The Differentiated Services Architecture

Author(s):  
Sergio Herrería-Alonso ◽  
Manuel Fernández Veiga ◽  
Andrés Suárez González ◽  
Miguel Rodríguez Pérez ◽  
Cándido López-García

IP networks only offer best-effort service to all kinds of traffic. This means that IP tries to deliver each packet as quickly as possible, but makes no service guarantees. However, as the diversity of applications increase, this simple model with no service guarantees cannot satisfy all of them. For example, novel interactive applications such as Internet telephony, video conferencing, or networked games expect some performance guarantees to operate right. The growing importance of these recent applications with stringent constraints behooves network service providers to differentiate among various types of traffic and provide a new range of service models able to accommodate heterogeneous application requirements and user expectations.

2021 ◽  
Vol 13 (3) ◽  
pp. 63
Author(s):  
Maghsoud Morshedi ◽  
Josef Noll

Video conferencing services based on web real-time communication (WebRTC) protocol are growing in popularity among Internet users as multi-platform solutions enabling interactive communication from anywhere, especially during this pandemic era. Meanwhile, Internet service providers (ISPs) have deployed fiber links and customer premises equipment that operate according to recent 802.11ac/ax standards and promise users the ability to establish uninterrupted video conferencing calls with ultra-high-definition video and audio quality. However, the best-effort nature of 802.11 networks and the high variability of wireless medium conditions hinder users experiencing uninterrupted high-quality video conferencing. This paper presents a novel approach to estimate the perceived quality of service (PQoS) of video conferencing using only 802.11-specific network performance parameters collected from Wi-Fi access points (APs) on customer premises. This study produced datasets comprising 802.11-specific network performance parameters collected from off-the-shelf Wi-Fi APs operating at 802.11g/n/ac/ax standards on both 2.4 and 5 GHz frequency bands to train machine learning algorithms. In this way, we achieved classification accuracies of 92–98% in estimating the level of PQoS of video conferencing services on various Wi-Fi networks. To efficiently troubleshoot wireless issues, we further analyzed the machine learning model to correlate features in the model with the root cause of quality degradation. Thus, ISPs can utilize the approach presented in this study to provide predictable and measurable wireless quality by implementing a non-intrusive quality monitoring approach in the form of edge computing that preserves customers’ privacy while reducing the operational costs of monitoring and data analytics.


Author(s):  
Richard Hartshorne ◽  
Haya Ajjan ◽  
Richard E. Ferdig

In this chapter, the authors provide evidence for the potential of various Web 2.0 applications in higher education through a review of relevant literature on both emerging educational technologies and social networking. Additionally, the authors report the results and implications of a study exploring faculty awareness of the potential of Web 2.0 technologies (blogs, wikis, social bookmarks, social networks, instant messaging, internet telephony, and audio/video conferencing) to support and supplement classroom instruction in higher education. Also, using the Decomposed Theory of Planned Behavior as the theoretical foundation, the authors discuss factors that influence faculty decisions to adopt specific Web 2.0 technologies. The chapter concludes with a discussion of the implications of the study and recommendations for future research.


Author(s):  
Christos Bouras ◽  
Apostolos Gkamas ◽  
Dimitris Primpas ◽  
Kostas Stamos

IP networks are built around the idea of best effort networking, which makes no guarantees regarding the delivery, speed, and accuracy of the transmitted data. While this model is suitable for a large number of applications, and works well for almost all applications when the network load is low (and therefore there is no congestion), there are two main factors that lead to the need for an additional capability of quality of service guarantees. One is the fact that an increasing number of Internet applications are related to real-time and other multimedia data, which have greater service requirements in order to be satisfying to the user. The other is that Internet usage is steadily increasing, and although the network infrastructure is also updated often, it is not always certain that network resource offerings will be ahead of usage demand. In order to deal with this situation, IETF has developed two architectures in order to enable QoS-based handling of data flows in IP networks. This article describes and compares these two architectures.


Author(s):  
R. J. Gibbens ◽  
S. K. Sargood ◽  
F. P. Kelly ◽  
H. Azmoodeh ◽  
R. Macfadyen ◽  
...  

2012 ◽  
Vol 44 (4) ◽  
pp. 995-1017 ◽  
Author(s):  
Souvik Ghosh ◽  
Soumyadip Ghosh

Cloud-computing shares a common pool of resources across customers at a scale that is orders of magnitude larger than traditional multiuser systems. Constituent physical compute servers are allocated multiple ‘virtual machines' (VMs) to serve simultaneously. Each VM user should ideally be unaffected by others’ demand. Naturally, this environment produces new challenges for the service providers in meeting customer expectations while extracting an efficient utilization from server resources. We study a new cloud service metric that measures prolonged latency or delay suffered by customers. We model the workload process of a cloud server and analyze the process as the customer population grows. The capacity required to ensure that the average workload does not exceed a threshold over long segments is characterized. This can be used by cloud operators to provide service guarantees on avoiding long durations of latency. As part of the analysis, we provide a uniform large deviation principle for collections of random variables that is of independent interest.


1999 ◽  
Vol 5 (4) ◽  
pp. 1-16 ◽  
Author(s):  
Pamela A. Kennett ◽  
Kenneth L. Bernhardt ◽  
Julie Z. Sneath

10.28945/2653 ◽  
2003 ◽  
Author(s):  
Marlenne Angulo ◽  
Marco Turrubiartes

One of the options that telephonic systems have to optimize the required bandwidth is to migrate to statistical multiplexing systems. These multiplexing schemes also allow the convergence with current data communications systems. This work is based in the interconnection of telephonic systems through the data network using digitalization and voice compression techniques. The optimization of the resources is achieved using the voice compression standard of 8 kbps instead of the 64 kbps standard, thus, introducing differentiated services in IP networks. The performance of the voice packages is improved in terms of delay and package losses. The results presented were obtained by means of computer simulations using COMNET. The merit of this work is the usage of real telephonic traffic information obtained from the call recorders of the UABC, the information was acquired for several months, which allowed the consideration of effects such as rush hours and variable call duration. The performance of the system was evaluated, obtaining very satisfactory results in terms of resources utilization.


Author(s):  
Harry G. Perros

When we call someone over the internet using a service such as Skype or Google talk, we may experience certain undesirable problems. For instance, we may not be able to hear the other person very well, or even worse, the call may be dropped. In order to eliminate these problems, the underlying IP network has to be able to provide quality of service guarantees. Several schemes have been developed that enable the IP network to provide such guarantees. Of these schemes, the multi-protocol label switching (MPLS) and the differentiated services (DiffServ) are the most widely used. In this chapter, some of the salient features of MPLS and DiffServ are reviewed.


Author(s):  
Dwayne Stevens ◽  
David T. Green

Voice over Internet Protocol (VoIP) networks signal an evolution in telecommunications that is accelerating the convergence of the Internet and the public switched telephone network (PSTN). Offering decreased costs and other benefits, VoIP is poised to transform telecommunications and the organizations that use them. However, some consider VoIP a security nightmare, combining the worst vulnerabilities of IP networks and voice networks. DOS attacks, crash attacks, packet spoofing, buffer overflow attacks, spam over Internet telephony (SPIT), and word injection all pose threats to commercial enterprise networks and the mission critical operations that they support.


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