Enhancing DCCP congestion control mechanism for long delay link

Most of the multimedia applications use the User Datagram Protocol (UDP) as a transport layer protocol because it is suitable for the delivery of multimedia data over the Internet.However, the use of UDP could endanger the stability of the network because there is no congestion control applied.To a certain extent, the network can collapse if too many applications deliberately use this protocol.Subsequently, instead of using the UDP, the applications have choices to use the Datagram Congestion Control Protocol (DCCP), which has a built-in congestion control that can provide a better network.Nevertheless, the congestion control mechanism in the Congestion Control Identifier (CCID)-2 TCP-like can cause problems when delivering multimedia data over a long delay link.To alleviate the problems, such as longer time taken for achieving maximum throughput, and throughput fluctuation during a congestion avoidance phase, two approaches have been used, i.e. setting of an appropriate slow-start threshold value and adjusting congestion window during a congestion avoidance phase.“TCP-like Threshold Window (TCP-like TW)” has been developed and modeled in the Network Simulator 2 (ns-2).For a long delay link, the TCP-like TW congestion control mechanism is able to minimize the time taken to achieve the maximum throughput. It also can smooth the fluctuation of throughput after achieving the maximum throughput

Reducing Internet Latency : A Survey of Techniques and their Merit

Bob Briscoe, Anna Brunstrom, Andreas Petlund, David Hayes, David Ros, Ing-Jyh Tsang, Stein Gjessing, Gorry Fairhurst, Carsten Griwodz, Michael WelzlPeer reviewedPreprin

Dual-Mode Congestion Control Mechanism for Video Services

Recent studies have shown that video services represent over half of Internet traffic, with a growing trend. Therefore, video traffic plays a major role in network congestion. Currently on the Internet, congestion control is mainly implemented through overprovisioning and TCP congestion control. Although some video services use TCP to implement their transport services in a manner that actually works, TCP is not an ideal protocol for use by all video applications. For example, UDP is often considered to be more suitable for use by real-time video applications. Unfortunately, UDP does not implement congestion control. Therefore, these UDP-based video services operate without any kind of congestion control support unless congestion control is implemented on the application layer. There are also arguments against massive overprovisioning. Due to these factors, there is still a need to equip video services with proper congestion control.Most of the congestion control mechanisms developed for the use of video services can only offer either low priority or TCP-friendly real-time services. There is no single congestion control mechanism currently that is suitable and can be widely used for all kinds of video services. This thesis provides a study in which a new dual-mode congestion control mechanism is proposed. This mechanism can offer congestion control services for both service types. The mechanism includes two modes, a backward-loading mode and a real-time mode. The backward-loading mode works like a low-priority service where the bandwidth is given away to other connections once the load level of a network is high enough. In contrast, the real-time mode always demands its fair share of the bandwidth.The behavior of the new mechanism and its friendliness toward itself, and the TCP protocol, have been investigated by means of simulations and real network tests. It was found that this kind of congestion control approach could be suitable for video services. The new mechanism worked acceptably. In particular, the mechanism behaved toward itself in a very friendly way in most cases. The averaged TCP fairness was at a good level. In the worst cases, the faster connections received about 1.6 times as much bandwidth as the slower connections

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

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

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 real-time interactive multimedia streams

The Internet is getting richer, and so the services. The richer the services, the more the users demand. The more they demand, the more we guarantee(1). This thesis investigates the congestion control mechanisms for interactive multimedia streaming applications. We start by raising a question as to why the congestion control schemes are not widely deployed in real-world applications, and study what options are available at present. We then discuss and show some of the good reasonings that might have made the control mechanism, specifically speaking the rate-based congestion control mechanism, not so attractive. In an effort to address the problems, we identify the existing problems from which the rate-based congestion control protocol cannot easily escape. We therefore propose a simple but novel windowbased congestion control protocol that can retain smooth throughput property while being fair when competing with TCP, yet still being responsive to the network changes. Through the extensive ns-2 simulations and the real-world experiments, we evaluate TFWC, our proposed mechanisms, and TFRC, the proposed IETF standard, in terms of network-oriented metrics (fairness, smoothness, stability, and responsive), and end-user oriented metrics (PSNR and MOS) to throughly study the protocol’s behaviors. We then discuss and conclude the options of the evaluated protocols for the real application. (1)We as congestion control mechanisms in the Internet