Video streaming is an increasingly popular Internet application. However, despite its popularity, real-time video streaming still remains a challenge in many scenarios. Limited home broadband bandwidth and mobile phone 3G bandwidth means many users stream videos at low quality and compromise on their user experience. To overcome this problem, we propose CStream, a system that aggregates bandwidth from multiple co-operating users in a neighborhood environment for better video streaming. CStream exploits the fact that wireless devices have multiple network interfaces and connects co-operating users with a wireless ad-hoc network to aggregate their unused downlink Internet bandwidth to improve video quality. CStream dynamically generates a streaming plan to stream a single video using multiple connections and continuously adapts to changes in the neighborhood and variations in the available bandwidth. We have built CStream and evaluated it on a controlled test-bed of computers with various performance measures. The results show linear increase in throughput and improved video streaming quality as the number of cooperating users in a neighborhood increase.

Thesis (M.S.)--Worcester Polytechnic Institute. / Keywords: video streaming, motion detection, adaptive scaling. Includes bibliographical references (p. 48-51).

Video streaming has become increasingly popular with commercial video streaming applications such as YouTube accounting for a large quantity of Internet traffic. While streaming video is sensitive to bandwidth jitter, a receiver buffer can ameliorate the effects of jitter by adjusting to the difference between the transmission rate and the playback rate. Unfortunately, there are few studies to determine the best size of the receiver buffer for TCP streaming. In this work, we investigate how the buffer size of video streaming applications changes with respect to variation in bandwidth. We model the video streaming system over TCP using simulation to develop our buffering algorithm. We propose using a dynamic client buffer size based on measured bandwidth variation to achieve fewer interruptions in video streaming playback. To evaluate our approach, we implement an application to run experiments comparing our algorithm with the buffer size of commercial video streaming.

The objectives of this thesis are to investigate the challenges on video streaming, to explore and compare different video streaming mechanisms, and to develop video streaming algorithms that maximize visual quality. To achieve these objectives, we first investigate scalable video multicasting schemes by comparing layered video multicasting with replicated stream video multicasting. Even though it has been generally accepted that layered video multicasting is superior to replicated stream multicasting, this assumption is not based on a systematic and quantitative comparison. We argue that there are indeed scenarios where replicated stream multicasting is the preferred approach. We also consider the problem of providing perceptually good quality of layered VBR video. This problem is challenging, because the dynamic behavior of the Internet's available bandwidth makes it difficult to provide good quality. Also a video encoded to provide a consistent quality exhibits significant data rate variability. We are, therefore, faced with the problem of accommodating the mismatch between the available bandwidth variability and the data rate variability of the encoded video. We propose an optimal quality adaptation algorithm that minimizes quality variation while at the same time increasing the utilization of the available bandwidth. Finally, we investigate the transmission control protocol (TCP) for a transport layer protocol in streaming packetized media data. Our approach is to model a video streaming system and derive relationships under which the system employing the TCP protocol achieves desired performance. Both simulation results and the Internet experimental results validate this model and demonstrate the buffering delay requirements achieve desired video quality with high accuracy. Based on the relationships, we also develop realtime estimation algorithms of playout buffer requirements.

In this paper, we propose an early retransmission scheme to improve TCP's performance in delivering time-sensitive media. Our extensive ns2 simulations show significant improvement. When integrated into a traditional TCP variant, namely TCP-SACK, the early retransmission scheme can substantially reduce the latency caused by retransmission timeout. As a result, it can help TCP-SACK achieve a considerably higher success rate in delivering real time media. Early Retransmission also enhances the performance of a delay-based TCP variant, namely PERT. Furthermore, we also explore the improvement brought by employing a fine-grained retransmission timer, and compare it with ER. We find out that ER outperforms the fine grained timer in a variety of conditions and the combination of the two can further improve performance.

As the success of P2P file sharing systems such as BitTorrent and eMule, P2P has become a promising technology to provide video streaming services over the Internet. The P2P technology is shown to be capable of significantly reducing the transmission overhead of video server. However, due to the dynamic nature of peers, a P2P streaming system suffers from bursty packet loss caused by peer departures. Furthermore, as the packet being forwarded peer by peer, the situation becomes worse and worse. This problem is recognized as packet loss accumulation problem. To overcome bursty packet loss and eliminate packet loss accumulation problems caused by peer departures in P2P streaming systems, a multi-source structure combining with a distributed FEC scheme for P2P streaming systems is proposed. In the proposed structure, each peer connects to multiple parents according to the pre-specified FEC packets ensemble and each parent forwards partial streaming packets to the peer. If one or few parents fail, other parents can still provide most of remaining part of streaming packets that can be used to recover the missing packets by using packet level FEC scheme. To evaluate the performance of P2P streaming systems using the proposed multi-source structure, we first propose a Continuous-Time Markov Chain to model the arrival/departure behavior of parents in P2P systems. Based on the Markov Chain, we further derived equations to calculate packet loss probabilities for both single-source and multi-source P2P systems. The mathematical analyses show how the packet loss accumulation occurs in P2P systems and how the proposed multi-source structure eliminates packet loss accumulation problem. In addition, simulations are conducted using NS2 to evaluate the proposed multi-source structure. Simulation results verify that the proposed multi-source structure combining with an appropriate FEC protection is capable of overcoming burst packet loss and eliminating packet loss accumulation problems. The simulation results also show that the proposed multi-source structure performs better than the single-source and the PROMISE/CollectCast P2P systems in terms of packet loss, end-to-end delay, and PSNR. A prototype system is implemented to conduct a real experiment over the Internet to validate the effectiveness of the proposed scheme.

No description available.

Enabling efficient media streaming over wireless channels requires efficient utilization of the limited wireless spectrum while satisfying multimedia applications' quality of service (QoS) requirements. In this dissertation, we provide insights into network and application-centric approaches for media streaming over wireless channels. In network-centric approaches, the fundamental problem is how to model network variations at the different layers and optimize the total quality across these layers. We use Finite-state Markov chain (FSMC) models to investigate the packet loss and delay performance over a wireless link. We propose a new method for partitioning the received SNR space that results in a FSMC model with tractable queueing performance. We then use this model to derive closed-form expressions for the {\em Effective Bandwidth\/} subject to either packet loss or packet delay constraints. In application-centric approaches, we take into account the VBR nature of video frames and channel dynamics and integrate in the analysis the dynamics of the playback buffer occupancy. We introduce a mixture of sourec/channel rate adaptation schemes that target efficient utilization of the wireless spectrum and safeguard the continuity of media streaming over wireless channels. First, we propose two source-rate control schemes for streaming video over wireless channels that provide gracefully degraded quality and soft guarantees on frame delay. The schemes are designed to maximize the source bit rate at the encoder while preventing/reducing events of starvation at the decoder. Second, we present a novel cycle-based rate adaptation scheme. The scheme is designed to maximize the source bit rate at the encoder while guaranteeing an upper bound on the probability of starvation at the playback buffer. This approach can be applied to both {\em one-way} and {\em interactive} video. Finally, we propose a playback-adaptive source/channel rate control (SCRC) for video streaming over wireless channels. We exploit the so-called playback adaptation margin and the playback buffer occupancy to control the source and channel rates. The SCRC scheme is designed to limit potential playback discontinuities that may occur due to variations in the wireless link.

Drones can be used for several civil applications including search and rescue, coverage, and aerial imaging. Newer applications like construction and delivery of goods are also emerging. Performing tasks as a team of drones is often beneficial but requires coordination through communication. In this thesis, the communication requirements of video streaming drone applications based on existing works are studied. The existing communication technologies are then analyzed to understand if the communication requirements posed by these drone applications can be met by the available technologies. The shortcomings of existing technologies with respect to drone applications are identified and potential requirements for future technologies are suggested. The existing communication and routing protocols including ad-hoc on-demand distance vector (AODV), location-aided routing (LAR), and greedy perimeter stateless routing (GPSR) protocols are studied to identify their limitations in context to the drone networks. An application scenario where a team of drones covers multiple areas of interest is considered, where the drones follow known trajectories and transmit continuous streams of sensed traffic (images or video) to a ground station. A route switching (RS) algorithm is proposed that utilizes both the location and the trajectory information of the drones to schedule and update routes to overcome route discovery and route error overhead. Simulation results show that the RS scheme outperforms LAR and AODV by achieving higher network performance in terms of throughput and delay. Video streaming drone applications such as search and rescue, surveillance, and disaster management, benefit from multicast wireless video streaming to transmit identical data to multiple users. Video multicast streaming using IEEE 802.11 poses challenges of reliability, performance, and fairness under tight delay bounds. Because of the mobility of the video sources and the high data-rate of the videos, the transmission rate should be adapted based on receivers' link conditions. Rate-adaptive video multicast streaming in IEEE 802.11 requires wireless link estimation as well as frequent feedback from multiple receivers. A contribution to this thesis is an application-layer rate-adaptive video multicast streaming framework using an 802.11 ad-hoc network that is applicable when both the sender and the receiver nodes are mobile. The receiver nodes of a multicast group are assigned with roles dynamically based on their link conditions. An application layer video multicast gateway (ALVM-GW) adapts the transmission rate and the video encoding rate based on the received feedback. Role switching between multiple receiver nodes (designated nodes) cater for mobility and rate adaptation addresses the challenges of performance and fairness. The reliability challenge is addressed through re-transmission of lost packets while delays under given bounds are achieved through video encoding rate adaptation. Emulation and experimental results show that the proposed approach outperforms legacy multicast in terms of packet loss and video quality.

Submitted by Maria Aparecida (cidazen@gmail.com) on 2017-08-16T12:47:03Z No. of bitstreams: 1 Sadick.pdf: 2266912 bytes, checksum: f7c0795c47de171be894492b7a3698c6 (MD5) / Made available in DSpace on 2017-08-16T12:47:03Z (GMT). No. of bitstreams: 1 Sadick.pdf: 2266912 bytes, checksum: f7c0795c47de171be894492b7a3698c6 (MD5) Previous issue date: 2008-02-14 / The Internet has experienced a considerable increase in the use of audio and video applications, which provoke a large consumption of the resources available in the network and servers. Therefore, the monitoring and analysis of those resources becomes an essential task in order to enhance the service delivered to users. This work depicts a monitoring module implemented in a video server architecture, which is used to track the transmission of some popular video formats. Our experiments have demonstrated that one of the formats delivers a performance considerably better than the other, regarding the bandwidth allocated to each user session, what only reassures the importance of having such a monitoring module available in a server architecture. / A Internet tem experimentado, nos últimos anos, um considerável aumento no uso de aplicações de áudio e vídeo, as quais promovem um acentuado consumo dos recursos disponíveis na rede e nos servidores. Desta forma, torna-se essencial o monitoramento e análise da utilização desses recursos, a fim de melhorar os serviços prestados aos usuários. Este trabalho descreve um módulo de monitoramento implementado em um servidor de vídeo, o qual é utilizado para acompanhar a transmissão de diferentes formatos populares de vídeo via streaming. Observou-se, por meio de experimentos, que um dos formatos apresenta um desempenho consideravelmente melhor que o outro, no que se refere à largura de banda alocada a cada sessão de usuário, o que corrobora a importância de módulos de monitoramento, tal como o desenvolvido nesta dissertação.