This demo will present both real and mocked-up usecases of a software application that enables accurate probabilistic modeling of new IP based services and their impact on network resources. The demonstration should be of general interest to Network Service Providers (NSP), device manufacturers, and application developers. Network planning and service analysis is a very important issue for today’s NSP’s; new communications devices and services emerge quickly (e.g., VoIP, Peer-to-peer filesharing, etc.) and end-users demand network resources that are up to the task. For the analyst, our tool leverages familiar UML style views, used to capture prototypical service “sessions”. It then allows annotation of the service model with a rich grammar, model conversion into semi-Markov form, and algorithm execution for network resource usage estimation. Ultimately, analysts gain insight into the impact of new consumer devices and services on their networks.

We present a detailed statistical analysis of a 2-hour long empirical sample of VBR video. The sample was obtained by applying a simple intraframe video compression code to an action movie. The main findings of our analysis are (1) the tail behavior of the marginal bandwidth distribution can be accurately described using “heavy-tailed” distributions (e.g., Pareto); (2) the autocorrelation of the VBR video sequence decays hyperbolically (equivalent to long-range dependence) and can be modeled using self-similar processes. We combine our findings in a new (non-Markovian) source model for VBR video and present an algorithm for generating synthetic traffic. Trace-driven simulations show that statistical multiplexing results in significant bandwidth efficiency even when long-range dependence is present. Simulations of our source model show long-range dependence and heavy-tailed marginals to be important components which are not accounted for in currently used VBR video traffic models.

The interaction of congestion control with the partitioning of source information into components of varying importance for variable-bit-rate packet voice and packet video is investigated. High-priority transport for the more important signal components results in substantially increased objective service quality. Using a Markov chain voice source model with simple PCM speech encoding and a priority queue, simulation results show a signal-to-noise ratio improvement of 45 dB with two priorities over an unprioritized system. Performance is sensitive to the fraction of traffic placed in each priority, and the optimal partition depends on network loss conditions. When this partition is optimized dynamically, quality degrades gracefully over a wide range of load values. Results with DCT encoded speech and video samples show similar behavior. Variations are investigated such as further partition of low-priority information into multiple priorities. A simulation with delay added to represent other network nodes shows general insensitivity to delay of network feedback information. A comparison is made between dropping packets on buffer overflow and timeout based on service requirements.>

In dual unidirectional bus networks, packets usually occupy fixed-length slots form the sending station to the end of the network. An erasure node is a specialized station which recognizes packets which have passed their destination stations and releases the slots for subsequent use. The authors derive the optimal locations for erasure nodes and show analytically, for uniform traffic, that only several erasure nodes are needed to achieve throughput close to twice the nominal network bandwidth. The results are tested by simulation of the DQDB (distributed queue dual bus) protocol, which demonstrates a realistic improvement of 40% with only three erasure nodes. Fair access among the stations is improved as well. The authors generalize the analytic results by providing an algorithm for determining the optimal erasure node locations and the throughput improvement, given any arbitrary traffic pattern. The application of this methodology to the related problem of bridged subnetworks is briefly discussed.>

It is shown analytically that for uniform traffic, throughput close to twice the nominal network bandwidth may be achieved with only several erasure nodes. The optimal erasure node locations are calculated. The results are tested by simulation of the DQDB (distributed queue dual bus) protocol, yielding a realistic improvement of 40% with only three erasure nodes. Fair access among the stations is improved as well. The analytic results are generalized by providing an algorithm for determining the optimal erasure node locations and the throughput improvement, given any arbitrary traffic pattern. The application of this methodology to the related problem of bridged subnetworks is discussed.>