Evaluation of QoS in internet accesses for multimedia applications (EQoSIM) (original) (raw)
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Control System for Internet Bandwidth Based on Java Technology
Journal of Theoretical and Applied information …, 2008
This paper presents a Java-based real-time Internet access estimation tool for Quality of Service (QoS) in Internet accesses for Multimedia applications (JEQoSIM). It is specially aimed for real-time multimedia applications which use the User Datagram Protocol (UDP). The system is capable of estimating access capacity, available bandwidth and delay as the critical end-to-end QoS parameters for this kind of applications. The algorithm used for QoS estimations is one-way, and is based on the packet train technique. Real-time QoS estimation elements are distributed among a central server and the Internet end user. The central server contains a UDP packet bursts server and a web server that hosts the Java applet that implements the UDP packet bursts client. JEQoSIM has been validated using several commercial Internet accesses with different technologies: Asymmetric Digital Subscriber Loop (ADSL), General Packet Radio Service (GPRS) and Universal Mobile Telecommunications System (UMTS).
The deployment of end-to-end available bandwidth estimation mechanism in web-based application
2010 IEEE Symposium on Industrial Electronics and Applications (ISIEA), 2010
Bandwidth is one of the most common terms in digital communication as it represents the supported amount of data that a link or network path can send per unit of time. Multimedia application's performance is directly impacted by available bandwidth because these applications have diverse Quality of Service (QoS) requirements. Internet is now a necessary technology in our daily life, and because of mobile communication systems people can access the network anytime and anywhere, resulting in an increase in the number of users. Such evolution brings new services that will require more data rates. As a result of this load, some services and users in the network will demand for certain data rate which cannot be guaranteed. In the past, there have been several proposals that provide some mechanism for bandwidth estimation based on the size of web message and TCP/IP stack. However, these solutions tend to provide unreliable bandwidth estimation due to TCP/IP buffering. This paper proposes a bandwidth estimation mechanism (BEM) that is capable of providing reliable estimation of the available bandwidth to users or applications. The experimental results validate the intrusiveness and the ability of the proposed BEM to provide accurate bandwidth estimates.
Implementation of on-demand QoS allocation system over IP for multimedia applications
2003
To support diverse transmission requirements of multimedia applications, Quality of Service (QoS) should be provided in the Internet, where only the best-effort service is available. Based on the bandwidth broker model for realizing the IETF differentiated service (DiffServ), in this paper, we describe our recent effort on the implementation and verification of an extendable and flexible QoS allocation and resource management system. Focusing on the bandwidth issue over single administrative domain, the implemented system provides real-time resource reservation and allocation, delayed call admission control, simple QoS negotiation between server and user, and simple resource monitoring. The implemented system is verified by evaluating the performance of a resource-intensive application over the real-world testbed network.
Performance analysis of multimedia based web traffic with QoS constraints
Journal of Computer and System Sciences, 2008
During the recent years, there has been a tremendous growth in the development and deployment of multimedia based networked applications such as video streaming, IP telephony, interactive games, among others. These applications, in contrast to elastic applications such as email and data sharing, are delay and delay jitter sensitive but can tolerate certain level of packet loss. A vital element of end-to-end delay and delay jitter is the random queueing delays in network switches and routers. Analysis of robust mechanisms for buffer management at network routers needs to be carried out in order to reduce end-to-end delay for traffic generated by multimedia applications. In this context, a threshold based buffer management scheme for accommodating multiple class multimedia traffic in network routers has been analysed. This technique effectively controls the allocation of buffer to various traffic classes according to their delay constraints. The forms of the joint state probabilities, as well as basic performance measures such as blocking probabilities are analytically established at equilibrium. Typical numerical experiments are included to illustrate the credibility of the proposed mechanism in the context of different quality of service (QoS) grades for various network traffic classes. This model, therefore, can be used as a powerful tool to provide a required grade of service to a particular class of multimedia based web traffic in any heterogeneous network.
Performance Modeling of HTTP and RTP Streaming for QoS Support over Next Generation Networks
IJCA Proceedings on International …, 2012
Attempts to display media on computers date back to the earliest days of computing in the mid-20th century. However, little progress was made for several decades, primarily due to the high cost and limited capabilities of computer hardware. From the late 1980s through the 1990s, consumer-grade personal computers became powerful enough to display various media. The primary technical issues related to streaming was having enough CPU power and bus bandwidth to support the required data rates and creating low-latency interrupt paths in the operating system (OS) to prevent buffer under run. However, computer networks were still limited, and media was usually delivered over non-streaming channels, such as by downloading a digital file from a remote server and then saving it to a local drive on the end user's computer or storing it as a digital file and playing it back from CD-ROMs. The challenges of new communication architecture are to offer better quality of service (QoS) in internet Network. A large diversity of services based on packet switching in 3G network and beyond 3G leads dramatic changes in the characteristics and parameter of data traffic. Through this paper we propose a streaming solution to offer better QoS over 3G and 4G networks. A comparative analysis of HTTP streaming and RTP streaming has been done and packet loss, average bandwidth utilization and total streaming time has been measured. Result shows RTP provides better quality of service over HTTP for multimedia based content transfer over various network. Simulation is done using Java Media Framework.
Investigating factors influencing QoS of Internet phone
Proceedings IEEE International Conference on Multimedia Computing and Systems, 1999
An increasing number of applications are using the Internet for voice transmission, and hence there is a growing demand for the Internet phone. However, the Internet was not originally designed for such isochronous trafic, and it is not clear that it is fully capable of providing the desired Quality of Service (QoS) for this application. In this paper we describe a number of QoS parameters to measure the perceptual quality of audio, including measures of delay, rate, and loss. We investigate the factors influencing the QoS of Internet voice, such as packet period and packet size, threshold delay, buffer delay, time slots and Internet sites. We present results from our experiments on how these factors influence the QoS of Internet voice. Based on the results of our experiments, we propose a QoS driven dynamic scheduling algorithm for the real-time transmission of Internet voice packets.
Quality of Service for Multimedia and Real-Time Services
New web technologies have encouraged the deployment of various network applications that are rich with multimedia and real-time services. These services demand stringent requirements are defined through Quality of Service (QoS) parameters such as delay, jitter, loss, etc. To guarantee the delivery of these services QoS routing algorithms that deal with multiple metrics are needed. Unfortunately, QoS routing with multiple metrics is considered an NP-complete problem that cannot be solved by a simple algorithm. This paper proposes three source based QoS routing algorithms that find the optimal path from the service provider to the user that best satisfies the QoS requirements for a particular service. The three algorithms use the same filtering technique to prune all the paths that do not meet the requirements which solves the complexity of NP-complete problem. Next, each of the three algorithms integrates a different Multiple Criteria Decision Making method to select one of the paths that have resulted from the route filtering technique. The three decision making methods used are the Analytic Hierarchy Process (AHP), Multi-Attribute Utility Theory (MAUT), and Kepner-Tregoe KT. Results show that the algorithms find a path using multiple constraints with a high ability to handle multimedia and real-time applications.
Communications, 2015
Basically Internet it's a connection of small and large networks. From the beginning Internet used only for transferring some information between two computers but nowadays Internet are huge commercial communication network. As usual Internet used for communication between applications. The first Internet applications weren't so sensitive for packets loss and delays. But in the modern Internet and networking applications requires more high level of connection quality, application become more sensitive for delays and packets loss and this is a reason to provide levels of quality (Quality of Service).The goal of experiment is studying applying QoS on network router (Access Router) to conform SLA (Service Layer Agreement). SLA includes End-To-End delay and Packet Delay Variation for Video Demonstration traffic. The approach to this paper is to use QoS functions to fulfill the specifications of the performance parameters in SLA between the operator and the customer. From the study it is found that using RSVP (Resource Reservation Protocol) gives better delay and delay variation results at the expense of more bandwidth.