Dynamic parallel TCP for scalable video streaming over MIMO wireless networks

Wireless networks have dynamic transmission properties as the wireless channel conditions randomly change over time due to several environmental factors. When transmitting real-time applications such as video, the dynamic properties of wireless networks can have detrimental effects on the quality of the video delivered to a user. Many Quality of Service (QoS) guarantee techniques to improve video quality in wireless networks are available. However, these mainly focus on traffic shaping based on bandwidth requirements and prioritization and do not take into account the effects a changing wireless channel might have on the transmitted video. With an increasing amount of video applications being transmitted over wireless networks, alternative methods to provide QoS guarantees should be investigated. This research proposes an adaptive Forward Error Control (FEC) mechanism that dynamically adjusts the number of error control packets based on the traffic load in the network and the state of the wireless channel. When the wireless channel is bad, such as during a rain storm, the proposed mechanism reduces the amount of FEC packets added thereby reducing computational overhead in the network. This enables the receiver to recover the information sent by the transmitter faster with fewer errors than it otherwise would. Simulation results using video trace files show an improvement over existing solutions in terms of the number of FEC packets added while overall the user-perceived quality of the video streaming session is maintained at the same level, even when the network is experiencing a high Packet Error Rate (PER).

Fractional Link scheduling is one of the most fundamental problems in wireless networks. The prevailing approach for shortest fractional link scheduling is based on a reduction to the maximum-weighted independent set problem, which itself may not admit efficient approximation algorithms. In addition, except for the wireless networks under the protocol interference model, none of the existing scheduling algorithms can produce a link schedule with explicit upper bounds on its length in terms of the link demands. As the result, the polynomial approximate capacity regions in these networks remain blank. This paper develops a purely combinatorial paradigm for fractional link scheduling in wireless networks. In addition to the superior efficiency, it is able to provide explicit upper bounds on the lengths of the produced link schedule. By exploiting these upper bounds, polynomial approximate capacity regions are derived. The effectiveness of this new paradigm is demonstrated by its applications in wireless networks under the physical interference model and wireless MIMO networks under the protocol interference model.

MIMO is an emerging technology which can substantially increase the link capacity of wireless communications. MIMO has also been shown to enhance the throughput in the coexistence of heterogeneous wireless networks due to its capability of performing spatial-multiplexing (SM) and interference cancellation (IC). Practical protocols such as 802.11n+ have been proposed to address the coexistence problem in MIMO wireless networks. However, all the previous work focused on the physical-layer or MAC-layer design. None of them studied the networks' incentives to comply with or violate the low-level protocols. In this work, we study the coexistence of two single-link MIMO wireless networks with IC and SM capabilities, by analyzing their incentives of cooperation or non-cooperation (whether they will use their available degree-of-freedoms to cancel the interference to each other). We propose a novel two-round game framework, based on which we derive two networks' strategies in equilibrium and the corresponding closed-form utilities. We obtain numerical results of the equilibrium strategies under different network scenarios. Several insights are gained, which can guide us the design of real-world MIMO wireless systems in the future.

Managing the Quality-of-Experience (QoE) of video streaming for wireless clients is becoming increasingly important due to the rapid growth of video traffic on wireless networks. The inherent variability of the wireless channel as well as the Variable Bit Rate (VBR) of the compressed video streams make QoE management a challenging problem. Prior work has studied this problem in the context of transmitting a single video stream. In this paper, we investigate multiplexing schemes to transmit multiple video streams from a base station to mobile clients that use number of playout stalls as a performance metric. In this context, we present an epoch-by-epoch framework to fairly allocate wireless transmission slots to streaming videos. In each epoch our scheme essentially reduces the vulnerability to stalling by allocating slots to videos in a way that maximizes the minimum `playout lead' across all videos. Next, we show that the problem of allocating slots fairly is NP-complete even for a constant number of videos. We then present a fast lead-aware greedy algorithm for the problem. Our choice of greedy algorithm is motivated by the fact that this algorithm is optimal when the channel quality of a user remains unchanged within an epoch (but different users may experience different channel quality). Moreover, our experimental results based on public MPEG-4 video traces and wireless channel traces that we collected from a WiMAX test-bed show that the lead-aware greedy approach performs a fair distribution of stalls across the clients when compared to other algorithms, while still maintaining similar or lower average number of stalls per client.

In this study, we evaluate the statistical characteristics of packet transmission routes and the hops in wireless multi-route and multi-hop network. Although the wireless network is based on cellar system, we consider the use of forwarding nodes for relaying the packets transmitted by user nodes. The network includes use of multiple routes from a user to base stations. Few researches have been done for statistical evaluations on the network, so far. In our route building method, nodes determine the connectivity based on SNR (signal-to-noise power ratio) of received signals. Our model employs free space propagation power loss model to derive SNR. We observe that the number of routes increases with the number of forwarding nodes and their transmit power. On the other hand, the number of hops per route is saturated with increases of those parameters. We conclude that an adequate selection of the parameters enables to suppress the number of hops

Adaptive and ubiquitous video streaming over Wireless Mesh Networks

The present paper describes the set-up for the measurement of MIMO (multi-input-multi-output) radio channels as part of the European project METRA (Multi Element Transmit and Receive Antenna). Inputs for the stochastic model described by Pedersen, Andersen, Kermoal and Mogensen (see IEEE Vehicular Technology Conference VTC 2000 Fall, Boston, USA, 2000) are extracted from the measurement results and fed into a COSSAP(R) block implementing this model. A good matching between the eigenanalysis performed on both measured and simulated signals is shown.

Streaming video applications are gaining popularity over not only the fixed Internet but also mobile hand-held devices. Growth in wireless access technologies along with the advanced coding schemes show the promise of better streaming video service for the mobile users. However, wireless channels are characterized by fluctuating bandwidth, thereby making it challenging for streaming applications to ensure good quality of user experience (QoE). In this paper, we adopt an objective approach and propose a cross-layer adaptive video streaming technique, where we make use of the channel loss information to update the application layer video encoding at a slower rate and the link layer modulation and coding scheme at a faster rate. In our adaptive scheme, we focus on a pause-free playback by preventing buffer underflow at the receiver. Our simulation experiments show that, the proposed adaptive technique improves the QoE of the streaming video significantly by gracefully degrading video quality in the face of pathological cases such as wireless channel error, while ensuring an uninterrupted video playback.

Video communication over wireless networks is still a considerably challenging task due to the burst of the video stream and the time-varying of the wireless channel. In this paper, we propose a cross-layer optimization approach for multiuser video streaming over wireless networks. We focus on the optimization among application layer, MAC layer and physical layer. Unlike recent schemes which analyze the video stream in the pixel domain, we define a frame relative importance index (FRII) using the compression domain information to express the video characteristics. No need to fully decompress the video stream, it can reduce the computational complexity substantially. Simulation results show that, the proposed framework outperforms the recent scheme with no cross-layer optimization by 1.6 dB in terms of the average PSNR of all users.

In this paper, we propose and evaluate the Trace Aloha protocol for distributed medium access control in wireless MIMO networks. We study the case in which a node only knows the MIMO channel matrix from the node to the access point and the total number of nodes in the network. Given a MIMO channel matrix, a node could calculate the corresponding singular values. According to the Trace Aloha protocol, a node transmits data if and only if the sum of the squared singular values exceeds a predetermined threshold. We use both analytical results and simulation results to justify the usage of the proposed protocol.

Wireless networks have a key role in operational deployment, mobility, the flexibility of network organization and the variety of possible applications, in many cases being the only economically justified solution. Based on the above, for the deployment of wireless networks, the correct theoretical design of the wireless network is an important issue: the required number of devices, their characteristics, placement, energy characteristics of channels, distribution routes and coverage areas, etc. Correct theoretical analysis will ensure error-free construction of the network for the successful implementation of the tasks and will not lead to the need to attract additional material costs. One of the key tasks of wireless communication construction is a reliable assessment of the energy characteristics of a wireless communication channel. Modern global manufacturers of telecommunications equipment build both backbone equipment and equipment to cover WLAN (wireless local area networks) areas, based on IEEE 802.11 standards. One of the most modern standards of the IEEE 802.11 group is the IEEE 802.11ac standard, which is characterized by several improved technical characteristics compared to its predecessor IEEE 802.11n and older versions. That is why the task of evaluating the performance of modern wireless communication equipment based on IEEE 802.11ac standards and analyzing the evaluation of the energy characteristics of the communication channel to predict the parameters and performance characteristics of a point-to-point, point-to-multipoint or other wireless transmission systems within a certain wireless communication network. The following results of solving this problem are presented in this paper: – theoretical evaluation of the energy characteristics of the wireless communication channel. – laboratory studies of wireless communication equipment of the 802.11ac standard based on simulating a communication line using attenuators. – verification of the results of laboratory studies in field conditions based on the model for assessing the quality of communication in a wireless channel created based on the 802.11ac standard.

TFRC protocol is suitable for video transmission in a wire network, and quality assessment is also essential in a video transmission system. In this paper, a real-time video transmission system based on TFRC protocol is proposed, and the evaluation model about the system is improved in the framework of Evalvid. It assesses the quality and efficiency of the video transmission according to the actual video file, and analyzes losses frame in different video types during transmission as well as the video quality in receiver. The results of simulation experime nt in NS-2 show that when real-time video transmitted in wire network environment using this system, the receiver can get satisfactory video quality by reason of the TFRC protocol friendliness and the smoothness of sending rate. Based on the characteristic of high error rate in wireless network, the novel TCP friendly congestion control algorithm TFRC-JI proposed in our previous work (1) was adopted, which introduces the latency vibration to distinguish the link congestion from code error, thus different speed control mechanism is feedback to the transmitting end. Simulation experiment results indicated that compared with the traditional TFRC, the TFRC-JI suites well for real-time service transmission. Index Terms—TFRC; Quality evaluate; NS2; Congestion control; Streaming video

The transmission of video materials by digital compression has been widely applied in recent years. The following problems are involved in this approach. (1) A duplex system composed of main and backup links is often used in the transmission of video materials. Thus, the transmission capacity of the backup system becomes redundant in the normal state. (2) No automatic image quality monitoring method for video transmission has been established. On the other hand, the coding/decoding techniques for digital video transmission are almost fully established, and various devices are already widely used. Thus, it is highly desirable to reduce the above redundancy, that is, to achieve highly efficient automatic picture quality monitoring. Consequently, this paper proposes a flat multiscalable video transmission scheme in which highly efficient and highly reliable video transmission with automatic picture quality monitoring can be realized by simply adding pre‐ and postprocessing, without modifying the existing coding/decoding devices. In the proposed scheme, noise cancellation by averaging and high picture quality are realized by processing that reduces the correlation of coding noise in the decoded video pictures of the two systems. The SN ratios of the received video and the original pictures can be estimated from the residual correlation. The proposed scheme can be applied not only to improving the efficiency of video materials transmission, but also to improving the reliability of picture transmission over the Internet. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 86(7): 42–53, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.1168

Internet video streaming is a hot topic in multimedia systems. A large variety of devices (computers, mobile phones, TVs, etc.) are connected to the Internet via wired or wireless networks and are capable of receiving and decoding HD video content. To enable new services like HD video streaming (e.g., online video rental), the Internet’s infrastructure was enhanced. But the Internet is still a best-effort network, which does not implement quality-of-service or admission control, resulting in time-varying bandwidth and packet delay, packet loss and network congestion. Because video streaming accounts for a considerable amount of the Internet’s traffic, video streaming needs additionally to be congestion-aware, to avoid a congestion collapse of the Internet. The Transmission Control Protocol (TCP) can adapt to changing network conditions and is currently the de facto standard protocol for congestion-aware and reliable data transmission in the Internet. This fact gave TCP-based video streaming a huge momentum. Consequently, this thesis investigates TCP-based adaptive video streaming for the Internet. The main goal is to provide a solution for congestion-aware video streaming, while still being able to achieve a reasonable performance in error-prone networks. To complement existing work on congestion-aware adaptive streaming, this thesis makes six contributions. (1) The baseline performance of TCP-based adaptive streaming is identified by means of an evaluation of different adaptive streaming approaches. The results represent a reference for further investigations. (2) An investigation on the influence of TCP’s behavior in presence of packet loss on the video streaming performance. (3) To overcome the shortcomings of TCP-based video streaming (single TCP connections fail to deliver a good performance in case of packet loss), a new approach to video streaming based on multiple request-response streams was introduced. The novelty of this system is that it is able to make use of multiple HTTP-based request-response streams while still providing TCP-friendliness. (4) A performance model of the HTTP-based request-response streams was developed, to estimate the influence of the system parameters and the network characteristics on the throughput performance. (5) A comprehensive evaluation of the HTTP-based request-response streams under diverse network conditions was conducted, to validate the model’s estimations. Additionally, the TCP-friendliness was evaluated, showing that request-response streaming systems can be configured to achieve TCP-friendliness. (6) A cellular network with high bandwidth fluctuations and RTTs was used to investigate the performance of the request-response streaming system in a mobile video streaming scenario. The results indicate that the streaming system can make good use of the available bandwidth, while the number of quality switches is kept low. While aggregating multiple TCP connections to improve the TCP streaming performance is quite common, usually the improvement comes at the cost of high deployment effort. By placing the streaming logic at the client, request-response streams can avoid this complexity. Additionally, this client-driven approach responds faster to changing network conditions and enables easy recovery from connection stalls or aborts, because the control loop is at the client. To improve the network efficiency and the scalability in terms of number of clients served, HTTP-based request-response streams can utilize HTTP proxies and caches.

Ultrasonic communication is proven to be reliable for transmitting data, images, and video streams through solid channels to overcome certain physical barriers which prevent conventional wired or wireless communication networks. In this study, we propose a novel encryption framework for securing the video stream transmission while taking into consideration the characteristics and constraints associated with ultrasonic communication through solid channels. In this framework, a Chaotic encryption algorithm employing the 2-D Around Cat Map and 1-D Logistic Map is used to confuse and diffuse the video stream. A benchmark system was designed to examine the performance of the Chaotic encryption algorithm applied to ultrasonic video transmission. The horizontally adjacent pixels in a video frame and the same pixel in two consecutive frames are randomly selected to evaluate the frame-to-frame pixel correlation before and after encryption. The scatter plots and the correlation coefficient reveal that the Chaotic encryption fully scrambles and diffuses the original video stream.