Interactive 3D simulated objects are crucial in AR/VR, animations, and robotics, driving immersive experiences and advanced automation. However, creating these articulated objects requires extensive human effort and expertise, limiting their broader applications. To overcome this challenge, we present Articulate-Anything, a system that automates the articulation of diverse, complex objects from many input modalities, including text, images, and videos. Articulate-Anything leverages vision-language models (VLMs) to generate code that can be compiled into an interactable digital twin for use in standard 3D simulators. Our system exploits existing 3D asset datasets via a mesh retrieval mechanism, along with an actor-critic system that iteratively proposes, evaluates, and refines solutions for articulating the objects, self-correcting errors to achieve a robust out- come. Qualitative evaluations demonstrate Articulate-Anything's capability to articulate complex and even ambiguous object affordances by leveraging rich grounded inputs. In extensive quantitative experiments on the standard PartNet-Mobility dataset, Articulate-Anything substantially outperforms prior work, increasing the success rate from 8.7-11.6\% to 75\% and setting a new bar for state-of-art performance. We further showcase the utility of our generated assets by using them to train robotic policies for fine-grained manipulation tasks that go beyond basic pick and place.

Simultaneous detection and estimation is important in many engineering applications. In particular, there are many applications where it is important to perform signal detection and Signal-to-Noise-Ratio (SNR) estimation jointly. Application of existing frameworks in the literature that handle simultaneous detection and estimation is not straightforward for this class of application. This paper therefore aims at bridging the gap between an existing framework, specifically the work by Middleton et al., and the mentioned application class by presenting a jointly optimal detector and signal and noise power estimators. The detector and estimators are given for the Gaussian observation model with appropriate conjugate priors on the signal and noise power. Simulation results affirm the superior performance of the optimal solution compared to the separate detection and estimation approaches.

In many detection applications with battery powered or energy-harvesting sensors, energy constraints preclude the use of the optimal detector all the time. Optimal energy-performance trade-off is therefore needed in such situations. In many applications, the signal and noise power may vary greatly over time, and this can be exploited to constrain energy consumption while maintaining the best possible performance. A detector scheduling algorithm based on the signal and noise power information is developed in this work. The resulting algorithm is simple due to its threshold-test structure and can be easily implemented with almost no overhead. A detection system with two detectors using the proposed scheduling scheme is estimated to greatly reduce the energy consumption for a wildlife monitoring application.

In recent years, wireless mesh networks (WMNs) have attracted considerable interest from both academia and industry. WMNs offer attractive advantages such as easy installation, fast deployment, and low operating expense. For this reason, many cities and network operators have built WMNs to provide wireless Internet access in metropolitan areas. Despite their advantages, WMNs face numerous challenges such as limited and fluctuating bandwidth of wireless links, interference among wireless links, and reliability issues due to broken links and faulty nodes caused by external impacts. Multipath routing can address these challenges and provides several benefits such as bandwidth aggregation, load balancing, optimal resource utilization, and high fault-tolerance. This paper presents a multipath routing design for WMNs. We evaluate our multipath routing design using simulations and compare it with the popular routing protocol Optimized Link State Routing Protocol (OLSR) for WMNs. Our evaluation shows promising results for our multipath routing design.

Beacon and event-driven warning messages are two important messages that are used by many Intelligent Transportation Systems (ITS) applications for improving road safety and traffic efficiency. Beacon messages are regularly broadcast by each vehicle to notify other vehicles of its presence and its status. Event-driven warning messages are emitted when a vehicle or the road infrastructure discerns a critical situation. When congestion occurs, it is necessary to reduce the beacon load to keep a certain amount of the available bandwidth for the event-driven messages because they carry time- critical information of high importance. This paper considers three beacon congestion control algorithms that control the beacon load below a certain threshold by adjusting the transmit rate and/or transmit power for the beacon messages. We perform a large set of simulations to evaluate these algorithms. Our simulation results reveal promising results for these algorithms.

In recent years, multi-channel MACs have attracted considerable attention in wireless networking research. Motivation for research into multi-channel MACs is their potential to increase the capacity of wireless ad hoc networks by exploiting multiple orthogonal frequency bands. This potential has been confirmed by several simulation studies that reported promising results for multi-channel MACs. However, the analytical performance of these protocols is not well understood. In this work, we present an analytical framework for modeling the performance multi-channel MACs in multi-hop networks. We employ our framework to investigate the performance of three representative multi channel MACs: Asynchronous Multichannel Coordination Protocol (AMCP), Multi-Channel MAC (MMAC), and Slotted Seeded Channel Hopping (SSCH). The important findings of our analysis are: (1) SSCH outperforms AMCP and MMAC in both 802.11b and 802.11a scenario, (2) AMCP performs worse than MMAC in 802.11b because the overhead of AMCP's control channel is higher than that of MMAC's control interval, and (3) the overhead of AMCP's control channel becomes negligible in an 802.11a scenario and is outweighed by that of MMAC's control interval. For this reason, AMCP delivers better performance than MMAC in 802.11a scenario.

Multi-channel MACs have recently attracted significant attention in wireless networking research because they possess the ability to exploit multiple frequency bands in order to increase the capacity of wireless ad hoc networks. A number of multi-channel MACs have been proposed in the last few years. However, these protocols rely on two rather impractical assumptions: low channel switching latency and fine-grained clock synchronization. This paper presents Asynchronous Multi-channel Hopping Protocol (AMHP), a multi-channel MAC protocol that does not build on these two impractical assumptions. We conduct an extensive performance evaluation and compare AMHP with 802.11 MAC, Asynchronous Multi-channel Coordination Protocol (AMCP), Multi-Channel MAC (MMAC), and Slotted Seeded Channel Hopping (SSCH). The major findings of our performance evaluation are: (1) all multi-channel MACs underperform 802.11 MAC at low load but AMHP comes very close to 802.11 MAC's performance at this load, (2) all multi-channel MACs give better performance than 802.11 MAC at medium and high load, (3) SSCH attains good results in one-hop scenarios or at low loads but loses its effectiveness at high loads and in multi-hop scenarios, and (4) AMHP outperforms all other multi-channel MACs at all loads.

The Internet Engineering Task Force (IETF) recently approved the proxy mobile IPv6 standard for network- based localized mobility management. Unlike host-based mobility management, the network handles mobility signaling on behalf of mobile nodes in proxy mobile IPv6, i.e., the mobile nodes are not involved in mobility management, hence have no control on handover-related signaling. The current specification of proxy mobile IPv6 defines basic operations for registration, deregistration, and tunnel management. As mobile nodes have no control on mobility management, the timing for handover signaling is not synchronized with the mobile node's ability to send and receive packets on the handover target link. This turns out to be an issue for mobile nodes being equipped with multiple network interfaces and performing handover between these access technologies (inter-technology handover), as the configuration of the target network interface and the associated link takes time. In this work, we investigate technical issues that occur during an inter-technology handover in proxy mobile IPv6. We demonstrate through experimental performance evaluation that an inter-technology handover in proxy mobile IPv6 can cause considerable handover delay and packet loss. We propose an efficient extension to proxy mobile IPv6 to eliminate delay and packet loss during an inter-technology handover. We demonstrate the effectiveness of the proposed extension through an experimental performance evaluation.

Various standardization development organizations consider network-based mobility management in future mobile communication networks to complement host-based mobility management. Proxy mobile IPv6 is currently being standardized to suit basic requirements of network-based mobility management, such as no-host involvement in the location update procedure. In this paper, we want to motivate advanced operation of multi-mode mobile terminals, which attach to a proxy mobile IPv6 enabled network through multiple network interfaces simultaneously. We propose extensions to proxy mobile IPv6 enabled networks to allow simultaneous operation of a device's multiple interfaces, which have the same IPv6 address configured, while being registered with a single mobility anchor. Along with a description of these extensions, this paper comprises results of measurements we could perform in a real test-bed to show the performance gain of two target scenarios for flow distribution and bandwidth aggregation.

We present a MAC protocol for opportunistic spectrum access (OSA-MAC) in cognitive wireless networks. The proposed MAC protocol works in a multi-channel environment which is capable of performing channel sensing to discover spectrum opportunities. For this MAC protocol, two channel selection methods are considered which trade the implementation complexity with throughput improvement. We then analyze the saturation throughput performance of the proposed MAC protocol under scenarios where the probabilities for each channel to be available to different secondary flows are the same or different. We then derive analytically the probability of collision of secondary users with primary users due to sensing errors. This analysis can be used, for example, to determine the requirement of sensing accuracy for secondary users, and to design an admission control method. We present numerical results to demonstrate the throughput performance of the OSA-MAC protocol and applications of the proposed analytical model.

Limited resources and time-varying nature of wireless ad hoc networks demand optimized use of resources across layers. Cross-layer optimization (CLO) for wireless networks, an approach that coordinates protocol behaviors at different layers with a goal to maximize a utility function, has received considerable attention lately. However, most existing work remains as theory and no practical CLO based on utility optimization exists today. The main difficulties in implementing theoretical CLO designs arise often from impractical assumptions about the characteristics of the wireless medium and also from computational and communication overhead of proposed solutions to achieve or approximate the optimality. In contrast, many existing practical approaches for CLO (not necessarily utility optimization) are rather ad hoc in nature and developed mostly based on intuitions. Thus, a clear gap between theory and practice in CLO exists. This paper addresses this dichotomy to close the gap by taking an optimal solution from utility-based CLO and applying practical approximation to enable a practical implementation in a wireless mesh network where nodes are statically positioned in an ad hoc fashion. We focus on the utility of maximizing throughput. We identify the impractical or computationally-intensive components of a theoretically-derived optimal throughput-maximizing solution and then propose, in most cases, practical approximation with O(1) complexity for MAC, scheduling, routing and congestion control. The result is a practical CLO solution that approximates the theoretically-derived optimal solution, but achieves much improved performance over existing practical CLO implementations.

We consider QoS-aware spectrum sharing in cognitive wireless networks where secondary users are allowed to access the spectrum owned by a primary network provider. The interference from secondary users to primary users is constrained to be below the tolerable limit. Also, signal to interference plus noise ratio (SINR) of each secondary user is maintained higher than a desired level for QoS insurance. When network load is high, admission control needs to be performed to satisfy both QoS and interference constraints. We propose an admission control algorithm which is performed jointly with power control such that QoS requirements of all admitted secondary users are satisfied while keeping the interference to primary users below the tolerable limit. When all secondary users can be supported at minimum rates, we allow them to increase their transmission rates and share the spectrum in a fair manner. We formulate the joint power/rate allocation with max-min fairness criterion as an optimization problem. We show how to transform it into a convex optimization problem so that its globally optimal solution can be obtained. Numerical results show that the proposed admission control algorithm achieves performance very close to the optimal solution. Also, impacts of different system and QoS parameters on the network performance are investigated for both admission control and rate/power allocation problems.

The widespread deployment of the IEEE 802.11 protocol has made it the de facto standard for wireless network access and allows Internet users to move freely (at least within a hotspot coverage). The convenience brought about by the IEEE 802.11 protocol is also accompanied by new technical challenges such as poor performance of the widely used transport protocol TCP due to the limited and varying bandwidth resources of the wireless medium. In this work, we seek to enhance efficiency of bandwidth usage for TCP over wireless links and improve TCP goodput. Our insight is that TCP acknowledgments can be spared on wireless links to save precious bandwidth resource of these links. We propose a simple technique that leverages the layer coordination between TCP and MAC to suppress TCP acknowledgments. Our technique is transparent to both TCP and MAC and does not require their modification or replacement.

Active queue management (AQM) in routers has been proposed as a solution to some of the scalability issues associated with TCP’s pure end-to-end approach to congestion control. However, beyond congestion control, controlling queues in routers is important because unstable router queues can cause poor application performance. Existing AQM schemes explicitly try to control router queues by probabilistically dropping (or marking) packets. We argue that while controlling router queues is important, this control needs to be tempered by a consideration of the overall lossrate at the router. Solely attempting to control queue length can induce loss-rates that have as negative an effect on application and network performance as the large queues that existing AQM schemes were trying to avoid. Thus controlling queue length without regard to loss-rate can be counterproductive. In this work we demonstrate that by jointly controlling queue length and loss-rate, both network and application performance are improved. We present a novel AQM design that attempts to simultaneously optimize queue length and loss-rate. Our algorithm, called loss and queuing delay control (LQD), is a control theoretic scheme that explicitly treats loss-rate as a control parameter. LQD is shown to provide stable control analytically and is evaluated empirically by comparing its performance against other control theoretic AQM designs (PI and REM). The results of evaluation in a laboratory testbed under realistic traffic mixes and loads show that LQD results in lower overall loss rates and that applications see lower average queue lengths than with PI or REM.

Active queue management (AQM) in routers has been proposed as a solution to some of the scalability issues associated with TCP's pure end-to-end approach to congestion control. A recent study of AQM demonstrated its effectiveness in reducing the response times of Web request/response exchanges as well as increasing link throughput and reducing loss rates [L. Le et al., 2003]. However, use of the ECN (explicit congestion notification) signaling protocol was required to outperform drop-tail queuing. Since ECN is not currently widely deployed on end-systems, we investigate an alternative to ECN, namely applying AQM differentially to flows based on a heuristic classification of the flow's transmission rate. Our approach, called differential congestion notification (DCN), distinguishes between "small" flows and "large" high-bandwidth flows and only provides congestion notification to large high-bandwidth flows. We compare DCN to other prominent AQM schemes and demonstrate that for Web and general TCP traffic, DCN outperforms all the other AQM designs, including those previously designed to differentiate between flows based on their size and rate.

We present an empirical study of the effects of active queue management (AQM) on the distribution of response times experienced by a population of web users. Three prominent AQM schemes are considered: the Proportional Integrator (PI) controller, the Random Exponential Marking (REM) controller, and Adaptive Random Early Detection (ARED). The effects of these AQM schemes were studied alone and in combination with Explicit Congestion Notification (ECN). Our major results are:

We present a general analytical model for predicting the reconstructed frame rate of an MPEG stream. Our model captures the temporal relationships between I-, P, and B-frames but is independent of the channel and media characteristics. We derive an adaptive FEC scheme from the general model and verify it by comparing it to the results of a simulation. The prediction error of the model compared to the simulation for a wide array of parameter values is less than 5%. We then use the derived adaptive FEC scheme to study the optimal rate allocation (i.e., between generating a higher frame rate or increasing the protection for a lower frame rate) when equation-based TCP rate control is used to couple packet rates to channel characteristics such as round trip time and packet loss probabilities. Surprisingly, we find that optimal protection levels for I- and P-frames are relatively static as loss rates increase from 1% to 4% while changes in the frame type pattern are used to ameliorate the effects of the increased loss. The study demonstrates how our model can be used to reveal joint source/channel coding tradeoffs and how they relate to encoding and transmission parameters.

"Best effort" packet-switched networks, like the Internet, do not offer a reliable transmission of packets to applications with real-time constraints such as voice. Thus, the loss of packets impairs the application-level utility. For voice this utility impairment is twofold: on one hand, even short bursts of lost packets may decrease significantly the ability of the receiver to conceal the packet loss and the speech signal playout is interrupted. On the other hand, some packets may be particular sensitive to loss as they carry more important information in terms of user perception than other packets.We first develop an end-to-end model based on loss run-lengths with which we can describe the loss distribution within a flow. These packet-level metrics are then linked to user-level objective speech quality metrics. Using this framework, we find that for low-compressing sample-based codecs (PCM) with loss concealment isolated packet losses can be concealed well, whereas burst losses have a higher perceptual impact. For high-compressing frame-based codecs (G.729) on one hand the impact of loss is amplified through error propagation caused by the decoder filter memories, though on the other hand such coding schemes help to perform loss concealment by extrapolation of decoder state. Contrary to sample-based codecs we show that the concealment performance may "break" at transitions within the speech signal however.We then propose mechanisms which differentiate between packets within a voice data flow to minimize the impact of packet loss. We designate these methods as "intra-flow" loss recovery and control. At the end-to-end level, identification of packets sensitive to loss (sender) as well as loss concealment (receiver) takes place. Hop-by-hop support schemes then allow to (statistically) trade the loss of one packet, which is considered more important, against another one of the same flow which is of lower importance. As both packets require the same cost in terms of network transmission, a gain in user perception is obtainable. We show that significant speech quality improvements can bem achieved and additional data and delay overhead can be avoided while still maintaining a network service which is virtually identical to best effort in the long term.