Increasing availability of autonomous small-size aerial vehicles leads to a variety of applications for aerial exploration and surveillance, transport, and other domains. Many of these applications rely on networks between aerial nodes, that will have high mobility dynamics with vehicles moving in all directions in 3D space and positioning in different orientations, leading to restrictions on network connectivity. In this paper, we propose a simple antenna extension to 802.11 devices to be used on aerial nodes. Path loss and small-scale fading characteristics of air-to-ground links are analyzed using signal strength samples obtained via real-world measurements at 5 GHz. Finally, network performance in terms of throughput and number of retransmissions are presented. Results show that a throughput of 12Mbps can be achieved at distances in the order of 300m.

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.

Universal Multimedia Access (UMA) calls for solutions where content is created once and subsequently adapted to given requirements. With regard to UMA and scalability, which is required often due to a wide variety of end clients, the best suited codecs are wavelet based (like the MC-EZBC) due to their inherent high number of scaling options. However, most transport technologies for delivering videos to end clients are targeted toward the H.264/AVC standard or, if scalability is required, the H.264/SVC. In this paper we will introduce a mapping of the MC-EZBC bitstream to existing H.264/SVC based streaming and scaling protocols. This enables the use of highly scalable wavelet based codecs on the one hand and the utilization of already existing network technologies without accruing high implementation costs on the other hand. Furthermore, we will evaluate different scaling options in order to choose the best option for given requirements. Additionally, we will evaluate different encryption options based on transport and bitstream encryption for use cases where digital rights management is required.

HTTP streaming has gained significant attraction in the last few years. Currently many commercial as well as standardized streaming systems are already offering adaptive streaming. In most cases, the adaptation is achieved by switching between separately encoded video streams in different qualities. In contrast to that, this paper focuses on the applicability of scalable video coding based on the H.264/SVC standard for adaptive HTTP streaming. Recent work has already highlighted the conceptual advantages like better cache utilization, fine-grained bit rate scalability, and lower storage requirements. This paper discusses the actual realization and design options for implementing priority streaming using the ISO Base Media File Format (BMFF). We propose three different strategies for organizing the scalable video bit stream that consider both the possibilities as well as limitations of the ISO BMFF. The proposed strategies are discussed and evaluated both conceptually and quantitatively. For that purpose, we provide a detailed analysis based on modeling both the overhead of the file format and the HTTP encapsulation. The results for all three priority streaming strategies show that the limitations of the ISO BMFF result in a high relative overhead in the case of low bit rate content. However, when applied to high quality content, priority streaming of H.264/SVC can be implemented at a very low cost. Depending on the number of layers and the offered scalability dimensions, different strategies should be chosen to minimize the overhead. Based on the analytical model and the discussion, this paper provides guidance for selecting the most efficient strategy.

We analyze unmanned aerial vehicle (UAV)-to-ground links for an 802.11a-based small quadrotor UAV network with two on-board antennas via a set of field experiments. The paper presents our first results toward modeling the uplink and downlink channel and provide the path loss exponents for an open field and a campus scenario. We illustrate the impact of antenna orientation on the received signal strength and UDP throughput performance for different heights, yaws, and distances. When both antennas are horizontal (parallel to the flight direction plane), yaw differences can be handled, whereas a vertical antenna can assist against signal loss due to tilting of the UAV during acceleration/deceleration. Further work is required to analyze fading as well as UAV-UAV links in a multi-UAV network.

Adaptive video streaming based on TCP/HTTP is becoming popular because of its ability to adapt to changing network conditions. We present an in-depth experimental analysis of the use of HTTP-based request-response streams for video streaming. In this scheme, video fragments are fetched by a client from the server, in smaller units called chunks, potentially via multiple parallel HTT P requests (TCP connections). A model for the achievable throughput is formulated.The model is validated by a broad range of streaming experiments, including an evaluation of TCP-friendliness. Our findings include that request-response streams are able to scale with the available bandwidth by increasing the chunk size or the number of concurrent streams. Several combinations of system parameters exhibiting TCP-friendliness are presented. We also evaluate the video streaming performance in terms of video quality in the presence of packet loss. Multiple request-response streams are able to maintain satisfactory performance, while a single TCP connection deteriorates rapidly with increasing packet loss. The results provide experimental evidence that HTTP-based request-response streams are a good alternative to classical TCP streaming

In this paper, we compare deterministic and probabilistic path planning strategies for an autonomous unmanned aerial vehicle (UAV) network, where the objective is to explore a given area with obstacles and provide an overview image. We present both online and offline implementations of the algorithms as alternative solutions, where applicable, and analyze the performance of the offline implementations. Results illustrate the benefits and drawbacks of different planning strategies and provide insight into which strategy should be taken, given the constraints of the application of interest.

In this paper, we present an approach for in-network adaptation of H.264/SVC in the context of 802.11 wireless networks. It builds upon our previous work on an adaptive RTSP/RTP proxy which allows to adapt video streams on Linux-based home router platforms. The proposed approach tackles the throughput variations that occur as a consequence of the physical rate adaptation in 802.11 equipment caused by the mobility of clients. By combining monitoring information available exclusively on the wireless router with the ability to adapt scalable video streams on-the-fly, the proposed in-network adaptation approach allows to quickly adjust the video bit rate to the current link conditions. Instead of reacting on packet loss, our approach uses an increase in queueing delay at the router to detect phases of throughput degradation. This allows a higher responsiveness compared to traditional end-to-end approaches that rely solely on RTCP feedback. The behavior of our novel approach was evaluated in several mobility scenarios in an experimental test bed. The results obtained by streaming and adapting high-definition content clearly demonstrate the feasibility and benefits of this approach.

Wireless router platforms based on the Linux operating system are becoming popular in consumers' home networks. The transmission of multimedia data or their use as media-aware network elements imposes high traffic and computational loads on these devices. Thus, it is interesting to evaluate the networking and processing capabilities of such home router platforms in order to assess their usefulness for improved multimedia services such as in-network H.264/SVC video stream adaptation. This paper presents a performance evaluation of three home router platforms representative for low-end, mid-range, and high-end devices. The scope of the evaluation is the performance of the Linux networking stack on these routers; results for both application-layer (TCP and UDP) transmission and kernel-level (UDP) traffic routing are given. The results show that both TCP and UDP throughputs are significantly below (less than half of) the outgoing (wired) links' nominal capacities and depend very much on the sizes of the transmitted data blocks. This clearly indicates that the networking performance is limited by the platforms' processing capabilities and the lack of mechanisms that offload networking tasks from the CPUs. This behaviour cannot be observed on today's PC systems and has to be considered when deploying multimedia services on these network devices. Furthermore, a detailed analysis of the Linux networking stack reveals that the performance is heavily impacted by the netfilter code, even when no packet filtering or network address translation is being performed. Considerable performance gains can be achieved when this netfilter code is bypassed.

Recent work in TCP video streaming indicates that multimedia streaming via TCP provides satisfactory performance when the achievable TCP throughput is approximately twice the media bit rate. However, these conditions may not be achievable on the Internet, e.g., when the delivery path offers insufficient bandwidth or becomes congested due to competing traffic. Therefore, adaptive streaming for videos over TCP is required and a number of rate-control algorithms for video streaming have been proposed and evaluated in the literature.\\ In this paper, we evaluate and compare three existing rate-control algorithms for TCP streaming in terms of the (PSNR) quality of the delivered video and in terms of the timeliness of delivery. The contribution of the paper is that, to the best of our knowledge, this is the first evaluation of TCP-based streaming in an Internet-like setting making use of the scalability features of the H.264/SVC video codec. Two simple bandwidth estimation algorithms and a priority-/deadline-driven approach are described to adapt the bit rates of, and transmit, the H.264/SVC video in a rate-distortion optimal manner. The results indicate that the three algorithms perform robustly in terms of video quality and timely delivery, both on under-provisioned links and in case of competing TCP flows. The priority-/deadline-driven technique is even more stable in terms of packet delays and jitter; thus, client buffers can be dimensioned more easily.

TCP-based video streaming encounters difficulties in unreliable networks with unanticipated packet loss. In combination with high round trip times, the effective throughput deteriorates rapidly and TCP connection resets or stalls may occur. In this paper, we propose a client-driven video transmission scheme which utilizes multiple HTTP/TCP streams. The scheme is largely insensitive to unanticipated packet loss and thereby reduces throughput fluctuations. Since it is based on HTTP, the scheme can easily be deployed in existing network infrastructures. It fosters scalability on the server side by shifting complexity from the server to the clients. Certain features of request-response schemes allow maintaining fairness, despite of using multiple HTTP streams. Making use of TCP, the scheme inherently adapts to congested network links.

This paper presents a novel approach that combines both in-network, application-layer adaptation and network-layer traffic control of scalable video streams based on the H.264/SVC standard. In the IPTV/VoD scenario considered, an intercepting RTSP/RTP proxy performs admission control of the requested video, based on the signaled scalability information, and decides whether the content can be streamed without changes or in an adapted version. The proxy configures the network layer appropriately in order to separate the video stream from best-effort traffic on the same link. Rather than performing fixed bandwidth allocation, our proxy approach uses the hierarchical token bucket (HTB) queuing discipline to allow for borrowing bandwidth between traffic classes. In that setting, two different allocation policies are introduced. The hard reservation policy (HRP) performs admission control and adaptation on the video streams and does not modify video bandwidth allocation after admission. In contrast, the flexible borrowing policy (FBP) restricts the admission control to the base layer of the SVC stream. The packets carrying MGS enhancement layer data are marked with priorities by the proxy and are handled at the network layer by a priority-based queuing mechanism. Both a qualitative comparison and an experimental evaluation of the two policies are given.

This paper addresses the efficient adaptation of encrypted scalable video content (H.264/SVC). RTP-based in-network adaptation schemes on a media aware network element (MANE) in an IPTV and VoD scenario are considered. Two basic alternatives to implement encryption and adaptation of H.264/SVC content are investigated: (i) full, format-independent encryption making use of Secure RTP (SRTP); (ii) SVC-specific encryption that leaves the metadata relevant for adaptation (NAL unit headers) unencrypted. The SRTP-based scheme (i) is straightforward to deploy, but requires the MANE to be in the security context of the delivery, i.e., to be a trusted node. For adaptation, the content needs to be decrypted, scaled, and re-encrypted. The SVC-specific approach (ii) enables both full and selective encryption, e.g., of the base layer only. SVC-specific encryption is based on own previous work, which is substantially extended and detailed in this paper. The adaptation MANE can now be an untrusted node; adaptation becomes a low-complexity process, avoiding full decryption and re-encryption of the content. This paper presents the first experimental comparison of these two approaches and evaluates whether multimedia-specific encryption can lead to performance and application benefits. Potential security threats and security properties of the two approaches in the IPTV and VoD scenario are elementarily analyzed. In terms of runtime performance on the MANE our SVC-specific encryption scheme significantly outperforms the SRTP-based approach. SVC-specific encryption is also superior in terms of induced end-to-end delays. The performance can even be improved by selective application of the SVC-specific encryption scheme. The results indicate that the efficient adaptation of SVC-encrypted content on low-end, untrusted network devices is feasible.

One of the most active research topics in the field of video signal processing is scalable video coding (SVC). The recently published extension of the H.264/AVC video coding standard introduces scalability features by employing a layered encoding of the video stream. In our work we investigated the usage of this scalable extension of H.264/AVC for in-network multimedia adaptation. We developed an RTSP/RTP-based proxy which exploits the layered encoding of the video and can perform real-time video adaptation on an inexpensive off-the-shelf WiFi router. This is achieved by applying a stateful, packet-based adaptation approach that keeps the computational costs at a minimum. With that approach it is possible to simultaneously adapt multiple video streams to varying network conditions or to the capabilities of the consumers' end-devices. In our demonstration we show the streaming of two scalable video streams from a server to a client and the in-network adaptation of the video at the WiFi router. The adaptation can be controlled interactively in the temporal, spatial and SNR domains.

This paper explores design options and evaluates implementations of in-network, RTP/RTSP based adaptation MANEs (Media Aware Network Elements) for H.264/SVC content streaming. The obvious technique to be employed by such an adaptation MANE is to perform SVC specific bitstream extraction or truncation. Another mechanism that can be used is description (metadata) driven, coding format independent adaptation based on generic Bitstream Syntax Descriptions (gBSD), as specified within MPEG-21 Digital Item Adaptation (DIA). Adaptation MANE architectures for both approaches are developed and presented, implemented in end-to-end streaming/adaptation prototype systems, and experimentally evaluated and compared. For the gBSD based solution, open issues like the granularity of bitstream descriptions and of bitstream adaptation, metadata overhead, metadata packetization and transport options, and error resilience in case of metadata losses, are addressed. The experimental results indicate that a simple SVC specific adaptation MANE does clearly outperform the gBSD based adaptation variants. Yet, the conceptual advantages of the description driven approach, like coding format independence and flexibility, may outweigh the performance drawbacks in specific applications.

Recent advances in video coding technology like the scalable extension of the MPEG-4 AVC/H.264 video coding standard pave the way for computationally cheap adaptation of video content. In this paper we present our work on a lightweight RTSP/RTP proxy that enables in-network stream processing. Based on an off-the-shelf wireless router that runs a Linux-based firmware we demonstrate that the video adaptation can be performed on-the-fly directly on a network device. The paper covers design and implementation details of the proxy as well as a discussion about the actual adaptation of the SVC stream. Based on experimental evaluations we show that our approach can handle a reasonable number of concurrent sessions for a typical home deployment scenario. Furthermore, the paper covers possible applications in which adaptation on the network device can be beneficial.

This chapter covers the topic of making use of scalable video content in streaming frameworks and applications. Specifically, the recent standard H.264/SVC, i.e., the scalable extension of the widely used H.264/AVC coding scheme, and its deployment for adaptive streaming, the combined activities of content adaptation and streaming, are considered. H.264/SVC is regarded as a promising candidate to enable applications to cope with bandwidth variations in networks and heterogeneous usage environments, mainly diverse end device capabilities and constraints. The relevant coding and transport principles of H.264/SVC are reviewed first. Subsequently, a general overview of H.264/SVC applications is given. The chapter then focuses on presenting architectural/implementation options and applications of H.264/SVC for adaptive streaming, emphasizing the aspect of where, i.e., on which network node and on which layer in the networking stack, in the video delivery path the content adaptation can take place; also, methods of content adaptation are covered. This pragmatic perspective is seen as complementing more general discussions of scalable video adaptation issues in the existing literature.