In this paper, we revisit the idea to remove Slow Start from congestion control. To do so, we build upon the newly gained freedom of transport protocol extendability offered by QUIC to hint bandwidth estimates from a typical web client to a server. Using this bandwidth estimate, we bootstrap congestion windows of new connections to quickly utilize available bandwidth. This custom flow initialization removes the common early exit of Slow Start and thus fuels short flow fairness with long-running connections. Our results indicate that we can drastically reduce flow completion time accepting some losses and thereby an inflated transmission volume. For example, for a typical DSL client, loading a 2 MB YouTube video chunk is accelerated by nearly 2x. In the worst case, we find an inflation of the transfer volume by 12 due to losses.
While past connections can hint at receiver network conditions, we rather see them as a building block that can be used in combination with a receiver-side bandwidth hint. In our work, we highlighted possible angles in which to tackle bandwidth estimates but do not go into detail. @cite_7 shows how to utilize cellular (4G) PHY-layer information at the client for bandwidth estimations, they even implement it in the Google QUIC framework but do not show the effects on other flows, fairness and losses nor focus on the startup phase. Further bandwidth estimations @cite_1 @cite_24 utilize client context information ( location, signal quality, speed) in cellular networks to learn and predict channel throughput, which could be used in an implementation of our system.
@cite_24 @cite_1 @cite_7
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With lower latency and higher bandwidth than its predecessor 3G networks, the latest cellular technology 4G LTE has been attracting many new users. However, the interactions among applications, network transport protocol, and the radio layer still remain unexplored. In this work, we conduct an in-depth study of these interactions and their impact on performance, using a combination of active and passive measurements. We observed that LTE has significantly shorter state promotion delays and lower RTTs than those of 3G networks. We discovered various inefficiencies in TCP over LTE such as undesired slow start. We further developed a novel and lightweight passive bandwidth estimation technique for LTE networks. Using this tool, we discovered that many TCP connections significantly under-utilize the available bandwidth. On average, the actually used bandwidth is less than 50 of the available bandwidth. This causes data downloads to be longer, and incur additional energy overhead. We found that the under-utilization can be caused by both application behavior and TCP parameter setting. We found that 52.6 of all downlink TCP flows have been throttled by limited TCP receive window, and that data transfer patterns for some popular applications are both energy and network unfriendly. All these findings highlight the need to develop transport protocol mechanisms and applications that are more LTE-friendly. Today, video distribution platforms use adaptive video streaming to deliver the maximum Quality of Experience to a wide range of devices connected to the Internet through different access networks. Among the techniques employed to implement video adaptivity, the stream-switching over HTTP is getting a wide acceptance due to its deployment and implementation simplicity. Recently it has been shown that the client-side algorithms proposed so far generate an on-off traffic pattern that may lead to unfairness and underutilization when many video flows share a bottleneck. In this paper we propose ELASTIC (fEedback Linearization Adaptive STreamIng Controller), a client-side controller designed using feedback control theory that does not generate an on-off traffic pattern. By employing a controlled testbed, allowing bandwidth capacity and delays to be set, we compare ELASTIC with other client-side controllers proposed in the literature. In particular, we have checked to what extent the considered algorithms are able to: 1) fully utilize the bottleneck, 2) fairly share the bottleneck, 3) obtain a fair share when TCP greedy flows share the bottleneck with video flows. The obtained results show that ELASTIC achieves a very high fairness and is able to get the fair share when coexisting with TCP greedy flows. How to keep the probability of hand-off drops within a prespecified limit is a very important quality-of-service (QoS) issue in cellular networks because mobile users should be able to maintain ongoing sessions even during their hand-off from one cell to another. We design and evaluate predictive and adaptive schemes for bandwidth reservation for the hand-offs of ongoing sessions and the admission control of new connections. We first develop a method to estimate user mobility based on an aggregate history of hand-offs observed in each cell. This method is then used to probabilistically predict mobiles' directions and hand-off times in a cell. For each cell, the bandwidth to be reserved for hand-offs is calculated by estimating the total sum of tractional bandwidths of the expected hand-offs within a mobility-estimation time window. Three different admission-control schemes for new connection requests using this bandwidth reservation are proposed. We also consider variations that utilize the path location information available from the car navigation system or global positioning system. Finally, we evaluate the performance of the proposed schemes extensively to show that they meet our design goal and outperform the static reservation scheme under various scenarios. With the advent of high-speed cellular access and the overwhelming popularity of smartphones, a large percent of today's Internet content is being delivered via cellular links. Due to the nature of long-range wireless signal propagation, the capacity of the last hop cellular link can vary by orders of magnitude within a short period of time (e.g., a few seconds). Unfortunately, TCP does not perform well in such fast-changing environments, potentially leading to poor spectrum utilization and high end-to-end packet delay. In this paper we revisit seminal work in cross-layer optimization in the context of 4G cellular networks. Specifically, we leverage the rich physical layer information exchanged between base stations (NodeB) and mobile phones (UE) to predict the capacity of the underlying cellular link, and propose nCQIC, a cross-layer congestion control design. Experiments on real cellular networks confirm that our capacity estimation method is both accurate and precise. A CQIC sender uses these capacity estimates to adjust its packet sending behavior. Our preliminary evaluation reveals that CQIC improves throughput over TCP by 1.08-2.89x for small and medium flows. For large flows, CQIC attains throughput comparable to TCP while reducing the average RTT by 2.38-2.65x. The wireless channel is time-varying where burst packet losses often occur during the fading or lossy handovers. In order to avoid unaccepted quality degradation of video streaming over 3G cellular networks, we propose and analyze a client-driven scalable cross-layer (CSC) retransmission scheme. Considering the perceptual importance of different video partitions under the real-time and bandwidth constraints, the proposed scheme uses the radio link-layer retransmission with priority to adapt conventional packet losses in wireless channels; furthermore, it uses the adaptive transport-layer retransmission to provide end-to-end quality-of-service (QoS) guarantees over cellular networks. The simulation experiments show that the proposed scheme can effectively improve the perceptual quality of 3G video streaming as compared to the traditional deadline-based scheme without the prioritized link-layer retransmission. Input-Output buffered crossbars are popular building blocks for scalable high-speed switching because they require minimum speed-up of memory bandwidth. Scaling the design of crossbar switches to large capacities is limited by technology issues such as the reconfiguration of high-speed fabrics or power consumption. In addition, these crossbar architechtures typically schedule and transfer in terms of fixed size envelopes. Thus when they are used in the context of IP networks where packets are of variable size, the incoming packets need to be fragmented into fixed size envelopes. This fragmentation can lead to, possibly large [1], loss of bandwidth and even instability. This paper proposes a new method for switching variable sized packets over a crossbar switch that i) allows maximum utilization of switch bandwidth and avoids the fragmentation effect, and ii) allows designers to use much larger envelopes for transfering data over the fabric and thus minimizes the reconfiguration frequency of the fabric. Reducing the scheduling frequency makes implementation of complex schedulers practical and enables us to build ultra-fast switches incorporating optical technology that can provide bandwidth and delay guarantees. Providing packet-level quality of service (QoS) is critical to support both rate-sensitive and delay-sensitive applications in bandwidth-constrained, shared-channel, multihop wireless networks. Packet scheduling has been a very popular paradigm to ensure minimum throughput and bounded delay access for packet flows. This work describes a packet scheduling approach to QoS provisioning in multihop wireless networks. Besides minimum throughput and delay bounds for each flow, our scheduling disciplines seek to achieve fair and maximum allocation of the shared wireless channel bandwidth. However, these two criteria can potentially be in conflict in a generictopology multihop wireless network where a single logical channel is shared among multiple contending flows and spatial reuse of the channel bandwidth is possible. In this paper, we propose a new scheduling model that addresses this conflict. The main results of this paper are the following: (a) a two-tier service model that provides a minimum "fair" allocation of the channel bandwidth for each packet flow and additionally maximizes spatial reuse of bandwidth, (b) an ideal centralized packet scheduling algorithm that realizes the above service model, and (c) a practical distributed backoff-based channel contention mechanism that approximates the ideal service within the framework of the CSMA CA protocol. The millimeter wave (mmWave) bands offer the possibility of orders of magnitude greater throughput for fifth-generation (5G) cellular systems. However, since mmWave signals are highly susceptible to blockage, channel quality on any one mmWave link can be extremely intermittent. This paper implements a novel dual connectivity protocol that enables mobile user equipment devices to maintain physical layer connections to 4G and 5G cells simultaneously. A novel uplink control signaling system combined with a local coordinator enables rapid path switching in the event of failures on any one link. This paper provides the first comprehensive end-to-end evaluation of handover mechanisms in mmWave cellular systems. The simulation framework includes detailed measurement-based channel models to realistically capture spatial dynamics of blocking events, as well as the full details of Medium Access Control, Radio Link Control, and transport protocols. Compared with conventional handover mechanisms, this paper reveals significant benefits of the proposed method under several metrics. 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. Efficient server selection algorithms reduce retrieval time for objects replicated on different servers and are an important component of Internet cache architectures. This paper empirically evaluates six client-side server selection algorithms. The study compares two statistical algorithms, one using median bandwidth and the other median latency, a dynamic probe algorithm, two hybrid algorithms, and random selection. The server pool includes a topologically dispersed set of United States state government Web servers. Experiments were run on three clients in different cities and on different regional networks. The study examines the effects of time-of day, client resources, and server proximity. Differences in performance highlight the degree of algorithm adaptability and the effect that network upgrades can have on statistical estimators. Dynamic network probing performs as well or better than the statistical bandwidth algorithm and the two probe bandwidth hybrid algorithms. The statistical latency algorithm is clearly worse, but does outperform random selection. Bandwidth is an extremely valuable and scarce resource in mobile networks; therefore, efficient mobility-aware bandwidth reservation is necessary to support multimedia applications (e.g., video streaming) that require quality of service (QoS). In this paper, we propose a distributed bandwidth reservation scheme called the mobility-prediction-aware bandwidth reservation (MPBR) scheme. The objective of MPBR is to reduce handoff call dropping rate and maintain acceptable new call blocking rate while providing efficient bandwidth utilization. MPBR consists of 1) a handoff time estimation (HTE) scheme that aims to estimate the time windows when a user will perform handoffs along the path to his destination, 2) an available bandwidth estimation (ABE) scheme that aims to estimate in advance available bandwidth, during the computed time windows in the cells to be traversed by the user to his destination, and 3) an efficient call admission control (ECaC) scheme that aims to control bandwidth allocation in the network cells. The simulation results show that MPBR outperforms existing schemes in terms of reducing handoff call dropping rate. As local area wireless networks based on the IEEE 802.11 standard see increasing public deployment, it is important to ensure that access to the network by different users remains fair. While fairness issues in 802.11 networks have been studied before, this paper is the first to focus on TCP fairness in 802.11 networks in the presence of both mobile senders and receivers. In this paper, we evaluate extensively through analysis, simulation, and experimentation the interaction between the 802.11 MAC protocol and TCP. We identify four different regions of TCP unfairness that depend on the buffer availability at the base station, with some regions exhibiting significant unfairness of over 10 in terms of throughput ratio between upstream and downstream TCP flows. We also propose a simple solution that can be implemented at the base station above the MAC layer that ensures that different TCP flows share the 802.11 bandwidth equitably irrespective of the buffer availability at the base station. In this paper, we present a simulation-based study of BitTorrent. Our results confirm that BitTorrent performs near-optimally in terms of uplink bandwidth utilization and download time, except under certain extreme conditions. On fairness, however, our work shows that low bandwidth peers systematically download more than they upload to the network when high bandwidth peers are present. We find that the rate-based tit-for-tat policy is not effective in preventing unfairness. We show how simple changes to the tracker and a stricter, block-based tit-for-tat policy, greatly improves fairness, while maintaining high utilization. Communication at millimeter wave (mmWave) frequencies is defining a new era of wireless communication. The mmWave band offers higher bandwidth communication channels versus those presently used in commercial wireless systems. The applications of mmWave are immense: wireless local and personal area networks in the unlicensed band, 5G cellular systems, not to mention vehicular area networks, ad hoc networks, and wearables. Signal processing is critical for enabling the next generation of mmWave communication. Due to the use of large antenna arrays at the transmitter and receiver, combined with radio frequency and mixed signal power constraints, new multiple-input multiple-output (MIMO) communication signal processing techniques are needed. Because of the wide bandwidths, low complexity transceiver algorithms become important. There are opportunities to exploit techniques like compressed sensing for channel estimation and beamforming. This article provides an overview of signal processing challenges in mmWave wireless systems, with an emphasis on those faced by using MIMO communication at higher carrier frequencies. Internet video makes up a significant part of the Internet traffic and its fraction is constantly growing. In order to guarantee best user experience throughout different network access technologies with dynamically varying network conditions, it is fundamental to adopt technologies enabling a proper delivery of the media content. One of such technologies is adaptive streaming. It allows to dynamically adapt the bit-rate of the stream to varying network conditions. There are various approaches to adaptive streaming. In our work, we focus on the receiver-driven approach where the media file is subdivided into segments, each of the segments is provided at multiple bit-rates, and the task of the client is to select the appropriate bit-rate for each of the segments. With this approach, the challenges are (i) to properly estimate the dynamics of the available network throughput, (ii) to control the filling level of the client buffer in order to avoid underflows resulting in playback interruptions, (iii) to maximize the quality of the stream, while avoiding unnecessary quality fluctuations, and, finally, (iv) to minimize the delay between the user's request and the start of the playback. During our work, we designed and implemented a receiver-driven adaptation algorithm for adaptive streaming that does not rely on cross-layer information or server assistance. We integrated the algorithm with a prototype implementation of a streaming client based on the MPEG DASH (Dynamic Adaptive Streaming over HTTP) standard. We evaluated the implemented prototype in real-world scenarios and found that it performes remarkably well even under challenging network conditions. Further, it exhibits stable and fair operation if a common link is shared among multiple clients. We discuss the relevance of fairness as a design objective for congestion control mechanisms in the Internet. Specifically, we consider a backbone network shared by a dynamic number of short-lived flows, and study the impact of bandwidth sharing on network performance. In particular, we prove that for a broad class of fair bandwidth allocations, the total number of flows in progress remains finite if the load of every link is less than one. We also show that provided the bandwidth allocation is "sufficiently" fair, performance is optimal in the sense that the throughput of the flows is mainly determined by their access rate. Neither property is guaranteed with unfair bandwidth allocations, when priority is given to one class of flow with respect to another. This suggests current proposals for a differentiated services Internet may lead to suboptimal utilization of network resources. BBR is a recently proposed congestion control. Instead of using packet loss as congestion signal, like many currently used congestion controls, it uses an estimate of the available bottleneck link bandwidth to determine its sending rate. BBR tries to provide high link utilization while avoiding to create queues in bottleneck buffers. The original publication of BBR shows that it can deliver superior performance compared to Cubic TCP in some environments. This paper provides an independent and extensive experimental evaluation of BBR at higher speeds. The experimental setup uses BBR's Linux kernel 4.9 implementation and typical data rates of 10Gbit s and 1 Gbit s at the bottleneck link. The experiments vary the flows' round-trip times, the number of flows, and buffer sizes at the bottleneck. The evaluation considers throughput, queuing delay, packet loss, and fairness. On the one hand, the intended behavior of BBR could be observed with our experiments. On the other hand, some severe inherent issues such as increased queuing delays, unfairness, and massive packet loss were also detected. The paper provides an in-depth discussion of BBR's behavior in different experiment setups. In this survey, we present state-of-the-art bitrate adaptation algorithms for HTTP adaptive streaming (HAS). As a key distinction from other streaming approaches, the bitrate adaptation algorithms in HAS are chiefly executed at each client, i.e. , in a distributed manner. The objective of these algorithms is to ensure a high quality of experience (QoE) for viewers in the presence of bandwidth fluctuations due to factors like signal strength, network congestion, network reconvergence events, etc. While such fluctuations are common in public Internet, they can also occur in home networksor even managed networks where there is often admission control and QoS tools. Bitrate adaptation algorithms may take factors like bandwidth estimations, playback buffer fullness, device features, viewer preferences, and content features into account, albeit with different weights. Since the viewer’s QoE needs to be determined in real-time during playback, objective metrics are generally used including number of buffer stalls, duration of startup delay, frequency and amount of quality oscillations, and video instability. By design, the standards for HAS do not mandate any particular adaptation algorithm, leaving it to system builders to innovate and implement their own method. This survey provides an overview of the different methods proposed over the last several years. Wireless local area networks (W-LANs) have become increasingly popular due to the recent availability of affordable devices that are capable of communicating at high data rates. These high rates are possible, in part, through new modulation schemes that are optimized for the channel conditions bringing about a dramatic increase in bandwidth efficiency. Since the choice of which modulation scheme to use depends on the current state of the transmission channel, newer wireless devices often support multiple modulation schemes, and hence multiple datarates, with mechanisms to switch between them Users are given the option to either select an operational datarate manually or to let the device automatically choose the appropriate modulation scheme (data rate) to match the prevailing conditions. Automatic rate selection protocols have been studied for cellular networks but there have been relatively few proposals for W-LANs. In this paper we present a rate adaptive MAC protocol called the Receiver-Based AutoRate (RBAR) protocol. The novelty of RBAR is that its rate adaptation mechanism is in the receiver instead of in the sender. This is in contrast to existing schemes in devices like the WaveLAN II [15]. We show that RBAR is better because it results in a more efficient channel quality estimation which is then reflected in a higher overall throughput Our protocol is based on the RTS CTS mechanism and consequently it can be incorporated into many medium access control protocols including the widely popular IEEE 802.11 protocol. Simulation results of an implementation of RBAR inside IEEE 802.11 show that RBAR performs consistently well. The demand for 360° Virtual Reality (VR) videos is expected to grow in the near future, thanks to the diffusion of VR headsets. VR Streaming is however challenged by the high bandwidth requirements of 360° videos. To save bandwidth, we spatially tile the video using the H.265 standard and stream only tiles in view at the highest quality. The video is also temporally segmented, so that each temporal segment is composed of several spatial tiles. In order to minimize quality transitions when the user moves, an algorithm is developed to predict where the user is likely going to watch in the near future. Consequently, predicted tiles are also streamed at the highest quality. Finally, the server push in HTTP 2 is used to deliver the tiled video. Only one request is sent from the client; all the tiles of a segment are automatically pushed from the server. This approach results in a better bandwidth utilization and video quality compared to traditional streaming over HTTP 1.1, where each tile has to be requested independently by the client. We showcase the benefits of our framework using a prototype developed on a Samsung Galaxy S7 and a Gear VR, which supports both tiled and non-tiled videos and streaming over HTTP 1.1 and HTTP 2. Under limited bandwidth conditions, we demonstrate how our framework can improve the quality watched by the user compared to a non-tiled solution where all of the video is streamed at the same quality. This result represents a major improvement for the efficient streaming of VR videos.