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A dynamic rate control mechanism for source coded traffic in a fast packet network

机译:快速分组网络中源代码流量的动态速率控制机制

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摘要

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.
机译:为了在语音和视频业务中获得更好的统计增益并缓解快速分组网络中的拥塞,提出了一种动态速率控制机制。开发了一种分析模型来评估此控制机制对语音流量的性能。模型中表示了源节点获得网络拥塞信息的反馈延迟。该研究表明,对于具有较小容量的链接(例如,可容纳少于24个语音呼叫的链接),可以在合理的反馈时间(约100毫秒)内实现统计增益的显着提高。增加统计增益的权衡是语音质量暂时降低到较低的速率。示出了反馈延迟是指数分布的还是恒定的,就分数分组丢失和平均接收编码率而言,不会显着影响性能。还显示出,使用话音突峰中的呼叫数或数据包队列长度作为拥塞度量可以提供相当的性能。

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