Packet reordering, high speed networks and transport protocol performance

We performed end-to-end measurements of UDP/IP flows across an Internet backbone network. Using this data, we characterized the packet reordering processes seen in the network. Our results demonstrate the high prevalence of packet reordering relative to packet loss, and show a strong correlation between packet rate and reordering on the network we studied. We conclude that, given the increased parallelism in modern networks and the demands of high performance applications, new application and protocol designs should treat packet reordering on an equal footing to packet loss, and must be robust and resilient to both in order to achieve high performance

Mitigating the impact of packet reordering to maximize performance of multimedia applications

We propose a solution to mitigate the performance degradation and corresponding Quality of Experience (QoE) reduction caused by packet reordering for multimedia applications which utilise unreliable transport protocols like the Datagram Congestion Control Protocol (DCCP). We analytically derive the optimum buffer size based on the applications data rate and the maximum delay tolerated by the multimedia application. We propose a dynamically adjustable buffer in the transport protocol receiver which uses this optimum buffer size. We demonstrate, via simulation results, that our solution reduces the packet loss rate, increases the perceived bandwidth and does not increase jitter in the received applications packets while still being within the application's delay limits, therefore resulting in an increased QoE of multimedia applications

DoS protection for a Pragmatic Multiservice Network Based on Programmable Networks

Proceedings of First International IFIP TC6 Conference, AN 2006, Paris, France, September 27-29, 2006.We propose a scenario of a multiservice network, based on pragmatic ideas of programmable networks. Active routers are capable of processing both active and legacy packets. This scenario is vulnerable to a Denial of Service attack, which consists in inserting false legacy packets into active routers. We propose a mechanism for detecting the injection of fake legacy packets into active routers. This mechanism consists in exchanging accounting information on the traffic between neighboring active routers. The exchange of accounting information must be carried out in a secure way using secure active packets. The proposed mechanism is sensitive to the loss of packets. To deal with this problem some improvements in the mechanism has been proposed. An important issue is the procedure for discharging packets when an attack has been detected. We propose an easy and efficient mechanism that would be improved in future work.Publicad

Online detection of pathological TCP flows with retransmissions in high-speed networks

Online Quality of Service (QoS) assessment in high speed networks is one of the key concerns for service providers, namely to detect QoS degradation on-the-fly as soon as possible and avoid customers’ complaints. In this regard, a Key Performance Indicator (KPI) is the number of TCP retransmissions per flow, which is related to packet losses or increased network and/or client/server latency. However, to accurately detect TCP retransmissions the whole sequence number list should be tracked which is a challenging task in multi-Gb/s networks. In this paper we show that the simplest approach of counting as a retransmission a packet whose sequence number is smaller than the previous one is enough to detect pathological flows with severe retransmissions. Such a lightweight approach eliminates the need of tracking the whole TCP flow history, which severely restricts traffic analysis throughput. Our findings show that low False Positive Rates (FPR) and False Negative Rates (FNR) can be achieved in the detection of such pathological flows with severe retransmissions, which are of paramount importance for QoS monitoring. Most importantly, we show that live detection of such pathological flows at 10 Gb/s rate per processing core is feasibleThis work has been partially funded by the Spanish Ministry of Economy and Competitiveness and the European Regional Development Fund under the projects TRÁFICA (MINECO/ FEDER TEC2015-69417-C2-1-R), Preproceso Inteligente de Tráfico (MINECO / FEDER TEC2015-69417-C2-2-R) and RACING DRONES (MINECO / FEDER RTC-2016-4744-7

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 in shallow-buffered switches. 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. While multiple end-to-end transport-layer solutions have been proposed, none of them have 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.Experiment results show that PABO can provide prominent advantage of mitigating transient congestions and can achieve significant gain on end-to-end delay

Master of Science

thesisDummynet is a link emulator that can be used by itself, as well as integrated within testbeds such as Emulab. Despite its popularity in the research community, Dummynet still lacks the ability to precisely emulate certain real network effects. In particular, it has no support for packet reordering. Since reordering is a common and prevalent network phe- nomenon just like packet loss or delay, it cannot be ignored when implementing emulators if we want to provide realistic emulation. It has been observed that networks suffer from reordering caused by packet striping, retransmissions, load balancing, multipath forwarding, etc. This has significant nega- tive effects on the performance of both Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). With the increase in prevalence of real-time streaming UDP applications such as video conferencing and Internet Protocol Television (IPTV), it has become important to focus on this problem which affects the performance of all these applications. Research into models and tools to diagnose and understand reordering requires that a sophisticated metric be used to describe it. So, in this thesis, I make two contributions: improving the realism of traffic shaping in Dummynet emulator by adding support for emulation of reordering, and an algorithm, a max-flow solver, that generates reordered sequences to be used by Dummynet, from a sophisticated reordering metric called Reorder Density (RD). My implementation enables the user to specify the desired amount of reordering in a metric, such as RD (or even others), and have Dummynet generate traffic that is reordered according to the input metric's value. This is accomplished within Dummynet by the use of a newly implemented scheduler. I conclude my thesis with an evaluation using real and software generated network traces to show that the algorithm is scalable and the implementation works correctly. Also, a datapath evaluation to show that my modifications to Dummynet do not result in any unnecessary increase in emulation running time is included

Enhancing wireless TCP a serialized-timer approach

IEEE INFOCOM Proceedings, 2010, p. 1-5In wireless networks, TCP performs unsatisfactorily since packet reordering and random losses may be falsely interpreted as congestive losses. This causes TCP to trigger fast retransmission and fast recovery spuriously, leading to under-utilization of available network resources. In this paper, we propose a novel TCP variant, known as TCP for noncongestive loss (TCP-NCL), to adapt TCP to wireless networks by using more reliable signals of packet loss and network overload for activating packet retransmission and congestion response, separately. TCP-NCL can thus serve as a unified solution for effective congestion control, sequencing control, and loss recovery. Different from the existing unified solutions, the modifications involved in the proposed variant are limited to sender-side TCP only, thereby facilitating possible future wide deployment. The two signals employed are the expirations of two serialized timers. A smart TCP sender model has been developed for optimizing the timer expiration periods. Our simulation studies reveal that TCP-NCL is robust against packet reordering as well as random packet loss while maintaining responsiveness against situations with purely congestive loss. ©2010 IEEE.published_or_final_versio