1,699 research outputs found

    Skew detection and compensation for Internet audio applications

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    Long lived audio streams, such as music broadcasts, and small differences in clock rates lead to buffer underflow or overflow events in receiving applications that manifest themselves as audible interruptions. We present a low complexity algorithm for detecting clock skew in network audio applications that function with local clocks and in the absence of a synchronization mechanism. A companion algorithm to perform skew compensation is also presented. The compensation algorithm utilises the temporal redundancy inherent in audio streams to make inaudible playout adjustments. Both algorithms have been implemented in a simulator and in a network audio application. They perform effectively over the range of observed clock rate differences and beyond

    Evaluating and improving firewalls for ip-telephony environments

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    Firewalls are a well established security mechanism for providing access control and auditing at the borders between different administrative network domains. Their basic architecture, techniques and operation modes did not change fundamentally during the last years. On the other side new challenges emerge rapidly when new innovative application domains have to be supported. IP-Telephony applications are considered to have a huge economic potential in the near future. For their widespread acceptance and thereby their economic success they must cope with established security policies. Existing firewalls face immense problems here, if they - as it still happens quite often - try to handle the new challenges in a way they did with "traditional applications". As we will show in this paper, IP-Telephony applications differ from those in many aspects, which makes such an approach quite inadequate. After identifying and characterizing the problems we therefore describe and evaluate a more appropriate approach. The feasibility of our architecture will be shown. It forms the basis of a prototype implementation, that we are currently working on

    A user perspective of quality of service in m-commerce

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    This is the post-print version of the Article. The official published version can be accessed from the link below - Copyright @ 2004 Springer VerlagIn an m-commerce setting, the underlying communication system will have to provide a Quality of Service (QoS) in the presence of two competing factors—network bandwidth and, as the pressure to add value to the business-to-consumer (B2C) shopping experience by integrating multimedia applications grows, increasing data sizes. In this paper, developments in the area of QoS-dependent multimedia perceptual quality are reviewed and are integrated with recent work focusing on QoS for e-commerce. Based on previously identified user perceptual tolerance to varying multimedia QoS, we show that enhancing the m-commerce B2C user experience with multimedia, far from being an idealised scenario, is in fact feasible if perceptual considerations are employed

    Congestion Control for Network-Aware Telehaptic Communication

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    Telehaptic applications involve delay-sensitive multimedia communication between remote locations with distinct Quality of Service (QoS) requirements for different media components. These QoS constraints pose a variety of challenges, especially when the communication occurs over a shared network, with unknown and time-varying cross-traffic. In this work, we propose a transport layer congestion control protocol for telehaptic applications operating over shared networks, termed as dynamic packetization module (DPM). DPM is a lossless, network-aware protocol which tunes the telehaptic packetization rate based on the level of congestion in the network. To monitor the network congestion, we devise a novel network feedback module, which communicates the end-to-end delays encountered by the telehaptic packets to the respective transmitters with negligible overhead. Via extensive simulations, we show that DPM meets the QoS requirements of telehaptic applications over a wide range of network cross-traffic conditions. We also report qualitative results of a real-time telepottery experiment with several human subjects, which reveal that DPM preserves the quality of telehaptic activity even under heavily congested network scenarios. Finally, we compare the performance of DPM with several previously proposed telehaptic communication protocols and demonstrate that DPM outperforms these protocols.Comment: 25 pages, 19 figure

    A Methodology for Characterizing Real-Time Multimedia Quality of Service in Limited Bandwidth Network

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    This paper presents how to characterize the quality of multimedia which consists of audio and video that are transmitted in real-time communication through the Internet with limited bandwidth. We developed a methodology of characterizing the multimedia Quality-of-Service (QoS) by measuring network parameters (i.e., bandwidth capacity, packet loss rate (PLR), and end-to-end delay) of testbed network and simulating the audio-video delivery according to the measured network parameters. The analysis of network parameters was aimed to describe the network characteristics. Multimedia QoS was characterized by conducting a simulation using data which was collected from the previous network characterization. A simulation network model was built using OMNet++ representing a delivery of audio-video in real-time while a background traffic was generated to represent a real condition of the network. Apllying the methodology in a network testbed in Indonesia’s rural area, the simulation results showed that audio-video could be delivered with accepted level of user satisfaction

    A network analysis on cloud gaming: Stadia, GeForce Now and PSNow

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    Cloud gaming is a new class of services that promises to revolutionize the videogame market. It allows the user to play a videogame with basic equipment while using a remote server for the actual execution. The multimedia content is streamed through the network from the server to the user. This service requires low latency and a large bandwidth to work properly with low response time and high-definition video. Three of the leading tech companies, (Google, Sony and NVIDIA) entered this market with their own products, and others, like Microsoft and Amazon, are planning to launch their own platforms in the near future. However, these companies released so far little information about their cloud gaming operation and how they utilize the network. In this work, we study these new cloud gaming services from the network point of view. We collect more than 200 packet traces under different application settings and network conditions for 3 cloud gaming services, namely Stadia from Google, GeForce Now from NVIDIA and PS Now from Sony. We analyze the employed protocols and the workload they impose on the network. We find that GeForce Now and Stadia use the RTP protocol to stream the multimedia content, with the latter relying on the standard WebRTC APIs. They result in bandwidth-hungry and consume up to 45 Mbit/s, depending on the network and video quality. PS Now instead uses only undocumented protocols and never exceeds 13 Mbit/s

    Distributed Rate Allocation Policies for Multi-Homed Video Streaming over Heterogeneous Access Networks

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    We consider the problem of rate allocation among multiple simultaneous video streams sharing multiple heterogeneous access networks. We develop and evaluate an analytical framework for optimal rate allocation based on observed available bit rate (ABR) and round-trip time (RTT) over each access network and video distortion-rate (DR) characteristics. The rate allocation is formulated as a convex optimization problem that minimizes the total expected distortion of all video streams. We present a distributed approximation of its solution and compare its performance against H-infinity optimal control and two heuristic schemes based on TCP-style additive-increase-multiplicative decrease (AIMD) principles. The various rate allocation schemes are evaluated in simulations of multiple high-definition (HD) video streams sharing multiple access networks. Our results demonstrate that, in comparison with heuristic AIMD-based schemes, both media-aware allocation and H-infinity optimal control benefit from proactive congestion avoidance and reduce the average packet loss rate from 45% to below 2%. Improvement in average received video quality ranges between 1.5 to 10.7 dB in PSNR for various background traffic loads and video playout deadlines. Media-aware allocation further exploits its knowledge of the video DR characteristics to achieve a more balanced video quality among all streams.Comment: 12 pages, 22 figure
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