Multimedia Streaming Applications Research Articles

Real-World Multimedia Streaming for Software Defined Vehicular Ad Hoc Networks

Multimedia services with required Quality of Service (QoS) is one of the most critical challenges in Software Defined Network (SDN) based Vehicular Ad-Hoc Networks (VANETs). It forms an essential part of the Intelligent Transport System (ITS), where infotainment services play an essential role.Streaming multimedia is one of the most popular applications and has a high demand for VANET infotainment services. The major issues for multimedia streaming on VANET are scalability, mobility of vehicles, frequent connection failures, frequent change in network topology, and distributed architecture with heterogeneous devices. To overcome these problems and provide a better QoS, we propose using a hybridarchitecture with a combination of VANET and SDN called Software-Defined Vehicular Networks (SDVN). This work presents a modified POX controller-based SDN framework for VANETs, especially for multimedia streaming applications in realistic traffic patterns. The proposed work has a real-world setup developed using Simulation of Urban Mobility (SUMO), where iPerf generates multimedia traffic. Also, streaming standard-definition YouTube videos in real-time between the vehicular nodes was done. The modified POX controller could take advantage of the centralised perspective of the network for action determination, and the integrated spanning tree algorithm reduced the redundancy. Despite the dynamic nature of the testing environments, the proposed Modified POX Controller consistently outperformed VANET, with up to 21 to 42% better packet delivery ratio for higher data transfer rates. The overall improvement in QoS parameters also accompanies an improvement in the consumers Quality of Experience (QoE) factors.

Streaming Erasure Codes Over Multi-Access Relayed Networks

Many emerging multimedia streaming applications involve multiple users communicating under strict latency constraints. In this paper we study streaming codes for a network involving two source nodes, one relay node and a destination node. In this paper’s setting, each source node transmits a stream of messages, through the relay, to a destination, who is required to decode the messages under a strict delay constraint. For the case of a single source node, a class of streaming codes has been proposed by Fong et al., using the concept of delay-spectrum. The current paper presents a novel framework, which constructs streaming codes for a relayed multi-user setting by sequentially constructing the codes for each link. This requires a characterization of the set of all achievable delay spectra for a given rate, blocklength and number of erasures, beyond the specific choice considered by Fong et al. This characterization is presented in the paper for systematic codes. Using this novel framework, the first proposed scheme involves greedily selecting the rate on the link from relay to destination and using properties of the delay-spectrum to find feasible streaming codes that satisfy the required delay constraints. A closed form expression for the achievable rate region is provided, and conditions for when the proposed scheme is optimal are established by a natural outer bound. The second proposed scheme builds upon this approach, but uses a numerical optimization-based approach to improve the achievable rate region over the first scheme. Experimental results show that the proposed schemes achieve significant improvements over baseline schemes based on single-user codes.

An efficient congestion control in multimedia streaming using adaptive BRR and fuzzy butterfly optimization

AbstractRecently, various video streaming platforms such as YouTube, Netflix and so forth have achieved great growth in the media industry. However, uncertain dynamic activities such as low bandwidth, packet loss and rapid fluctuations, quality, and interrupted flow are still creating several complications in multimedia streaming applications and it is necessary to provide enhanced throughput and minimum frame delivery delay. Therefore, the significant objective of this article is to attain enhanced bandwidth allocation as well as a high packet scheduling rate. In this article, an adaptive BBR technique is proposed for transmitting real‐time video with minimum transmission delay and packet loss. The adaptive BBR mainly uses the pacing gain update mechanism to maintain different flow rates and minimal data in the bottleneck queue. Also, a fuzzy butterfly optimization (FBO) with a ranking strategy is proposed to attain optimal bandwidth allocation. The FBO algorithm with ranking strategy computes the minimal bandwidth needed by the model to satisfy the service requirement of every queue in a particular round. Then, the packet scheduling is done by employing an intelligent time slice‐based round‐robin (ITSBRR) technique. The ITSBRR algorithm is employed to enhance the throughput and end‐to‐end delay by implementing the time quantum‐based scheduling mechanism. Finally, the experimental analysis is carried out for various parameters namely single path transmission delay, packet loss; frame delay is computed for various techniques to compute the efficiency of the proposed system.

SDN Enabled QoE and Security Framework for Multimedia Applications in 5G Networks

The technologies for real-time multimedia transmission and immersive 3D gaming applications are rapidly emerging, posing challenges in terms of performance, security, authentication, data privacy, and encoding. The communication channel for these multimedia applications must be secure and reliable from network attack vectors and data-contents must employ strong encryption to preserve privacy and confidentiality. Towards delivering secure multimedia application environment for 5G networks, we propose an SDN/NFV (Software-Defined-Networking/Network-Function-Virtualization) framework called STREK , which attempts to deliver highly adaptable Quality-of-Experience (QoE), Security, and Authentication functions for multi-domain Cloud to Edge networks. The STREK architecture consists of a holistic SDNFV dataplane, NFV service-chaining and network slicing, a lightweight adaptable hybrid cipher scheme called TREK, and an open RESTful API for applications to deploy custom policies at runtime for multimedia services. For multi-domain/small-cell deployments, the key-generation scheme is dynamic at flow/session-level, and the handover authentication scheme uses a novel method to exchange security credentials with the Access Points (APs) of neighborhood cells. This scheme is designed to improve authentication function during handover with low overhead, delivering the 5G ultra-low latency requirements. We present the experiments with both software and hardware-based implementations and compare our solution with popular lightweight cryptographic solutions, standard open source software, and SDN-based research proposals for 5G multimedia. In the microbenchmarks, STREK achieves smaller hardware, low overhead, low computation, higher attack resistance, and offers better network performance for multimedia streaming applications. In real-time multimedia use-cases, STREK shows greater level of quality distortion for multimedia contents with minimal encryption bitrate overhead to deliver data confidentiality, immunity to common cryptanalysis, and significant resistance to communication channel attacks, in the context of low-latency 5G networks.

Energy-Aware Scheduling of Streaming Applications on Edge-Devices in IoT-Based Healthcare

The reliance on Network-on-Chip (NoC)-based Multiprocessor Systems-on-Chips (MPSoCs) is proliferating in modern embedded systems to satisfy the higher performance requirement of multimedia streaming applications. Task level coarse grained software pipeling also called re-timing when combined with Dynamic Voltage and Frequency Scaling (DVFS) has shown to be an effective approach in significantly reducing energy consumption of the multiprocessor systems at the expense of additional delay. In this article we develop a novel energy-aware scheduler considering tasks with conditional constraints on Voltage Frequency Island (VFI)-based heterogeneous NoC-MPSoCs deploying re-timing integrated with DVFS for real-time streaming applications. We propose a novel task level re-timing approach called R-CTG and integrate it with non linear programming-based scheduling and voltage scaling approach referred to as ALI-EBAD. The R-CTG approach aims to minimize the latency caused by re-timing without compromising on energy-efficiency. Compared to R-DAG, the state-of-the-art approach designed for traditional Directed Acyclic Graph (DAG)-based task graphs, R-CTG significantly reduces the re-timing latency because it only re-times tasks that free up the wasted slack. To validate our claims we performed experiments on using 12 real benchmarks, the results demonstrate that ALI-EBAD out performs CA-TMES-Search and CA-TMES-Quick task schedulers in terms of energy-efficiency.

Non-Orthogonal Unicast and Broadcast Transmission via Joint Beamforming and LDM in Cellular Networks

Limited bandwidth resources and higher energy efficiency requirements motivate incorporating multicast and broadcast transmission into the next-generation cellular network architectures, particularly for multimedia streaming applications. Layered division multiplexing (LDM), a form of non-orthogonal multiple access (NOMA), can potentially improve unicast throughput and broadcast coverage with respect to traditional orthogonal frequency division multiplexing (FDM) or time division multiplexing (TDM), by simultaneously using the same frequency and time resources for multiple unicast or broadcast transmissions. In this paper, the performance of LDM-based unicast and broadcast transmission in a cellular network is studied by assuming a single frequency network (SFN) operation for the broadcast layer, while allowing arbitrarily clustered cooperation among the base stations (BSs) for the transmission of unicast data streams. Beamforming and power allocation between unicast and broadcast layers, the so-called injection level in the LDM literature, are optimized with the aim of minimizing the sum-power under constraints on the user-specific unicast rates and on the common broadcast rate. The effects of imperfect channel coding and imperfect channel state information (CSI) are also studied to gain insights into robust implementation in practical systems. The non-convex optimization problem is tackled by means of successive convex approximation (SCA) techniques. Performance upper bounds are also presented by means of the S-procedure followed by semidefinite relaxation (SDR). Finally, a dual decomposition-based solution is proposed to facilitate an efficient distributed implementation of LDM in each of the SCA subproblems, where the unicast beamforming vectors can be obtained locally by the cooperating BSs. Numerical results are presented, which show the tightness of the proposed bounds and hence the near-optimality of the proposed solutions.

QoS-Aware Multicast for Scalable Video Streaming in Software-Defined Networks

Scalable Video Coding (SVC) is a promising coding technique that enables flexible video transmission to support heterogeneous devices. However, the current best-effort delivery model characterized by the Internet cannot fully exploit the scalability feature of SVC because the network nodes are transparent for multimedia streaming applications. Software-defined networking (SDN) has emerged as an innovative network paradigm that decouples the control and forwarding planes while routing. This architecture with the OpenFlow protocol enables network operators to obtain per-flow QoS control in a more scalable, flexible, and finely granular manner compared to the conventional network architecture. Inspired by these facts, this paper designs a comprehensive QoS-aware multicast scheme for scalable video streaming over SDN networks. First, we present the designed multimedia streaming multicast system architecture and formulate the QoS-aware multicast routing problem as a non-linear programming model. We then propose a fundamental tree construction algorithm that decomposes the video streaming requests into subrequests and serves them in a bottom-up manner. Furthermore, we design a layer switching strategy based on the video distortion model to mitigate network-induced video quality distortion. Finally, the experimental results are presented to validate the performance of our methods in terms of scalability, network utility, video playout quality, and playback interruption ratio.

Interoperability and Data Storage in Internet of Multimedia Things: Investigating Current Trends, Research Challenges and Future Directions

Internet of Things (IoT) has widely been accepted as a promising paradigm for connecting a large number of resource-constrained miniature sensor nodes that have the ability to sense the deployed environment. They have found their applications in various aspect of our daily lives. However, these nodes are mostly restricted to sense only the scalar data. Nowadays, multimedia sensor nodes are gaining significant attention due to their ability to collect scalar as well as multi-dimensional data. These nodes are considered as the foundation of Internet of Multimedia Things (IoMT) and are shaping the perception of IoT. Multimedia data have stringent requirements in terms of reliability, latency, storage, bandwidth, and Quality of Service (QoS). To provide seamless and interoperable communication in IoMT, the underlying protocol stacks need to fulfill these stringent requirements. However, the heterogeneous nature of multimedia sensors makes interoperability a challenging task to fulfill. To understand the challenges faced by seamless and interoperable communication in IoMT, we provide a comprehensive review of the existing protocol stacks of IoMT and analyze their feasibility for multimedia streaming applications. Data storage of multimedia applications is another area that requires immediate attention of the research community. For this purpose, we study cloud as an entity to facilitate multimedia applications of IoMT. The instances of multimedia cloud are analyzed and a number of shortcomings are identified that pave the way for edge computing in IoMT. Finally, we present a case study that shows the significance of our work. The case study portrays an in-home patient monitoring system with an interoperable communication among the connected multimedia streaming devices at home, and healthcare practitioners at hospital. The case study also highlights the importance of uninterrupted data storage and retrieval at the network edge and multimedia sensor nodes.

Autonomous framework simulation tools for real-time multimedia streaming over wireless ad-hoc networks

Real-time video communication has become one of the most significant applications extensively used by homogeneous/heterogeneous wireless network technologies, such as Wi-Fi, the Internet of things, the wireless sensor network (WSN), 5G, etc. This leads to enhanced deployment of multimedia streaming applications over wireless network technologies. In order to accomplish the optimal performance of real-time multimedia streaming applications over the homogeneous/heterogeneous wireless network, it is therefore necessary to develop a simulation tool-set that effectively measures the quality of service (QoS) for different multimedia streaming applications over transport layer protocols. This paper proposes an autonomous simulation tool (AST) that is entirely independent from the source code of transport layer protocols. Furthermore, the AST is integrated into NS-2 to evaluate the QoS of real-time video streaming over numerous transport layer protocols and it uses new QoS measurement tools to test the video delivery quality based on I-frames to speeds up the assessment of multimedia streaming quality and ensure high accuracy of performance metrics. The simulation results show that using the AST to simulate real-time multimedia stream results in between 13% and 36% higher delivery ratio and 150–250% less cumulative jitter delay compared with using baseline simulation tools. Also, the AST guarantees an optimal QoS performance measurements in terms of the peak signal-to-noise Ratio and visual quality of the received video.

A Buffer-Aware QoS Streaming Approach for SDN-Enabled 5G Vehicular Networks

With the progress of network technology in recent years, multimedia streaming applications have become increasingly popular. However, it is difficult to achieve quality of service and efficiency for multimedia streaming over vehicular networks because of the high mobility feature. Over the existing network architecture, it is difficult to immediately analyze the status of the entire network, and then establish the rules of allocation and management. However, the novel network architecture, software-defined networking, offers other options for making network management more efficient, especially for the 5G network environment. Hence, a buffer-aware streaming approach is proposed to allow users to play multimedia streaming over vehicular 5G networks, in the case of handover between different eNodeBs, to achieve minimum delay and have better quality of service. According to the user's mobility information, the status of the player buffer, and the current strength of the network signal, the proposed approach can provide the transmission strategy of multimedia streaming to the SDN controller. Finally, the experimental results proved that the proposed approach is able to not only adjust the priority of streaming content segments with the buffer and mobility status of user equipment to effectively retain overall streaming services quality, but also avoid the delay of streaming content transmission for 5G vehicular networks.

Simulated performance of TCP, SCTP, DCCP and UDP protocols over 4G network

Forth Generation (4G) network support for wide geographical locations proves its use as a more advanced wireless technology. The Long-Term Evolution (LTE) is a 4G mobile communications standard set by International Communication Union (ITU), specifically ITU Radio communication Sector (ITU-R). At the moment, video traffic and telecommunications grow under the expansion of LTE, which is considered as the actual motivating access technology of 4G network. Throughout the deployment of LTE, various transport protocols are advised and broadly experimented, for instance, TCP, SCTP, DCCP and UDP that may execute differently on 4G networks subject to the network scenarios and parameter settings. Even though the deployment of LTE is swiftly enhanced, there is a lack of performance evaluation of its protocols. Hence, a widespread scrutiny is required for the evaluation of the operation of numerous protocols for high-end applications such as multimedia. Adopting these applications with flexible quality of service constraints with improved usage of resources is a challenging task. In this paper, the output results of different transport protocols for multimedia streaming applications, e.g., video, through extensive simulations are analyzed. The performance of an MPEG-4 video streaming is evaluated using NS-3. The performance metrics used are delay, jitter, throughput, and packet loss. These metrics are evaluated at the base station via TCP, SCTP, DCCP and UDP protocols over the 4G-LTE technology. The obtained results show that the DCCP performs the best in throughput improvement with the minimization of delay and jitter as compared to UDP, TCP, and SCTP.

Packet Delay Distribution Model for Investigating Delay of Network Speech Recognition

Unlike multimedia streaming applications that require a smooth playback at the client, application of network speech recognition (NSR) that recognizes speech signal in a sentence-by-sentence manner might tolerate an acceptable delay. The acceptable delay is a user-defined time in which the entire sentence data should be received by the server. We proposed a calculation method to investigate the acceptable delay of network speech recognition that employs a speech segmenter to send speech signal sentence-by-sentence over TCP channel to the server. The calculation multiplies the mean packet delay of TCP flow at steady-state with the number of created packets. For validation we implemented a MATLAB program and solved it using 2500 Indonesian speech sentences. The results were then compared with the results of our previous model that used a transient analysis method. It was found that this calculation method is not appropriate due to the transient behavior of the streaming sentences.

Media-Friendly and TCP-Friendly Rate Control Protocols for Multimedia Streaming

This paper describes a design framework for TCP-friendly and media-friendly rate control algorithms for multimedia streaming applications. The idea of this framework is to start from a TCP-friendly Rate Control (TFRC) transmission rate and then alter this transmission rate so that it tracks the media characteristics of the stream (e.g., bitrate) or other application characteristics like the client buffer fill level. In this way, the media-friendly property of the algorithm is achieved. We give three rules that guide how the TFRC throughput should track the evolution of the stream’s media characteristics and remain TCP-friendly in the long term. We also present, as proof of concept, four simple media-friendly and TCP-friendly congestion control algorithms built using the aforementioned framework. These congestion control algorithms are better suited for multimedia streaming applications than traditional TCP congestion control or smooth congestion control algorithms like TFRC. We have performed evaluations of two of the four proposed media-friendly and TCP-friendly congestion control algorithms under various network conditions and validated that they represent viable transport solutions, better than TFRC, for variable bitrate video streams. More specifically, our two media-friendly and TCP-friendly congestion control algorithms maintained a TCP-friendly throughput in the long term in all experiments and avoided an empty buffer at the client side in situations when TFRC could not achieve this.

Evaluating QoS parameters for video streaming transmission in mobile ad hoc networks

Recently, multimedia streaming applications are deployed anywhere (homes, campuses, etc.). Delivering these applications over mobile ad hoc networks is a challenging task: first, mobile ad hoc networks are infrastructure less, with dynamic topology, and limited resources. Secondly these applications have strict quality of service (QoS) requirements given with delay, jitter, bandwidth, loss rate, etc. Several architectures have been developed. However, an important part like QoS in routing level is still omitted. In this paper, many different coding techniques for video streaming are reviewed and two routing protocols (reactive and proactive) are used to evaluate which of them can improve QoS for multimedia streaming applications. Results show that reactive protocol performs better than proactive protocol in terms of throughput and network load in high mobility, but roles are reversed in terms of loss rate and network load for large-scale networks. PNSR and Jitter are influenced by nodes numbers independently of used protocol.

A First Design Approach to a Multimedia SDK Based on a Hybrid P2P Architecture

The Internet multimedia streaming increased proportional to the number of streaming users and from 2005 peer-to-peer media streaming received a substantial amount of research attention and was applied for both live and on-demand video streaming. This technique succeeded to provide a large number of multimedia streams while consuming less bandwidth than in the case of a client-server architecture. Multimedia streaming is a complex subject, it widens over various computer science fields as the networking area, multimedia compression area and the security area. Due to the increasing need of multimedia streaming applications and the need for continuous communication with harsh constraints such as real-time communication, low bandwidth and content security, the need for a flexible and extensible tool is justified, and the main purpose of such a tool is to facilitate the development of applications such as Goober (9), IConf (10), Ekiga (11) or Skype (12). The responsibilities of such SDK are to capture efficiently multimedia information from a web camera and/or a microphone and send them to its peer. The proposed SDK was built on the .NET Framework 4.5 based on a hybrid peer-to-peer architecture. The SDK can be integrated on multiple .NET platforms such .NET Framework 4.5, Silverlight, and Windows Phone 8, and due to its flexibility it can be used by desktop clients, web clients and mobile clients. From a communication perspective, the SDK starts several independent services which capture incoming data, and uses dynamic proxy objects to send data to its peers, services which assure the necessary degree of parallelism needed to have a responsive application with real-time communication.

Intelligent Bandwidth Estimation for Variable Bit Rate Traffic

A novel neuro-evolutionary Cartesian genetic programming-based frame size estimator for multimedia streaming applications has been proposed in this letter. The frame size obtained from the proposed estimator is used to calculate and allocate the bandwidth required for frame transmission. Bandwidth calculation and allocation is done via various probabilistic and linear regression methods. To obtain conclusive results regarding the feasibility of the proposed system, different test case scenarios have been exploited. The bandwidth allocation efficiency for the technique has been compared with previously proposed methods to evaluate its effectiveness in precise bandwidth utilization. Compared to other contemporary techniques, our technique gives approximately 98% efficient frame size prediction and bandwidth allocation.

Investigating a Mobility-Aware QoS Model for Multimedia Streaming Rate Adaptation

Supporting high quality multimedia streaming on wireless devices poses several challenges compared to wired networks due to the high variance in network performance encountered in the mobile environment. Although rate adaptation is commonly used in multimedia applications to compensate for fluctuations in network performance, it is a reactive mechanism which is not aware of the frequently changing connectivity that may occur on mobile devices. This paper proposed a performance evaluation model for multimedia streaming applications that is aware of user mobility and network performance. We presented an example of mathematical solution to the model and demonstrated the functionality using common mobility and connectivity examples that may be found in an urban environment. The proposed model is evaluated based on this functionality and how it may be used to enhance application performance.

Mobile multimedia streaming techniques: QoE and energy saving perspective

Multimedia streaming to mobile devices is challenging for two reasons. First, the way content is delivered to a client must ensure that the user does not experience a long initial playback delay or a distorted playback in the middle of a streaming session. Second, multimedia streaming applications are among the most energy hungry applications in smartphones. The energy consumption mostly depends on the delivery techniques and on the power management techniques of wireless access technologies (Wi-Fi, 3G, and 4G). In order to provide insights on what kind of streaming techniques exist, how they work on different mobile platforms, their efforts in providing smooth quality of experience, and their impact on energy consumption of mobile phones, we did a large set of active measurements with several smartphones having both Wi-Fi and cellular network access. Our analysis reveals five different techniques to deliver the content to the video players. The selection of a technique depends on the mobile platform, device, player, quality, and service. The results from our traffic and power measurements allow us to conclude that none of the identified techniques is optimal because they take none of the following facts into account: access technology used, user behavior, and user preferences concerning data waste. We point out the technique with the best playback buffer configuration, which provides the most attractive trade-offs in particular situations.

Dynamic Backlight Scaling Optimization: A Cloud-Based Energy-Saving Service for Mobile Streaming Applications

With the increasing variety of mobile applications, reducing the energy consumption of mobile devices is a major challenge in sustaining multimedia streaming applications. This paper explores how to minimize the energy consumption of the backlight when displaying a video stream without adversely impacting the user's visual experience. First, we model the problem as a dynamic backlight scaling optimization problem. Then, we propose algorithms to solve the fundamental problem and prove the optimality in terms of energy savings. Finally, based on the algorithms, we present a cloud-based energy-saving service. We have also developed a prototype implementation integrated with existing video streaming applications to validate the practicability of the approach. The results of experiments conducted to evaluate the efficacy of the proposed approach are very encouraging and show energy savings of 15-49 percent on commercial mobile devices.

Traffic Pattern-Based Content Leakage Detection for Trusted Content Delivery Networks

Due to the increasing popularity of multimedia streaming applications and services in recent years, the issue of trusted video delivery to prevent undesirable content-leakage has, indeed, become critical. While preserving user privacy, conventional systems have addressed this issue by proposing methods based on the observation of streamed traffic throughout the network. These conventional systems maintain a high detection accuracy while coping with some of the traffic variation in the network (e.g., network delay and packet loss), however, their detection performance substantially degrades owing to the significant variation of video lengths. In this paper, we focus on overcoming this issue by proposing a novel content-leakage detection scheme that is robust to the variation of the video length. By comparing videos of different lengths, we determine a relation between the length of videos to be compared and the similarity between the compared videos. Therefore, we enhance the detection performance of the proposed scheme even in an environment subjected to variation in length of video. Through a testbed experiment, the effectiveness of our proposed scheme is evaluated in terms of variation of video length, delay variation, and packet loss.