Network Impairments Research Articles

Perceptual Evaluation Of Playout Buffer Algorithm For Enhancing Perceived Quality Of Voice Transmission Over Ip Network

Voice over Internet Protocol (VoIP) is a technology that allows you to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line. Voice over Internet Protocol (VoIP) has led human speech to a new level, where conversation across continents can be much cheaper & faster. However, as IP networks are not designed for real-time applications, the network impairments such as packet loss, jitter and delay have a severe impact on speech quality. The playout buffer at the receiver side is used to compensate jitter at a trade-off of delay and loss. We found the characteristics of delay and loss are dependent on IP network and sudden variable delay (spike) often performs both regular and irregular characteristics. Different playout buffer algorithms can have different impacts on the achievement speech quality. It is important to design a playout buffer algorithm which can help achieve an optimum speech quality. In this paper, we investigate to the understanding how network impairments and existing adaptive buffer algorithms affect the speech quality and further to design a modified buffer algorithm to obtain an optimized voice quality. We conduct experiments to existing algorithms and compared their performance under different network conditions with high and low network delay variations. Preliminary results show that the new algorithm can enhance the perceived speech quality in most network conditions and it is more efficient and suitable for real buffer mechanism.

SVC-Based HTTP Adaptive Streaming

HTTP adaptive streaming (HAS) is rapidly evolving into a key video delivery technology, supported by implementations from Microsoft, Apple, and Adobe, and actively pursued by standardization organizations. Using segments in multiple video qualities, distributed via an already available Hypertext Transfer Protocol (HTTP) delivery infrastructure, a HAS client is able to seamlessly adapt to the available bandwidth in the network. However, existing HAS solutions have a number of disadvantages such as the additional storage and bandwidth requirements, a large playout buffer to absorb network impairments, and a non-optimal quality selection under fluctuating network conditions. In this paper, we investigate the opportunity of combining HAS with scalable video coding. We show that this combination creates possibilities to reduce the client buffer, which implies improvements for live and interactive video, and reduces storage requirements, increases the cache hit-ratio for supporting content delivery network (CDN) nodes, and demonstrates more robust behavior in the HAS client, ultimately improving the quality of experience (QoE) for the viewer. © 2012 Alcatel-Lucent.

Subjective Video Quality Assessment of HTTP Adaptive Streaming Technologies

HTTP adaptive steaming (HAS) is becoming ubiquitous as a reliable method of delivering video content over the open Internet to a variety of devices from personal computers (PCs), to tablets, game consoles, and smartphones. HAS is able to adapt to both the available bandwidth and the display requirements by trading-off video quality. This paper describes two experiments to test end user subjective response to this varying quality. First, we tested three commercially available HAS products in our viewing room. This allowed us to control the introduction of network impairments and to record the mean opinion score (MOS). In a second experiment, we generated clips with impairments typical of HAS. These were downloaded and commented on by a group of young people. This provided insight into the response of users to different types of visual impairments and hence what steps can be taken to improve the end user experience. © 2012 Alcatel-Lucent.

Non-Intrusive Speech Quality with Different Time Scale

Speech quality evaluation is an extremely important problem in modern communication networks. Service providers always strive to achieve a certain Quality of Service (QoS) in order to ensure customer satisfaction. Modeling the speech quality becomes an urgent issue. In this project a computable model for different time scale speech quality evaluation, called E-Model is developed. The results indicate that subjects can monitor speech quality variations very accurately with a delay of approximately 1 second. Non-intrusive speech quality is measured at the receiver from a degraded signal using G.107 (E-model) which is a parameter based model and calculate MOS values with quality rating factor. The quality rating factor is calculated by network impairments (loudness rating) of a speech. The output from the model described here is a scalar quality rating value, R, which varies directly with the overall conversational quality. The key contribution of this paper is to explore the use of G-107 (E-model) based features for different time scale non-intrusive speech quality evaluation using time varying loudness of a speech for long stimuli. Sectional speech quality is obtained by E- model, which is called instantaneous quality of the section that will be constant for each section. Overall perceived quality can be calculated by using average of instantaneous speech quality.

Parametric assessment of video quality in content

IP-based packet-switched networks have become one of the main content distribution platforms for emerging multimedia services such as IPTV, thanks to the rapidly growing bandwidth and exclusive inter-networking and interactivity features of IP-based networks. Meanwhile, high quality video content services are becoming particularly popular within content delivery networks (CDN). During content distribution, packets of encoded video streams can be delayed, corrupted or dropped due to network impairments in packet-switched networks. This leads to perceptual quality degradations of the delivered content at receiver. Although network impairments are rare in commercial managed networks, any distortion caused by impairments can be highly detrimental to end users' experience. Consequently the ability to meet customer expectations on video quality has become a critical service differentiator. Quality of Experience (QoE) that was merely recognised as a value-added service of traditional content distribution services is now one of the fundamental requirements and challenges of providing high quality video services. In order to maintain a high level of user experience throughout the life-cycle of a video service, a service quality measurement and management service must be established.

Resilient Inter-Carrier Traffic Engineering for Internet Peering Interconnections

We present a novel resilient routing policy for controlling the routing across peering links between Internet carriers. Our policy is aimed at offering more dependability and better performance to the routing decision with respect to the current practice (e.g., hot-potato routing). Our work relies on a non-cooperative game framework, called Peering Equilibrium MultiPath (PEMP), that has been recently proposed. PEMP allows two carrier providers to coordinate a multipath route selection for critical flows across peering links, while preserving their respective interests and independence. In this paper, we propose a resilient PEMP execution policy accounting for the occurrence of potential impairments (traffic matrix variations, intra-AS and peering link failures) that may occur in both peering networks. We mathematically define how to produce robust equilibrium sets and describe how to appropriately react to unexpected network impairments that might take place. The results from extensive simulations show that, under a realistic failure scenario, our policy adaptively prevents from peering link congestions and excessive route deviations after failures.

Multi-interface Extension to a Scalable Video Streaming Architecture

Video service adaptation capabilities are essential for the efficient utilization of the network resources in heterogeneous multi-access environments and ensuring sufficient perceived service quality for the end users. Multihoming is seen as one important enabler for high-quality and reliable video streaming services and it is expected to be supported extensively in the future Internet. Already today, terminal devices are equipped with multiple network interfaces and able to use them simultaneously. In the future, mobile multimedia services such as video streaming can maximize the user experienced quality by utilizing the available network resources, concurrently. However, intelligent decision-making as well as dynamic mapping of traffic to network interfaces are required for optimal operation. In this paper, we introduce an overall architecture for adaptive video streaming exploiting the novel scalable video coding technology. We also propose a multi-interface streaming extension to the architecture to allow dynamic and concurrent usage of the available access networks in the video service delivery. We present a prototype implementation of the proposed multi-interface video streaming system and test its operation in the presence of network impairments in an experimental evaluation.

Network in a Box: Facilitating problem‐based learning through network emulation

AbstractProblem‐based learning (PBL) is a pedagogical method that challenges students to “learn how to learn” by working cooperatively in groups to develop solutions to real world problems. Network in a Box (NiB) is a unique, engineering‐oriented effort to employ PBL in core computer networks curriculum. Developed as an open‐ended exercise in PBL, NiB sets a learning environment in which student teams design and implement a network emulator using off‐the‐shelf hardware and open‐source software. The aims of this article are twofold. The first goal of this article is to catalogue our efforts at San Diego State University in implementing PBL in a computer networks engineering course. The second goal of this article is to introduce the NiB project as a viable example of PBL in the field of computer networks. This project not only arms students with a good understanding of network impairments but also provides the knowledge of low level packet processing, packet creation, packet injection into the network, meeting real time processing requirements of packet processing and packet filtering. By arming students with an understanding of network impairments, a working knowledge of low‐level packet processing, packet creation, packet injection, and their processing requirements, NiB serves as a model introduction of PBL into the field of computer networks engineering. In this article we disseminate the product specification and system design as an educational and research utility along with an assessment of this project as perceived by students over several semesters. © 2009 Wiley Periodicals, Inc. Comput Appl Eng Educ 19: 433–446, 2011

QoT-aware lightpath set-up in GMPLS-controlled WDM networks: A survey

GMPLS-controlled Wavelength Switched Optical Networks (WSON) often undergo upgrade in bit rate or in network element switching capability. In such a scenario many wavelength paths cannot be anymore considered pre-validated from the original network design phase. Thus, it is required to verify the quality of the optical path physical signal during path computation and signaling by considering the optical network impairments (e.g., attenuation, amplified spontaneous emission, etc.). In this paper, the general problem of impairment-aware path computation is presented and then solutions based on GMPLS extensions to be used during path set-up are considered. Specifically, an overview is presented where different approaches for implementing Quality of Transmission (QoT) estimation and QoT measurements in transparent networks are outlined. In addition, techniques for exploiting sparse regeneration in translucent networks are illustrated. Numerical results show that the proposed schemes are effective in finding or designating QoT-validated paths in only few set-up attempts when a fully transparent path is available. If the transparent path becomes unfeasible, the translucent approach is adopted showing how to optimally designate intermediate regenerators to satisfy the end-to-end QoT constraints.

Discrete quality assessment in IPTV content distribution networks

Maintaining the quality of videos in resource-intensive IPTV services is challenging due to the nature of packet-based content distribution networks (CDN). Network impairments are unpredictable and highly detrimental to the quality of video content. Quality of the end user experience (QoE) has become a critical service differentiator. An efficient real-time quality assessment service in distribution networks is the foundation of service quality monitoring and management. The perceptual impact of individual impairments varies significantly and is influenced by complex impact factors. Without differentiating the impact of quality violation events to the user experience, existing assessment methodologies based on network QoS such as packet loss rate cannot provide adequate supports for the IPTV service assessment. A discrete perceptual impact evaluation quality assessment (DEQA) framework is introduced in this paper. The proposed framework enables a real-time, non-intrusive assessment service by efficiently recognising and assessing individual quality violation events in the IPTV distribution network. The discrete perceptual impacts to a media session are aggregated for the overall user level quality evaluation. With its deployment scheme the DEQA framework also facilitates efficient network diagnosis and QoE management. To realise the key assessment function of the framework and investigate the proposed advanced packet inspection mechanism, we also introduce the dedicated evaluation testbed—the LA2 system. A subjective experiment with data analysis is also presented to demonstrate the development of perceptual impact assessment functions using analytical inference, the tools of the LA2 system, subjective user tests and statistical modelling.

Perceptual-Based Playout Mechanisms for Multi-Stream Voice over IP Networks

Packet loss and delay are the major network impairments for transporting real-time voice over IP networks. In the proposed system, multiple descriptions of the speech are used to take advantage of the packet path diversity. A new objective method is presented for predicting the perceived quality of multi-stream voice transmission. Also proposed is a multi-stream playout buffer algorithm, together with an adaptive parameter adjustment scheme, that maximizes the perceived speech quality via delay-loss trading. Experimental results showed that, compared to FEC-protected single-path transmission, the proposed multi-stream transmission scheme achieves significant reductions in delay and packet loss rates as well as improved speech quality.

Adaptive IPTV services based on a novel IP Multimedia Subsystem

Heterogeneous communication devices are emerging and changing the way of communication. Innovative multimedia applications are now accessible through these embedded systems. The 3GPP IP Multimedia Subsystem (IMS) provides a basic architecture framework for the Next Generation Network (NGN) supporting the convergence platform for service provisioning in heterogeneous networks. ETSI TISPAN standardization effort focuses on delivering IPTV services on such platform. Nevertheless, IPTV on IMS standardization suffers from a lack of efficient user-centric network management mechanisms as the end-user may consume IPTV service from different access networks, on different mobile devices, at anytime. User's Perceived Quality of Service (PQoS) or Quality of Experience (QoE) of IPTV service may also suffer from wireless access network impairments. This paper introduces new functionalities in IPTV over IMS architecture which optimize satisfaction of the end-user and resource utilization of the operator's networks. A context-sensitive User Profile (UP) model is used to deliver IPTV streams adapted to the user's environment. In order to optimize the operator network usage, the impact of spatiotemporal dynamics of the video content on the deduced perceptual quality is considered. A Multimedia Content Management System (MCMS) is proposed to perform dynamic cross-layer adaptation of the IPTV stream based on PQoS measurements at the end-user side.

Fast and adaptive impairment aware routing and wavelength assignment algorithm optimized by offline simulations

In all-optical networks, signals are transmitted through optical physical layer with no regeneration. Therefore, noise accumulation along lightpath can severely impair optical signal-to-noise ratio. Impairment aware routing and wavelength assignment algorithms (IA-RWA) can take into account these effects, improving the network performance. In this paper we describe a fast and high performance adaptive weight function to be used as the metric for the routing algorithm in optical networks constrained by physical impairments. The input information for this function are link availability, route length and two adjustable parameters. These two parameters provide information about the network impairments. An offline simulation must be run to adjust them prior to the online network operation. The main advantage of this approach is the use of simple network parameters during the routing process instead of a complex optical noise based formulation, which renders it a better performance in terms of both, time to find a route and blocking probability. In our simulations we considered three physical layer effects: ASE noise generation, Optical Amplifier gain and ASE saturation and OXC crosstalk. We performed a performance comparison between this new metric and other metrics previously described in the literature.

Could Skype be more satisfying? a QoE-centric study of the FEC mechanism in an internet-scale VoIP system

The phenomenal growth of Skype in recent years has surpassed all expectations. Much of the application's success is attributed to its FEC mechanism, which adds redundancy to voice streams to sustain audio quality under network impairments. Adopting the quality of experience approach (i.e., measuring the mean opinion scores), we examine how much redundancy Skype adds to its voice streams and systematically explore the optimal level of redundancy for different network and codec settings. This study reveals that Skype's FEC mechanism, not so surprisingly, falls in the ballpark, but there is surprisingly a significant margin for improvement to ensure consistent user satisfaction.

Influence of informational content of background noise on speech quality evaluation for VoIP application

With the rise of mobile communication systems, the background noise in the speaker's environment and its interaction with VoIP network impairment affects speech quality perception. This effect should be taken into account in nonintrusive models in order to improve accuracy of end user perception measurement. The aim of this study is to determine the impact of information contained in background noise (background speech, environmental sources) on speech quality perception. A subjective test on the speech quality perception in real network conditions has been done comparing the effect of stationary background noise mainly due to transmission equipment (electrical humming or blowing noises) with the effect of nonstationary environmental noise (public place, traffic noise, background conversation). Interactions between these different background noise condition and realistic network impairments (coders and packets loss) were also studied. The speech quality was evaluated through the Mean Opinion Score determined from an Absolute Category Rating method.

On the impact of active queue management on VoIP quality of service

In this paper, we assess the impact of active queue management (AQM) on the quality of service of voice over IP (VoIP). Some of the most representative AQM schemes are analyzed through extensive simulations; evaluating its effect on the perceived quality of voice calls. We investigate different network scenarios, changing network loads and scheme control parameters. Network impairments are related to user perception by means of well known algorithmic models, expressing user satisfaction in the MOS scale. Our main results shows that the use of active queue management schemes significantly improve the perceived quality of voice calls, offering adequate user satisfaction.

Code-Empowered Lightwave Networks

In this paper, an architecture for code-empowered optical CDMA (OCDMA) lightwave networks is presented. The architecture is based on reconfigurable optically transparent paths among users of the network to provide high-bandwidth optical connections on demand over small areas such as local area networks or access networks. The network operates on the transmission of incoherent OCDMA codes, each network station being equipped with an OCDMA encoder and decoder. The routing at a network node is based on the OCDMA code itself. The destination address, as well as the next node on the path, is given by the code as in a code-empowered network. A node consists of an OCDMA router built from parallel code converter routers that perform switching, routing, and code conversion. The latter enables a virtual code path for increased scalability. Commonly available delay lines enable the tunability of the encoder, decoder, and router for a reconfigurable and flexible network. Flexibility and granularity are also accentuated by OCDMA encoding. An OCDMA lightwave network can therefore respond to changes in traffic load, traffic conditions, failure, and other network impairments. We describe the possible architectures and the routing constraints of such OCDMA lightwave networks. We present a power analysis and focus on the performance issues of dynamic routing. The effect of coding, topology, load condition, and traffic demand is analyzed using simulations. The obtained results show that the flexibility of OCDMA and the large offered cardinality can be a solution to the needs of local area and access networks.

Metabolic–dopaminergic mapping of the 6-hydroxydopamine rat model for Parkinson’s disease

The unilateral 6-hydroxydopamine (6-OHDA) lesion rat model is a well-known acute model for Parkinson's disease (PD). Its validity has been supported by invasive histology, behavioral studies and electrophysiology. Here, we have characterized this model in vivo by multitracer imaging [glucose metabolism and dopamine transporter (DAT)] in relation to behavioral and histological parameters. Eighteen female adult Wistar rats (eight 6-OHDA-lesioned, ten controls) were investigated using multitracer [(18)F]-fluoro-2-deoxy-D: -glucose (FDG) and [(18)F]-FECT {2'-[(18)F]-fluoroethyl-(1R-2-exo-3-exe)-8-methyl-3-(4-chlorophenyl)-8-azabicyclo(3.2.1)-octane-2-carboxylate} small animal positron emission tomography (PET). Relative glucose metabolism and parametric DAT binding images were anatomically standardized to Paxinos space and analyzed on a voxel-basis using SPM2: , supplemented by a template-based predefined volumes-of-interest approach. Behavior was characterized by the limb-use asymmetry test; dopaminergic innervation was validated by in vitro tyrosine hydroxylase staining. In the 6-OHDA model, significant glucose hypometabolism is present in the ipsilateral sensory-motor cortex (-6.3%; p = 4 x 10(-6)). DAT binding was severely decreased in the ipsilateral caudate-putamen, nucleus accumbens and substantia nigra (all p < 5 x 10(-9)), as confirmed by the behavioral and histological outcomes. Correlation analysis revealed a positive relationship between the degree of DAT impairment and the change in glucose metabolism in the ipsilateral hippocampus (p = 3 x 10(-5)), while cerebellar glucose metabolism was inversely correlated to the level of DAT impairment (p < 3 x 10(-4)). In vivo cerebral mapping of 6-OHDA-lesioned rats using [(18)F]-FDG and [(18)F]-FECT small animal PET shows molecular-functional correspondence to the cortico-subcortical network impairments observed in PD patients. This provides a further molecular validation supporting the validity of the 6-OHDA lesion model to mimic multiple aspects of human PD.

Adaptive Voice Smoothing with Optimal E-Model Method for VoIP Services

VoIP, one of emerging technologies, offers high quality of real time voice services over IP-based broadband networks; however, the quality of voice would easily be degraded by IP network impairments such as delay, jitter and packet loss, hereon initiate the presence of new technologies to help solve out the problems. Among those, playout buffer at the receiving end can compensate for the jitter effects by its function of tradeoff between delay and loss. Adaptive smoothing algorithms are capable of the dynamical adjustment of smoothing size by introducing a variable delay based on the use of the network parameters so as to avoid the quality decay problem. This paper introduces an efficient and feasible perceived quality method for buffer optimization to achieve the best voice quality. This work formulates an online loss model which incorporates buffer sizes and applies the ITU-T E-model approach to optimize the delay-loss problem. Distinct from other optimal smoothers, the proposed optimal smoother can be applied for most codecs and carries the lowest complexity. Since the adaptive smoothing scheme introduces variable playback delays, the buffer re-synchronization between the capture and the playback becomes essential. This work also presents a buffer re-synchronization algorithm based on silence skipping to prevent unacceptable increase in the buffer preloading delay and even buffer overflow. Simulation experiments validate that the proposed adaptive smoother achieves significant improvement in the voice quality.

The effects of audio and video correlation and lip synchronization

This paper investigates the interaction effect of audio and video, and studies lip synchronization (lip sync). The study shows a comprehensive evaluation of achievable audio and video quality undertaken based upon different sets of impairments between audio and video, prior to transmission. The tests have been conducted on two different task scenarios, i.e. passive communication and interactive communication (person to person). The research concentrates on quantifying the effects of network impairments (packet loss) on perceived audio and video quality, as well as finding the correlations between audio and video in multimedia applications. The results presented in this paper show the strong interaction dependency between audio and video. It was justified that video has a unique benefit on multimedia quality for its psychological effects. The findings also concluded that the sensory interactions, and the attention given to a particular aspect of performance, are clearly content‐dependent.