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]). 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. 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. 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. 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. The theory of network coding promises significant benefits in network performance, especially in lossy networks and in multicast and multipath scenarios. To realize these benefits in practice, we need to understand how coding across packets interacts with the acknowledgment (ACK)-based flow control mechanism that forms a central part of today's Internet protocols such as transmission control protocol (TCP). Current approaches such as rateless codes and batch-based coding are not compatible with TCP's retransmission and sliding-window mechanisms. In this paper, we propose a new mechanism called TCP NC that incorporates network coding into TCP with only minor changes to the protocol stack, thereby allowing incremental deployment. In our scheme, the source transmits random linear combinations of packets currently in the congestion window. At the heart of our scheme is a new interpretation of ACKs-the sink acknowledges every degree of freedom (i.e., a linear combination that reveals one unit of new information) even if it does not reveal an original packet immediately. Thus, our new TCP ACK rule takes into account the network coding operations in the lower layer and enables a TCP-compatible sliding-window approach to network coding. Coding essentially masks losses from the congestion control algorithm and allows TCP NC to react smoothly to losses, resulting in a novel and effective approach for congestion control over lossy networks such as wireless networks. An important feature of our solution is that it allows intermediate nodes to perform re-encoding of packets, which is known to provide significant throughput gains in lossy networks and multicast scenarios. Simulations show that our scheme, with or without re-encoding inside the network, achieves much higher throughput compared to TCP over lossy wireless links. We present a real-world implementation of this protocol that addresses the practical aspects of incorporating network coding and decoding with TCP's window management mechanism. We work with TCP-Reno, which is a widespread and practical variant of TCP. Our implementation significantly advances the goal of designing a deployable, general, TCP-compatible protocol that provides the benefits of network coding. The authors present random early detection (RED) gateways for congestion avoidance in packet-switched networks. The gateway detects incipient congestion by computing the average queue size. The gateway could notify connections of congestion either by dropping packets arriving at the gateway or by setting a bit in packet headers. When the average queue size exceeds a present threshold, the gateway drops or marks each arriving packet with a certain probability, where the exact probability is a function of the average queue size. RED gateways keep the average queue size low while allowing occasional bursts of packets in the queue. During congestion, the probability that the gateway notifies a particular connection to reduce its window is roughly proportional to that connection's share of the bandwidth through the gateway. RED gateways are designed to accompany a transport-layer congestion control protocol such as TCP. The RED gateway has no bias against bursty traffic and avoids the global synchronization of many connections decreasing their window at the same time. Simulations of a TCP IP network are used to illustrate the performance of RED gateways. > This paper presents traffic measurement of a Massively Multiplayer On-line Role Playing Game (MMORPG). This analysis characterizes the MMORPG traffic and shows its implications for future research issues. The target game is 'Lineage II' developed by NCsoft, which is one of the world's largest MMORPGs in terms of the number of concurrent users. We collected about 1 tera bytes of packets for four consecutive days including a weekend. The MMORPG traffic consists of two kinds of packets: client-generated packets and server-generated packets. We observe that the client packet has an average of 19 bytes payload size, while the average payload size of server packets is about 318 bytes. This asymmetry is due to the fact that the server transmits all the information to construct the visual environment for the clients in the same region. Likewise, the bandwidth usage of the server traffic is about ten times larger than that of the client traffic. The analysis of RTT reveals that client packets and server packets are transmitted mostly at the interval of 200 milliseconds due to TCP's delayed ACK. We find that there is a linear relationship between the number of users and the bandwidth usage except when the number of users is around 5000. The network scenario is that of an infrastructure IEEE 802.11 WLAN with a single AP with which several stations (STAs) are associated. The AP has a finite size buffer for storing packets. In this scenario, we consider TCP-controlled upload and download file transfers between the STAs and a server on the wireline LAN (e.g., 100Mbps Ethernet) to which the AP is connected. In such a situation, it is well known that because of packet losses due to finite buffers at the AP, upload file transfers obtain larger throughputs than download transfers. We provide an analytical model for estimating the upload and download throughputs as a function of the buffer size at the AP. We provide models for the undelayed and delayed ACK cases for a TCP that performs loss recovery only by timeout, and also for TCP Reno. The models are validated in comparison with NS2 simulations. Several high-bandwidth network interfaces use Interrupt Coalescence (IC), i.e., they generate a single interrupt for multiple packets sent or received in a short time interval. IC decreases the per-packet interrupt processing overhead. However, IC also introduces queueing delays and alters the “dispersion” (i.e., interarrival time spacing) of packet pairs or trains. In this work, we first explain how IC works in two popular Gigabit Ethernet controllers. Then, we identify the negative effects of IC on active and passive network measurements. Specifically, we show that IC can affect active bandwidth estimation techniques, causing erroneous measurements. It can also alter the packet interarrivals in passive monitors that use commodity network interfaces. Based on the “signature” that IC leaves on the dispersion and one-way delays of packet trains, we show how to detect IC and how to filter its effects from raw measurements. Finally, we show that IC can be detrimental to TCP self-clocking, causing bursty delivery of ACKs and subsequent bursty transmission of data segments. HTTP adaptive streaming (HAS) is quickly becoming a popular mechanism for delivering on-demand video content over the Internet. The chunked transmission and application-layer adaptation create a very different traffic pattern than traditional progressive video downloads where the entire video is downloaded with a single request. In this paper, we investigate experimentally the interplay between HAS and the network transport control protocol (TCP). We investigate the impact of network delay on achievable throughput and discover that HAS streams cannot fully utilize the available bandwidth due to the start and stop nature of HAS traffic patterns and its interaction with TCP. We investigate TCP pacing as a potential solution to this issue, particularly for packet losses that occur as a result of bursting packets into the network at the start of a transmission. We find that pacing can significantly increase a TCP flow's congestion window but it does not necessarily translate into higher throughput. Instead, we find that packet losses at the end of chunk transmission have a greater impact on throughput. YouTube traffic is bursty. These bursts trigger packet losses and stress router queues, causing TCP's congestion-control algorithm to kick in. In this paper, we introduce Trickle, a server-side mechanism that uses TCP to rate limit YouTube video streaming. Trickle paces the video stream by placing an upper bound on TCP's congestion window as a function of the streaming rate and the round-trip time. We evaluated Trickle on YouTube production data centers in Europe and India and analyzed its impact on losses, bandwidth, RTT, and video buffer under-run events. The results show that Trickle reduces the average TCP loss rate by up to 43 and the average RTT by up to 28 while maintaining the streaming rate requested by the application. While the 802.11 power saving mode (PSM) and its enhancements can reduce power consumption by putting the wireless network interface (WNI) into sleep as much as possible, they either require additional infrastructure support, or may degrade the transmission throughput and cause additional transmission delay. These schemes are not suitable for long and bulk data transmissions with strict QoS requirements on wireless devices. With increasingly abundant bandwidth available on the Internet, we have observed that TCP congestion control is often not a constraint of bulk data transmissions as bandwidth throttling is widely used in practice. In this paper, instead of further manipulating the trade-off between the power saving and the incurred delay, we effectively explore the power saving potential by considering the bandwidth throttling on streaming downloading servers. We propose an application-independent protocol, called PSM-throttling. With a quick detection on the TCP flow throughput, a client can identify bandwidth throttling connections with a low cost Since the throttling enables us to reshape the TCP traffic into periodic bursts with the same average throughput as the server transmission rate, the client can accurately predict the arriving time of packets and turn on off the WNI accordingly. PSM-throttling can minimize power consumption on TCP-based bulk traffic by effectively utilizing available Internet bandwidth without degrading the application's performance perceived by the user. Furthermore, PSM-throttling is client-centric, and does not need any additional infrastructure support. Our lab-environment and Internet-based evaluation results show that PSM-throttling can effectively improve energy savings (by up to 75 ) and or the QoS for a broad types of TCP-based applications, including streaming, pseudo streaming, and large file downloading, over existing PSM-like methods. In this paper, optimal control of linear time-invariant (LTI) systems over unreliable communication links is studied. The motivation of the problem comes from growing applications that demand remote control of objects over Internet-type or wireless networks where links are prone to failure. Depending on the availability of acknowledgment (ACK) signals, two different types of networking protocols are considered. Under a TCP structure, existence of ACK signals is assumed, unlike the UDP structure where no ACK packets are present. The objective here is to mean-square (m.s.) stabilize the system while minimizing a quadratic performance criterion when the information flow between the controller and the plant is disrupted due to link failures, or packet losses. Sufficient conditions for the existence of stabilizing optimal controllers are derived. We consider an IEEE 802.11 based wireless LAN where an access point is used to connect a fixed number of users to the Web or to a shared file system. Users alternate between activity periods (corresponding to the download of a file) and idle periods (corresponding to think times). We first consider the interaction of TCP with the IEEE 802.11 MAC protocol, and get approximate expressions for the TCP throughput in the presence of n"c competing persistent connections. This result is then used to develop a queueing model, which provides the mean session delay in the presence of short-lived flows. The analysis also accounts for the TCP delayed ACK option. Comparison with simulation outcomes is provided, validating the model and providing guidelines for network designers. A particular emphasis is devoted to the impact of the TCP's advertised window size; in particular, it is proved that setting it to a small value leads to insensitivity of mean file transfer times to the file size distribution. Abstract We discuss a packet network architecture called a cognitive packet network (CPN), in which intelligent capabilities for routing and flow control are moved towards the packets, rather than being concentrated in the nodes and protocols. Our architecture contains “smart” and “dumb” packets, as well as acknowledgement packets. Smart CPN packets route themselves, and learn to avoid congestion and losses from their own observations about the network and from the experience of other packets. They use a reinforcement learning algorithm to route themselves based on a goal function which has been assigned to them for each connection. Dumb CPN packets of a specific quality of service (QoS) class use routes which have been selected by the smart packets (SPs) of that class. Acknowledgement (ACK) packets are generated by the destination when an SP arrives there; the ACK heads back to the source of the SP along the inverse route and is used to update mailboxes in CPN routers, as well as to provide source routing information for dumb packets. We first summarize the basic concepts behind CPN, and present simulations illustrating their performance for different QoS goals, and analytical results for best and worst case performance. We then describe a test-bed network we have designed and implemented in order to demonstrate these ideas. We provide measurement data on the test-bed to illustrate the capacity of the network to adapt to changes in traffic load and to failures of links. Finally, we use measurements to evaluate the impact of the ratio of smart to dumb packets on the end-to-end delay experienced by all of the packets. We consider several WLAN stations associated at rates r"1,r"2,...,r"k with an Access Point. Each station (STA) is downloading a long file from a local server, located on the LAN to which the Access Point (AP) is attached, using TCP. We assume that a TCP ACK will be produced after the reception of d packets at an STA. We model these simultaneous TCP-controlled transfers using a semi-Markov process. Our analytical approach leads to a procedure to compute aggregate download, as well as per-STA throughputs, numerically, and the results match simulations very well. TCP is the ubiquitous transport protocol in the Internet. However, in a wireless ad-hoc environment where links are unreliable, TCP causes a number of performance issues. The key reason behind this is that TCP considers all packet losses to be due to congestion and reduces its send rate, which is not necessarily appropriate in a lossy ad-hoc environment. In prior work, we have designed Loss Tolerant TCP (LT-TCP) [1], [2], which extends TCP towards making it more efficient and robust in a wireless ad-hoc environment. LT-TCP uses Explicit Congestion Notification (ECN) and forward error correction (FEC) to mitigate the effects of random packet losses. The protocol uses these mechanisms to both distinguish between congestion and other losses, and recover from losses due to lossy wireless links. In this paper, we describe the implementation of LT-TCP in the Linux operating system kernel, and present and analyze initial performance results for the protocol on lossy links. Results show that LT-TCP provides much improved performance over TCP over lossy connections that model ad-hoc networks. In addition, it shows that performance of LT-TCP is nearly linear with loss rate, whereas TCP suffers disproportionately as loss rate increases. These promising implementation results point to further experimentation for LT-TCP, including a push towards Internet standards bodies. Video streaming represents a large fraction of Internet traffic. Surprisingly, little is known about the network characteristics of this traffic. In this paper, we study the network characteristics of the two most popular video streaming services, Netflix and YouTube. We show that the streaming strategies vary with the type of the application (Web browser or native mobile application), and the type of container (Silverlight, Flash, or HTML5) used for video streaming. In particular, we identify three different streaming strategies that produce traffic patterns from non-ack clocked ON-OFF cycles to bulk TCP transfer. We then present an analytical model to study the potential impact of these streaming strategies on the aggregate traffic and make recommendations accordingly. 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.