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.>

The leaky bucket algorithm, which has been proposed as a monitor/enforcer in the rate-based management of bandwidth in high-performance, integrated packet communication networks, is considered. The effectiveness of the unbuffered leaky bucket algorithm in protecting the quality of service (QOS) experienced by connections passing through a common internodal link queue is studied. It is shown that excess traffic generated by a source can pass through the leaky bucket without marked or discarded and cause unacceptable packet delay/loss to other connections sharing the queue. It is also shown that average bursts that are larger, or average rate that is higher, than that assumed at connection establishment can deteriorate the queue performance even with the leaky bucket enforcing the agreed throughput QOS by discarding packets. On the basis of this analysis, a strategy for alleviating these problems is devised, namely, allocating network bandwidth above the leaky bucket permit rate and configuring the leaky bucket with a loss/mark probability greater than 10/sup -2/.>

When congestion occurs in a packet queuing system, packets can be dropped from the rear or the front of the queue. It is demonstrated that the probability of a packet being dropped is the same in systems with rear and front packet dropping. It is shown that the probability of a packet being delayed longer than a given value in a system with front dropping is less than or equal to that in a system with rear dropping. It is further illustrated that front dropping not only improves the delay performance on an internodal link, but also provides the overall loss performance for time constrained traffic such as packet voice.>

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.>

To achieve better statistical gain for voice and video traffic and to relieve congestion in fast packet networks, a dynamic rate control mechanism is proposed. An analytical model is developed to evaluate the performance of this control mechanism for voice traffic. The feedback delay for the source node to obtain the network congestion information is represented in the model. The study indicates that significant improvement in statistical gain can be realized for smaller capacity links (e.g., links that can accommodate less than 24 voice calls) with a reasonable feedback time (about 100 ms). The tradeoff for increasing the statistical gain is temporary degradation of voice quality to a lower rate. It is shown 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. It is also shown that using the number of calls in talkspurt or the packet queue length as measures of congestion provides comparable performance.>

Voice packet loss behavior at both the destination and internodal links in a packet-switched network is investigated. 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. The authors' work indicates that packets with excessive delay should be discarded at the internodal links, instead of blocking them at the destination. The relation between the internodal link buffer size and end-to-end permissible queueing delay is established.>

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.>

In order to reduce the time delays as well as multiplexer memory requirements in packet voice systems, a family of congestion control schemes is proposed. They are all based on the selective discarding of packets whose loss will produce the least degradation in quality of the reconstructed voice signal. A mathematical model of the system is analyzed and queue length distributions are derived. These are used to compute performance measures, including mean waiting time and fractional packet loss. Performance curves for some typical systems are presented, and it is shown that the control procedures can achieve significant improvement over uncontrolled systems, reducing the mean waiting time and total packet loss (at transmitting and receiving ends). Congestion control with a resume level is also analyzed, showing that without increasing the fractional packet loss, the mean and variance of the queue can be reduced by selecting an appropriate resume level. The performance improvements are confirmed by the results of some informal subjective testing.>