Theory and experiments show that as the per-flow product of bandwidth and latency increases, TCP becomes inefficient and prone to instability, regardless of the queuing scheme. This failing becomes increasingly important as the Internet evolves to incorporate very high-bandwidth optical links and more large-delay satellite links. To address this problem, we develop a novel approach to Internet congestion control that outperforms TCP in conventional environments, and remains efficient, fair, scalable, and stable as the bandwidth-delay product increases. This new eXplicit Control Protocol, XCP, generalizes the Explicit Congestion Notification proposal (ECN). In addition, XCP introduces the new concept of decoupling utilization control from fairness control. This allows a more flexible and analytically tractable protocol design and opens new avenues for service differentiation. Using a control theory framework, we model XCP and demonstrate it is stable and efficient regardless of the link capacity, the round trip delay, and the number of sources. Extensive packet-level simulations show that XCP outperforms TCP in both conventional and high bandwidth-delay environments. Further, XCP achieves fair bandwidth allocation, high utilization, small standing queue size, and near-zero packet drops, with both steady and highly varying traffic. Additionally, the new protocol does not maintain any per-flow state in routers and requires few CPU cycles per packet, which makes it implementable in high-speed routers.

Theory and experiments show that as the per-flow product of bandwidth and latency increases, TCP becomes inefficient and prone to instability, regardless of the queuing scheme. This failing becomes increasingly important as the Internet evolves to incorporate very high-bandwidth optical links and more large-delay satellite links. To address

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Abstract — The paper considers the design of explicit rate-based flow control for ABR sources in an ATM network. The goal is to share the available capacity “fairly ” among many sources while maintaining queue length at a bottleneck node at a desired level. This problem is formulated as a stochastic control problem, and in this framework rate-control mechanisms are developed, which stabilize the queue length even thou,gh different sources may have different round-trip de-lays to the bottleneck node. Various robustness properties of the solution are illustrated through simulation experi-ments.

Achieving efficient and fair bandwidth allocation while minimizing packet loss and bottleneck queue in high bandwidthdelay product networks has long been a daunting challenge. Existing end-to-end congestion control (e.g., TCP) and traditional congestion notification schemes (e.g., TCP+AQM/ ECN) have significant limitations in achieving this goal. While the XCP protocol addresses this challenge, it requires multiple bits to encode the congestion-related information exchanged between routers and end-hosts. Unfortunately, there is no space in the IP header for these bits, and solving this problem involves a non-trivial and time-consuming standardization process. In this paper, we design and implement a simple, lowcomplexity protocol, called Variable-structure congestion Control Protocol (VCP), that leverages only the existing two ECN bits for network congestion feedback, and yet achieves comparable performance to XCP, i.e., high utilization, negligible packet loss rate, low persistent queue length, and reasonable fairness. On the downside, VCP converges significantly slower to a fair allocation than XCP. We evaluate the performance of VCP using extensive ns2 simulations over a wide range of network scenarios and find that it significantly outperforms many recently-proposed TCP variants, such as HSTCP, FAST, and CUBIC. To gain insight into the behavior of VCP, we analyze a simplified fluid model and prove its global stability for the case of a single bottleneck shared by synchronous flows with identical round-trip times. 1.

This paper presents TCP Rate Control, a new technique for transparently augmenting end-to-end TCP performance by controlling the sending rate of a TCP source. The "rate" of a TCP source is determined by its window size, the round trip time and the rate of acknowledgments (or acks). TCP rate control controls these aspects by modifying the ack number and receiver window fields in acknowledgments and by modulating the acknowledgment rate.

We present a framework for the provision of deterministic end-toend bandwidth guarantees in wireless ad hoc networks. Guided by a set of local feasibility conditions, multi-hop sessions are dynamically offered allocations, further translated to link demands. Using a distributed Time Division Multiple Access (TDMA) protocol nodes adapt to the demand changes on their adjacent links by local, conflict-free slot reassignments. As soon as the demand changes stabilize, the nodes must incrementally converge to a TDMA schedule that realizes the global link (and session) demand allocation. We first derive sufficient local feasibility conditions for certain topology classes and show that trees can be maximally utilized. We then introduce a converging distributed link scheduling algorithm that exploits the logical tree structure that arises in several ad hoc network applications. Decoupling bandwidth allocation to multi-hop sessions from link scheduling allows support of various end-to-end Quality of Service (QoS) objectives. We focus on the max-min fairness (MMF) objective and design an end-to-end asynchronous distributed algorithm for the computation of the session MMF rates. Once the end-to-end algorithm converges, the link scheduling algorithm converges to a TDMA schedule that realizes these rates. We demonstrate the applicability of this framework through an implementation over an existing wireless technology. This implementation is free of restrictive assumptions of previous TDMA approaches: it does not require any a-priori knowledge on the number of nodes in the network nor even network-wide slot synchronization.

Abstract--We propose a novel explicit rate-flow-control algo-rithm intended for available-bit-rate (ABR) service on an ATM net-work subject to loss and fairness constraints. The goal is to guar-antee low cell loss in order to avoid throughput collapse due tore-transmission by higher level protocols. The mechanism draws on measuring the current queue length and bandwidth availability, as well as tracking tile current number of active sessions contending for capacity, to adjust an explicit bound on the source transmis-sion rates. We identify the factors that affect queue overflows and propose simple design rules aimed at achieving transmission with controlled loss in a dynamic environment. We also discuss how con-servative design rules might be relaxed by accounting for statistical multiplexing in bandwidth sharing among bursty ABR sources and variable-bit-rate (VBR) sources. Index Terms--ABR service, ATM networks, delay differential equations, explicit rate flow control. A

In this paper, we study several properties of binary-feedback congestion control in rate-based applications. We first derive necessary conditions for generic binary-feedback congestion control to converge to fairness monotonically (which guarantees asymptotic stability of the fairness point) and show that AIMD is the only TCP-friendly binomial control with monotonic convergence to fairness. We then study steady-state behavior of binomial controls with n competing flows on a single bottleneck. Our main result here shows that combined probing for new bandwidth by all flows results in significant overshoot of the available bandwidth and rapid (often super-linear as a function of n) increase in packet loss. We also show that AIMD has the best scalability and lowest packet loss increase among all TCP-friendly binomial schemes. We conclude the paper by deriving the conditions necessary to achieve constant packet loss regardless of the number of competing flows n and examine one new scheme with such constant packet loss called Ideally Scalable Congestion Control (ISCC) in both simulation and streaming experiments.

This paper gives a new definition of general weighted (GW) fairness and shows how this can achieve various fairness definitions, such as those mentioned in the ATM Forum TM 4.0 specifications. The GW fairness can be achieved by calculating the ExcessFairshare (weighted fairshare of the left over bandwidth) for each VC. We show how a switch algorithm can be modified to support the GW fairness by using the ExcessFairshare term. We use ERICA+ as an example switch algorithm and show how it can be modified to achieve the GW fairness. For simulations, the weight parameters of the GW fairness are chosen to map a typical pricing policy. Simulation results are presented to demonstrate that, the modified switch algorithm achieves GW fairness. An analytical proof for convergence of the modified ERICA+ algorithm is given in the appendix.