Multiplexing regulated traffic streams: design and performance

The main network solutions for supporting QoS rely on traf- fic policing (conditioning, shaping). In particular, for IP networks the IETF has developed Intserv (individual flows regulated) and Diffserv (only ag- gregates regulated). The regulator proposed could be based on the (dual) leaky-bucket mechanism. This explains the interest in network element per- formance (loss, delay) for leaky-bucket regulated traffic. This paper describes a novel approach to the above problem. Explicitly using the correlation structure of the sources’ traffic, we derive approxi- mations for both small and large buffers. Importantly, for small (large) buffers the short-term (long-term) correlations are dominant. The large buffer result decomposes the traffic stream in a stream of constant rate and a periodic impulse stream, allowing direct application of the Brownian bridge approximation. Combining the small and large buffer results by a concave majorization, we propose a simple, fast and accurate technique to statistically multiplex homogeneous regulated sources. To address heterogeneous inputs, we present similarly efficient tech- niques to evaluate the performance of multiple classes of traffic, each with distinct characteristics and QoS requirements. These techniques, applica- ble under more general conditions, are based on optimal resource (band- width and buffer) partitioning. They can also be directly applied to set GPS (Generalized Processor Sharing) weights and buffer thresholds in a shared resource system

An Optimal Lower Bound for Buffer Management in Multi-Queue Switches

In the online packet buffering problem (also known as the unweighted FIFO variant of buffer management), we focus on a single network packet switching device with several input ports and one output port. This device forwards unit-size, unit-value packets from input ports to the output port. Buffers attached to input ports may accumulate incoming packets for later transmission; if they cannot accommodate all incoming packets, their excess is lost. A packet buffering algorithm has to choose from which buffers to transmit packets in order to minimize the number of lost packets and thus maximize the throughput. We present a tight lower bound of e/(e-1) ~ 1.582 on the competitive ratio of the throughput maximization, which holds even for fractional or randomized algorithms. This improves the previously best known lower bound of 1.4659 and matches the performance of the algorithm Random Schedule. Our result contradicts the claimed performance of the algorithm Random Permutation; we point out a flaw in its original analysis

Improved Competitive Guarantees for QoS Buffering

textabstractWe consider a network providing Differentiated Services (Diffserv), which allow Internet Service Providers (ISP s) to offer different levels of Quality of Service (QoS) to different traffic streams. We study two types of buffering policies that are used in network switches supporting QoS. In the FIFO type, packets must be transmitted in the order they arrive. In the uniform bounded-delay type, there is a maximum delay time associated with the switch and each packet must be transmitted within this time, or otherwise it is dropped. In both models, the buffer space is limited, and packets are lost when the buffer overflows. Each packet has an intrinsic value, and the goal is to maximize the total value of transmitted packets. Our main contribution is an algorithm for the FIFO model with arbitrary packet values that for the first time achieves a competitive ratio better than 2, namely 2 - epsilon for a constant epsilon > 0. We also describe an algorithm for the uniform bounded delay model which simulates our algorithm for the FIFO model, and show that it achieves the same competitive ratio

Performance Analysis in IP-Based Industrial Communication Networks

S rostoucím počtem řídicích systémů a jejich distribuovanosti získávájí komunikační sítě na důležitosti a objevují se nové výzkumné trendy. Hlavní problematikou v této oblasti, narozdíl od dřívějších řídicích systémů využívajících dedikovaných komunikačních obvodů, je časově proměnné zpoždění měřicích a řídicích signálů způsobené paketově orientovanými komunikačními prostředky, jako např. Ethernet. Aspekty komunikace v reálném čase byly v těchto sítích již úspěšně vyřešeny. Nicméně, analýzy trendů trhu předpovídají budoucí využití také IP sítí v průmyslové komunikaci pro časově kritickou procesní vyměnu dat. IP komunikace má ovšem pouze omezenou podporu v instrumentaci pro průmyslovou automatizace. Tato výzva byla nedávno technicky vyřešena v rámci projektu Virtual Automation Networks (virtuální automatizační sítě - VAN) zapojením mechanismů kvality služeb (QoS), které jsou schopny zajistit měkkou úroveň komunikace v reálném čase. Předložená dizertační práce se zaměřuje na aspekty výkonnosti reálného času z analytického hlediska a nabízí prostředek pro hodnocení využitelnosti IP komunikace pro budoucí průmyslové aplikace. Hlavním cílem této dizertační práce je vytvoření vhodného modelovacího rámce založeného na network calculus, který pomůže provést worst-case výkonnostní analýzu časového chování IP komunikačních sítí a jejich prvků určených pro budoucí použití v průmyslové automatizaci. V práci byla použita empirická analýza pro určení dominantních faktorů ovlivňujících časového chování síťových zařízení a identifikaci parametrů modelů těchto zařízení. Empirická analýza využívá nástroj TestQoS vyvinutý pro tyto účely. Byla navržena drobná rozšíření rámce network calculus, která byla nutná pro modelování časového chování používaných zařízení. Bylo vytvořeno několik typových modelů zařízení jako výsledek klasifikace různých architektur síťových zařízení a empiricky zjištěných dominantních faktorů. U modelovaných zařízení byla využita nová metoda identifikace parametrů. Práce je zakončena validací časových modelů dvou síťových zařízení (přepínače a směrovače) oproti empirickým pozorováním.With the growing scale of control systems and their distributed nature, communication networks have been gaining importance and new research challenges have been appearing. The major problem, contrary to previously used control systems with dedicated communication circuits, is time-varying delay of control and measurement signals introduced by packet-switched networks, such as Ethernet. The real-time issues in these networks have been tackled by proper adaptations. Nevertheless, market trend analyses foresee also future adoptions of IP-based communication networks in industrial automation for time-critical run-time data exchange. IP-based communication has only a limited support from the existing instrumentation in industrial automation. This challenge has recently been technically tackled within the Virtual Automation Networks (VAN) project by adopting the quality of service (QoS) architecture delivering soft-real-time communication behaviour. This dissertation focuses on the real-time performance aspects from the analytical point of view and provides means for applicability assessment of IP-based communication for future industrial applications. The main objective of this dissertation is establishment of a relevant modelling framework based on network calculus which will assist worst-case performance analysis of temporal behaviour of IP-based communication networks and networking devices intended for future use in industrial automation. Empirical analysis was used to identify dominant factors influencing the temporal performance of networking devices and for model parameter identification. The empirical analysis makes use of the TestQoS tool developed for this purpose. Minor extensions to the network calculus framework were proposed enabling to model the required temporal behaviour of networking devices. Several exemplary models were inferred as a result of classification of different networking device architectures and empirically identified dominant factors. A novel method for parameter identification was used with the modelled devices. Finally, two temporal models of networking devices (a switch and a router) were validated against empirical observations.

The Longest Queue Drop Policy for Shared-Memory Switches is 1.5-competitive

We consider the Longest Queue Drop memory management policy in shared-memory switches consisting of $N$ output ports. The shared memory of size $M\geq N$ may have an arbitrary number of input ports. Each packet may be admitted by any incoming port, but must be destined to a specific output port and each output port may be used by only one queue. The Longest Queue Drop policy is a natural online strategy used in directing the packet flow in buffering problems. According to this policy and assuming unit packet values and cost of transmission, every incoming packet is accepted, whereas if the shared memory becomes full, one or more packets belonging to the longest queue are preempted, in order to make space for the newly arrived packets. It was proved in 2001 [Hahne et al., SPAA '01] that the Longest Queue Drop policy is 2-competitive and at least $\sqrt{2}$-competitive. It remained an open question whether a (2-\epsilon) upper bound for the competitive ratio of this policy could be shown, for any positive constant \epsilon. We show that the Longest Queue Drop online policy is 1.5-competitive

Congestion Control for Streaming Media

The Internet has assumed the role of the underlying communication network for applications such as file transfer, electronic mail, Web browsing and multimedia streaming. Multimedia streaming, in particular, is growing with the growth in power and connectivity of today\u27s computers. These Internet applications have a variety of network service requirements and traffic characteristics, which presents new challenges to the single best-effort service of today\u27s Internet. TCP, the de facto Internet transport protocol, has been successful in satisfying the needs of traditional Internet applications, but fails to satisfy the increasingly popular delay sensitive multimedia applications. Streaming applications often use UDP without a proper congestion avoidance mechanisms, threatening the well-being of the Internet. This dissertation presents an IP router traffic management mechanism, referred to as Crimson, that can be seamlessly deployed in the current Internet to protect well-behaving traffic from misbehaving traffic and support Quality of Service (QoS) requirements of delay sensitive multimedia applications as well as traditional Internet applications. In addition, as a means to enhance Internet support for multimedia streaming, this dissertation report presents design and evaluation of a TCP-Friendly and streaming-friendly transport protocol called the Multimedia Transport Protocol (MTP). Through a simulation study this report shows the Crimson network efficiently handles network congestion and minimizes queuing delay while providing affordable fairness protection from misbehaving flows over a wide range of traffic conditions. In addition, our results show that MTP offers streaming performance comparable to that provided by UDP, while doing so under a TCP-Friendly rate