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In this paper we have addressed the topic of an evolutionary integration of light fields into standard image/video processing chains by pre-processing light fields with superpixel-based and structurally adaptive Gaussian pre-filters and circular pseudo-temporal sequencing to feed them into an HEVC-codec with low-delay predictive coding configuration. We could show significant bit rate reductions of up to 27% compared to pseudo-temporal sequencing without pre-processing. The paper includes experimental results showing that not only the perceived visual quality, but also the cornucopia of post-processing options is preserved.

Networked control applications for cyber-physical networks demand predictable and reliable real-time communication. Applications of this domain have to cooperate with network protocols, the operating system, and the hardware to improve safety properties and increase resource efficiency. In consequence, a cross-layer approach is necessary for the design and holistic optimisation of cyber-physical systems and networks. This paper presents X-L ap , a cross-layer, inter-host timing analysis tool tailored to the needs of real-time communication. We use X-L ap to evaluate the timing behaviour of a reliable real-time communication protocol. Our analysis identifies parts of the protocol which are responsible for unwanted jitter. To system designers, X-L ap provides useful support for the design and evaluation of networked real-time systems.

The dynamic environment of a vehicular communication system poses a difficult task of estimating the chan­nel at minimal complexity. A time-varying multipath channel is estimated by computationally intensive algorithms that are generally not suitable for implementation on resource limited consumer hardware. Compressed Sensing (CS) schemes have been established to provide an accurate estimate by exploiting the inherent sparsity of a wireless communication channel. Correspondingly, the Rake-Matching Pursuit (RMP) and its low complexity variant, the Gradient Rake-Matching Pursuit (GRMP) algorithm, first identify different delay taps in the environment. The Doppler is then implicitly estimated by a tracking stage of respective tap coefficients. Although their performance is encouraging even under high Doppler shifts, its adoption for a static multipath environment is excessive due to the required computational resources. A low complexity scheme, like Least Squares (LS), is sufficient to estimate and compensate such channels. The cognitive framework envisages the switch between a high mobility scheme, like RMP, and a low mobility scheme, like LS, based on the channel conditions. In this paper, an enhanced cognitive framework is proposed to interchange between the channel estimation schemes to provide an adequate Bit Error Rate (BER) performance at optimum complexity. Even though the experimentation is performed for the IEEE 802.11p standard, the proposed metrics are relevant for any Orthogonal Frequency- Division Multiplexing (OFDM) based wireless communication system.

Interacting with real-time rendered 3D content from powerful machines on smaller devices is becoming ubiquitous through commercial products that enable in-home streaming within the same local network. However, support for high resolution, low latency in-home streaming at high image quality is still a challenging problem. To enable this, we enhance an existing open source framework for in-home streaming. We add highly optimized DXT1 (DirectX Texture Compression) support for thin desktop and notebook clients. For rendered light fields, we improve the encoding algorithms for higher image quality. Within a 10 Gigabit Ethernet (10 GbE) network, we achieve streaming up to 5K resolution at 55 frames per second. Through new low-level algorithmic improvements, we increase the compression speed of ETC1 (Ericsson Texture Compression) by a factor of 5. We are the first to bring ETC2 compression to real-time speed, which increases the streamed image quality. Last, we reduce the required data rate by more than a factor of 2 through foveated compression with real-time eye tracking.

Vehicular Communication Systems also known as Vehicle-to-Vehicle communication systems (V2V) specify the use of Orthogonal Frequency-Division Multiplexing (OFDM) in the physical layer. High mobility in vehicular communication systems results in a time-varying multipath or a doubly selective channel. Such channels exhibit selectivity in both the time as well as the frequency domain. The estimation of the channel parameters plays a vital role in overcoming the effects of high mobility at the receiver. Moreover, it is known that the estimation of such a channel is a non-trivial task. The Matching Pursuit (MP) algorithm is a Compressed Sensing (CS) scheme that works well in scenarios of high mobility. However, it is shown that under less severe channel conditions, simpler channel estimation and equalization schemes can be more power-efficient and hence increase battery-life without reducing the quality of the equalization. In this paper, we propose a cognitive framework optimizing the estimation scheme based on the channel conditions and certain measurements from the receiver chain. The result is a channel estimation scheme that is robust, precise and reliable in all channel conditions while adapting the complexity to be optimal for the channel being estimated. In addition to this, a simulation tool with the proposed scheme is provided for the IEEE 802.11p standard. The proposed cognitive framework is implemented for the IEEE 802.11p standard and is applicable in any OFDM based wireless system that is expected to work in highly mobile environments.

In this paper we present a scalable multi-view video streaming system with client-side view interpolation. Most parts of the system have been shown to be able to perform in real-time, but never as a complete system. Even though our algorithms are not yet highly optimized, we show that running a complete multi-view streaming system video on consumer hardware is possible today. Since multi-view coding and streaming has not reached the consumer market yet due to its high requirements for computational power and Internet bandwidth, the proof that those problems can be solved without specialized hardware can lead to new and more immersive video experiences without the need for 3D glasses or VR goggles.

High mobility has become a mandatory consideration in the design and development of wireless communication systems today. It results in a doubly selective or a time-varying multipath channel that is arduous to estimate. Compressed Sensing (CS) schemes like the Rake Matching Pursuit (RMP) algorithm exploit the inherent sparsity in these channels and are often able to resolve the multipath environments into their respective sparse representations, even under the presence of large Doppler shifts. However, the complexity involved is substantial and might imperil practical implementation on resource limited consumer device hardware. We propose a novel low-complexity CS scheme, called as Gradient Rake MP (GRMP) that optimizes the search related to the multipath delays resulting in a complexity that is significantly lower than all CS based channel estimation schemes. Additionally, the results confirm that the Bit Error Rate (BER) performance of GRMP is comparable to that of the more complex RMP algorithm. The dictionary is an imperative requirement of CS schemes and plays a decisive role in the quality of the channel estimate. Often in literature, details regarding the generation of the dictionary and its complexity is ignored and instead a suitable dictionary is assumed to be available at the receiver. This paper investigates the complexity and memory demands associated with the dictionary and presents a novel scheme to build it using the concept of wavelets. The ideas proposed in this paper are targeted towards reducing the complexity associated with the estimation of a doubly selective channel with a goal to enable implementation on consumer hardware. Although implemented for the IEEE 802.11p standard, the proposed ideas are applicable to any Orthogonal Frequency-Division Multiplexing (OFDM) based wireless system that is expected to work in highly mobile environments.

Traditionally, network core devices are simple and the complexity is in the end-hosts. With the rise of Software- Defined Networking, this changes and complex functions are moved into the network core. This paper presents Transparent Transmission Segmentation (TTS), which is able to improve performance by executing parts of network functions at the core. An implementation for segmenting TCP connections in SDNs is presented, including the network integration and traffic steering process, as well as evidence on its positive effects on latency. Deutsche Forschungsgemeinschaft (DFG)

High mobility along with multipath propagation in mobile reception results in a doubly selective channel that exhibits selectivity in both the time and frequency domain. The estimation and subsequent compensation of such a channel is a non-trivial task. The DVB-T2 standard defines the use of Orthogonal Frequency-Division Multiplexing (OFDM) in the physical layer that along with many other inherent advantages helps in reducing Inter-Symbol Interference (ISI) by the use of a Cyclic Prefix (CP). However, Doppler shift caused due to high mobility in these systems leads to the time selectivity of this channel and results in a loss of orthogonality between the sub-carriers. Thus, it is appropriate to estimate the Doppler shift in the time domain. Although the DVB-T2 standard specifies different pilot patterns in the frequency domain, a pilot/reference pattern is not available in the time domain. In this paper, we propose time-domain pilots in DVB-T2, a novel approach that aids in the estimation of a doubly selective channel by any suitable channel estimation algorithm. The proposed technique retains the structure of the transmitter chain and is computationally feasible from an implementation standpoint. Moreover, it would enable a precise estimate of the channel that would improve the performance of the DVB-T2 receiver in challenging channel conditions.

Light fields are gaining more impact in today's filming technology due to post-processing possibilities such as software refocus, perspective shift, reconstruct 3D images, and depth scaling. Light fields are captured light rays at every point in space travelling in every direction, therefore a large amount of data is recorded for representation. For exchange, broadcast and storing of such imagery data, we lack efficient compression techniques. This paper proposes a light field compression approach based on low-complexity preprocessing by pseudo-temporal sequencing frames suitable for state-of-the-art codecs. An efficient novel approach of circular reordering is introduced where the redundancy between frames is maximized and exploited by low-delay HEVC predictive coding. Experimental results showcase a consistent increase in both PSNR and SSIM for several light fields and outperforms available schemes with pseudo-temporal sequencing using HEVC. An increase upto 1 dB gain compared to state-of-the-art reordering techniques using HEVC is achieved.

Adaptive video streaming is today's predominating video delivery model in the Internet. However, existing bitrate adaptation achieves satisfactory performance only under buffering of several tens of seconds. The major challenge is the stabilization of client buffer dynamics, because a small buffer lacks resilience to the buffer fluctuation imposed by the mismatch between the network throughput and the average bitrate of the selected video chunk. This study tackles this fundamental challenge by developing a novel adaptation algorithm that minimizes buffer deviation from the desired level despite a small client buffer and jointly reduces the rate of bitrate switching. Our evaluation methodology combines a bitrate-adaptive streaming simulation with a trace-driven simulation of 3G mobile network in the context of video streaming. Simulation results show that our algorithm effectively supports low-latency adaptive video streaming with a buffering delay as small as a chunk-duration.

Traditional services like video broadcast and telephony are more and more implemented on general-purpose networks. This is usually the Internet, which has different properties compared to special-purpose infrastructures as the public switched telephone network. This paper highlights three major causes why this migration is currently providing user experiences that are worse than with traditional networks. In particular, challenges caused by changed latency characteristics are addressed. For every cause, approaches are highlighted to tackle them, including novel transport protocols as well as paradigms to optimize transmissions at the network core. A presentation of new network infrastructures following the concept of Software-Defined Networking (SDN) completes the picture and allows to implement flow-oriented forwarding. Finally, the Open Networking Testbed operated by the Telecommunications Chair at Saarland University is presented, including its means to evaluate novel protocols, paradigms, and concepts in a controlled and realistic environment. Having international partners in place, it is possible to test new approaches on a global scale.

Vehicular Communication Systems also known as Vehicle-to-Vehicle communication systems (V2V) are standardized by the IEEE 802.11p standard which specifies the use of Orthogonal Frequency-Division Multiplexing (OFDM) in the physical layer. Doppler shift due to high mobility along with multipath propagation in vehicular environments result in a time-varying multipath or a doubly selective channel. The estimation of such a channel and its subsequent equalization is a non-trivial task [1]. Several techniques for the estimation and equalization of a doubly selective channel have been proposed in literature. However, these techniques either perform poorly or work with a complexity that is prohibitive for consumer hardware [2]. We propose the Matching Pursuit (MP) algorithm, a Compressed Sensing (CS) scheme to perform the task of channel estimation. We also propose modifications to this algorithm that are focused to reduce the computational complexity. We further show that the channel estimate from the MP algorithm can be used by both conventional and state-of-art equalizers. The proposed algorithms are implemented as a Software-Defined Radio (SDR), that provides an excellent platform for design, development and the ability to perform simulation as well as real-world experiments. Simulation results assess the performance gains of the proposed channel estimation and equalization schemes.

HTTP-based adaptive streaming is today's predominating video delivery model over Internet. However, state-of-the-art bitrate adaptation algorithms are not suitable for low-latency adaptive video streaming due to a distinct lack of buffer dynamic stabilization at the client. A small buffer size lacks resilience to the buffer fluctuation imposed by the mismatch between the network throughput and the average bitrate of the selected video chunk. This paper introduces a bitrate adaptation algorithm that minimizes buffer deviation from the desired level despite a buffering delay as small as a chunk-duration. We demonstrate the performance of our algorithm with respect to the quality-of-experience (QoE) using two evaluation frameworks and compare it against two well-known streaming algorithms, FESTIVE and DASH VLC plugin. The preliminary results show that our algorithm achieves at least 11% higher QoE in the simulation framework and 4% in the emulation framework.

In 2015, NetFlix made the headlines with nearly 37% of Internet traffic being caused by their streaming applications1. The trend towards similar applications is further accelerated by increasing number of mobile devices used to consume high resolution video and interact with low latency. Hence, new challenges arise that can no longer be solved by solely altering the network's edge. Software-Defined Networking (SDN) and its most popular incarnation OpenFlow [1] cause a trend towards intelligent devices at the network's core. The end-to-end (E2E) principle, as proposed by Saltzer et al. [2] in 1984 and further discussed by Moors [3] in 2002, is one of the paradigms to be reconsidered in this new world. Theory and practice show that segmenting multiple transmission domains can provide benefits for the overall network performance. Current approaches, as e.g. Split-TCP [4], require changes to the end-hosts and applications to explicitly use this protocol, which can be avoided when using SDN technologies. We propose the novel networking pattern Transparent Transmission Segmentation (TTS) that allows to separate certain domains and lead to better performance, especially in heterogeneous environments. With most of multimedia streaming traffic being transmitted using Dynamic Adaptive Streaming (DASH) [5], it is straightforward to consider TCP transmissions. We identify network functions of TCP that can perform better when segmentation is employed and give a differentiated view on this using simulations measuring transmission times. The evaluation shows that although we have performance degradations of maximally 1% for mean, median and standard deviation for certain scenarios, our approach can achieve more than 14% improvement on certain scenarios and 3 – 10% on most scenarios. This shows that TTS has to be applied thoughtfully, but can provide significant improvements when employed in the scenarios with high loss and jitter.

This paper deals with a browser based approach of multi-reality interfaces in the context of e-maintenance in modular factory environments, specifically remote monitoring and maintenance scenarios. Through their rich visualization capabilities (stereoscopic video & 3D graphics) multi-reality interfaces offer a way to aggregate and structure the wide-spread variety of information from different sources that is required to analyze such complex scenarios. Additionally, they serve the purpose of making collaboration with on-site personal easier and more efficient in time and cost. Finally, the browser based implementation fits very well into the web based world that a modular factory environment represents and make the presented solution extremely adaptive to (non-)industrial HMI.

Internet Protocol Television (IPTV) over IEEE 802.11 Wireless LANs, which are considered today as the de-facto wireless access network for local distribution (home networks, hotspots), brings forth a big challenge to guarantee high quality multicast/broadcast delivery over unreliable and time-variant wireless channels. However, in IEEE 802.11 Wireless LANs, current standard MAC layer protocols do not provide any error correction scheme for broadcast/multicast. In our previous work, we enhanced a Leader Based Protocol (LBP) and proposed a Beacon-driven Leader Based Protocol (BLBP) for the MAC layer multicast error control. However, as pure Automatic Repeat reQuest (ARQ) schemes, both LBP and BLBP are not efficient for large multicast groups. In this paper, we combine BLBP and packet level Forward Error Correction (FEC) and propose a Hybrid Leader Based Protocol (HLBP) for the MAC layer multicast error control. HLBP transmits the original data packets using raw broadcast and retransmits parity packets using an improved BLBP which is based on block feedback. To guarantee the required Packet Loss Ratio (PLR) under strict delay constraints for multicast delivery over a Gilbert-Elliott (GE) channel, we analyze LBP, BLBP and HLBP, develop analytic performance models that allow optimizing the configurations. The performance models are verified via simulation experiments. Both the theoretical analysis and simulation results show that HLBP is much more efficient than LBP and BLBP especially for large multicast groups and is also more efficient than the best application layer multicast error correction scheme. BLBP needs the minimum number of redundancy transmissions among all pure ARQ based schemes while HLBP needs the near-minimum number of redundancy transmissions among all schemes. Due to the simplicity and efficiency, BLBP is a good choice for IPTV multicast delivery for small groups in Wireless LANs while HLBP is a better choice for large multicast groups.

The subject of quality of service (QoS) and admission control for wireless home networks is of great interest now for both research and industry communities. Many standard developing organizations consider admission control as the only means to prevent overload of the network and a basis of a well thought through QoS solution. In this paper, we present a post-admission control solution for wireless home networks. By analysing the traffic load of the network the admission control decisions are made with respect to the already running applications. With the help of a dynamic prioritization mechanism coupled with the policy-based user-centric QoS solution, the presented approach provides flexible regulation of traffic on the network and improves the overall user experience.

This special issue is a collection of recent innovative research results, albeit inevitably limited, on most aspects of IPTV in multimedia broadcasting, including architectural standardization, content protection, quality of service and quality of experience, middleware, content consumption measurement, application and content platform, as well as network control, etc. There are 21 articles in this issue.

State-of-the-art rate adaptation is not suitable for low-latency dynamic streaming due to a lack of explicit stabilization of client buffer dynamics. Specifically, when the client buffer is at its maximum level, a biased throughput feedback is induced that leads to suboptimal adaptation decisions. We propose a rate adaptation algorithm that minimizes buffer deviation from the desired level, and implement the algorithm within a server-based architecture to avoid delayed feedback. We evaluate the performance of our solution using video quality metrics derived from subjective user tests and compare it against two well-known streaming architectures, DASH VLC plugin and QAC. The results show that our solution improves user experience in dynamic streaming with buffering delays as low as the chunk duration. Furthermore, we explore the impact of the server-based architecture on the streaming performance of the proposed solution by experimental analysis. Within a client-based streaming architecture, our rate adaptation algorithm also achieves at least 65% lower impairment of playback stalls.

In order to meet very low packet loss rate (PLR) requirements under strict delay constraints for IP based DVB services over wireless home networks (WHN), an optimized real-time transport protocol (RTP) level hybrid error correction (HEC) scheme is proposed in this paper. The scheme is compared with the Adaptive Forward Error Correction (AFEC) scheme using Reed-Solomon code. By the HEC scheme, the needed redundancy information (RI) for repairing missing packets can be minimized, especially in DVB systems with a single user. The result of quantitative analysis shows that the needed RI of the proposed scheme can even achieve the Shannon limit with only three retransmission rounds. On the other hand, our studies show that the performance of the HEC scheme is better than that of the AFEC scheme if both the average link PLR and group size are small. However, if both the average link PLR and group size are large enough, the AFEC scheme performs better than the HEC scheme. At last, the simulation results for one typical scenario are also provided in the paper to validate the result of quantitative analysis for the HEC scheme. As a result, because of the simplicity and efficiency of the proposed HEC scheme, it is a very good choice for guaranteeing the Quality of Service for DVB systems in WHN with the small group size and small average link PLR. However, for scenarios with the large average link PLR and large group size, the AFEC scheme should be taken into account

The development of wireless communications created broad opportunities in mobile communications. However, poor channel estimation and lack of proper Doppler shift compensation techniques limit the mobility considerably. Therefore, a high quality and low-complexity channel estimation is one of the most important conditions for high mobility. In this paper we propose a low-complexity delay-Doppler search (DDS) for matching pursuit algorithms. The proposed search algorithm can be applied to MP algorithms which estimates the delay and the Doppler features of the channel in two dimensions efficiently. Two dimensional approach lowers the complexity significantly. At the end we show results for the DVB-T2 systems.

A high mobility algorithm is introduced for OFDM systems based on the FFT which can compensate Doppler shifts in time-variant channels with low complexity. We first propose a channel matrix inversion algorithm with a complexity similar to FFT and then introduce a channel matrix construction based on matching pursuit algorithm.

Dynamic video streaming performs video rate selection based on streaming client's throughput estimate. In practice, the accuracy of the estimated throughput is limited due to feedback delay and unawareness of the dynamics of the underlying HTTP/TCP transport layer. Furthermore, state-of-the-art rate selection methods inherently introduce a two-fold quantization error, as will be detailed in this paper. As a result, streaming adaptation is often erroneous and causes client buffer instability. In this paper, we present Open-Loop rAte Control (OLAC), an adaptive streaming architecture designed to support low-latency streaming applications. The key components of our architecture are a server-side simulation of the streaming client's buffer, which provides a low-delay feedback for the video rate selection, and a hybrid adaptation logic based on throughput and buffer information, which stabilizes the adaptive response to dynamics of transport and application layers. We evaluate the performance of OLAC in wireless streaming scenarios by comparing it against two well-known streaming architectures, QAC and vlc-plugin DASH. The results show that our approach achieves at least 10% higher average quality and at least 73% lower rebuffering duration - achieving zero rebuffering duration in most scenarios - for dynamic live streaming with buffering delays as low as two seconds.

Adaptive video delivery approaches perform video rate selection based on the streaming client's throughput estimate. In practice, the accuracy of the estimated throughput is limited due to feedback delay and the unawareness of the dynamics of the underlying HTTP/TCP transport layer. As a result, streaming applications employ large playback delays in the order of tens of seconds so as to maintain continuous video rendering with good quality and bandwidth utilization. In this paper, we introduce DASP, an advanced video rate adaptation, to achieve a low-latency adaptive video streaming. The key components of our solution are a server-side mirroring of the streaming client's buffer, which provides a low-delay feedback for the video rate selection, and a hybrid rate adaptation logic based on goodput and buffer information, which stabilizes the adaptive response to the dynamics of transport layer.We demonstrate the performance of our rate selection algorithm by evaluating the stability of the receiver buffer under low-latency and high-definition adaptive video streaming with variable bit rate encoding over an emulated wide-area network link. The results also show that our approach is promising and applicable for dynamic live streaming with a playback delay as low as the chunk duration.

Multi-view video (MVV) offers many possibilities for video communication systems. The creation of virtual views allows to change the viewer's perspective interactively on the receiver side. In this paper we propose a real time algorithm capable of rendering virtual views with a high image quality without the support of scene geometry information. We utilize the highly parallel architecture of recent GPUs for online depth estimation and a sophisticated but generally available shader architecture for efficient rendering. Using these methods we are able to render virtual views with resolutions of up to 3600×3000 with a frame rate of more than 25 frames per second.

Accurate, dynamic rate selection is considered a hard problem in stream switching video applications over Internet Protocol. Not only the feedback delay but also the unawareness about the dynamics of the underlying HTTP/TCP transport layer require streaming applications to implement huge receiver buffers in the order of tens of seconds so as to maintain continuous video rendering with acceptable quality and bandwidth utilization. In order to achieve low-latency adaptive video streaming with a buffering delay as low as the chunk size, we present a streaming architecture under the implementation of media transport with predictably reliable transport and delay-based congestion control. A key component of our solution is a server-side simulation of the streaming client's buffer, which provides low-delay feedback for the video rate selection. The receiver buffer is being controlled via an open-loop synchronization mechanism relying on a reference clock that is transmitted within the media packets. We demonstrate the performance of our approach by evaluating the stability of the receiver buffer under low-latency and highdefinition adaptive video streaming with variable bit rate encoding over an emulated wide area network link.

In Orthogonal Frequency Division Multiplexing (OFDM) mobile communication systems time-varying channels cause an inter-carrier interference (ICI) and degrade the performance due to the frequency orthogonality loss. Doppler shift compensation or equalization is very complex and often prohibitive in consumer receiver equipment. We propose a Doppler effect compensation / equalization technique for these kinds of scenarios. Using this technique we can compensate fast time-varying channels perfectly within an OFDM symbol with uniform Doppler shift and perfect channel knowledge. The computational complexity of the technique is the same as of FFT (N log(N)).

Large IP networks tend to have a highly heterogeneous structure with many different transmission segments, each endowed with individual characteristics like data rate or delay. Obviously, applications find it difficult to precisely adjust their sending behavior to the underlying topology. Basically, a higher network utilization without congestion is desirable. Server applications demand for reliable information about the available data rate in the network in order to provide a sufficient quality to the clients. Traditionally, this information is completely gathered by source and sink without interaction with the network infrastructure. The rise of software-defined networks (SDNs) enables the integration of the network into this process. In this paper we propose a rate control mechanism for software-defined networks to avoid congestion situations. The approach enables clients to communicate with the network to bidirectionally exchange information about available and occupied data rate. We found that this customized communication approach enables better network load distribution and application transmission reaction - which can be easily implemented with SDNs. Ultimately, the controller finds an efficient forwarding path in the network and the clients have a reliable upper bound for their sending data rate.

Despite the fact that non-interactive Internet multimedia applications are prevalently deployed via HTTP/TCP, dedicated real-time protocols remain the essential basis for interactive voice and video services. This is particularly pronounced by the standardization of WebRTC, which enables peer-to-peer realtime communication between web browsers via the RTP protocol suite. However, a compound solution that contributes error and congestion control for RTP-based communications services is missing by now. In our previous work we presented a novel transport protocol - Predictably Reliable Real-time Transport (PRRT) - a protocol layer that efficiently supports the reliability required by interactive multimedia services under their specific time constraint. In the following paper we integrate PRRT's workflow and header format into a bi-directional RTP session by enhancement of the profile for generic forward error correction. We particularly focus on a low messaging overhead and the preservation of backward compatibility in the protocol's error control.

Due to the increasing number of portable and mobile consumer electronics devices, today's Internet paths include generally at least one wireless segment. Predominant architectures are IEEE 802.11 wireless as well as 3G/4G mobile networks. Dynamic stream switching schemes such as DASH (Dynamic Adaptive Streaming over HTTP) have been developed in order to adapt the rigid bandwidth requirement of real-time video services to the dynamic capacity of such Internet channels. Meanwhile, the actual bandwidth provisioning on wireless and mobile networks is no longer the limiting factor for high definition media streaming. However, the quality of such HTTP/TCP-based stream switching services is severely affected by the inefficiency of TCP's loss-based congestion control, the ACK-clocked window control as well as the ARQ-only error control in presence of physical packet loss and large propagation delay. In this paper we address this quality bottleneck on wireless and mobile networks via a novel dynamic streaming architecture that proactively transmits packetized multimedia services under delay-based congestion control. The presented scheme builds upon the Predictably Reliable Real-time Transport protocol (PRRT) that efficiently provides the reliability required by multimedia services under their specific delay constraint. Under the delay-based congestion control the transport protocol implements opportunistic TCP-friendliness, which allows media streams to acquire additional bandwidth that remains unutilized by inefficient TCP sessions. We evaluate the bandwidth allocation of our dynamic streaming approach and demonstrate a significant increase of the video bit rate compared to HTTP-DASH on mobile broadband and wireless home network infrastructures as well as an emulated broadband access network path.

Internet multimedia traffic currently occupies more than half of the total Internet traffic and it continues to expand tremendously. Targeting to meet strict constraints imposed by the requirements of real-time multimedia applications appropriate error-correction techniques should be implemented within the data dissemination network. We propose to introduce multipurpose relay nodes called Mediators into several positions within the tree networks typical for multicasting and broadcasting scenarios. By utilizing the error-correction domain separation paradigm in combination with selective insertion of the supplementary data from parallel networks, when the corresponding content is available, the proposed mechanism reduces the total network load and improves scalability of multicast/broadcast transmission. We share our view on how the existing application frameworks could benefit from the incremental deployment of the proposed mechanism. Experimental results confirm suitability and applicability of our assumptions.

When it comes to creating a very large combined screen comprised of multiple individual displays, a so-called tiled display wall, dedicated graphics and synchronization hardware is commonly used for two reasons: firstly, without compromising display resolution neither spatially nor in the time domain, each display needs to be driven by one discrete frame buffer at the full native display resolution. Secondly, synchronization of the individual displays to a common clock is required for smooth and undistorted reproduction of motion spanning multiple displays. One example for synchronization of multiple frame buffers is digital television broadcast. Herein, audio-visual bitstreams are generated including clock information in order to enable receivers to align playback frequency and phase accordingly, which is termed genlock. In this paper we present an architecture for IP-based reverse genlock, whereas a display clock signal is used both for inter-display synchronization as well as for generator synchronization. The accuracy of the synchronization enables multi-HD active stereoscopy at 120 Hz across many solely IP-interconnected displays.

As audiovisual communications services become the dominant traffic source in the Future Internet, their stable coexistence and their fairness towards background traffic is an active research topic. The Internet implements flow multiplexing in a decentralized fashion via congestion control. Audiovisual media streams, however, are classified as inelastic traffic flows that do not allow rate control. Today's Internet streaming applications implement dynamic HTTP streaming in order to adapt to the transport protocol's congestion window size. Nevertheless, the infeasibility of TCP's congestion and error control for low-latency, high-rate communications services leads to significant underutilization of the available bandwidth. As a result, the multimedia quality suffers unnecessarily. In order to address TCP's drawbacks, we implement a novel transport protocol — Predictably Reliable Real-time Transport (PRRT), a protocol layer that efficiently supports the reliability required by multimedia services under their specific time constraint. PRRT is based on hybrid error coding that allows for both proactive and reactive reliability mechanisms. Specifically, it is designed to achieve the optimal trade-off between those mechanisms under strict delay constraints in order to optimize bandwidth utilization. Based on this transport protocol we present a dynamic media streaming architecture with near-optimal bandwidth utilization under equation-based congestion control. We evaluate PRRT's bandwidth utilization and compare it to recent media streaming standards such as HTTP-DASH (Dynamic Adaptive Streaming over HTTP).

Erasure error correction by retransmission based on feedback jamming is a mechanism that has been proposed for wireless networks to reduce the overhead of collecting feedback about reception of the same data from multiple receivers (multicast). This is achieved by aggregating feedback from all receivers within the same time slot. Cancellation of a single positive feedback message by negative feedback within the same time slot triggers a retransmission. Yet, this feedback jamming may not be perfectly reliable. In this paper, we evaluate the residual error of a low-delay feedback aggregation and jamming multicast retransmission protocol in the presence of i.i.d. erasures and medium access collisions in large OFDM based MBMS systems by example of 802.11 wireless LANs. We show by simulation how the residual error due to an imperfect jamming can be tuned to meet a certain residual error rate, as e.g. required by audio-visual codecs, and that it becomes negligible for large groups of receivers.

The Internet is experiencing a change in paradigm: once determined by text-based applications, currently, various types of audio-visual content already adopt the major part of the bandwidth. Audio-visual applications and multimedia data impose individual demands on reliability and delivery time, which are not sufficiently covered by available transport protocols. In addition, a more efficient use of network resources such as link bandwidth or the spatial distribution of network routes is essential. Therefore, a split of a given transmission path into several independent areas, where each contains a set of similar, adjacent network links is appropriate. An individual transmission and error correction scheme can be applied to such a group of related links. In previous work it has been shown that there is an optimum number of segments. Too few or too many limit the achievable efficiency, i.e. increase the required redundancy information. The search for this optimum, however, is a complex task not applicable to real-time adaptivity. This paper proposes a decentralized, metric based mechanisms to decide where a network path should be intersected, leading to a lower complexity for the retrieval of these areas. We also investigate an example where the proposed clustering approach leads to an overall redundancy information of about 1/4 of the redundancy information obtained by a traditional end-to-end approach and thus is very close to the Shannon limit.

Leader-based protocols have been proposed for wireless networks such as 802.11 as an alternative approach to MAC layer ARQ. If based on feedback cancellation, such protocols may achieve low latency and high throughput at predictable reliability and superior scalability with respect to the number of subscribers. Especially for delay-bounded multicast A/V transmission these properties are highly desirable. The authors have proposed such schemes to the IEEE task group TGaa, which is dedicated to introducing more reliable multicast to 802.11 wireless LANs. This paper provides results on wireless feedback cancellation probabilities derived from real measurements with consumer hardware as well as simulation data.

Overlay architectures [1] are the information theoretical playgrounds of the Internet. They provide significant gains in throughput by shifting complexity into the network nodes. A consequence is the requirement for smart routers between adjacent network segments. Since meanwhile audiovisual media allocate the major part of the Internet traffic, protocols as well as overlay structures have to serve Predictable Reliability under Predictable Delay (PRPD). The link-level operation mode of the AHEC architecture [6] is an approach to fit individual channel coding parameters to inhomogeneous network segments. All link segments of a chosen network route share an overall time and reliability budget determined by the respective constraints of the application. Whereas some structures require support by all intermediate nodes, just the contribution from a limited number of routers is reasonable in the AHEC overlay: The budget distribution hits a saturation point above which configurations are suboptimal. This paper derives the saturation point with respect to the distribution of the delay constraint of audiovisual applications in the link-level operation mode of the AHEC. The result has major impact on the complexity of coding as well as the protocol parametrization.

In this paper, we address the QoS issue of reliable real-time multimedia multicast services over packet-switched networks. Using a generalized architecture of delay limited erasure error recovery, we propose an efficient greedy algorithm for optimizing the parameters of the general architecture. Through the proposed greedy algorithm, the total needed redundant information can be minimized by choosing the best scheme automatically among those entire schemes integrated in the architecture. As a result, the greedy algorithm can be a good candidate technique for guaranteeing QoS requirements for real-time multimedia multicast services over packet-switched, especially over wireless networks.

In Wireless LANs MAC layer multicast error control schemes lead to shorter delays compared with application layer schemes. In this paper, we enhance a Leader Based Protocol (LBP) and propose a Beacon-driven Leader Based Protocol (BLBP) for the MAC layer multicast error control. Like LBP, BLBP selects one of the multicast group receivers as the leader and allows acknowledgement and negative acknowledgements from the leader receiver and non-leader receivers respectively. The use of the beacon frame is to lead the non-leader receivers to set timers and to announce the sequence number of the following data frame. Thanks to the sequence check based on the beacon frame, BLBP avoids the unnecessary transmissions in LBP. Both the theoretical analysis and simulation experiments show that BLBP can correct nearly all the errors for all receivers in the MAC layer and is more efficient than LBP. BLBP is even more efficient than the application layer automatic repeat request (ARQ) scheme and the total multicast delay is much shorter. BLBP is very good for real-time multicast applications with strict delay constraints.

This paper presents a framework for humanoid binaural source separation in a non-ideal reverberant environment and investigates the ideal position of a human head for several separation algorithms. A movable human dummy head residing in a normal office room is used to model the conditions humans experience in a complex auditory scene. Prior to source separation the dummy head analyzes the auditory scene, estimates several parameters and aligns to the best separation position. The estimated information is used to enhance the succeeding source separation: Several parallel separation paths estimate independent time-frequency masks to demix the target source. A combination stage infers a final estimate of the ideal binary mask. The presented approach outperforms fixed beamforming and the DUET source separation algorithm in reverberant auditory scenes consisting of two and three speech sources by up to 17 dB SIR gain.

Wireless networking technologies are becoming very popular for distribution of digital content throughout the home. However, media distribution is not ready yet to satisfy users' quality expectations in wireless networks. Due to bandwidth variations and high error rates, a number of quality of service (QoS) mechanisms have to be deployed in the network in order to guarantee a certain level of quality. We propose a user-centric QoS management approach based on user preferences. Unlike traditional QoS approaches that always give multimedia applications a pre-determined high priority due to their strict latency and loss requirements (e.g. Wi-Fi multimedia), the proposed approach classifies traffic according to user preferences, thus, ensuring a certain quality regardless of the application requirements. The application priorities are assigned according to user profiles and automatically updated in case of changes in the network or user environment. The implementation details and experimental results demonstrate a practical application of the described approach in the wireless home network. The paper suggests that the introduced management approach will significantly contribute to the overall user experience in digital home.

In this paper, an application layer hybrid error correction scheme with Reed-Solomon (HEC-RS) code is proposed to satisfy the very low packet loss ratio (PLR) requirement under a strict latency limit for DVB services over wireless LANs. The PLR performance of the HEC-RS scheme is analyzed and compared with the HEC scheme based on the packet repetition (HEC-PR) technique and an adaptive forward error correction (AFEC) scheme based on RS codes. It is found that the needed redundancy information (RI) of the HEC-RS scheme is much less than that of the AFEC scheme. However, if the group size and link PLR are small enough, the needed RI of the HEC-RS scheme is proven to be much more than that of the HEC-PR scheme. As a result, the optimum performance can be achieved when combining the HEC-PR and the HEC-RS scheme.

Modern home networking technologies provide great possibilities for distributing digital content throughout the home. However, media distribution is not ready yet to satisfy users' quality expectations, especially in wireless networks. That is why a number of quality of service (QoS) mechanisms have to be deployed in the network in order to guarantee a certain level of quality. We propose a user-centric QoS management approach based on user preferences. Unlike traditional QoS approaches that always give multimedia applications a pre-determined high priority due to their strict latency and loss requirements, the proposed approach classifies traffic according to user preferences, thus, ensuring a certain quality regardless of the application requirements. The application priorities are assigned according to user profiles and automatically updated in case of changes in the network or user environment. The paper suggests that the introduced management approach will significantly contribute to the overall user experience in digital home.

We propose a model for adaptive QoS management that dynamically adjusts application priorities based on user requirements. Unlike traditional QoS approaches that always give multimedia applications a pre-determined high priority due to their strict latency and loss requirements, the proposed model classifies traffic based on user preferences, thus, ensuring a certain level of quality regardless of the application requirements. Additionally, the model employs usability rules to detect and penalize concurrent use of conflicting applications in order to make fair use of the available bandwidth. The example of possible implementation of the model based on 802.11e standard demonstrates effectiveness of the proposed model with respect to the typical QoS schemes.

Visual CAPCTCHAs, like gimpy images, are commonly used to screen human users from automated computer scripts. We propose using them for direct, visual authentication of digital documents. The basic assumption is that if it is hard or impossible for the computer to recognize a document, then it cannot manipulate it. If such a document is still recognizable by a human, he or she can be confident that it is authentic, or at least that no automated process has manipulated it. Such authentication is highly desirable in the context of digital signatures, which, due to the computational complexity, have to be produced by a computer. It can be a specialized, trusted cryptographic computer, e.g. a smartcard, but the path to it is generally not secure. Visual CAPTCHAs can be used for securing the path without the need for additional hardware.