David J. Houck

Call admission control and load balancing for voice over IP

IP networks are traditionally designed to support a best-effort service, with no guarantees on the reliable and timely delivery of packets. With the migration of real-time applications such as voice onto IP-based platforms, the existing IP network capabilities become inadequate to provide the quality-of-service (QoS) levels that the end-users are accustomed to. While new protocols such as DiffServ and MPLS allow some amount of traffic prioritization, guaranteed QoS requires call admission control. This paper reviews several possible implementations and shows simulation results for one promising method that makes efficient use of the network and is scalable to large networks.

Wireless and wireline network interactions in disaster scenarios

The fast and unrelenting spread of wireless telecommunication devices has changed the landscape of the telecommunication world, as we know it. Today we find that most users have access to both wireline and wireless communication devices. This widespread availability of alternate modes of communication is adding, on one hand, to a redundancy in networks, yet, on the other hand, has cross network impacts during overloads and disruptions. This being the case, it behooves network designers and service providers to understand how this redundancy works so that it can be better utilized in emergency conditions where the need for redundancy is critical. In this paper, we examine the scope of this redundancy as expressed by telecommunications availability to users under different failure scenarios. We quantify the interaction of wireline and wireless networks during network failures and traffic overloads. Developed as part of a Department of Homeland Security Infrastructure Protection (DHS IP) project, the network simulation modeling and analysis research tool (N-SMART) was used to perform this study. The product of close technical collaboration between the National Infrastructure Simulation and Analysis Center (NISAC) and Lucent Technologies, N-SMART supports detailed wireline and wireless network simulations and detailed user calling behavior

Call admission control for voice over IP

Voice over Internet protocol (VoIP) is a technology that enables the transmission of voice over an IP network. Recent years have witnessed heavy investment in this area in the commercial world. For VoIP to replace Public Switched Telephone Network (PSTN), it should provide voice quality comparable to circuit-switched PSTN networks. This paper addresses the mechanisms to guarantee VoIP quality of service (QoS). The focus is given to the call admission control, which blocks voice calls when the required resources are not available to guarantee the QoS for the call. We review call admission control approaches that can be applied to VoIP, and describe the advantages and disadvantages of each approach. In the second part of the paper, we present a measurement-based admission control scheme that achieves QoS in an efficient and scalable manner. The scheme uses voice traffic load measurements at each router link to compute link-level blocking policies for new call attempts. Then, these policies are translated into path-level blocking policies, which are applied to new call set-up requests. The performance of the scheme is presented for single and multiple-priority voice calls. Copyright

A measurement-based admission control algorithm for VoIP

We propose an admission control algorithm for voice over Internet protocol (VoIP) calls and present the results of an extensive performance evaluation study. The proposed algorithm operates in VoIP gateways and rejects call requests when congestion is indicated in the path to the specified destination gateway. The algorithm derives the network congestion information from the call quality reports received from the current real-time transport protocol (RTP) connections between the source and destination gateways. The performance of the algorithm has been investigated using simulation experiments and the results show very low packet loss ratios (PLRs) for the voice calls. Moreover, the call blocking ratio is very similar to that of an ideal call admission control that knows the exact number of calls at every network link.

Centralized call admission control and load balancing for voice over IP

IP networks are traditionally designed to support a best- effort service, with no guarantees on the reliable and timely delivery of packets. With the migration of real-time applications such as voice onto IP-based platforms, the existing IP network capabilities become inadequate to provide the quality-of-service (QoS) levels that the end- users are accustomed to. While new protocols such as DiffServ and MPLS allow some amount of traffic prioritization, guaranteed QoS requires call admission control. This paper reviews several possible implementations and shows simulation results for one promising method that makes efficient use of the network and is scalable to large networks.

Data Synchronization Methods Based on ShuffleNet and Hypercube for Networked Information Systems

In contrast to a typical single source of data updates in Internet applications, data files in a networked information system are often distributed, replicated, accessed and updated by multiple nodes. Due to concurrent updates, replicated data files must be synchronized. For certain applications, stringent concurrency control must be employed to ensure data integrity, while for other applications, periodic data synchronization may enable very efficient data sharing. For the latter applications, this paper devises the ShuffleNet and hypercube schemes for data synchronization in such networked information systems. Their performance in terms of update delay, processing complexity, failure tolerance and growth complexity is examined. Our results reveal that the ShuffleNet and hypercube scheme provide identical maximum update delay and similar processing complexity. However, as the number of nodes in the system changes (e.g., due to failure or temporary out of service for maintenance), the hypercube scheme maintains all existing synchronization sessions and greatly simplifies system administration overhead such as moving files from node to node for the purpose of data synchronization. The ShuffleNet scheme does provide a higher degree of failure tolerance for global data files, but the hypercube scheme provides more than adequate failure tolerance. Lastly, a generalization of the hypercube scheme, based on the ideas of shift registers, is also proposed for systems where the number of nodes is a perfect power of 2.