This paper presents VisIoT, a system that tracks the location of a wireless transmitter in IoT devices and displays it in the screen of an AR device such as smart glasses and tablet. The proposed system benefits existing IoT systems by enabling intuitive interaction between a user and IoT devices and further enhancing visualization of the data collected from IoT sensors. VisIoT achieves them through a combination of wireless sensing and camera motion tracking. By using the azimuth and elevation angles between the wireless transmitter and the camera-equipped mobile device, VisIoT can instantly identify the location of the IoT device from the camera image. This paper introduces novel azimuth and elevation estimation algorithms that leverage the phase difference of the signals from two antennas together with the tracked camera rotation. We prototype VisIoT using a tablet PC and a USRP software radio, and develop a software that tracks and visualizes the location of ZigBee nodes in real time. The evaluation results show that VisIoT can accurately track the nodes with the median position error of 6%.

With the availability of inexpensive and powerful drones, it is possible to let drones automatically follow a user for video taping. This can not only reduce cost, but also support video taping in situations where otherwise not possible (e.g., during private moments or at inconvenient locations like indoor rock climbing). While there have been many follow-me drones on the market for outdoors, which rely on GPS, enabling indoor follow-me function is more challenging due to the lack of an effective approach to track users in indoor environments. To this end, we develop a holistic system that lets a mobile phone carried by a user accurately track the drone's relative location and control it to maintain a specified distance and orientation for automatic video taping. We develop a series of techniques to (i) track a drone's location using acoustic signals with sub-centimeter errors even under strong propeller noise from the drone and complicated multipath in indoor environments, and (ii) solve practical challenges in applying model predictive control (MPC) framework to control the drone. The latter consists of developing measurement-based flight models, designing measurement techniques to provide feedback to the controller, and predicting the user's movement. We implement our system on AR Drone 2.0 and Samsung S7. The extensive evaluation shows that our drone can follow a user effectively and maintain a specified following distance and orientation within 2-3 cm and 1-3 degree errors, respectively. The videos taped by the drone during flight are smooth according to the jerk metric.

Next generation devices, such as virtual reality (VR), augmented reality (AR), and smart appliances, demand a simple and intuitive way for users to interact with them. To address such needs, we develop a novel acoustic based device-free tracking system, called Strata, to enable a user to interact with a nearby device by simply moving his finger. In Strata, a mobile (e.g., smartphone) transmits known audio signals at inaudible frequency, and analyzes the received signal reflected by the moving finger to track the finger location. To explicitly take into account multipath propagation, the mobile estimates the channel impulse response (CIR), which characterizes signal traversal paths with different delays. Each channel tap corresponds to the multipath effects within a certain delay range. The mobile selects the channel tap corresponding to the finger movement and extracts the phase change of the selected tap to accurately estimate the distance change of a finger. Moreover, it estimates the absolute distance of the finger based on the change in CIR using a novel optimization framework. We then combine the absolute and relative distance estimates to accurately track the moving target. We implement our tracking system on Samsung Galaxy S4 mobile phone. Through micro-benchmarks and user studies, we show that our system achieves high tracking accuracy and low latency without extra hardware.

Wireless video traffic has grown at an unprecedented rate and put significant burden on wireless networks. Multicast can significantly reduce traffic by sending a single video to multiple receivers simultaneously. On the other hand, wireless receivers are heterogeneous due to both channel and antenna heterogeneity, the latter of which is rapidly increasing with the emergence of 802.11n and 802.11ac. In this paper, we develop optimized layered integrated video encoding (LIVE) to guarantee reasonable performance to weaker receivers (with worse channel and/or fewer antennas) and allow stronger receivers to enjoy better quality. Our approach has three distinct features: (i) It uses a novel layered coding to naturally accommodate the heterogeneity of different video receivers; (ii) It uses an optimization framework to optimize the amount of time used for transmission and the amount of information to transmit at each layer under the current channel condition; and (iii) It uses an integrated modulation, where most video data are transmitted using soft modulation to enjoy efficiency and resilience while the most important video data are transmitted using a combination of soft modulation and conventional hard modulation to further enhance their reliability. To our knowledge, this is the first approach that handles MIMO antenna heterogeneity in wireless video multicast. We demonstrate its effectiveness through extensive Matlab simulation and USRP testbed experiments.

Motivated by the recent push to deploy LTE in unlicensed spectrum, this paper develops a novel system to enable co-existence between LTE and WiFi. Our approach leverages LTE and WiFi antennas already available on smartphones to let LTE and WiFi transmit together and successfully decode the interfered signals. Our system offers several distinct advantages over existing MIMO work: (i) it can decode all the interfering signals under cross technology interference even when the interfering signals have similar power and occupy similar frequency, (ii) it does not need clean reference signals from either WiFi or LTE transmission, (iii) it can decode interfering WiFi MIMO and LTE transmissions, and (iv) it has a simple yet effective carrier sense mechanism for WiFi to access the medium under interfering LTE signals while avoiding other WiFi transmissions. We use USRP implementation and experiments to show its effectiveness.

No abstract available.

A mouse has been one of the most successful user interfaces due to its intuitive use. As more devices are equipped with displays and offer rich options for users to choose from, a traditional mouse that requires a surface to operate is no longer sufficient. While different types of air mice are available in the market, they rely on accelerometers and gyroscopes, which significantly limit the accuracy and ease of use.

Distributed multiple-input multiple-output (MIMO) promises a dramatic capacity increase. While significant theoretical work has been done on distributed MIMO at the physical layer, how to translate the physical layer innovation into tangible benefits to real networks remains open. In particular, realizing multi-point to multi-point MIMO involves the following challenges: (i) how to accurately synchronize multiple APs in phase and time in order to successfully deliver precoded signals to the clients, and (ii) how to develop a MAC protocol to effectively support multi-point to multi-point MIMO. In this paper, we develop a practical approach to address the above challenges. We implement multi-point to multi-point MIMO for both uplink and downlink to enable multiple APs to simultaneously communicate with multiple clients. We examine a number of important MAC design issues, such as how to access the medium, perform rate adaptation, support acknowledgments in unicast traffic, deal with losses/collisions, and schedule transmissions. We demonstrate its feasibility and effectiveness through a prototype implementation on USRP and SORA, two of the most well-known software defined radio platforms.

Explosive growth of WiFi traffic calls for new technologies to dramatically improve spectrum efficiency. In this paper, we propose an approach to adapt the spectrum on a per-frame basis. It consists of three major components: (i) a fine-grained spectrum access design that allows a sender and receiver to change their transmission and reception spectrum on demand, (ii) fast and accurate spectrum detection that allows a receiver to determine which spectrum is used by its sender on a per-frame basis by exploiting the IEEE 802.11 preamble structure, and (iii) an efficient spectrum allocation algorithm that determines which spectrum to use for each transmission by taking into account frequency diversity and interference. It can further be adapted to perform a joint assignment of spectrum, schedule, and access point (AP) for each frame. Using a SORA implementation and trace-driven simulation, we demonstrate the feasibility of per-frame spectrum adaptation and its significant benefit over existing channel assignment approaches. To our knowledge, this is the first per-frame spectrum adaptation prototype for WiFi networks.

We present SYNC, a physical layer transmission scheme that drastically reduces the cost of retransmission by introducing network coding concepts to symbol level operation. It piggybacks a new packet on each retransmitted packet, and exploits the previously received packet (possibly with error) at the receiver to recover both the retransmitted packet and the piggybacked packet. The piggybacking is achieved through higher modulation, but it does not decrease the decodability of the mixed packets owing to the previously received packet at the receiver, which can be analytically shown. SYNC works independently of other PHY level performance boosting schemes such as channel coding and spatial diversity. The proof-of-concept SYNC prototype has been implemented on a software defined radio (SDR) platform. The measurement data shows that under the same channel condition SYNC achieves 110 percent and 42 percent median throughput gain over traditional retransmission and SOFT, respectively. We also show that SYNC can be used proactively where the feedback as to the success of the previous transmission is not available, such as in broadcast.

An inherent limitation of the existing digital wireless network coding is that the relay node has to settle for a single broadcast rate for the coded packet transmission. Since the selected rate should be supported on the worst quality links to the intended receivers, the throughput gain by network coding is essentially bound to the capacity of the worst link. Worse yet, the bottleneck capacity diminishes as the diversity of links increases, which generally happens when the nodes participating in network coding operation grow in number. In this paper, we solve this "broadcast bottleneck" by using a novel symbol-level network coding scheme called Rate Diverse Network Coding (RDNC). With RDNC, the relay node can deal with receivers under disparate channel conditions with a single coded data stream, eliminating the single-rate broadcast bottleneck. Through extensive simulation, we find that RDNC significantly boosts the coding gain and the throughput, more when the given topology provides richer opportunities for coding. Specifically, RDNC is as good as COPE in the worst case, but can achieve up to 2.5 times the coding gain if the network topology permits.

The IEEE 802.11 wireless LAN (WLAN) is a time-tested technology, but it is still evolving towards even higher speeds and richer features. For multi-room IPTV delivery, where multiple IPTVs are fed from a main set-top box (STB), the IEEE 802.11 WLAN is an attractive choice for wirelessly connecting the STB and the IPTVs. In this paper, however, we first show that both the standard 802.11 unicast and broadcast/multicast delivery mechanisms have efficiency and reliability issues when they are applied to the multi-room IPTV delivery. In order to resolve the issues, we propose to use so called the pseudo-broadcast, where a unicast transmission is promiscuously received by multiple receivers. By dynamically changing the unicast destination based on the 802.11k measurement, the scheme also minimizes losses experienced by the promiscuous receivers. The scheme can be implemented without any change in the 802.11 standard, and provides a cost-effective solution to high-reliability and high-efficiency multi-room IPTV delivery.

For the WAVE applications built on safety beacons, it is imperative that the neighboring vehicle information be collected as quickly and efficiently as possible. But the unreliability of broadcast transmission and the randomness of channel access in the IEEE 802.11p MAC hamper the collection process. Specifically, the process suffers from essentially a form of the classic Coupon Collector's Problem, where it takes longer and longer to obtain the remaining information. In this paper, we solve the problem by introducing the application-level acknowledgement of the safety beacons. We demonstrate that this optimization drastically reduces the collection completion time, eventually contributing to the safety and efficiency in WAVE- based systems.

Although predicting the RFID tag distribution before a read cycle begins would be generally difficult and even futile, a likely and interesting scenario is where the tags have a sequential arrangement. In large-scale applications such as supply chain management, for instance, hundreds or thousands of the same type of product made by the same manufacturer can be stacked together in the stock. In this paper, we propose a simple RFID tag anti-collision algorithm that exploits such sequential structure of the given tag space to achieve a higher read efficiency. The proposed algorithm is also designed to still achieve the level of performance comparable to that of existing algorithms if tags exhibit little sequential structure. Through exhaustive simulations over various tag distributions, we demonstrate that the proposed algorithm achieves at least 10% improvement even in the worst case (i.e., when tags are completely randomly distributed), but yields significantly higher improvement when the number of tags and/or the correlation increases.

The already severe inefficiency problem in running VoIP on the IEEE 802.11-based wireless LAN is exacerbated in multi-hop networks due to spatial interference. This paper presents a novel scheme that significantly mitigates the effect of the interference by combining the inter-call aggregation and the pseudo-broadcast technique as used in network coding. As such, the proposed scheme works effectively not just for calls traveling the same routed path but also for calls crossing each other inside the multi-hop networks. We demonstrate through extensive simulation that the IEEE 802.11-based multi-hop network can be boosted up to 700% in terms of the number of supportable VoIP calls, with acceptable QoS, through the proposed scheme.

The technology based on the IEEE 802.11 standard has been hugely successful, and is evolving towards even higher speeds and richer features. In particular, the 802.11n amendment of the standard aims to achieve the physical layer (PHY) rate of 600 Mbps. Although the 802.11n offers sufficient bandwidth to support high-resolution video applications such as high definition TV (HDTV), we find that the number of video streams that can be supported on the IEEE 802.11n networks depends heavily on how the frame aggregation is implemented. In addition to the frame aggregation scheme stipulated in the amendment, we explore a multiple-receiver frame aggregation scheme for video traffic. The comparative study reveals through extensive simulation that the proposed multiple-receiver aggregation scheme increases the number of supported video streams by a factor of 2 or higher. We also shed light on the qualitative difference in the dynamics of the two approaches. Whereas the aggregation efficiency worsens with traffic increase in the point-to-point aggregation (which is highly undesirable), the proposed multiple- receiver aggregation exhibits resiliency against congestion, by matching the aggregation efficiency to the traffic load.

In this paper, we explore the feasibility of vehicle-to-vehicle (V2V) secure communication using public key cryptography. We implement a prototype and test it extensively in real driving conditions. Measurement data indicate that the system is feasible in terms of delay and loss, and that we can usually expect better performance than typical simulation works predict. However, we find that traffic congestion and the performance of public key cryptography component can pose significant challenges.

The bandwidth efficiency of voice over IP (VoIP) traffic on the IEEE 802.11 WLAN is notoriously low. VoIP over 802.11 incurs high bandwidth cost for voice frame packetization and MAC/PHY framing, which is aggravated by channel access overhead. For instance, 10 calls with the G.729 codec can barely be supported on 802.11b with acceptable QoS - less than 2% efficiency. As WLANs and VoIP services become increasingly widespread, this inefficiency must be overcome. This paper proposes a solution that boosts the efficiency high enough to support a significantly larger number of calls than existing schemes, with fair call quality. The solution comes in two parts: adaptive frame aggregation and uplink/downlink bandwidth equalization. The former reduces the absolute number of MAC frames according to the link congestion level, and the latter balances the bandwidth usage between the access point (AP) and wireless stations. When used in combination, they yield superior performance, for instance, supporting more than 100 VoIP calls over an IEEE 802.11b link. The authors demonstrate the performance of the proposed approach through extensive simulation, and validate the simulation through analysis.

Algorithmic approach to link adaptation for IEEE 802.11 networks such as Automatic Rate Fallback (ARF) is known to suffer from the inability to differentiate between collision and channel-induced error. In this paper, we propose a novel algorithm called COLA that overcomes the shortcoming and achieves near-optimal throughput over wide range of channel and load conditions. The result is significant since the throughput is achieved without any hardware support. Moreover, COLA does not require any optional or extra-protocol mechanisms support, either, such as RTS/CTS exchange, Clear Channel Assessment (CCS), and promiscuous channel monitoring. Finally, the COLA algorithm has a short critical path of just 10 instructions, and it is free of heuristic parameters, which will facilitate practical use.

The throughput performance of the distributed coordination function (DCF) of the IEEE 802.11 MAC protocol quickly degrades as the number of contending stations increases. To solve this problem, it has been shown recently that adaptive contention window modulation based on channel idle time tracking can be used, generating near optimal throughput. In this paper, we extend the approach for the IEEE 802.11e network, where different QoS classes are defined. We show how to find the class-specific optimal contention window sizes that yield the maximum aggregate throughput while maintaining the target throughput difference between classes

In multi-hop wireless networks, the number of supportable VoIP calls can be surprisingly small due to the increased spatial interference. To mitigate the interference, voice frame aggregation can be used. In this paper, we depart from the traditional approaches that perform aggregation at the voice source, and propose a technique called the self-controlled frame aggregation (SCFA) that runs at wireless routers. The core idea of SCFA is to let the congestion itself control the degree of aggregation. Unlike existing frame aggregation approaches, SCFA does not incur fixed delay cost, since it is used only when and by exactly as much as it is needed. In this paper, we take the example of 802.11-based multi-hop network to show the impact of SCFA, since many emerging multi-hop networks are built on the 802.11 technology. The result shows that SCFA on 802.11-based multi-hop network can boost the number of calls approximately twofold, or extends the hop distance threefold for a given number of calls to carry.

To solve the poor bandwidth efficiency problem of VoIP over IEEE 802.11 wireless LANs (WLANs), the literature frequently suggests elongating voice payload size. Such "voice frame aggregation" can be performed (1) over a single call or multiple calls, (2) at the voice source or at the wireless access point (AP), and (3) with additional repacketization delay or without it. This paper shows that over IEEE 802.11 WLANs, voice frame aggregation should better be done at the AP, over a single call, and without additional delay. We demonstrate through simulation and analysis that this particular implementation choice enables us to support far more VoIP calls on 802.11 WLANs than other choices. For one instance, we show that 100 or more G.729 calls are possible on 802.11b within ITU-T recommended delay bounds.