includes an experimental evaluation of the proposed architecture and a comparison between this architecture with a well-known alternative NAT-based solution. The experimental outcomes confirm that the proposed technique ensures higher per-connection throughputs than the NAT-based solution. This thesis also examines the problems encountered by TCP over multi-hop ad hoc networks. Research on efficient transport protocols for ad hoc networks is one of the most active topics in the MANET community. Such a great interest is basically motivated by numerous observations showing that, in general, TCP is not able to efficiently deal with the unstable and very dynamic environment provided by multi-hop ad hoc networks. This is because some assumptions, in TCP design, are clearly inspired by the characteristics of wired networks dominant at the time when it was conceived. More specifically, TCP implicitly assumes that packet loss is almost always due to congestion phenomena causing buffer overflows at intermediate routers. Furthermore, it also assumes that nodes are static (i.e., they do not change their position over time). Unfortunately, these assumptions do not hold in MANETs, since in this kind of networks packet losses due to interference and link-layer contentions are largely predominant, and nodes may be mobile. The typical approach to solve these problems is patching TCP to fix its inefficiencies while preserving compatibility with the original protocol. This thesis explores a different approach. Specifically, this thesis presents a new transport protocol (TPA) designed from scratch, and address TCP interoperability at a late design stage. In this way, TPA can include all desired features in a neat and coherent way. This thesis also includes an experimental, as well as, a simulative evaluation of TPA, and a comparison between TCP and TPA performance (in terms of throughput, number of unnecessary transmissions and fairness). The presented analysis considers several of possible configurations of the protocols parameters, different routing protocols, and various networking scenarios. In all the cases taken into consideration TPA significantly outperforms TCP

Evaluation Study for Delay and Link Utilization with the New-Additive Increase Multiplicative Decrease Congestion Avoidance and Control Algorithm

As the Internet becomes increasingly heterogeneous, the issue of congestion avoidance and control becomes ever more important. And the queue length, end-to-end delays and link utilization is some of the important things in term of congestion avoidance and control mechanisms. In this work we continue to study the performances of the New-AIMD (Additive Increase Multiplicative Decrease) mechanism as one of the core protocols for TCP congestion avoidance and control algorithm, we want to evaluate the effect of using the AIMD algorithm after developing it to find a new approach, as we called it the New-AIMD algorithm to measure the Queue length, delay and bottleneck link utilization, and use the NCTUns simulator to get the results after make the modification for the mechanism. And we will use the Droptail mechanism as the active queue management mechanism (AQM) in the bottleneck router. After implementation of our new approach with different number of flows, we expect the delay will less when we measure the delay dependent on the throughput for all the system, and also we expect to get end-to-end delay less. And we will measure the second type of delay a (queuing delay), as we shown in the figure 1 bellow. Also we will measure the bottleneck link utilization, and we expect to get high utilization for bottleneck link with using this mechanism, and avoid the collisions in the link