In order to accommodate the new generation of Internet applications, routers need to be designed to support QoS, policy based management, and mobility. This paper presents architecture for the control plane of a software router that integrates signaling protocols and control mechanisms for QoS, policy based management, and mobile IP. The architecture has the flexibility of using the policy control to dynamically configure routers to act as nodes in an IntServ of DiffServ domain. Within each domain the routers can be configured to control their resource usage or to implement service level agreements. Implementation of a software router based on Linux and the experimental results are presented. The results show that with software routers this architecture can be used to meet the end-to-end QoS performance for most Internet applications.

In future wireless networks, different traffic classes will exhibit a large variety of characteristics and QoS requirements, such as transmission rate, maximum tolerable bit error rate and timeout specifications. However, currently there is no standard way of guaranteeing QoS in wireless access networks like wireless LAN based on IEEE 802.11. In this paper, we propose medium access control protocol enhancements and a distributed scheduler for QoS guarantees and fairness over IEEE 802.11 WLAN. The proposed scheme uses distributed scheduling at both the access point and the user terminals to schedule the transmission of packets according to their delay requirements. The algorithms used for both scheduling are the same. A flexible and fair resource allocation method among the traffic classes and the user terminals is provided by this scheme. Its performance has been evaluated using the UCB/LBNL/VINT Network Simulator and an implementation in the Linux kernel.

As communication networks provide newer services, signaling is becoming more and more compute intensive compared to present day networks. It is known that under overload conditions, the call throughput (goodput) and the network revenue drops to zero even when transport resources are available. A distributed overload control algorithm for delay-bounded call setup is proposed. The end-to-end delay bound is budgeted among the switching nodes involved in call setup, and these nodes apply a local overload control with a deterministic delay threshold and drop call requests experiencing higher delays. This algorithm does not depend on feedback on network conditions and makes use of only parameters that can be instrumented locally by the switching node. Using an M/M/1 queueing model with first-in-first-out (FIFO) service discipline at a switching node, two optimized control schemes are considered for local overload control and compared their performance through analysis: one with arrival rate limit and the other with buffer size limit. Though both the schemes minimize the unproductive call processing at heavy load, the latter is found to yield higher call throughput and lower average call setup delays compared to the former. Also, the buffer size required for the scheme with buffer size limit is typically small and call throughput close to the server capacity can be achieved during overload. The performance of the distributed overload control algorithm in a network is evaluated through simulation experiments, using the scheme with buffer size limit for the local overload control. It shows that the average end-to-end delay could be much less than the end-to-end delay bound, providing room for overprovisioning of the delay bounds. The tradeoff between the number of nodes, call throughput, and average end-to-end delay needs to be considered while deciding the route budgeting the end-to-end delay bound among different nodes along the route. These performance results are expected to serve as lower bounds to more sophisticated local call rejection mechanisms such as push-out or time-out along with a last-in-first-out (LIFO) service discipline.

Real-time MPEG-2 video transport applications do not usually have the luxury of a reverse channel for recovering from any errors that might occur during communication. Degradation in quality of decoded video frames is immediately apparent in the presence of errors in headers. In this paper, we focus on protecting header information by replicating it in any free space that might be available in the defined MPEG-2 transport stream packets. We also present our implementation experience over wired ATM as well as wireless IEEE 802.11 LAN by incorporating this forward error recovery approach with a real-time MPEG-2 encoder. In our experiments, it is found that the free space available is generally more than adequate for replicating essential header information.

Article Free AccessA forward error recovery technique for MPEG-II video transport Share on Authors: R. Radhakrishna Pillai Kent Ridge Digital Labs, 21 Heng Mui Keng Terrace, Singapore 119613 Kent Ridge Digital Labs, 21 Heng Mui Keng Terrace, Singapore 119613View Profile , B. Prabhakaran School of Computing, National University of Singapore, Singapore 119260 School of Computing, National University of Singapore, Singapore 119260View Profile , Qui Qiang School of Computing, National University of Singapore, Singapore 119260 School of Computing, National University of Singapore, Singapore 119260View Profile Authors Info & Claims MULTIMEDIA '99: Proceedings of the seventh ACM international conference on Multimedia (Part 2)October 1999 Pages 59–61https://doi.org/10.1145/319878.319897Online:01 October 1999Publication History 3citation288DownloadsMetricsTotal Citations3Total Downloads288Last 12 Months1Last 6 weeks0 Get Citation AlertsNew Citation Alert added!This alert has been successfully added and will be sent to:You will be notified whenever a record that you have chosen has been cited.To manage your alert preferences, click on the button below.Manage my AlertsNew Citation Alert!Please log in to your account Save to BinderSave to BinderCreate a New BinderNameCancelCreateExport CitationPublisher SiteeReaderPDF

The paper discusses the motivation, requirements, and the emerging technologies to carry multimedia services over the Internet.