TCP throughput guarantee in the DiffServ Assured Forwarding service: what about the results?

Since the proposition of Quality of Service architectures by the IETF, the interaction between TCP and the QoS services has been intensively studied. This paper proposes to look forward to the results obtained in terms of TCP throughput guarantee in the DiffServ Assured Forwarding (DiffServ/AF) service and to present an overview of the different proposals to solve the problem. It has been demonstrated that the standardized IETF DiffServ conditioners such as the token bucket color marker and the time sliding window color maker were not good TCP traffic descriptors. Starting with this point, several propositions have been made and most of them presents new marking schemes in order to replace or improve the traditional token bucket color marker. The main problem is that TCP congestion control is not designed to work with the AF service. Indeed, both mechanisms are antagonists. TCP has the property to share in a fair manner the bottleneck bandwidth between flows while DiffServ network provides a level of service controllable and predictable. In this paper, we build a classification of all the propositions made during these last years and compare them. As a result, we will see that these conditioning schemes can be separated in three sets of action level and that the conditioning at the network edge level is the most accepted one. We conclude that the problem is still unsolved and that TCP, conditioned or not conditioned, remains inappropriate to the DiffServ/AF service

Managing Network Delay for Browser Multiplayer Games

Latency is one of the key performance elements affecting the quality of experience (QoE) in computer games. Latency in the context of games can be defined as the time between the user input and the result on the screen. In order for the QoE to be satisfactory the game needs to be able to react fast enough to player input. In networked multiplayer games, latency is composed of network delay and local delays. Some major sources of network delay are queuing delay and head-of-line (HOL) blocking delay. Network delay in the Internet can be even in the order of seconds. In this thesis we discuss what feasible networking solutions exist for browser multiplayer games. We conduct a literature study to analyze the Differentiated Services architecture, some salient Active Queue Management (AQM) algorithms (RED, PIE, CoDel and FQ-CoDel), the Explicit Congestion Notification (ECN) concept and network protocols for web browser (WebSocket, QUIC and WebRTC). RED, PIE and CoDel as single-queue implementations would be sub-optimal for providing low latency to game traffic. FQ-CoDel is a multi-queue AQM and provides flow separation that is able to prevent queue-building bulk transfers from notably hampering latency-sensitive flows. WebRTC Data-Channel seems promising for games since it can be used for sending arbitrary application data and it can avoid HOL blocking. None of the network protocols, however, provide completely satisfactory support for the transport needs of multiplayer games: WebRTC is not designed for client-server connections, QUIC is not designed for traffic patterns typical for multiplayer games and WebSocket would require parallel connections to mitigate the effects of HOL blocking

QoS Implementation with Triple-Metric Based Active Queue Management for Military Networks

For supporting Quality of Service (QoS) in a military network, applications of triple-metric priority of performance, importance, and urgency as well as autonomous and lightweight implementation are required. In the previous study, we analyzed a Korean military network’s QoS implementation in the perspective of the triple-metric, and presented some improvements in the simplification of service classes of Differentiated Services (DiffServ). To extend the simplified DiffServ in the previous research, this paper proposes Active Queue Management (AQM) algorithms to process the traffic of each service class differently based on importance and urgency, and shows the feasibility through some experiments

User-Centric Quality of Service Provisioning in IP Networks

The Internet has become the preferred transport medium for almost every type of communication, continuing to grow, both in terms of the number of users and delivered services. Efforts have been made to ensure that time sensitive applications receive sufficient resources and subsequently receive an acceptable Quality of Service (QoS). However, typical Internet users no longer use a single service at a given point in time, as they are instead engaged in a multimedia-rich experience, comprising of many different concurrent services. Given the scalability problems raised by the diversity of the users and traffic, in conjunction with their increasing expectations, the task of QoS provisioning can no longer be approached from the perspective of providing priority to specific traffic types over coexisting services; either through explicit resource reservation, or traffic classification using static policies, as is the case with the current approach to QoS provisioning, Differentiated Services (Diffserv). This current use of static resource allocation and traffic shaping methods reveals a distinct lack of synergy between current QoS practices and user activities, thus highlighting a need for a QoS solution reflecting the user services. The aim of this thesis is to investigate and propose a novel QoS architecture, which considers the activities of the user and manages resources from a user-centric perspective. The research begins with a comprehensive examination of existing QoS technologies and mechanisms, arguing that current QoS practises are too static in their configuration and typically give priority to specific individual services rather than considering the user experience. The analysis also reveals the potential threat that unresponsive application traffic presents to coexisting Internet services and QoS efforts, and introduces the requirement for a balance between application QoS and fairness. This thesis proposes a novel architecture, the Congestion Aware Packet Scheduler (CAPS), which manages and controls traffic at the point of service aggregation, in order to optimise the overall QoS of the user experience. The CAPS architecture, in contrast to traditional QoS alternatives, places no predetermined precedence on a specific traffic; instead, it adapts QoS policies to each individual’s Internet traffic profile and dynamically controls the ratio of user services to maintain an optimised QoS experience. The rationale behind this approach was to enable a QoS optimised experience to each Internet user and not just those using preferred services. Furthermore, unresponsive bandwidth intensive applications, such as Peer-to-Peer, are managed fairly while minimising their impact on coexisting services. The CAPS architecture has been validated through extensive simulations with the topologies used replicating the complexity and scale of real-network ISP infrastructures. The results show that for a number of different user-traffic profiles, the proposed approach achieves an improved aggregate QoS for each user when compared with Best effort Internet, Traditional Diffserv and Weighted-RED configurations. Furthermore, the results demonstrate that the proposed architecture not only provides an optimised QoS to the user, irrespective of their traffic profile, but through the avoidance of static resource allocation, can adapt with the Internet user as their use of services change.France Teleco

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

Quality of Service Management and Interoperability

Information technology: general issue

Fixed-point analysis of a network of routers with persistent TCP/UDP flows and class-based weighted fair queuing

Fixed-point models have already been successfully used to analytically study networks consisting of persistent TCP flows only, or mixed TCP/UDP flows with a single queue per link and differentiated buffer management for these two types of flows. In the current study, we propose a nested fixed-point analytical method to obtain the throughput of persistent TCP and UDP flows in a network of routers supporting class-based weighted fair queuing allowing the use of separate queues for each class. In particular, we study the case of two classes where one of the classes uses drop-tail queue management and is intended for only UDP traffic. The other class targeting TCP, but also allowing UDP traffic for the purpose of generality, is assumed to employ active queue management. The effectiveness of the proposed analytical method is validated in terms of accuracy using ns-3 simulations and the required computational effort. © 2016, Springer Science+Business Media New York

Analysis of RED-Family Active Queue Management

RED is an Active Queue Management (AQM) technique that is intended to achieve high link utilization with a low queuing delay. Recent studies show that RED is difficult to configure for some rapidly changing traffic mixes and loads [1]. Other studies show that under some conditions, the performance gains of RED and its variants over traditional drop-tail queue management is not significant given the additional complexity required for proper configuration [2], [3]. Recent variants of RED, such as Adaptive- RED [4], are designed to provide more robust RED performance under a wider-range of traffic conditions. This paper develops a general queue law for TCP-RED control systems that use packet dropping and/or Explicit Congestion Notification (ECN) marking as congestion signaling methods, and illustrates the impact of TCP traffic on the behavior of congested router queue. Furthermore, this paper provides additional analysis of RED and newer variants of RED including Adaptive-RED [4] that is designed to provide more robust RED performance under a wider-range of traffic conditions. Through careful simulation designs using the queue law and analysis, this paper confirms that RED-like AQM techniques that employ packet dropping do not significantly improve performance over that of drop-tail queue management. However, when AQM techniques use ECN marking, the performance gains of AQM in terms of goodput and delay can be significant over that of drop-tail queue management

Throughput and Delay on the Packet Switched Internet

The Internet has become a vital and essential part of modern everyday life. Services delivered by the Internet are used by people across the planet every moment of every day of the year. The Internet has proven a positive force for good improving the lives of billions of people worldwide. The power of the Internet to deliver this positive good to humanity relies on its ability to deliver life improving services. In my doctorate work culminating in this dissertation I have striven to sustain and increase the Internet's ability to deliver these services and to have a positive good effect upon humanity.The overarching purpose of this dissertation is to improve the Internet's ability to deliver life improving services. I have further divided this purpose into two goals. To improve the ability of applications operating in challenging network conditions to gain their fair share of the bandwidth resources and to reduce the delay with which these services are delivered. Every service delivered by the Internet consists of Internet objects that are delivered through communication paths across the Internet. The delivery of these objects is defined by the two characteristics; Throughput and delay. Throughput determines how much of an object can be delivered over a period of time and delay determines how long it takes to deliver an object.These two characteristics determine the Internet's ability to deliver objects across communication paths. Improving these two characteristics (bandwidth and delay) increase the ability of the Internet to deliver objects and thus improve the Internet's capability to deliver life improving services. To accomplish this goal I present projects along three areas of effort. These three areas of effort are: (1) Increase the ability of applications operating in challenging conditions to achieve their fair share of bandwidth. (2) Synthesize knowledge required to address the effort to reduce delay. (3) Develop protocols that reduce delay encountered in the communications paths of the Internet.In this dissertation I present projects along these three areas of effort that accomplish the two goals (increase bandwidth and reduce delay) to achieve the purpose of improving the Internet's ability to deliver essential and life improving services. These projects and their organization into areas of effort, goals and purpose are my contributions to the networking sciences

Dual Queue Coupled AQM: Deployable Very Low Queuing Delay for All

On the Internet, sub-millisecond queueing delay and capacity-seeking have traditionally been considered mutually exclusive. We introduce a service that offers both: Low Latency Low Loss Scalable throughput (L4S). When tested under a wide range of conditions emulated on a testbed using real residential broadband equipment, queue delay remained both low (median 100--300 $\mu$s) and consistent (99th percentile below 2 ms even under highly dynamic workloads), without compromising other metrics (zero congestion loss and close to full utilization). L4S exploits the properties of `Scalable' congestion controls (e.g., DCTCP, TCP Prague). Flows using such congestion control are however very aggressive, which causes a deployment challenge as L4S has to coexist with so-called `Classic' flows (e.g., Reno, CUBIC). This paper introduces an architectural solution: `Dual Queue Coupled Active Queue Management', which enables balance between Scalable and Classic flows. It counterbalances the more aggressive response of Scalable flows with more aggressive marking, without having to inspect flow identifiers. The Dual Queue structure has been implemented as a Linux queuing discipline. It acts like a semi-permeable membrane, isolating the latency of Scalable and `Classic' traffic, but coupling their capacity into a single bandwidth pool. This paper justifies the design and implementation choices, and visualizes a representative selection of hundreds of thousands of experiment runs to test our claims.Comment: Preprint. 17pp, 12 Figs, 60 refs. Submitted to IEEE/ACM Transactions on Networkin