A significant amount of connection establishments on the web require a prior domain name resolution by the client. Especially on high-latency access networks, these DNS lookups cause a significant delay on the client's connection establishment with a server. To reduce the overhead of QUIC's connection establishment with prior DNS lookup on these networks, we propose a novel QuicSocks proxy. Basically, the client delegates the domain name resolution towards the QuicSocks proxy. Our results indicate, that colocating our proxy with real-world ISP-provided DNS resolvers provides great performance gains. For example, 10 of our 474 sample nodes distributed across ISP's in Germany would save at least 30ms per QUIC connection establishment. The design of our proposal aims to be readily deployable on the Internet by avoiding IP address spoofing, anticipating Network Address Translators and using the standard DNS and QUIC protocols. In summary, our proposal fosters a faster establishment of QUIC connections for clients on high-latency access networks.
Furthermore, Miniproxy @cite_2 can be used to accelerate TCP's connection establishment. This approach places a proxy between the client and the web server, which doubles the number of required TCP handshakes. Miniproxy can provide a faster TCP connection establishment in case of a favorable network topology and significant RTTs between client and web server. The QUIC protocol includes computationally expensive cryptographic handshakes causing a significant delay compared to TCP's handshake @cite_20 . Therefore, this approach seems less feasible when QUIC is used.
@cite_20 @cite_2
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TCP proxies are basic building blocks for many advanced middleboxes. In this paper we present Miniproxy, a TCP proxy built on top of a specialized minimalistic cloud operating system. Miniproxy's connection handling performance is comparable to that of full-fledged GNU Linux TCP proxy implementations, but its minimalistic footprint enables new use cases. Specifically, Miniproxy requires as little as 6 MB to run and boots in tens of milliseconds, enabling massive consolidation, on-the-fly instantiation and edge cloud computing scenarios. We demonstrate the benefits of Miniproxy by implementing and evaluating a TCP acceleration use case. QUIC is a performance-optimized secure transport protocol and a building block of the upcoming HTTP 3 standard. To protect against denial-of-service attacks, QUIC servers need to validate the IP addresses claimed by their clients. So far, the QUIC protocol conducts address validation for each hostname separately using validation tokens. In this work, we review this practice and introduce a new QUIC transport parameter to allow a shared address validation across hostnames. This parameter indicates to the client, that an issued validation token can be used to abbreviate the address validation when connecting to specific other hostnames. Based on trust-relations between real-world hostnames we evaluate the performance benefits of our proposal. Our results suggest that a shared address validation saves a round-trip time on almost 60 of the required handshakes to different hosts during the first loading of an average website. Assuming a typical transatlantic connection with a round-trip time of 90ms. We find that deploying our proposal reduces the delay overhead to establish all required connections for an average website by 142.2ms. QUIC (Quick UDP Internet Connection) is a new multiplexed and secure transport atop UDP, designed from the ground up and optimized for HTTP 2 semantics. While built with HTTP 2 as the primary application protocol, QUIC builds on decades of transport and security experience, and implements mechanisms that make it attractive as a modern general- purpose transport. QUIC provides multiplexing and flow control equivalent to HTTP 2, security equivalent to TLS, and connection semantics, reliability, and congestion control equivalent to TCP. This draft documents the early deployment of the QUIC protocol prior to standardization. Small TCP flows make up the majority of web flows. For them, the TCP three-way handshake represents a significant delay overhead. The TCP Fast Open (TFO) protocol provides zero round-trip time (0-RTT) handshakes for subsequent TCP connections to the same host. In this paper, we present real-world privacy and performance limitations of TFO. We investigated its deployment on popular websites and browsers. We found that a client revisiting a web site for the first time fails to use an abbreviated TFO handshake about 40 of the time due to web server load-balancing. Our analysis further reveals significant privacy problems in the protocol design and implementation. Network-based attackers and online trackers can exploit these shortcomings to track the online activities of users. As a countermeasure, we introduce a novel protocol called TCP Fast Open Privacy (FOP). It overcomes the performance and privacy limitations of TLS over TFO by utilizing a custom TLS extension. TCP FOP prevents tracking by network attackers and impedes third-party tracking, while still allowing for 0-RTT handshakes as in TFO. As a proof-of-concept, we have implemented the proposed protocol. Our measurements indicate that TCP FOP outperforms TLS over TFO when websites are served from multiple IP addresses. Nowadays, network traffic classification plays an important role in many fields including network management, intrusion detection system, malware detection system, etc. Most of the previous research works concentrate on features extracted in the non-encrypted network traffic. However, these features are not compatible with all kind of traffic characterization. Google's QUIC protocol (Quick UDP Internet Connection protocol) is implemented in many services of Google. Nevertheless, the emergence of this protocol imposes many obstacles for traffic classification due to the reduction of visibility for operators into network traffic, so the port and payload- based traditional methods cannot be applied to identify the QUIC- based services. To address this issue, we proposed a novel technique for traffic classification based on the convolutional neural network which combines the feature extraction and classification phase into one system. The proposed method uses the flow and packet-based features to improve the performance. In comparison with current methods, the proposed method can detect some kind of QUIC-based services such as Google Hangout Chat, Google Hangout Voice Call, YouTube, File transfer and Google play music. Besides, the proposed method can achieve the microaveraging F1-score of 99.24 percent. We present our experience with QUIC, an encrypted, multiplexed, and low-latency transport protocol designed from the ground up to improve transport performance for HTTPS traffic and to enable rapid deployment and continued evolution of transport mechanisms. QUIC has been globally deployed at Google on thousands of servers and is used to serve traffic to a range of clients including a widely-used web browser (Chrome) and a popular mobile video streaming app (YouTube). We estimate that 7 of Internet traffic is now QUIC. We describe our motivations for developing a new transport, the principles that guided our design, the Internet-scale process that we used to perform iterative experiments on QUIC, performance improvements seen by our various services, and our experience deploying QUIC globally. We also share lessons about transport design and the Internet ecosystem that we learned from our deployment. Providing real-time cloud services to Vehicular Clients (VCs) must cope with delay and delay-jitter issues. Fog computing is an emerging paradigm that aims at distributing small-size self-powered data centers (e.g., Fog nodes) between remote Clouds and VCs, in order to deliver data-dissemination real-time services to the connected VCs. Motivated by these considerations, in this paper, we propose and test an energy-efficient adaptive resource scheduler for Networked Fog Centers (NetFCs). They operate at the edge of the vehicular network and are connected to the served VCs through Infrastructure-to-Vehicular (I2V) TCP IP-based single-hop mobile links. The goal is to exploit the locally measured states of the TCP IP connections, in order to maximize the overall communication-plus-computing energy efficiency, while meeting the application-induced hard QoS requirements on the minimum transmission rates, maximum delays and delay-jitters. The resulting energy-efficient scheduler jointly performs: (i) admission control of the input traffic to be processed by the NetFCs; (ii) minimum-energy dispatching of the admitted traffic; (iii) adaptive reconfiguration and consolidation of the Virtual Machines (VMs) hosted by the NetFCs; and, (iv) adaptive control of the traffic injected into the TCP IP mobile connections. The salient features of the proposed scheduler are that: (i) it is adaptive and admits distributed and scalable implementation; and, (ii) it is capable to provide hard QoS guarantees, in terms of minimum maximum instantaneous rates of the traffic delivered to the vehicular clients, instantaneous rate-jitters and total processing delays. Actual performance of the proposed scheduler in the presence of: (i) client mobility; (ii) wireless fading; and, (iii) reconfiguration and consolidation costs of the underlying NetFCs, is numerically tested and compared against the corresponding ones of some state-of-the-art schedulers, under both synthetically generated and measured real-world workload traces. In this paper, we focus on various experiments conducted to analyze the performance of Transmission Control Protocol (TCP) and Universal Datagram Protocol (UDP) based applications in a IEEE 802.16 deployed network. We analyze the effect of Medium Access Control (MAC) and Physical layer characteristics on the performance of TCP and UDP-based applications. Our key findings are: (i) throughput achieved by TCP-based applications is lower than that of UDP-based applications, (ii) slot utilization of TCP-based applications is lower than that of UDP-based applications and (iii) throughput of TCP-based applications suffers in the presence of UDP-based applications for similar channel states. We also observe that throughput achieved by both TCP and UDP-based applications with Automatic Repeat ReQuest (ARQ) set in system are higher as compared to that of without ARQ. The findings of these experiments can be adopted while designing efficient scheduling schemes for IEEE 802.16-based network, such that higher throughput, utilization and better delay performance can be achieved. 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. The current move to Cloud Computing raises the need for verifiable delegation of computations, where a weak client delegates his computation to a powerful server, while maintaining the ability to verify that the result is correct. Although there are prior solutions to this problem, none of them is yet both general and practical for real-world use. We demonstrate a relatively efficient and general solution where the client delegates the computation to several servers, and is guaranteed to determine the correct answer as long as even a single server is honest. We show: A protocol for any efficiently computable function, with logarithmically many rounds, based on any collision-resistant hash family. The protocol is set in terms of Turing Machines but can be adapted to other computation models. An adaptation of the protocol for the X86 computation model and a prototype implementation, called Quin, for Windows executables. We describe the architecture of Quin and experiment with several parameters on live clouds. We show that the protocol is practical, can work with nowadays clouds, and is efficient both for the servers and for the client. This work studies the potential and impact of the FRANC network coding protocol for delivering high quality Dynamic Adaptive Streaming over HTTP (DASH) in wireless networks. Although DASH aims to tailor the video quality rate based on the available throughput to the destination, it relies on the TCP protocol for reliability in data delivery. TCP is known to drop its throughput performance by several fold in the presence of even 1 or 2 packet losses, which are common in wireless systems. This will force DASH to settle at a much lower video resolution, thus reducing the user's quality of experience. We show that the use of FRANC, an adaptive network coding protocol that provides both low delay and high throughput to upper layers, as a reliability mechanism for TCP can significantly increase video quality. As part of our analysis, we benchmark the performance of various TCP versions, including CUBIC, Reno, Veno, Vegas, and Westwood+, under different packet loss rates in wireless systems using a real testbed with Raspberry Pi devices. Our goal was to choose the most promising TCP version in terms of delay performance, in this case TCP Reno, and make a fair comparison between TCP running alone and using FRANC underneath for reliability. Our demonstrator with DASH in Raspberry Pi devices using the DASH benchmark, shows that the video rate delivered is 4× higher when using FRANC. Even in harsh packet loss conditions, FRANC is able to deliver higher data rates (increase 4×), while experiencing significantly shorter (decrease 10×) video lags. 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. 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. The soft real-time nature of large scale web applications in today's datacenters, combined with their distributed workflow, leads to deadlines being associated with the datacenter application traffic. A network flow is useful, and contributes to application throughput and operator revenue if, and only if, it completes within its deadline. Today's transport pro- tocols (TCP included), given their Internet origins, are agnostic to such flow deadlines. Instead, they strive to share network resources fairly. We show that this can hurt application performance. Motivated by these observations, and other (previously known) deficiencies of TCP in the datacenter environment, this paper presents the design and implementation of D3, a deadline-aware control protocol that is customized for the datacenter environment. D3 uses explicit rate control to apportion bandwidth according to flow deadlines. Evaluation from a 19-node, two-tier datacenter testbed shows that D3, even without any deadline information, easily outper- forms TCP in terms of short flow latency and burst tolerance. Further, by utilizing deadline information, D3 effectively doubles the peak load that the datacenter network cansupport. Recent advances in signal processing have demonstrated in-band full-duplex capability at WiFi ranges. In addition to simultaneous two-way exchange between two nodes, full-duplex access points can potentially support simultaneous uplink and downlink flows. However, the atomic three-node topology, which allows simultaneous uplink and downlink, leads to inter-client interference. In this paper, we propose a random-access medium access control protocol using distributed power control to manage inter-client interference in wireless networks with full-duplex-capable access points that serve half-duplex clients. Our key contributions are two-fold. First, we identify the regimes in which power control provides sum throughput gains for the three-node atomic topology, with one uplink flow and one downlink flow. Second, we develop and benchmark PoCMAC, a full 802.11-based protocol that allows distributed selection of a three-node topology. The proposed MAC protocol is shown to achieve higher capacity as compared to an equivalent half-duplex counterpart, while maintaining similar fairness characteristics in single contention domain networks. We carried out extensive simulations and software-defined radio-based experiments to evaluate the performance of the proposed MAC protocol, which is shown to achieve a significant improvement over its half-duplex counterpart in terms of throughput performance. Multipath TCP (MPTCP) is a major modification to TCP that enables a single transport connection to use multiple paths. Smartphones can benefit from MPTCP by using both WiFi and 3G 4G interfaces for their data-traffic, potentially improving the performance and allowing mobility through vertical handover. However, MPTCP requires a modification of the end hosts, thus suffers from the chicken-and-egg deployment problem. A global deployment of MPTCP is therefore expected to take years. To increase the incentives for clients and servers to upgrade their system, we propose MiMBox an efficient protocol converter that can translate MPTCP into TCP and vice versa to provide multipath benefits to early adopters of MPTCP. MiMBox is application agnostic and can be used transparently or explicitly. Moreover, a close attention was paid to the implementation's design to achieve good forwarding performance. MiMBox is implemented entirely in the Linux kernel so that it is able to more easily circumvent the bottlenecks of a user-space implementation. Measurements show that we always outperform user-space solutions and that the performance is close to plain IP packet forwarding. For interactive networked applications like web browsing, every round-trip time (RTT) matters. We introduce ASAP, a new naming and transport protocol that reduces latency by shortcutting DNS requests and eliminating TCP's three-way handshake, while ensuring the key security property of verifiable provenance of client requests. ASAP eliminates between one and two RTTs, cutting the delay of small requests by up to two-thirds. 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. 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. 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.