Content-Centric Networking (CCN) is a concept being considered as a potential future alternative to, or replacement for, today's Internet IP-style packet-switched host-centric networking. One factor making CCN attractive is its focus on content distribution, which dominates current Internet traffic and which is arguably not well-served by IP. Named Data Networking (NDN) is a prominent example of CCN. It is also one of several on-going research efforts aiming to design and develop a full-blown candidate future Internet architecture. Although NDN's primary motivation is content distribution, it is envisioned to support other types of traffic, such as conferencing (audio, video) as well as more historical applications, such as remote login. However, it is unclear how suitable NDN is for applications that are not obviously content-centric. In this paper, we explore NDN in the context of a class of applications that involve low- latency bidirectional communication. Specifically, we propose a few architectural amendments to NDN that provide significantly better throughput and lower latency for this class of applications. The proposed approach is validated via both simulations and testbed experiments.
In @cite_11 , study the effect of bi-directional traffic on TCP congestion control algorithm. In particular, they observe that in the BSD Tahoe TCP implementation packets from a single connection are clustered together, similarly to what observed by the same authors in @cite_9 over one-way traffic. This causes ACK compression, which significantly reduces the available bandwidth for TCP connections. They also show that in case of two-way traffic, the issue of ACK compression is made worse by the interaction of ACKs and data packets in the queue.
@cite_9 @cite_11
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We use simulation to study the dynamics of the congestion cent rol algorithm embedded in the BSD 4.3-Tahoe TCP implementation. We investigate the simple case of a few TCP connections, originating and terminating at the same pair of hosts, using a single bottleneck link. This work is an extension of our earlier work ([16]), where one-way traffic (i.e., all of the sources are on the same host and all of the destinations are on the other host) was studied. In this paper we investigate the dynamics that results from two-way traffic (in which there are data sources on both hosts). We find that the one-way traffic clustering and loss-synchronization phenomena d cussed in [16] persist in this new situation, albeit in a slightly modified form. In addition, there are two new phenomena not present in the earlier study: (1) ACK-compression, which is due to the interaction of data and ACK packets and gives rise to rapid fluctuations in queue length, and (2) an out-of-phase queue-synchronization mode, which keeps link utilization less than optimal even in the limit of very large buffers. These phenomena are helpful in understanding results from an earlier study of network oscillations ([19]). We use simulation to make some observations about the behavior of the congestion control algorithm currently embedded in the 4.3-Tahoe BSD TCP implementation. We investigate a simple case of a few TCP connections, originating and terminating at the same pair of hosts, using a single bottleneck link. Our simulations reveal two unexpected phenomena. First, packets from the individual connections, rather than being mixed together, completely separate into individual clusters. Second, every connection loses a single packet during each congestion epoch. As a way of exploring the cause of these phenomena, we discuss how the behavior is altered by modifications to the congestion control algorithm and to the switch queue control algorithm. High-speed networks with large delays present a unique environment where TCP may have a problem utilizing the full bandwidth. Several congestion control proposals have been suggested to remedy this problem. The existing protocols consider mainly two properties: TCP friendliness and bandwidth scalability. That is, a protocol should not take away too much bandwidth from standard TCP flows while utilizing the full bandwidth of high-speed networks. This work presents another important constraint, namely, RTT (round trip time) unfairness where competing flows with different RTTs may consume vastly unfair bandwidth shares. Existing schemes have a severe RTT unfairness problem because the congestion window increase rate gets larger as the window grows ironically the very reason that makes them more scalable. RTT unfairness for high-speed networks occurs distinctly with drop tail routers for flows with large congestion windows where packet loss can be highly synchronized. After identifying the RTT unfairness problem of existing protocols, This work presents a new congestion control scheme that alleviates RTT unfairness while supporting TCP friendliness and bandwidth scalability. The proposed congestion control algorithm uses two window size control policies called additive increase and binary search increase. When the congestion window is large, additive increase with a large increment ensures square RTT unfairness as well as good scalability. Under small congestion windows, binary search increase supports TCP friendliness. The simulation results confirm these properties of the protocol. We present experimental results evaluating fairness of several proposals to change the TCP congestion control algorithm, in support of operation on high bandwidth-delay- product (BDP) network paths. We examine and compare the fairness of New Reno TCP BIC, cubic, hamilton-TCP, highspeed-TCP and Scalable-TCP. We focus on four different views of fairness: TCP-friendliness RTT-fairness, intra- and inter-protocol fairness. In this paper we investigate the potential benefits of coordinated congestion control for multipath data transfers, and contrast with uncoordinated control. For static random path selections, we show the worst-case throughput performance of uncoordinated control behaves as if each user had but a single path (scaling like log(log(N)) log(N) where N is the system size, measured in number of resources). Whereas coordinated control gives a throughput allocation bounded away from zero, improving on both uncoordinated control and on the greedy-least loaded path selection of e.g. Mitzenmacher. We then allow users to change their set of routes and introduce the notion of a Nash equilibrium. We show that with RTT bias (as in TCP Reno), uncoordinated control can lead to inefficient equilibria. With no RTT bias, both uncoordinated or coordinated Nash equilibria correspond to desirable welfare maximising states. Moreover, simple path reselection polices that shift to paths with higher net benefit can find these states. Several high-speed TCP variants are adopted by end users and therefore, heterogeneous congestion control has become the characteristic of newly emerging high-speed optical networks. In contrast to homogeneous TCP flows, fairness among heterogeneous TCP flows now depends on router parameters such as queue management scheme and buffer size. To the best of our knowledge, this is the first study of fairness performance among heterogeneous TCP variants over a 10 Gbps high-speed optical network environment. Our evaluation scenarios for heterogeneous TCP flows consist of TCP variants with substantial presence in current Internet; therefore, TCP-SACK, CUBIC TCP (CUBIC) and HighSpeed TCP (HSTCP) compete for bottleneck bandwidth. Experimental results for fairness are presented for different queue management schemes, such as Drop-tail, Random Early Detection (RED), CHOose and Keep for responsive flows CHOose and Kill for unresponsive flows (CHOKe), and Approximate Fair Dropping (AFD), with varying degree of buffer sizes. We observe heterogeneous TCP flows induce more unfairness than homogeneous ones. Active queue management (AQM) schemes, RED, CHOKe, and AFD, improve fairness for large buffer sizes as compared to Drop-tail, whereas AQM schemes lose the fairness advantage for small buffer sizes. This study provides preliminary results for fairness challenges to deploy all-optical routers with limited buffer size. We study the effects of bandwidth asymmetry on end-to-end TCP performance and suggest a mechanism to improve it. Bandwidth asymmetry has been considered a mismatch in the bandwidths in two directions of data transfer. It is common in wireless networks, ADSL networks, etc. In a bandwidth asymmetry network, throughput achieved is not only dependent on the traffic and link characteristics of the forward direction, but significantly determined by those of the backward direction. Some mechanisms, has been proposed to overcome this problem. In our study, we focus on the ACK filtering algorithm, which decrease the rate of ACK on a constrained reverse link, taking advantage of the fact that TCP ACKs are cumulative. The fair allocation of resources among different nodes is one of the critical problems in wireless mesh networks. Existing solutions mainly focus on rate-limitation policies or distributed fair MAC schemes at the potential expense of total network utilization. This paper investigates a special starvation problem among TCP flows that are different hops away from the BS, as well as its recently proposed solution, the ‘Minimum Content Window’ policy based on IEEE 802.11e. It is found that the aggregate throughput degrades sharply because the effect of this policy on the TCP congestion mechanism has been overlooked. This paper proposes a priority-based congestion control by using ‘Cross-Layer Explicit Congestion Notification’. Analysis and simulation results demonstrate that our scheme can improve the fairness of TCP flows while the aggregate throughput is at least 20 higher than the ‘Minimum Content Window’ policy. Copyright © 2011 John Wiley & Sons, Ltd. ACK thinning refers to the technique to discard or reduce TCP acknowledgements (ACKs) for the purpose of diverting scarce bandwidth to TCP data traffic. Delayed ACK and ACK filtering fall into the category. It has been shown that under some circumstances the technique is effective to boost the TCP throughput on wireless links, in particular the IEEE 802.11 wireless LAN (WLAN). In this paper, however, we show that ACK thinning backfires under congestion due to its cross-layer impact on the 802.11 MAC dynamics. With the ACK filtering example, we demonstrate the phenomenon and analyze the cause. Based on the analysis, we show how the IEEE 802.11 contention window size control solves the problem. Although only the ACK filtering and Delayed ACK are considered in this paper, any other techniques to save on TCP ACK bandwidth usage share the same fundamental issues. In 2016, Google published the bottleneck bandwidth and round-trip time (BBR) congestion control algorithm. Unlike established loss- or delay-based algorithms like CUBIC or Vegas, BBR claims to operate without creating packet loss or filling buffers. Because of these prospects and promising initial performance results, BBR has gained wide-spread attention. As such it has been subject to behavior and performance analysis, which confirmed the results, but also revealed critical flaws.Because BBR is still work in progress, measurement results have limited validity for the future. In this paper we present our publicly available framework for reproducible TCP measurements based on network emulation. In a case study, we analyze the TCP BBR algorithm, reproduce and confirm weaknesses of the current BBR implementation, and provide further insights. We also contribute an analysis of BBR’s inter-flow synchronization behavior, showing that it reaches fairness equilibrium for long lived flows. The traditional TCP-Reno is not capable of achieving a high throughput in high-speed networks due to its highly conservative and loss-dependent congestion control mechanism. Although TCP-Vegas achieves a good link utilization due to its proactive type of congestion control mechanism that uses RTT as an indication of network congestion, it lacks friendliness to TCP-Reno. Protocols such as HighSpeed-TCP and Scalable-TCP have been proposed to address this problem, but they behave non-friendly to TCP-Reno especially when sharing high bandwidth links. In order to improve the friendliness with TCP-Reno and throughput performance in high-speed networks, we propose a new congestion control mechanism, TCP-xHS (High-speed) that harmonizes the fast ramping and proactive congestion controlling approaches of HighSpeed-TCP and TCP-Vegas. Through simulation results we justify the effectiveness of TCP-xHS for a wider range of link bandwidths. Measurements of LAN and WAN traffic show that network traffic exhibits variability on different scales. We examine a mechanism that gives rise to self-similar network traffic and discuss performance. The mechanism we study is the transfer of files or messages whose size is drawn from a heavy-tailed distribution. In a realistic client server network the degree to which file sizes are heavy-tailed can directly determine the degree of traffic self-similarity at the link level. This causal relationship is robust relative to changes in network resources, network topology, the influence of cross-traffic, and the distribution of interarrival times. Properties of the transport layer play an important role in preserving and modulating this relationship. The reliable transmission and flow control mechanisms of TCP serve to maintain the long-range dependency structure induced by heavy-tailed file size distributions. In contrast, if a non-flow-controlled and unreliable (UDP-based) transport protocol is used, the resulting traffic shows little self-similarity: although still bursty at short time scales, it has little long-range dependence. Performance implications of self-similarity are discussed as represented by various performance measures. Increased self-similarity as expected, results in degradation of performance. Queueing delay, in particular is discussed. Throughput-related measures such as packet loss and retransmission rate, however increase only gradually with increasing traffic self-similarity as long as reliable, flow-controlled transport protocol is used. Multipath TCP, as proposed by the IETF working group mptcp, allows a single data stream to be split across multiple paths. This has obvious benefits for reliability, and it can also lead to more efficient use of networked resources. We describe the design of a multipath congestion control algorithm, we implement it in Linux, and we evaluate it for multihomed servers, data centers and mobile clients. We show that some 'obvious' solutions for multipath congestion control can be harmful, but that our algorithm improves throughput and fairness compared to single-path TCP. Our algorithmis a drop-in replacement for TCP, and we believe it is safe to deploy. Several works have established links between congestion control in communication networks and feedback control theory. In this paper, following this paradigm, the design of an AQM (active queue management) ensuring the stability of the congestion phenomenon at a router is proposed. To this end, a modified fluid flow model of TCP (transmission control protocol) that takes into account all delays of the topology is introduced. Then, appropriate tools from control theory are used to address the stability issue and to cope with the time-varying nature of the multiple delays. More precisely, the design of the AQM is formulated as a structured state feedback for multiple time delay systems through the quadratic separation framework. The objective of this mechanism is to ensure the regulation of the queue size of the congested router as well as flow rates to a prescribed level. Furthermore, the proposed methodology allows to set arbitrarily the QoS (quality of service) of the communications following through the controlled router. Finally, a numerical example and some simulations support the exposed theory. We propose a duality model of end-to-end congestion control and apply it to understanding the equilibrium properties of TCP and active queue management schemes. The basic idea is to regard source rates as primal variables and congestion measures as dual variables, and congestion control as a distributed primal-dual algorithm over the Internet to maximize aggregate utility subject to capacity constraints. The primal iteration is carried out by TCP algorithms such as Reno or Vegas, and the dual iteration is carried out by queue management algorithms such as DropTail, RED or REM. We present these algorithms and their generalizations, derive their utility functions, and study their interaction. In a wireless ad hoc network with multihop transmissions and interference-limited link rates, can we balance power control in the physical layer and congestion control in the transport layer to enhance the overall network performance, while maintaining the stability, robustness, and architectural modularity of the network? We present a distributive power control algorithm that couples with the original TCP protocols to increase the end-to-end throughput and energy efficiency of the network. Under the rigorous framework of nonlinearly constrained optimization, we prove the convergence of this coupled system to the global optimum of joint power control and congestion control, for both synchronized and asynchronous implementations. The rate of convergence is geometric and a desirable modularity between the transport and physical layers is maintained. In particular, when the congestion control mechanism is TCP Vegas, that a simple utilization in the physical layer of the router buffer occupancy information suffices to achieve the joint optimum of this cross layer design. Both analytic results and simulations illustrate other desirable properties of the proposed algorithm, including robustness to channel outage and to path loss estimation errors, and flexibility in trading-off performance optimality for implementation simplicity. Congestion control in wireless multi-hop networks is challenging and complicated because of two reasons. First, interference is ubiquitous and causes loss in the shared medium. Second, wireless multihop networks are characterized by the use of diverse and dynamically changing routing paths. Traditional end point based congestion control protocols are ineffective in such a setting resulting in unfairness and starvation. This paper adapts the optimal theoretical work of Tassiulas and Ephremedes on cross-layer optimization of wireless networks involving congestion control, routing and scheduling, for practical solutions to congestion control in multi-hop wireless networks. This work is the first that implements in real off-shelf radios, a differential backlog based MAC scheduling and router-assisted backpressure congestion control for multi-hop wireless networks. Our adaptation, called DiffQ, is implemented between transport and IP and supports legacy TCP and UDP applications. In a network of 46 IEEE 802.11 wireless nodes, we demonstrate that DiffQ far outperforms many previously proposed "practical" solutions for congestion control. 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. This paper investigates the interactions between two-way TCP connections over 3GPP LTE networks. In the LTE network, the two-way TCP flows share buffers on a common bottleneck, i.e., the radio access links. The behaviors of TCPs significantly influence the others in the opposite direction. Specifically, the radio links of LTE are asymmetric, which may induce drastic interactions of TCPs and rapid draining of downlink buffer. The periodic idleness of downlink is a huge waste of the precious radio bandwidth and results in considerable performance degradation. In the viewpoint of Coupled Queues, we thoroughly understand the interacting TCPs and explain the reason for performance degradation. Based on a straightforward modeling procedure, we formalize the evolution of two-way TCPs and model the bottleneck queue size in every slot. The model indicates the queues are close coupled, which is verified with simulations on NS2. If the uplink (queue) is fully utilized, the downlink (queue) will always be underutilized even idle, and vice versa. Furthermore, an effective solution called Preemptive ACK Queueing (PAQ) is designed to decouple the queues, which improves the performance of two-way TCPs over LTE networks. Legacy congestion controls including TCP and its variants are known to perform poorly over cellular networks due to highly variable capacities over short time scales, self-inflicted packet delays, and packet losses unrelated to congestion. To cope with these challenges, we present Verus, an end-to-end congestion control protocol that uses delay measurements to react quickly to the capacity changes in cellular networks without explicitly attempting to predict the cellular channel dynamics. The key idea of Verus is to continuously learn a delay profile that captures the relationship between end-to-end packet delay and outstanding window size over short epochs and uses this relationship to increment or decrement the window size based on the observed short-term packet delay variations. While the delay-based control is primarily for congestion avoidance, Verus uses standard TCP features including multiplicative decrease upon packet loss and slow start. Through a combination of simulations, empirical evaluations using cellular network traces, and real-world evaluations against standard TCP flavors and state of the art protocols like Sprout, we show that Verus outperforms these protocols in cellular channels. In comparison to TCP Cubic, Verus achieves an order of magnitude (> 10x) reduction in delay over 3G and LTE networks while achieving comparable throughput (sometimes marginally higher). In comparison to Sprout, Verus achieves up to 30 higher throughput in rapidly changing cellular networks.