Nanying Yin

On closed loop rate control for ATM cell relay networks

A closed-loop, rate-based congestion control mechanism for ATM cell relay networks is presented. The goal of the congestion control mechanism is to guarantee the negotiated quality of service during overload conditions, and to allow a connection to exceed its negotiated throughput when there is unused capacity along its path through the network. With this control mechanism, the rate of an end system is adjusted based on both the network congestion state and whether its rate is within the negotiated parameters. The performance of this control mechanism for file transfer traffic is illustrated by results from simulations, Using a simple analytical model, the worst case buffer requirement and file transfer delay are analysed. From the analysis, conditions are derived for the rate increment, rate decrease factor and initial rate that minimize the buffer requirement and file transfer delay. From the analytical results, an adaptive rate reduction scheme is suggested to improve performance.<<ETX>>

A dynamic rate control mechanism for integrated networks

To achieve better statistical gain for voice and video traffic and to relieve congestion in integrated networks, a dynamic rate control mechanism is proposed. This mechanism, using a variable rate coder, adjusts the source coding rate based on network feedback information. An analytical model is developed to evaluate the performance of the mechanism for voice traffic. The feedback delay for the source node to obtain the network congestion information is presented in the model. Significant improvement in statistical gain can be realized for smaller capacity links (e.g., links that can accommodate less than 24 calls) with a reasonable feedback time (about 100 ms); but this increase causes a temporary degradation in voice quality to a lower rate. The authors show that whether the feedback delay is exponentially distributed or constant does not significantly affect performance in terms of fractional packet loss and average received coding rate. Using the number of calls in talkspurt or the packet queue length as measures of congestion provides comparable performance.<<ETX>>

Voice packet loss: destination vs. internodal links

An investigation is made of voice packet loss behavior at both the destination and internodal links in a packet switched network. The fractional loss and blocking time periods for both are derived using a bivariate Markov model. The numerical results show that blocking due to the delay constraint at the destination can result in long periods of consecutive packet loss, which seriously degrade voice quality. This work indicates that the packets with excessive delay should be discarded at the internodal links, instead of blocked at the destination. The relation between the internodal link buffer size and end-to-end permissible queuing delay is established.<<ETX>>