Open RAN is an emerging paradigm in mobile networks where the Radio Access Network (RAN) functions are disaggregated and virtualized on commodity servers. Despite the importance of Open RAN research, existing platforms often lack the fidelity and stability required to address a wide range of research problems. In response to this limitation, we have developed an enterprise-scale Open RAN testbed aimed at conducting state-of-the-art research in key areas that have received limited attention due to the lack of suitable platforms. In this poster, we provide an overview of the testbed we have created and examples of the research it has enabled, with the hope of catalyzing future open RAN research and innovation.

Leaf chlorophyll content (LCC) is an important plant physiological trait and is critical for accurate modeling of vegetation photosynthesis over time and space. To date, there is still a lack of a global long time-series dataset of LCC. In this study, we developed an algorithm to retrieve global LCC from MODIS surface reflectance data from 2000–2020. An essential requirement for generating LCC time series is to capture its seasonal dynamics. This issue was addressed by using a matrix system with two pairs of vegetation indices to minimize the impacts of leaf area index and canopy non-photosynthetic material on LCC estimation in different seasons. The matrix system algorithm was applied to Landsat data and MODIS data, respectively. The validation based on Landsat data and ground measurements reveals the algorithm has the ability to catch the seasonal variations of LCC in different plant functional types, and the MODIS-derived LCC shows good agreement with Landsat-upscaled LCC (R2=0.77, RMSE=6.9 μg/cm2). The global 8-day LCC data at 500-m resolution in 2000–2020 was generated using the matrix system from MODIS and presented distinct temporal and spatial variations, which provides a new opportunity for analyzing vegetation physiological dynamics in climate change studies.

State-of-the-art distributed in-memory datastores (FaRM, FaSST, DrTM) provide strongly-consistent distributed transactions with high performance and availability. Transactions in those systems are fully general; they can atomically manipulate any set of objects in the store, regardless of their location. To achieve this, these systems use complex distributed transactional protocols. Meanwhile, many workloads have a high degree of locality. For such workloads, distributed transactions are an overkill as most operations only access objects located on the same server - if sharded appropriately.

The connection between solar-induced fluorescence (SIF) and vegetation gross primary productivity is being widely investigated across spatial, temporal, and biological scales, including: a) studies at the leaf [1], [2], plant canopy [2]–[4] or satellite pixel scale [5], [6], b) temporally with studies spanning from diurnal [7] to seasonal scales [1], [3], [5], and b) biologically with studies covering various plant functional types (PFTs), e.g., crops [4], [7], deciduous [8] or evergreen forests [1], [3], in response to different sources of stress.

Gross primary production (GPP) from photosynthesis by terrestrial vegetation is the largest sink of atmospheric CO2. Sun-induced chlorophyll fluorescence (SIF) has been shown a powerful proxy for photosynthetic activity and used to estimate GPP. However, both non-physiological and functional factors controlling the emission of canopy SIF. The non-physiological factors, especially the sun-viewer geometry, impact the relationships between SIF and GPP. In this study, we did near-surface observations of both carbon flux and multi-view-angle spectra above a wheat canopy. The carbon flux was used to calculate GPP and the canopy spectra were used to retrieve SIF. SIF is significantly correlated with the angle between sun and viewer than SIF (R2=0.63). The relationships of SIF with GPP are also changing with different view azimuth angles. Generally, SIF observed at 180o(pointing north) are more correlated to GPP than that at the other two angles. A model recently developed by He et al. (2017) was used to normalize multi-angle SIF to the hotspot direction (SIFh) and to compute the canopy-level total SIF (SIFt), in order to reduce the effects of sun-viewer geometry on SIF. Compared to the correlation of GPP with the original observed SIF (SIFobs), the coefficients of determination (R2) increase by ~0.1 and ~0.05 for those of GPP with SIFhand SIFt, respectively. These results would be helpful in estimating GPP of crops at large scales using remote sensing techniques.

Sun-induced fluorescence (SIF) is an electromagnetic signal emitted by the chlorophyll of green leaves which has been shown to be a good proxy for plant photosynthetic activity. Recent developments in spaceborne spectrosopy have led to the derivation of the first global maps of SIF from a number of space-based spectrometers. This contribution will provide an overview on the state-of-the-art of terrestrial SIF monitoring in terms of existing and upcoming satellite missions and application fields.

Sun-induced fluorescence (SIF) is a novel remote sensing method, which can be potentially used as a direct indicator of photosynthetic activity. In this study, we present an approach for generating a consistent long-term global SIF product (2003-2015) by combining Scanning Imaging Absorption Spectrometer for Atmospheric Chartography (SCIAMACHY) and the Global Ozone Monitoring Experiment -2 (GOME-2). The relationships between SCIAMACHY (2003-2011) and GOME-2 SIF (2007-2015) were established pixel by pixel using two data series during overlapped period (2007-2011). Then the SCIAMACHY SIF from 2012 to 2015 was estimated from GOME-2 SIF. With the consistent SIF product, the influence of extreme climate event such as high surface temperature to the vegetation of different terrestrial ecosystem will be studied. The vegetation phenological patterns will also be explored by SIF data.

Remote sensing of sun-induced chlorophyll fluorescence (SIF) is a novel optical tool for assessment of terrestrial photosynthesis or gross primary production (GPP). Along with the breakthroughs of global retrievals of SIF from space-borne sensors, exploitation of SIF in improving the representation of photosynthesis and its role in Earth System models became a very relevant and active field. Recent space-borne measurements of SIF can offer an observational constraint on photosynthesis simulations. Tailored to this special session, this presentation gives a discussion on recent advances in the retrievals of leaf biological traits from SIF, e.g., the maximum carboxylation rate (Vcmax), regarding the applications and problems.

Although cellular networks can be provisioned according to the peak demand, they usually experience large fluctuations in both channel conditions and traffic load level. Scheduling with both channel and load awareness allows us to exploit the delay tolerance of data traffic to alleviate network congestion, and thus reduce the peak. However, solving the optimal scheduling problem leads to a large-scale Markov decision process (MDP) with extremely high complexity. In this paper, we propose a scalable and distributed approach to this problem, called Coordinated Scheduling (CoSchd). CoSchd decomposes the large-scale MDP problem into many individual MDP problems, each of which can be solved independently by each user under a limited amount of coordination signals from the base station (BS). We show that CoSchd is close to optimal when the number of users becomes large. Furthermore, we propose an approximation of CoSchd that iteratively updates the scheduling policy based on online measurements. Simulation results demonstrate that exploiting channel and load awareness with CoSchd can effectively alleviate cellular network congestion.

In this paper we investigate single-hop multicast transmission in which randomly located multiple transmitters multicast packets to a cluster of receivers. Packet retransmission is known as a promising mechanism for improving the transmission reliability. Our focus is on evaluating (i) the minimum required delay (retransmission attempts), τ*, for establishing an outage-free multicast, where a transmitted packet is successfully decoded by entire nodes in the cluster, and (ii) Multicast Progress Radius (MPR) for a given delay constraint. MPR indicates how far, on average, a packet can successfully progress in a cluster without outage while the retransmission delay is restricted. Assuming general fading distribution, we derive closed-form expressions for the cumulative distribution function of τ*, and MPR. By simulations we confirmed our analysis and studied the impact of several system parameters on the MPR. Based on results of this paper we conclude that outage-free multicast requires a very large number of retransmission attempts, thus not practically achievable only based on retransmission.

Today's WLANs are struggling to provide desirable features like high efficiency, fairness and QoS because of the use of Distributed Coordination Function (DCF). In this paper we present OpenTDMF, an architecture to enable TDMA on commodity WLAN devices. Our hope is to provide the desirable features without entirely rebuilding the WLAN infrastructure. OpenTDMF is inspired by and architecturally similar to Software Defined Networking (SDN). Specifically, we leverage the backhaul of WLAN to coordinate all the stations for channel access. This fine-grained coordination is performed in a decoupled control plane which includes a central controller and programmable APs. To realize OpenTDMF on commodity WLAN devices, we develop several novel techniques to achieve μs-level time synchronization among all the APs. We also enable AP-triggered uplink transmission so that all the transmissions in the WLAN can be determined. We implemented a prototype of OpenTDMF based on commodity WLAN devices. Empirical results validate the OpenTDMF design and demonstrate its benefits.

Similar to TV bands, the guardband frequencies are not occupied therefore are whitespace that potentially allows additional communication activities. Considering the difference to TV whitespace, we propose independent communication for guardband whitespace. In this paper, we present the Pilotfish system which realizes independent communication and turns guardband whitespace into new communication channels. To address the big challenges of interference mitigation, we employ novel PHY design which includes specially customized FBMC and an Nulled Decoding technique to null the strong background signal in guardbands. We implemented Pilotfish using software radio system. Empirical evaluation results validate the Pilotfish design in both PHY and MAC.

The ever increasing popularity of social networks and the ever easier photo taking and sharing experience have led to unprecedented concerns on privacy infringement. Inspired by the fact that the Robot Exclusion Protocol, which regulates web crawlers' behavior according a per-site deployed robots.txt, and cooperative practices of major search service providers, have contributed to a healthy web search industry, in this paper, we propose Privacy Expressing and Respecting Protocol (PERP) that consists of a Privacy.tag -- a physical tag that enables a user to explicitly and flexibly express their privacy deal, and Privacy Respecting Sharing Protocol (PRSP) -- a protocol that empowers the photo service provider to exert privacy protection following users' policy expressions, to mitigate the public's privacy concern, and ultimately create a healthy photo-sharing ecosystem in the long run. We further design an exemplar Privacy.Tag using customized yet compatible QR-code, and implement the Protocol and study the technical feasibility of our proposal. Our evaluation results confirm that PERP and PRSP are indeed feasible and incur negligible computation overhead.

We have observed two trends, growing wireless capability at the physical layer powered by MIMO-OFDM and growing video traffic as the dominant application traffic. Both the video source and MIMO-OFDM channel components exhibit nonuniform energy distribution. This has motivated us to leverage the source data redundancy at the channel to achieve high video recovery performance. We propose ParCast+ that first separates the source and the channel into independent components, matches the more important source components with higher-gain channel components, allocates power weights with joint consideration to the source and the channel, and uses pseudo-analog modulation for transmission. Such a scheme achieves fine-grained unequal error protection across source components. We implemented ParCast+ in Matlab and on Sora. Extensive evaluation has shown that our scheme outperforms competing schemes by notable margins, sometimes up to 6.4 dB in PSNR for challenging scenarios.

Wi-Fi tethering (i.e., sharing the Internet connection of a mobile phone via its Wi-Fi interface) is a useful functionality and is widely supported on commercial smartphones. Yet, existing Wi-Fi tethering schemes consume excessive power: they keep the Wi-Fi interface in a high power state regardless if there is ongoing traffic or not. In this paper, we propose DozyAP to improve the power efficiency of Wi-Fi tethering. Based on measurements in typical applications, we identify many opportunities that a tethering phone could sleep to save power. We design a simple yet reliable sleep protocol to coordinate the sleep schedule of the tethering phone with its clients without requiring tight time synchronization. Furthermore, we develop a two-stage, sleep interval adaptation algorithm to automatically adapt the sleep intervals to ongoing traffic patterns of various applications. DozyAP does not require any changes to the 802.11 protocol and is incrementally deployable through software updates. We have implemented DozyAP on commercial smartphones. Experimental results show that, while retaining comparable user experiences, our implementation can allow the Wi-Fi interface to sleep for up to 88% of the total time in several different applications and reduce the system power consumption by up to 33% under the restricted programmability of current Wi-Fi hardware.

Email is a key application used on smartphones. Even when the phone is in stand-by mode, users expect the phone to continue syncing with an email server to receive new mes-sages. Each such sync operation wakes up the smartphone for data reception and processing. In this paper, we show that this "cost of email sync" in stand-by mode constitutes a significant source of energy consumption, and thus reduces battery life. We quantify the power performance of different existing email clients on two smartphone platforms, An-droid and Windows Phone, and study the impact of system parameters such as email size, inbox size, and pull vs. push. Our results show that existing email clients do not handle email sync in an energy efficient way. This is because the underlying protocols and architectures are not designed for the specific needs of operating in stand-by mode. Based on our findings, we derive general design principles for energy-efficient event handling on smartphones, and apply these principles to the case of email sync and implement our techniques on commercial smartphones. Experimental results show that our techniques are able to significantly reduce energy cost of email sync by 49.9% on average with our experiment settings.

Multi-user multiple-input multiple-output (MU-MIMO) is the latest communication technology that promises to linearly increase the wireless capacity by deploying more antennas on access points (APs). However, the large number of MIMO antennas will generate a huge amount of digital signal samples in real time. This imposes a grand challenge on the AP design by multiplying the computation and the I/O requirements to process the digital samples. This paper presents BigStation, a scalable architecture that enables realtime signal processing in large-scale MIMO systems which may have tens or hundreds of antennas. Our strategy to scale is to extensively parallelize the MU-MIMO processing on many simple and low-cost commodity computing devices. Our design can incrementally support more antennas by proportionally adding more computing devices. To reduce the overall processing latency, which is a critical constraint for wireless communication, we parallelize the MU-MIMO processing with a distributed pipeline based on its computation and communication patterns. At each stage of the pipeline, we further use data partitioning and computation partitioning to increase the processing speed. As a proof of concept, we have built a BigStation prototype based on commodity PC servers and standard Ethernet switches. Our prototype employs 15 PC servers and can support real-time processing of 12 software radio antennas. Our results show that the BigStation architecture is able to scale to tens to hundreds of antennas. With 12 antennas, our BigStation prototype can increase wireless capacity by 6.8x with a low mean processing delay of 860μs. While this latency is not yet low enough for the 802.11 MAC, it already satisfies the real-time requirements of many existing wireless standards, e.g., LTE and WCDMA.

Reliable Group Data Delivery (RGDD) is a pervasive traffic pattern in data centers. In an RGDD group, a sender needs to reliably deliver a copy of data to all the receivers. Existing solutions either do not scale due to the large number of RGDD groups (e.g., IP multicast) or cannot efficiently use network bandwidth (e.g., end-host overlays). Motivated by recent advances on data center network topology designs (multiple edge-disjoint Steiner trees for RGDD) and innovations on network devices (practical in-network packet caching), we propose Datacast for RGDD. Datacast explores two design spaces: 1) Datacast uses multiple edge-disjoint Steiner trees for data delivery acceleration. 2) Datacast leverages in-network packet caching and introduces a simple soft-state based congestion control algorithm to address the scalability and efficiency issues of RGDD. Our analysis reveals that Datacast congestion control works well with small cache sizes (e.g., 125KB) and causes few duplicate data transmissions (e.g., 1.19%). Both simulations and experiments confirm our theoretical analysis. We also use experiments to compare the performance of Datacast and BitTorrent. In a BCube(4, 1) with 1Gbps links, we use both Datacast and BitTorrent to transmit 4GB data. The link stress of Datacast is 1.01, while it is 1.39 for BitTorrent. By using two Steiner trees, Datacast finishes the transmission in 16.9s, while BitTorrent uses 52s.

With the increasing of physical-layer (PHY) data rate in modern wireless local area networks (WLANs) (e.g., 802.11n), the overhead of media access control (MAC) progressively degrades data throughput efficiency. This trend reflects a fundamental aspect of the current MAC protocol, which allocates the channel as a single resource at a time. This paper argues that, in a high data rate WLAN, the channel should be divided into separate subchannels whose width is commensurate with the PHY data rate and typical frame size. Multiple stations can then contend for and use subchannels simultaneously according to their traffic demands, thereby increasing overall efficiency. We introduce FICA, a fine-grained channel access method that embodies this approach to media access using two novel techniques. First, it proposes a new PHY architecture based on orthogonal frequency division multiplexing (OFDM) that retains orthogonality among subchannels while relying solely on the coordination mechanisms in existing WLAN, carrier sensing and broadcasting. Second, FICA employs a frequency-domain contention method that uses physical-layer Request to Send/Clear to Send (RTS/CTS) signaling and frequency domain backoff to efficiently coordinate subchannel access. We have implemented FICA, both MAC and PHY layers, using a software radio platform, and our experiments demonstrate the feasibility of the FICA design. Furthermore, our simulation results show FICA can improve the efficiency of WLANs from a few percent to 600% compared to existing 802.11.

Transport Control Protocol (TCP) incast congestion happens in high-bandwidth and low-latency networks when multiple synchronized servers send data to the same receiver in parallel. For many important data-center applications such as MapReduce and Search, this many-to-one traffic pattern is common. Hence TCP incast congestion may severely degrade their performances, e.g., by increasing response time. In this paper, we study TCP incast in detail by focusing on the relationships between TCP throughput, round-trip time (RTT), and receive window. Unlike previous approaches, which mitigate the impact of TCP incast congestion by using a fine-grained timeout value, our idea is to design an Incast congestion Control for TCP (ICTCP) scheme on the receiver side. In particular, our method adjusts the TCP receive window proactively before packet loss occurs. The implementation and experiments in our testbed demonstrate that we achieve almost zero timeouts and high goodput for TCP incast.

Most IP-geolocation mapping schemes [14], [16], [17], [18] take delay-measurement approach, based on the assumption of a strong correlation between networking delay and geographical distance between the targeted client and the landmarks. In this paper, however, we investigate a large region of moderately connected Internet and find the delay-distance correlation is weak. But we discover a more probable rule - with high probability the shortest delay comes from the closest distance. Based on this closest-shortest rule, we develop a simple and novel IP-geolocation mapping scheme for moderately connected Internet regions, called GeoGet. In GeoGet, we take a large number of webservers as passive landmarks and map a targeted client to the geolocation of the landmark that has the shortest delay. We further use JavaScript at targeted clients to generate HTTP/Get probing for delay measurement. To control the measurement cost, we adopt a multistep probing method to refine the geolocation of a targeted client, finally to city level. The evaluation results show that when probing about 100 landmarks, GeoGet correctly maps 35.4 percent clients to city level, which outperforms current schemes such as GeoLim [16] and GeoPing [14] by 270 and 239 percent, respectively, and the median error distance in GeoGet is around 120 km, outperforming GeoLim and GeoPing by 37 and 70 percent, respectively.

Reliable Group Data Delivery (RGDD) is a pervasive traffic pattern in data centers. In an RGDD group, a sender needs to reliably deliver a copy of data to all the receivers. Existing solutions either do not scale due to the large number of RGDD groups (e.g., IP multicast) or cannot efficiently use network bandwidth (e.g., end-host overlays).

There have been some serious concerns about the TCP performance in data center networks, including the long completion time of short TCP flows in competition with long TCP flows, and the congestion due to TCP incast. In this paper, we show that a properly tuned instant queue length based Explicit Congestion Notification (ECN) at the intermediate switches can alleviate both problems. Compared with previous work, our approach is appealing as it can be supported on current commodity switches with a simple parameter setting and it does not need any modification on ECN protocol at the end servers. Furthermore, we have observed a dilemma in which a higher ECN threshold leads to higher throughput for long flows whereas a lower threshold leads to more senders on incast under buffer pressure. We address this problem with a switch modification only scheme - dequeue marking, for further tuning the instant queue length based ECN to achieve optimal incast performance and long flow throughput with a single threshold value. Our experimental study demonstrates that dequeue marking is effective for increasing the maximum incast senders close to the performance limit of ECN, achieving a gain anywhere from 16% to 140%.

We observe two trends, growing wireless capability at the physical layer powered by MIMO-OFDM, and growing video traffic as the dominant application traffic. Both the source and MIMO-OFDM channel components exhibit non-uniform energy distribution. This motivates us to leverage the source data redundancy at the channel to achieve high video recovery performance. We propose ParCast that first separates the source and channel into independent components, matches the more important source components with higher-gain channel components, allocates power weights with joint consideration to the source and the channel, and uses analog modulation for transmission. Such a scheme achieves fine-grained unequal error protection across source components. We implemented ParCast in Matlab and on Sora. Extensive evaluation shows that our scheme outperforms competitive schemes by notable margins, sometimes up to 5~dB in PSNR for challenging scenarios.

Retransmissions reduce the efficiency of data communication in wireless networks because of: (i) per-retransmission packet headers, (ii) contention overhead on every retransmission, and (iii) redundant bits in every retransmission. In fact, every retransmission nearly doubles the time to successfully deliver the packet. To improve spectrum efficiency in a lossy environment, we propose a new in-frame retransmission scheme using uACKs. Instead of waiting for the entire transmission to end before sending the ACK, the receiver sends smaller uACKs for every few symbols, on a separate narrow feedback channel. Based on these uACKs, the sender only retransmits the lost symbols after the last data symbol in the frame, thereby adaptively changing the frame size to ensure it is successfully delivered. We have implemented uACK on the Sora platform. Experiments with our prototype validate the feasibility of symbol-level uACK . By significantly reducing the retransmistion overhead, the sender is able to aggressively use higher data rate for a lossy link. Both improve the overall network efficiency. Our experimental results from a controlled environment and an 9-node software radio testbed show that uACK can have up to 140% throughput gain over 802.11g and up to 60% gain over the best known retransmission scheme.

Wi-Fi tethering (i.e., sharing the Internet connection of a mobile phone via its Wi-Fi interface) is a useful functionality and is widely supported on commercial smartphones. Yet existing Wi-Fi tethering schemes consume excessive power: they keep the Wi-Fi interface in a high power state regardless if there is ongoing traffic or not. In this paper we propose DozyAP to improve the power efficiency of Wi-Fi tethering. Based on measurements in typical applications, we identify many opportunities that a tethering phone could sleep to save power. We design a simple yet reliable sleep protocol to coordinate the sleep schedule of the tethering phone with its clients without requiring tight time synchronization. Furthermore, we develop a two-stage, sleep interval adaptation algorithm to automatically adapt the sleep intervals to ongoing traffic patterns of various applications. DozyAP does not require any changes to the 802.11 protocol and is incrementally deployable through software updates. We have implemented DozyAP on commercial smartphones. Experimental results show that, while retaining comparable user experiences, our implementation can allow the Wi-Fi interface to sleep for up to 88% of the total time in several different applications, and reduce the system power consumption by up to 33% under the restricted programmability of current Wi-Fi hardware.

We present the design and implementation of BeepBeep, a high-accuracy acoustic-based system for ranging and localization. It is a pure software-based solution and uses the most basic set of commodity hardware -- a speaker, a microphone, and some form of interdevice communication. The ranging scheme works without any infrastructure and is applicable to sensor platforms and commercial-off-the-shelf mobile devices. It achieves high accuracy through three techniques: two-way sensing , self-recording , and sample counting . We further devise a scalable and fast localization scheme. Our experiments show that up to one-centimeter ranging accuracy and three-centimeter localization accuracy can be achieved.

Multicast benefits data center group communication in both saving network traffic and improving application throughput. The SLA (Service Level Agreement) of cloud service requires the computation correctness of distributed applications, translating to the requirement of reliable Multicast delivery. In this paper we present RDCM, a novel reliable Multicast approach for data center network. The key idea of RDCM is to minimize the impact of packet loss on the Multicast performance, by leveraging the rich link resource in data centers. A Multicast-tree-aware backup overlay is purposely built on group members for peer-to-peer packet repair. Riding on Unicast, packet repair not only achieves complete repair isolation, but also has high probability to bypass the pathological links in the Multicast tree where packet loss occurs. The backup overlay is organized in such a way that it causes little individual repair burden, control overhead, as well as overall repair traffic. We have implemented RDCM as a user-level library on Windows platform. The experiments on our test bed show that RDCM handles packet loss without obvious throughput degradation during high-speed data transmission.

The goal of data-center networking is to interconnect a large number of server machines with low equipment cost while providing high network capacity and high bisection width. It is well understood that the current practice where servers are connected by a tree hierarchy of network switches cannot meet these requirements. In this paper, we explore a new server-interconnection structure. We observe that the commodity server machines used in today's data centers usually come with two built-in Ethernet ports, one for network connection and the other left for backup purposes. We believe that if both ports are actively used in network connections, we can build a scalable, cost-effective interconnection structure without either the expensive higher-level large switches or any additional hardware on servers. We design such a networking structure called FiConn. Although the server node degree is only 2 in this structure, we have proven that FiConn is highly scalable to encompass hundreds of thousands of servers with low diameter and high bisection width. We have developed a low-overhead traffic-aware routing mechanism to improve effective link utilization based on dynamic traffic state. We have also proposed how to incrementally deploy FiConn.

In this paper, we propose virtual data center (VDC) as the unit of resource allocation for multiple tenants in the cloud. VDCs are more desirable than physical data centers because the resources allocated to VDCs can be rapidly adjusted as tenants' needs change. To enable the VDC abstraction, we design a data center network virtualization architecture called SecondNet. SecondNet achieves scalability by distributing all the virtual-to-physical mapping, routing, and bandwidth reservation state in server hypervisors. Its port-switching based source routing (PSSR) further makes SecondNet applicable to arbitrary network topologies using commodity servers and switches. SecondNet introduces a centralized VDC allocation algorithm for bandwidth guaranteed virtual to physical mapping. Simulations demonstrate that our VDC allocation achieves high network utilization and low time complexity. Our implementation and experiments show that we can build SecondNet on top of various network topologies, and SecondNet provides bandwidth guarantee and elasticity, as designed.

TCP incast congestion happens in high-bandwidth and low-latency networks, when multiple synchronized servers send data to a same receiver in parallel [15]. For many important data center applications such as MapReduce[5] and Search, this many-to-one traffic pattern is common. Hence TCP in-cast congestion may severely degrade their performances, e.g., by increasing response time.

This demonstration shows a novel virtualization architecture, called Multi-Purpose Access Point (MPAP), which can virtualize multiple heterogenous wireless standards based on software radio. The basic idea is to deploy a wide-band radio front-end to receive wireless signals from all wireless standards sharing the same spectrum band, and use separate software base-bands to demodulate information stream for each wireless standard. Based on software radio, MPAP consolidates multiple wireless devices into single hardware platform, and allows them to share the same general-purpose computing resource. Different software base-bands can easily communicate and coordinate with one another. Thus, it also provides better coexistence among heterogenous wireless standards. As an example, we demonstrate to use non-contiguous OFDM in 802.11g PHY to avoid the mutual interference with narrow-band ZigBee communication.

Modern communication technologies are steadily advancing the physical layer (PHY) data rate in wireless LANs, from hundreds of Mbps in current 802.11n to over Gbps in the near future. As PHY data rates increase, however, the overhead of media access control (MAC) progressively degrades data throughput efficiency. This trend reflects a fundamental aspect of the current MAC protocol, which allocates the channel as a single resource at a time.

Sora is a fully programmable, high performance software radio platform based on commodity general-purpose PC. In this demonstration, we illustrate the main features of the Sora platform that provide researchers flexible and powerful means to conduct wireless experiments at different levels with various goals. Specifically, the demonstrator will show four useful applications for wireless research that are built based on the Sora platform: 1) A capture tool that allows one to take a snapshot on a wireless channel; 2) a signal generation tool that allows one to transmit arbitrary baseband wave-form over the air, from a monophonic tone to a complex modulated frame; 3) an on-line real-time receiving application that uses the Sora User-Mode Extension; and 4) a fully featured Software radio WiFi driver (SoftWiFi) that can seamlessly inter-operate with commercial WiFi cards.

Mobile phones are truly personal devices loaded with personal data such as photos, contacts, and call history. Yet it is often necessary or desirable to share our phones with others. This is especially true as mobile phones are integrating features conventionally provided by other dedicated devices, from MP3 players to games consoles. Yet existing phones assume a single user and provide little protection for private data and applications when a phone is shared. That is, when we lend our phones to others, we give away complete access. In this work, we present xShare, a protection solution to address this problem. xShare allows phone owners to rapidly specify what they want to share and place the phone into a restricted mode where only the data and applications intended for sharing can be accessed. We first present two formative user studies and derive the design requirements of xShare. We then offer the design of xShare based on file-level access control. We describe the implementation of xShare on Windows Mobile and report a comprehensive evaluation, including performance measurements, usability, and a one-month field trial.

Shipping-container-based data centers have been introduced as building blocks for constructing mega-data centers. However, it is a challenge on how to interconnect those containers together with reasonable cost and cabling complexity, due to the fact that a mega-data center can have hundreds or even thousands of containers and the aggregate bandwidth among containers can easily reach tera-bit per second. As a new inner-container server-centric network architecture, BCube [9] interconnects thousands of servers inside a container and provides high bandwidth support for typical traffic patterns. It naturally serves as a building block for mega-data center.

In this paper, we present Structon, a novel approach that uses Web mining together with inference and IP traceroute to geolocate IP addresses with significantly better accuracy than existing automated approaches. Structon is composed of three ideas which we realize in three corresponding steps. First, we extract geolocation information of Web server IP addresses from Web pages. Second, we devise heuristic algorithms to improve both the accuracy and the coverage of the IP geolocation database using these Web server IP addresses and their geolocations as input. Third, for those segments that are not covered in the first two steps, we use IP traceroute to identify the access routers of those segments. When the location of the access router is known, we can deduce the location of the associated segment since it is co-located together with the access router. By mining 500-million Web pages collected in China in 2006 (11 percent of the total Web pages in China at that time), we are able to identify the geolocations for 103 million IP addresses. This represents nearly 88 percent IP addresses allocated to China in March 2008. Structon is 87.4 percent accurate at city granularity and up to 93.5 percent accurate at province level. We also used 10 day Windows Live client log to evaluate our client IP addresses coverage: Structon identified geolocations of 98.9 percent of client IP addresses.

Loaded with personal data, e.g. photos, contacts, and call history, mobile phones are truly personal devices. Yet it is often necessary or desirable to share our phones with others. This is especially true as mobile phones are integrating features conventionally provided by other dedicated devices, from MP3 players to games consoles. Unfortunately, when we lend our phones to others, we give away complete access because existing phones assume a single user and provide little protection for private data and applications. In this work, we present xShare, a protection solution to address this problem. xShare allows phone owners to rapidly specify what they want to share and place the phone into a restricted mode where only the data and applications intended for sharing can be accessed.

Point&Connect (P&C) offers an intuitive and resilient device pairing solution on standard mobile phones. Its operation follows the simple sequence of point-andconnect: when a user plans to pair her mobile phone with another device nearby, she makes a simple hand gesture that points her phone towards the intended target. The system will capture the user's gesture, understand the target selection intention, and complete the device pairing. P&C is intention-based, intuitive, and reduces user efforts in device pairing. The main technical challenge is to come up with a simple system technique to effectively capture and understand the intention of the user, and pick the right device among many others nearby. It should further work on any mobile phones or small devices without relying on infrastructure or special hardware. P&C meets this challenge with a novel collaborative scheme to measure maximum distance change based on acoustic signals. Using only a speaker and a microphone, P&C can be implemented solely in user-level software and work on COTS phones. P&C adds additional mechanisms to improve resiliency against imperfect user actions, acoustic disturbance, and even certain malicious attacks. We have implemented P&C in Windows Mobile phones and conducted extensive experimental evaluation, and showed that it is a cool and effective way to perform device pairing.

This paper presents BCube, a new network architecture specifically designed for shipping-container based, modular data centers. At the core of the BCube architecture is its server-centric network structure, where servers with multiple network ports connect to multiple layers of COTS (commodity off-the-shelf) mini-switches. Servers act as not only end hosts, but also relay nodes for each other. BCube supports various bandwidth-intensive applications by speeding-up one-to-one, one-to-several, and one-to-all traffic patterns, and by providing high network capacity for all-to-all traffic.

Rate adaptation is critical to the system performance of wireless networks. Typically, rate adaptation is considered as a MAC layer mechanism in IEEE 802.11. Most previous work relies only on frame losses to infer channel quality, but performs poorly if frame losses are mainly caused by interference. Recently SNR- based rate adaptation schemes have been proposed, but most of them have not been studied in a real environment. In this paper, we first conduct a systematic measurement-based study to confirm that in general SNR is a good prediction tool for channel quality, and identify two key challenges for this to be used in practice: (1) The SNR measures in hardware are often uncalibrated, and thus the SNR thresholds are hardware dependent. (2) The direct prediction from SNR to frame delivery ratio (FDR) is often over optimistic under interference conditions. Based on these observations, we present a novel practical SNR- Guided Rate Adaptation (SGRA) scheme. We implement and evaluate SGRA in a real test-bed and compare it with other three algorithms: ARF, RRAA and HRC. Our results show that SGRA outperforms the other three algorithms in all cases we have tested.

It has been a challenging problem to support VoIP-type delay sensitive applications in an 802.11 wireless LAN, because the standard handoff procedure implemented in many current 802.11 products occurs a delay deem unacceptable to VoIP users. To reduce this delay, we have developed a fast handoff scheme called Proactive Scan. It employs two new techniques. The first is to decouple the time-consuming channel scan from the actual handoff, and to eliminate channel scan delay by doing scan early and interleaving it with ongoing traffic in a non-intrusive way. The second technique is a smart trigger that takes into account both uplink and downlink quality and explicitly addresses the link asymmetry which has yet not been touched in previous work. Through implementation and experimentation study, we have shown that Proactive Scan does provide fast handoff and satisfactory performance to VoIP applications. Further, it is a software-only client-only solution that any mobile device can use in any existing 802.11 networks.

We envision a new better-together mobile application paradigm where multiple mobile devices are placed in a close proximity and study a specific together-viewing video application in which a higher resolution video isplayed back across screens of two mobile devices placed side by side. This new scenario imposes real-time, synchronous decoding and rendering requirements which are difficult to achieve because of the intrinsic complexity of video andthe resource constraints such as processing power and battery life of mobile devices. We develop a novel efficient collaborative half-frame decoding schemeand design a tightly coupled collaborative system architecture that aggregates resources of both devices to achieve the task. We have implemented the system and conducted experimental evaluation. Results confirm that our proposed collaborative and resource aggregation techniques can achieve our vision of better-together mobile experiences.

The demo, BeepBeep, shows a high-accuracy acoustic-based ranging system without relaying on any pre-planned infrastructure or inter-device time synchronization. Moreover, the BeepBeep is a pure software-based solution and readily applicable to many low-cost sensor platforms and to most commercial-off-the-shelf mobile devices. Our demo using two common cell phones shows that BeepBeep can achieve average two centimeters accuracy within a range of more than ten meters.

We present the design, implementation, and evaluation of BeepBeep, a high-accuracy acoustic-based ranging system. It operates in a spontaneous, ad-hoc, and device-to-device context without leveraging any pre-planned infrastructure. It is a pure software-based solution and uses only the most basic set of commodity hardware -- a speaker, a microphone, and some form of device-to-device communication -- so that it is readily applicable to many low-cost sensor platforms and to most commercial-off-the-shelf mobile devices like cell phones and PDAs. It achieves high accuracy through a combination of three techniques: two-way sensing, self-recording, and sample counting. The basic idea is the following. To estimate the range between two devices, each will emit a specially-designed sound signal ("Beep") and collect a simultaneous recording from its microphone. Each recording should contain two such beeps, one from its own speaker and the other from its peer. By counting the number of samples between these two beeps and exchanging the time duration information with its peer, each device can derive the two-way time of flight of the beeps at the granularity of sound sampling rate. This technique cleverly avoids many sources of inaccuracy found in other typical time-of-arrival schemes, such as clock synchronization, non-real-time handling, software delays, etc. Our experiments on two common cell phone models have shown that we can achieve around one or two centimeters accuracy within a range of more than ten meters, despite a series of technical challenges in implementing the idea.

Prior studies have shown that XCP D. Katabi et al., (2002), while showing high potential for effective congestion control, may have significant performance problems if it misestimates the bottleneck link capacity Y. Zhang et al., (2005). To explore the magnitude of this implication and ultimately to contain it, we have conducted a control theoretic analysis of the XCP protocol and have studied its properties in the presence of capacity estimation errors. Under such conditions, by using a revised fluid model, we have discovered that XCP will not settle at zero steady-state error. However, we found that the steady-state error is bounded by the estimation error, and this bound can be exploited in XCP router queue size planning. This preliminary analysis explains what we have observed experimentally in an implementation study of XCP.

The explicit control protocol (XCP) has been proposed as a multi-level network feedback mechanism for congestion control of Internet transport protocols. Theoretical and simulation results have suggested that the protocol is stable and efficient over high bandwidth-delay product paths, while being more scalable to deploy than mechanisms that require per-flow state in routers. However, there is little operational experience with the approach. Since the deployment of XCP would require changes to both the end hosts and routers, it is important to study the implications of this new architecture before advocating such wide scale changes to Internets. This paper presents the results of an experimental study of XCP. We first implemented XCP in the Linux kernel and solved various systems issues. After validating previously reported simulation results, we studied the sensitivity of XCP's performance to various environmental factors, and discovered issues with TCP/IP configuration, capacity misestimation due to link sharing, handling of non-congestion losses, and the partial deployment of XCP queues in the network. These sensitivities can significantly reduce XCP's ability to control congestion and achieve fairness. Our contributions are twofold. First, through implementation we have revealed the challenges in platforms that lack large native data types or floating point arithmetic, and the need to keep fractions in the XCP protocol header. Second, through experiment and analysis we have identified several possibilities for XCP to enter into incorrect feedback control loops and adversely affect the performance. The challenges identified are deployment challenges intrinsic to the XCP design, and they suggest that the current proposal requires additional development and extension.

Developing and evaluating secure MANET (mobile adhoc networks) in real systems is a complex process that involves careful design of attack test cases and security countermeasures, as well as meaningful performance measurements to evaluate both the impact of attacks and the performance of security solutions. It is desirable to have a development and testing environment that can automate this process. In this paper, we propose a software framework for such an environment and describe a system implementation in the secure MANET routing domain. This environment includes the following three major features. First, the environment is built upon a wireless network emulation tool to support repeatable experimentation. Second, it adds an attack emulation layer with necessary API for easy development and execution of attack test cases. Third, the extensible attack library includes a full set of basic attacks at its core and a way to compose complex attacks from the atomic elements. To demonstrate the usefulness of this tool, we show the development of an Intrusion Detection System (IDS) as a case study. Our successful experience confirms that the platform can greatly facilitate the development of security solutions on MANET.

Transmission control protocol (TCP) performance enhancement proxy (PEP) mechanisms have been proposed, and in some cases widely deployed, to improve TCP performance in all-Internet protocol (IP) wireless networks. However, this technique is conflicted with IP-security (IPsec)-a standard IP security protocol that will make inroad into wireless networks. This paper analyzes the fundamental problem behind this conflict and develops a solution called multilayer IP-security (ML-IPsec). The basic principle is to use a multilayer protection model and a fine grain access control to make IP security protocols compatible with TCP PEP. It allows wireless network operators or service providers to grant base stations or wireless routers limited and controllable access to the TCP headers for performance enhancement purposes. Through careful design, implementation, and evaluation, we show that we can easily add ML-IPsec to existing IPsec software and the overhead is low. We conclude that ML-IPsec can help wireless networks provide both security and performance.

This work explores several system issues regarding the design and implementation of routing protocols for ad-hoc wireless networks. We examine the routing architecture in current operating systems and find it insufficient on several counts, especially for supporting on-demand or reactive routing protocols. Examples include lack of mechanisms for queuing outstanding packets awaiting route discovery and mechanisms for communicating route usage information from kernel to userspace. We propose an architecture and a generic API for any operating system to augment the current routing architecture. Implementing the API may normally require kernel modifications, but we provide an implementation for Linux using only the standard Linux 2.4 kernel facilities. The API is provided as a shared user-space library called the Ad-hoc Support Library (ASL), which uses a small loadable kernel module. To prove the viability of our framework, we provide a full-fledged implementation of the AODV protocol using ASL, and a design for the DSR protocol. Through this study, we also reinforce our belief that it is profoundly important to consider system issues in ad-hoc routing protocol design. 1

Group subscription is a useful mechanism for multicast congestion control: RLM, RLC, FLID-DL, and WEBRC form a promising line of multi-group protocols where receivers provide no feedback to the sender but control congestion via group membership regulation. Unfortunately, the group subscription mechanism also o#ers receivers an opportunity to elicit self-beneficial bandwidth allocations. In particular, a misbehaving receiver can ignore guidelines for group subscription and choose an unfairly high subscription level in a multi-group multicast session. This poses a serious threat to fairness of bandwidth allocation. In this paper, we present the first solution for the problem of inflated subscription. Our design guards access to multicast groups with dynamic keys and consists of two independent components: DELTA (Distribution of ELigibility To Access) -- a novel method for in-band distribution of group keys to receivers that are eligible to access the groups according to the congestion control protocol, and SIGMA (Secure Internet Group Management Architecture) -- a generic architecture for key-based group access at edge routers.

No abstract available.

In today's heterogeneous Internet, bandwidth available to TCP flows is often variable. However, current TCP cannot perform optimally under such dynamically varying bandwidth conditions. This paper addresses this problem by introducing a new architecture to improve TCP performance with explicit bandwidth notification (EBN). It uses a normalized bandwidth feedback method to provide accurate and timely bandwidth estimations. Then, a new TCP control algorithm (TCP-EBN) is proposed to give a prompt response to any bandwidth changes. Our simulation results have shown that TCP-EBN performs much better than several other variations of TCP.

In a Mobile Ad Hoc Network (MANET), temporary link failures and route changes happen frequently. With the assumption that all packet losses are due to congestion, TCP performs poorly in such environment. While there has been some research on improving TCP performance over MANET, most of them require feedback from the network or the lower layer. In this research, we explore a new approach to improve TCP performance by detecting and responding to out-of-order packet delivery events, which are the results of frequent route changes. In our simulation study, this approach had achieved on average 50% performance improvement, without requiring feedback from the network or the lower layer.

Mobile Ad-Hoc Network (MANET) has become an increas-ingly active research area with a plethora of work in ad-hoc routing, media access, and protocols, etc. However, much of the effort so far has been in simulation with only a few systems that have ever been implemented and none that we know have been tried in a scale beyond a dozen nodes. One reason is the high complexity involved in implementing and testing actual ad-hoc networks, and the lack of soft-ware tools for doing so. We have thus built an inexpensive and flexible environment to support such tasks and to facil-itate network research. The core component is a mobility emulator to test an ad-hoc network of virtually any scale and with any mobility scenario without actually moving the nodes physically.

In heterogeneous networks such as today's Internet, TCP must be made to handle links and paths with vastly different characteristics, including longer delays and dynamically changing available bandwidth. In this research, we first analyze the performance of TCP under networks with TCP-unfriendly characteristics and introduce an architecture to tackle the performance problems in such networks.

We believe that a network, to be survivable, must be heterogeneous. Just like a species that draws on a small gene pool can succumb to a single environmental threat, so a homogeneous network is vulnerable to a malicious attack that exploits a single weakness common to all of its components. In contrast, in a network in which each critical functionality is provided by a diverse set of protocols and implementations, attacks that focus on a weakness of one such protocol or implementation will not be able to bring down the entire network, even though all elements are not be bulletproof and even if some of components are compromised.Following this survivability through heterogeneity philosophy, we propose a new survivability paradigm, called heterogeneous networking, for improving a network's defense capabilities. Rather than following the current trend of converging towards single solutions to provide the desired functionality at every element of the network architecture, this methodology calls for systematically increasing the network's heterogeneity without sacrificing its interoperability.

As the recent denial-of-service attacks on several major Internet sites have shown us, no open computer network is immune from intrusions. The wireless ad-hoc network is particularly vulnerable due to its features of open medium, dynamic changing topology, cooperative algorithms, lack of centralized monitoring and management point, and lack of a clear line of defense. Many of the intrusion detection techniques developed on a fixed wired network are not applicable in this new environment. How to do it differently and effectively is a challenging research problem. In this paper, we first examine the vulnerabilities of a wireless ad-hoc network, the reason why we need intrusion detection, and the reason why the current methods cannot be applied directly. We then describe the new intrusion detection and response mechanisms that we are developing for wireless ad-hoc networks.

To allow a seamless integration between wireless LANs and wireless WANs, we developed a full stack adaptation model and a simple subnet architecture that superimposes Mobile-IP on cellular-type wireless LANs. The idea is to use Mobile IP as an integrative layer atop different LAN/WAN networks. While Mobile-IP is widely used in wireless WANs, it is not known how well it performs under a wireless LAN environment, against native MAC-level handoff. Through experimentation using the 802.11 W-LAN, we found that under practical values of handoff frequencies, the performance of Mobile IP based W-LAN handoff is almost identical to the performance of W-LAN handoff. Further performance studies show the suitability of Mobile-IP as an integrative layer in this architecture.

This paper describes a mechanism, called "persistent connection" to preserve stream connections after the communicating peer exits and till it restarts. Such connections have many applications: to survive failures that crash one party, network partitions that cut off the two parties, and temporary disconnection in a mobile computing environment. They can also facilitate suspension of process execution in a limited resource environment and maintain connectivity when one party migrates from one machine to another. Persistent connection uses logical endpoints to hide disconnection from applications and to achieve location independence. It can be constructed from the normal "transient" connection that goes down with processes. Prototypes have been developed on Unix to provide persistent connections in both the TCP-socket level and the DCE RPC level. Many existing programs can benefit from this software to achieve transparence to disconnection and relocation. We conclude that persistent connection is a convenient communication abstraction for reliable, adaptable, and reconfigurable applications.

O-Raid [1, 2] uses a layered approach to provide support for objects on top of a distributed relational database system called RAID [3], It reuses the replication controller of RAID to allow replication of simple objects as well as replication of composite objects. In this paper, we first describe the experiments conducted on O-Raid that measure the overheads incurred in supporting objects through a layered implementation, and the overheads involved in replicating objects. The overheads are low (e.g. 4ms for an insert query involving objects). We present experiments that evaluate three replication strategies for composite objects, namely, full replication, selective replication and no replication in a two-site and a four-site O-Raid system. For composite object experiments, the selective replication strategy demonstrated the flexibility of tuning replication of member objects based on the patterns of access. The experimentation is performed in different networking environments (LANs and WANs) to further evaluate the replication schemes. The results indicate that selective replication scheme has greater benefits in WAN than in LAN.

Abstract Computing involving data in a logo‐syllabic oriental language like Chinese is more difficult than in English. Many attempts have been made to develop bilingual or multilingual processing systems.1–3 Most of them are based on microcomputers like the IBM‐PC. With the widespread use of workstations and window systems, much better multi‐lingual processing environments can be provided. This paper describes the design and implementation of the program cxterm, a Chinese terminal emulator for the X Window System. We discuss the representation of multi‐byte international characters, the problems of Chinese character input and output, the pros and cons of various approaches, and the design decisions for cxterm. A feature of cxterm is its independence of input methods. A user can incorporate new input methods into cxterm at run‐time, without changing the program code. We also compare our approach with related work in multi‐lingual input/output in X, and describe how cxterm performs better in terms of efficiency, flexibility, and user‐friendliness.