HDAX: Historical symbolic modelling of delay time series in a communications network

There are certain performance parameters like packet delay, delay variation (jitter) and loss, which are decision factors for online quality of service (QoS) traffic routing. Although considerable efforts have been placed on the Internet to assure QoS, the dominant TCP/IP - like the best-effort communications policy - does not provide sufficient guarantee without abrupt change in the protocols. Estimation and forecasting end-to-end delay and its variations are essential tasks in network routing management for detecting anomalies. A large amount of research has been done to provide foreknowledge of network anomalies by characterizing and forecasting delay with numerical forecasting methods. However, the methods are time consuming and not efficient for real-time application when dealing with large online datasets. Application is more difficult when the data is missing or not available during online forecasting. Moreover, the time cost in statistical methods for trivial forecasting accuracy is prohibitive. Consequently, many researchers suggest a transition from computing with numbers to the manipulation of perceptions in the form of fuzzy linguistic variables. The current work addresses the issue of defining a delay approximation model for packet switching in communications networks. In particular, we focus on decision-making for smart routing management, which is based on the knowledge provided by data mining (informed) agents. We propose a historical symbolic delay approximation model (HDAX) for delay forecasting. Preliminary experiments with the model show good accuracy in forecasting the delay time-series as well as a reduction in the time cost of the forecasting method. HDAX compares favourably with the competing Autoregressive Moving Average (ARMA) algorithm in terms of execution time and accuracy. © 2009, Australian Computer Society, Inc

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

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

Performance Evaluation of IPTV over WiMAX Networks Under Different Terrain Environments

Deployment Video on Demand (VoD) over the next generation (WiMAX) has become one of the intense interest subjects in the research these days, and is expected to be the main revenue generators in the near future. In this paper, the performance of Quilty of Service of video streaming (IPTV) over fixed mobile WiMax network is investigated under different terrain environments, namely Free Space, Outdoor to Indoor and Pedestrian. OPNET is used to investigate the performance of VoD over WiMAX. Our findings analyzing different network statistics such as packet lost, path loss, delay, network throughput.Comment: arXiv admin note: substantial text overlap with arXiv:1302.1409, and substantial text overlap with other internet sources by other author

Skew detection and compensation for Internet audio applications

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

TCP over CDMA2000 Networks: A Cross-Layer Measurement Study

Modern cellular channels in 3G networks incorporate sophisticated power control and dynamic rate adaptation which can have significant impact on adaptive transport layer protocols, such as TCP. Though there exists studies that have evaluated the performance of TCP over such networks, they are based solely on observations at the transport layer and hence have no visibility into the impact of lower layer dynamics, which are a key characteristic of these networks. In this work, we present a detailed characterization of TCP behavior based on cross-layer measurement of transport layer, as well as RF and MAC layer parameters. In particular, through a series of active TCP/UDP experiments and measurement of the relevant variables at all three layers, we characterize both, the wireless scheduler and the radio link protocol in a commercial CDMA2000 network and assess their impact on TCP dynamics. Somewhat surprisingly, our findings indicate that the wireless scheduler is mostly insensitive to channel quality and sector load over short timescales and is mainly affected by the transport layer data rate. Furthermore, with the help of a robust correlation measure, Normalized Mutual Information, we were able to quantify the impact of the wireless scheduler and the radio link protocol on various TCP parameters such as the round trip time, throughput and packet loss rate