The explosion of enhanced applications such as live video streaming, video gaming and Virtual Reality calls for efforts to optimize transport protocols to manage the increasing amount of data traffic on future 5G networks. Through bandwidth aggregation over multiple paths, the Multi-Path Transmission Control Protocol (MPTCP) can enhance the performance of network applications. MPTCP can split a large multimedia flow into subflows and apply a congestion control mechanism on each subflow. Segment Routing (SR), a promising source routing approach, has emerged to provide advanced packet forwarding over 5G networks. In this paper, we explore the utilization of MPTCP and SR in SDN-based networks to improve network resources utilization and end- user's QoE for delivering multimedia services over 5G networks. We propose a novel QoE-aware, SDN- based MPTCP/SR approach for service delivery. In order to demonstrate the feasibility of our approach, we implemented an intelligent QoE- centric Multipath Routing Algorithm (QoMRA) on an SDN source routing platform using Mininet and POX controller. We carried out experiments on Dynamic Adaptive video Steaming over HTTP (DASH) applications over various network conditions. The preliminary results show that, our QoE-aware SDN- based MPTCP/SR scheme performs better compared to the conventional TCP approach in terms of throughput, link utilization and the end-user's QoE.

Video resolution changes in an HTTP adaptive streaming session may negatively affect the viewer's quality of experience. Our goal is, through encoding, to make such resolution changes less noticeable for the viewers. This can be achieved by taking video complexity features into account during the encoding process. In this paper, we compare Constant Bitrate (CBR) versus Constrained Constant Rate Factor (CRF) coding approaches and their effects on the noticeability of video resolution changes. To this end, we conducted a dedicated subjective study with 20 subjects in a quasi -lab environment. Our results suggest that choices for fixed-bitrate encoding have to be improved by deeper analysis of video complexity and resolution change patterns.

Quality of Experience (QoE) assessment of multimedia services is a challenging task and an understanding of how the user perceives quality at the physiological level would facilitate this. Physiological signals, such as the electroencephalogram (EEG), have shown promise in revealing the subject's emotion or attention in quality assessment and the correlation of this with media service quality. This paper investigated the relationships between changes in EEG features and subjective quality test scores (i.e. MOS) for High Dynamic Range (HDR) images viewed with a mobile device. Results show that changes in the gamma and beta bands correlated negatively with MOS, whereas positive correlations were observed in the alpha band. Coupling between activities in the delta and beta bands (i.e. positive correlation between power in the fast beta and slow delta frequency bands) is related to anxiety and dissatisfaction. Thus, the results suggest that increases in the degree of coupling are associated with decreases in HDR quality. This also suggests that in the HDR image QoE assessment, human emotions play a significant role. Potentially, these findings may be exploited in objective QoE perception modelling.

HTTP adaptive streaming (HAS) has become the de-facto standard for video streaming to ensure continuous multimedia service delivery under irregularly changing network conditions. Many studies already investigated the detrimental impact of various playback characteristics on the Quality of Experience of end users, such as initial loading, stalling or quality variations. However, dedicated studies tackling the impact of resolution adaptation are still missing. This paper presents the results of an immersive audiovisual quality assessment test comprising 84 test sequences from four different video content types, emulated with an HAS adaptation mechanism. We employed a novel approach based on systematic creation of adaptivity conditions which were assigned to source sequences based on their spatio-temporal characteristics. Our experiment investigates the resolution switch effect with respect to the degradations in MOS for certain adaptation patterns. We further demonstrate that the content type and resolution change patterns have a significant impact on the perception of resolution changes. These findings will help develop better QoE models and adaptation mechanisms for HAS systems in the future.

The development of HDR imaging is seen as an important step towards improving the visual quality of experience (QoE) of the end user in many applications. In practice, Tone-mapping operators (TMOs) provide a useful means for converting a high dynamic range (HDR) image to a low dynamic range (LDR) image in order to achieve better visualization on standard displays. Although mobile devices are becoming popular, the techniques for displaying the content of HDR images on the screens of such devices are still in the early stages. While several studies have been conducted to evaluate TMOs on conventional displays, few studies have been carried out to date to evaluate TMOs on small screen displays, such as those used in mobile devices. In this paper we evaluate, using subjective and objective methods, the most popular Tone-mapping-operators in different mobile displays and resolutions under normal viewing conditions for the end-user. Preliminary results show that small screen displays (SSDs) have an impact on the performance of TMOs compared to computer displays. In general, the larger the mobile resolution, the better the subjective results. We also found clear differences between SSDs and LDRs performances. The best TMO for mobile displays is iCAM06 and for computer displays it is Photographic Reproduction.

Dynamic Adaptive Streaming over HTTP (DASH) or MPEG-DASH is a popular technique that allows video quality adaptation for high quality streaming over the Internet. However, with bandwidth fluctuations, DASH performs poorly due to annoying frequent number of stalls. Software Defined Networking (SDN) has emerged as an attractive technology which has found its way into datacentres. Since one of the main goals of the SDN architecture is to make the network programmable and accelerate network innovation by utilizing its control plane, this paper has used the SDN control plane to propose a video quality management scheme based on the traffic intensity under DASH. Experimental results obtained by using Mininet and OpenDaylight controller have shown that the proposed scheme can significantly reduce the number of frequently annoying stalls and their duration by at least 84% and 94%, respectively. This has been achieved by switching network flows from high to low congested network paths within an SDN architecture.

This paper proposes a QoE-driven LTE downlink scheduling algorithm which is able to maximise the number of users per cell while maintaining an acceptable QoE for VoIP applications. The QoE-driven scheduling algorithm was implemented in LTE-Sim and its performance was compared to common LTE scheduling algorithms. The main difference between QoE-driven scheduling and existing scheduling mechanisms was prioritisation of users for resource allocation by taking into account their QoE requirements for VoIP applications. Preliminary results have shown that QoE-driven LTE downlink scheduling improved the cell capacity by 75% compared to MLWDF and 250% compared to PF and EXP-PF.

A new reference-free, objective, video quality prediction model that takes into account video content type to predict the quality of streamed high efficiency video coding (HEVC) encoded video sequences is proposed. Research has shown that for the same encoder settings and network quality of service (NQoS), the video quality differs for different types of video content . This indicates that, in addition to encoder settings and NQoS, there may be other key parameters that impact video quality. In this work, we hypothesized that video content type is one of the key parameters that may impact the quality of streamed videos. Based on this assertion, temporal information is extracted from the motion vector (MV) information inherent in the encoded video bitstreams and spatial information is extracted from the quantisation parameter (QP) and the number of bits (Bits) of coded intra (I) and predictive (P) frames to develop a metric that quantifies the content type of different video sequences . The content type metric is subsequently used together with encoding QP setting and network packet loss rate (PLR) to develop a reference -free objective video quality prediction model for streamed HEVC encoded video sequences. This model has an accuracy of 92% when the model predicted values of sequences not used in model derivation are compared with mean opinion score (MOS) obtained through subjective method.

The paper presents novel content-aware adaptation schemes for QoE optimized HTTP-based adaptive video streaming services. In recent years, content type has been investigated and proofed to have a deep influence on user's acceptable quality of experience (QoE). However, most of the existing HTTP-based adaptive video strategies only conduct bitrate switching according to network resources without a consideration of video content types. This may cause a non-optimal network resource allocation which may not achieve the optimized QoE for delivered video streaming services. In this paper, we proposed two content-aware adaptation schemes, named as Content-aware Probe and Adapt (C-PANDA) and Content-aware Dynamic Programming (C-DP), which are able to decide the video bitrate for the next video segment, based on not only the available bandwidth, the existing buffer capacity, but also the video content type. We implemented both content-aware adaptation algorithms in player prototypes based on libdash 3.0 library. Preliminary results show that content-aware adaptation schemes achieve better QoE when compared with conventional PANDA scheme. The proposed schemes are able to optimize QoE according to contents. They can achieve higher than acceptable QoE for most attractive contents such as those with high motion intensity to improve user experience for the most interesting scenes; select same video representations for segments belonging to a scene to avoid quality oscillation; and reserve buffer resource during low motion scenes to avoid stalling for the following high motion scenes. The work will be helpful in designing QoE-aware DASH schemes in the future.

This paper deals with the performance evaluation of deploying an IMS-based media plane interoperability framework. The 3GPP standards describe two possible operating modes based on either a reactive or a proactive approach. We show that both approaches entail some advantages and drawbacks in terms of signaling overhead and call setup times at different ratios of incompatible calls. A prototype implementation of the required elements was carried out in order to use experimental delay contributions. Additionally, the evaluation focuses on the impact of deploying the interoperability solutions over current UMTS and LTE radio access networks.

This paper analyzes the scenario where multiple users are trying to transmit the same or related content to a common destination through a shared radio access network. A specific aspect of this study is that all users work for a common objective, as it would be the transmission of critical emergency-related media contents to a central operator. We analyze the performance achievable using an LTE cell, with different client-driven autonomous adaptation strategies that take both the network performance and the battery status as input parameters. Remarkable performance results are achieved in terms of number of acceptable uplink video streams in different simulation scenarios. The work would help for designing context-and network-aware adaptation schemes for emergency mobile applications.

With the all IP based Next Generation Networks being deployed around the world, the use of real-time multimedia service applications is being extended from normal daily communications to emergency situations. However, currently different emergency providers utilise differing networks and different technologies. As such, conversations could be terminated at the setup phase or data could be transmitted in plaintext should incompatibility issues exit between terminals. To this end, a novel security gateway that can provide the necessary security support for incompatible terminals was proposed, developed and implemented to ensure the successful establishment of secure real-time multimedia conversations. A series of experiments were conducted to evaluate the security gateway through the use 40 Boghe softphone acting as the terminals. The experimental results demonstrate that the best performance of the prototype was achieved by utilising a multithreading and multi-buffering technique, with an average of 582 microseconds processing overhead. Based upon the ITU-Ts 150 milliseconds one way delay recommendation for voice communications, it is envisaged that such a marginal overhead will not be noticed by users in practice.

IP Multimedia Subsystem (IMS) offers a framework which enables the provisioning of multimedia services with Quality of Service (QoS) and mobility support across heterogeneous networks. The aim of this paper is twofold. First, to present a new fuzzy logic based Sender Bitrate (SBR) adaptation scheme at pre-encoding stage that is Quality of Experience (QoE) driven for video applications. The scheme was tested and evaluated in the NS2 based simulation access networks of third generation Universal Mobile Telecommunication System (UMTS) networks. Second, to demonstrate the implementation of the proposed adaptation scheme in our developed open Android-based IMS test bed. The test bed was developed to fully understand and manipulate the effects of network conditions on perceptual quality. The SBR adaptive scheme is evaluated in terms of the Mean Opinion Score (MOS). Extensive simulation and test bed results demonstrate the effectiveness of the proposed adaptation scheme especially at UMTS bottleneck access networks where perceived video quality is most affected. The proposed scheme was responsive to available network bandwidth and congestion and adapted the SBR accordingly maintaining acceptable quality in terms of the MOS. The proposed scheme has applications in network planning and content provisioning for network/service providers.

The Next Generation Networks (NGNs) were proposed to provide ubiquitous access to multimedia services over the Internet. With the best effort nature of the Internet, the quality of real time multimedia services such as voice and video can rapidly degrade due to network impairments. In this paper, we propose a novel feedback-free Early VoIP Quality Adaptation Scheme (EVQAS) which can adapt voice send bit rate (e.g. Adaptive Multi-Rate (AMR) codec mode) automatically according to the buildup of queuing delay in the network. This differs from existing adaptation methods which adapt send bit rate according to either packet loss information or Perceived Quality of Service (PQoS) metrics (e.g. Mean Opinion Score (MOS)) derived from packet loss and delay. The proposed scheme can avoid congested packet loss which has an obvious impact on perceived voice quality. We implemented this adaptation mechanism in open Android mobile and tested in Open IMS Core testbed. Preliminary results show that the early adaptation mechanism can effectively release congestion before packet loss occurs. This has increased overall perceived voice quality when compared with existing adaptation method. This lightweight feedback-free early adaptation mechanism can be easily applied to other mobile devices to improve VoIP quality in current NGN VoIP applications.

Abstract A novel one‐to‐all broadcasting scheme is proposed for arbitrary static interconnection networks with bursty background traffic having head‐of‐line priorities, complete buffer sharing and repetitive service blocking with random destination. The scheme facilitates inter‐process communication amongst parallel computers and grid computing networks as well as asynchronous transfer mode space division architectures and it is applicable to a single‐port mode of message passing communication. The scheme utilizes a queue‐by‐queue decomposition algorithm for arbitrary open queueing network models, based on the principle of maximum entropy, in conjunction with an information theoretic decomposition criterion and graph theoretic concepts. Evidence based on empirical studies indicates the suitability of the scheme for achieving an optimal one‐to‐all broadcasting cost. Copyright © 2004 John Wiley & Sons, Ltd.