Cross layer techniques for flexible transport protocol using UDP-Lite over a satellite network

Traditional real-time multimedia and streaming services have utilised UDP over RTP. Wireless transmission, by its nature, may introduce a variable, sometimes high bit error ratio. Current transport layer protocols drop all corrupted packets, in contrast, protocols such as UDP-Lite allow error-resilient applications to be supported in the networking stack. This paper presents experimental quantitative performance metrics using H.264 and UDP Lite for the next generation transport of IP multimedia, and discusses the architectural implications for enhancing performance of a wireless and/or satellite environment

Full TCP/IP for 8-Bit architectures

We describe two small and portable TCP/IP implementations fulfilling the subset of RFC1122 requirements needed for full host-to-host interoperability. Our TCP/IP implementations do not sacrifice any of TCP's mechanisms such as urgent data or congestion control. They support IP fragment reassembly and the number of multiple simultaneous connections is limited only by the available RAM. Despite being small and simple, our implementations do not require their peers to have complex, full-size stacks, but can communicate with peers running a similarly light-weight stack. The code size is on the order of 10 kilobytes and RAM usage can be configured to be as low as a few hundred bytes

Rate Adaptation for Avoiding Congestion in the Use of Multimedia Over User Datagram Protocol

Multimedia applications have increased rapidly in the Internet today. However, multimedia communication suffers from bandwidth requirements problem. Therefore, it is important to optimize the network bandwidth utilization. Optimizing the network bandwidth utilization allows increasing the number of users who use multimedia applications that require guaranteed quality of service. The user experiences the performance during using the network service, which is the important factor to determine the users’ satisfaction. With the limitation of the network bandwidth, multimedia traffic can cause congestion which degrades the performance experienced by the network users. Therefore, there is an essential need to reduce the occurrence of congestion situations in a network to optimize the utilization of network resources to provide the network users with suitable performance. For most of multimedia applications, UDP is used as transport protocol. Current UDP implementation helps in increasing the traffic as it does not have flow or congestion control mechanisms. Congestion can be avoided when the traffic arrival rate to a gateway maintained close to the outgoing link capacities and the gateways' queue lengths kept small to guarantee the availability of buffer capacity for successful buffering and consequent forwarding of temporary traffic upsurges which could otherwise cause buffer overflows and packet loss. Congestion management is the combined responsibility of network gateways and end-point hosts. Gateways are invested with the ability to delay or drop the packets inside the network. Gateways are responsible for congestion detection & notification delivery, queue's traffic arrival rate control, and queue length control. Traffic sources are responsible for the adjustment of their data transmission rates to enable the gateways to achieve their goals. Building a new responsive multimedia application and protocol, based on the UDP concept, can decrease the congestion occurrence and enhance the performance of the network, especially in the real-time environment

Studies on the readability and on the detection rate in a Mach-Zehnder interferometer-based implementation for high-rate, long-distance QKD protocols

We study the way that chromatic dispersion affects the visibility and the synchronization on Quantum Key Distribution (QKD) protocols in a widely-used setup based on the use of two fiber-based Mach-Zehnder (MZ) interferometers at transmitter/receiver stations. We identify the necessary conditions for the path length difference between the two arms of the interferometers for achieving the desired visibility given the transmission distance -- where the form of the detector's window can be considered. We also associate the above limitations with the maximum detection rate that can be recorded in our setup, including the quantum non-linearity phenomenon, and to the maximum time window of the detector's gate. Exploiting our results we provide two methods, depending on the clock rate of the setup, to perform chromatic dispersion compensation techniques to the signal for keeping the correct order of the transmitted symbols. At the end, we apply our theoretical outcomes in a more realistic QKD deployment, considering the case of phase-encoding BB84 QKD protocol, which is widely used. Our proposed methods, depending on the transmission distance and on the photon emission rate at transmitter station, can be easily generalized to every fiber-optic QKD protocol, for which the discrimination of each symbol is crucial.Comment: 14 pages, 12 figure

Minimal TCP/IP implementation with proxy support

Over the last years, interest for connecting small devices such as sensors to an existing network infrastructure such as the global Internet has steadily increased. Such devices often has very limited CPU and memory resources and may not be able to run an instance of the TCP/IP protocol suite. In this thesis, techniques for reducing the resource usage in a TCP/IP implementation is presented. A generic mechanism for offloading the TCP/IP stack in a small device is described. The principle the mechanism is to move much of the resource demanding tasks from the client to an intermediate agent known as a proxy. In particular, this pertains to the buffering needed by TCP. The proxy does not require any modifications to TCP and may be used with any TCP/IP implementation. The proxy works at the transport level and keeps some of the end to end semantics of TCP. Apart from the proxy mechanism, a TCP/IP stack that is small enough in terms of dynamic memory usage and code footprint to be used in a minimal system has been developed. The TCP/IP stack does not require help from a proxy, but may be configured to take advantage of a supporting proxy

Learning Algorithm Design for Human-Robot Skill Transfer

In this research, we develop an intelligent learning scheme for performing human-robot skills transfer. Techniques adopted in the scheme include the Dynamic Movement Prim- itive (DMP) method with Dynamic Time Warping (DTW), Gaussian Mixture Model (G- MM) with Gaussian Mixture Regression (GMR) and the Radical Basis Function Neural Networks (RBFNNs). A series of experiments are conducted on a Baxter robot, a NAO robot and a KUKA iiwa robot to verify the effectiveness of the proposed design.During the design of the intelligent learning scheme, an online tracking system is de- veloped to control the arm and head movement of the NAO robot using a Kinect sensor. The NAO robot is a humanoid robot with 5 degrees of freedom (DOF) for each arm. The joint motions of the operator’s head and arm are captured by a Kinect V2 sensor, and this information is then transferred into the workspace via the forward and inverse kinematics. In addition, to improve the tracking performance, a Kalman filter is further employed to fuse motion signals from the operator sensed by the Kinect V2 sensor and a pair of MYO armbands, so as to teleoperate the Baxter robot. In this regard, a new strategy is developed using the vector approach to accomplish a specific motion capture task. For instance, the arm motion of the operator is captured by a Kinect sensor and programmed through a processing software. Two MYO armbands with embedded inertial measurement units are worn by the operator to aid the robots in detecting and replicating the operator’s arm movements. For this purpose, the armbands help to recognize and calculate the precise velocity of motion of the operator’s arm. Additionally, a neural network based adaptive controller is designed and implemented on the Baxter robot to illustrate the validation forthe teleoperation of the Baxter robot.Subsequently, an enhanced teaching interface has been developed for the robot using DMP and GMR. Motion signals are collected from a human demonstrator via the Kinect v2 sensor, and the data is sent to a remote PC for teleoperating the Baxter robot. At this stage, the DMP is utilized to model and generalize the movements. In order to learn from multiple demonstrations, DTW is used for the preprocessing of the data recorded on the robot platform, and GMM is employed for the evaluation of DMP to generate multiple patterns after the completion of the teaching process. Next, we apply the GMR algorithm to generate a synthesized trajectory to minimize position errors in the three dimensional (3D) space. This approach has been tested by performing tasks on a KUKA iiwa and a Baxter robot, respectively.Finally, an optimized DMP is added to the teaching interface. A character recombination technology based on DMP segmentation that uses verbal command has also been developed and incorporated in a Baxter robot platform. To imitate the recorded motion signals produced by the demonstrator, the operator trains the Baxter robot by physically guiding it to complete the given task. This is repeated five times, and the generated training data set is utilized via the playback system. Subsequently, the DTW is employed to preprocess the experimental data. For modelling and overall movement control, DMP is chosen. The GMM is used to generate multiple patterns after implementing the teaching process. Next, we employ the GMR algorithm to reduce position errors in the 3D space after a synthesized trajectory has been generated. The Baxter robot, remotely controlled by the user datagram protocol (UDP) in a PC, records and reproduces every trajectory. Additionally, Dragon Natural Speaking software is adopted to transcribe the voice data. This proposed approach has been verified by enabling the Baxter robot to perform a writing task of drawing robot has been taught to write only one character

Mixed streaming of video over wireless networks

In recent years, transmission of video over the Internet has become an important application. As wireless networks are becoming increasingly popular, it is expected that video will be an important application over wireless networks as well. Unlike wired networks, wireless networks have high data loss rates. Streaming video in the presence of high data loss can be a challenge because it results in errors in the video.Video applications produce large amounts of data that need to be compressed for efficient storage and transmission. Video encoders compress data into dependent frames and independent frames. During transmission, the compressed video may lose some data. Depending on where the packet loss occurs in the video, the error can propagate for a long time. If the error occurs on a reference frame at the beginning of the video, all the frames that depend on the reference frame will not be decoded successfully. This thesis presents the concept of mixed streaming, which reduces the impact of video propagation errors in error prone networks. Mixed streaming delivers a video file using two levels of reliability; reliable and unreliable. This allows sensitive parts of the video to be delivered reliably while less sensitive areas of the video are transmitted unreliably. Experiments are conducted that study the behavior of mixed streaming over error prone wireless networks. Results show that mixed streaming makes it possible to reduce the impact of errors by making sure that errors on reference frames are corrected. Correcting errors on reference frames limits the time for which errors can propagate, thereby improving the video quality. Results also show that the delay cost associated with the mixed streaming approach is reasonable for fairly high packet loss rates