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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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. TLS can resume previous connections via abbreviated resumption handshakes that significantly decrease the delay and save expensive cryptographic operations. For that, cryptographic TLS state from previous connections is reused. TLS version 1.3 recommends to avoid resumption handshakes, and thus the reuse of cryptographic state, when connecting to a different hostname. In this work, we reassess this recommendation, as we find that sharing cryptographic TLS state across hostnames is a common practice on the web. We propose a TLS extension that allows the server to inform the client about TLS state sharing with other hostnames. This information enables the client to efficiently resume TLS sessions across hostnames. Our evaluation indicates that our TLS extension provides huge performance gains for the web. For example, about 58.7 of the 20.24 full TLS handshakes that are required to retrieve an average website on the web can be converted to resumed connection establishments. This yields to a reduction of 44 of the CPU time consumed for TLS connection establishments. Furthermore, our TLS extension accelerates the connection establishment with an average website by up to 30.6 for TLS 1.3. Thus, our proposal significantly reduces the (energy) costs and the delay overhead in the encrypted web. 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. 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. 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. We present and evaluate an implementation of a prototype scalable web server consisting of a load-balanced cluster of hosts that collectively accept and service TCP connections. The host IP addresses are advertised using round robin DNS (RR-DNS) technique, allowing any host to receive requests from any client. Once a client attempts to establish a TCP connection with one of the hosts, a decision is made as to whether or not the connection should be redirected to a different host-namely, the host with the lowest number of established connections. We use the low-overhead Distributed Packet Rewriting (DPR) technique to redirect TCP connections. In our prototype, each host keeps information about the remaining hosts in the system. Load information is maintained using periodic multicast amongst the cluster hosts. Performance measurements suggest that our prototype outperforms both pure RR-DNS and the stateless DPR solutions. Scalable reliable multicast protocols have been the focus of recent research, tackling the problem of efficient reliable data delivery to an arbitrarily large number of receivers. Yet, the common applications of multicast, such as multi-point file delivery, or video streaming from a media server, typically only involve a moderate number of receivers, such as a thousand or fewer. Moreover, because of the limited deployment of these specialized multicast protocols, it is common, when feasible, for applications to use multiple TCP connections instead, one for each receiver, to implement multi-point delivery, causing a significant demand on the transmission server and the downstream links. We describe a multicast extension to TCP, called single-source multicast optimization (SMO), that optimizes this case of multipoint delivery, providing the benefits of multicast together with the familiar features and API of TCP. Our results from experiments based on a Linux implementation and performed on a testbed show that TCP-SMO requires just a modest extension to the TCP implementation and provides scalable performance of multicast up to over a thousand receivers, thereby satisfying the common case requirements. In addition, used with TCP-RTM (real-time mode), TCP-SMO also supports real-time multimedia multicast applications well. 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. 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 mobile devices, the wireless network interface card (WNIC) consumes a significant portion of overall system energy. One way to reduce energy consumed by a device is to transition its WNIC to a lower-power steep mode when data is not being received or transmitted. In this paper, we investigate client-centered techniques for energy efficient communication, using IEEE 802.11b, within the network layer. The basic idea is to conserve energy by keeping the WNIC in high-power mode only when necessary. We track each connection, which allows us to determine inactive intervals during which to transition the WNIC to sleep mode. Whenever necessary, we also shape the traffic from the client side to maximize sleep intervals-convincing the server to send data in bursts. This trades lower WNIC energy consumption for an increase in transmission time. Our techniques are compatible with standard TCP and do not rely on any assistance from the server or network infrastructure. Results show that during Web browsing, our client-centered technique saved 21 percent energy compared to PSM and incurred less than a 1 percent increase in transmission time compared to regular TCP. For a large file download, our scheme saved 27 percent energy on average with a transmission time increase of only 20 percent IP technology for resource-constrained devices en- ables transparent end-to-end connections between a vast variety of devices and services in the Internet of Things (IoT). To protect these connections, several variants of traditional IP security protocols have recently been proposed for standardization, most notably the DTLS protocol. In this paper, we identify significant resource requirements for the DTLS handshake when employing public-key cryptography for peer authentication and key agree- ment purposes. These overheads particularly hamper secure com- munication for memory-constrained devices. To alleviate these limitations, we propose a delegation architecture that offloads the expensive DTLS connection establishment to a delegation server. By handing over the established security context to the con- strained device, our delegation architecture significantly reduces the resource requirements of DTLS-protected communication for constrained devices. Additionally, our delegation architecture naturally provides authorization functionality when leveraging the central role of the delegation server in the initial connection establishment. Hence, in this paper, we present a comprehensive, yet compact solution for authentication, authorization, and secure data transmission in the IP-based IoT. The evaluation results show that compared to a public-key-based DTLS handshake our delegation architecture reduces the memory overhead by 64 , computations by 97 , network transmissions by 68 . An accurate mapping of traffic to applications is important for a broad range of network management and measurement tasks. Internet applications have traditionally been identified using well-known default server network-port numbers in the TCP or UDP headers. However this approach has become increasingly inaccurate. An alternate, more accurate technique is to use specific application-level features in the protocol exchange to guide the identification. Unfortunately deriving the signatures manually is very time consuming and difficult.In this paper, we explore automatically extracting application signatures from IP traffic payload content. In particular we apply three statistical machine learning algorithms to automatically identify signatures for a range of applications. The results indicate that this approach is highly accurate and scales to allow online application identification on high speed links. We also discovered that content signatures still work in the presence of encryption. In these cases we were able to derive content signature for unencrypted handshakes negotiating the encryption parameters of a particular connection. This document specifies the BBR congestion control algorithm. BBR uses recent measurements of a transport connection's delivery rate and round-trip time to build an explicit model that includes both the maximum recent bandwidth available to that connection, and its minimum recent round-trip delay. BBR then uses this model to control both how fast it sends data and the maximum amount of data it allows in flight in the network at any time. Relative to loss-based congestion control algorithms such as Reno [RFC5681] or CUBIC [draft-ietf-tcpm-cubic], BBR offers substantially higher throughput for bottlenecks with shallow buffers or random losses, and substantially lower queueing delays for bottlenecks with deep buffers (avoiding "bufferbloat"). This algorithm can be implemented in any transport protocol that supports packet-delivery acknowledgment (thus far, open source implementations are available for TCP [RFC793] and QUIC [draft-ietf- quic-transport-00]). 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. 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. Facebook is one of the most famous social network sites hosting a number of users approaching a billion [1]. Facebook is a cloud-based web site which contains a number of advanced technologies behind the scene. Despite the fact that it is the web site that most users open for all-day and, in some places, all-night long, and its traffic is generally part of all types of networks — wired and wireless, PAN, LAN, MAN, and WAN —, there is virtually no research work or technical paper that report on client's perspective of what happen when users login onto their Facebook homepages. How many and in what order components loaded are, which and how many servers those components loaded are, or how many TCP streams used, are examples of questions which users and network administrators have a little knowledge of. Our work tries to answer the questions by examining every of over 2,000 packets per a single Facebook homepage retrieval using the Wireshark packet capturing software. From the captured packets, we thoroughly examined all objects that Facebook retrieved and categorized them into groups based on the characteristics. Our investigations found that Facebook retrieves these objects from both international and domestic servers, and over a number of parallel TCP streams. The results also showed that the Facebook traffic exhibited a two-spike bursty pattern, regardless of the loading time. 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. 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. The popularity of smartphones and smartphone applications means that data is the dominant traffic type in current mobile networks. In this paper we present our work on a systematic investigation into facets of the LTE EPC architecture that impact the performance of TCP as the predominant transport layer protocol used by applications on mobile networks. We found that (1) load increase in a cell causes dramatic bandwidth reduction on UEs and significantly degrades TCP performance, (2) seamless handover causes significant TCP losses while lossless handover increases TCP segments' delay. 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.