网络抖动(jitter)的计算
2009-08-30 13:50
嗯,最近在做一些很古老的Congestion Control的工作,用UDP设计一套仿照TCP AIMD的机制,最重要的几点就在于: 1. 如何判断拥塞发生 2. 用什么样的策略来调整窗口(对UDP来说是sending rate) 对于1来说,使用两个threshold, 一个就是jitter,另一个收到两个feedback之间的时间差。这里转载两篇节选的关于jitter计算的文章。一篇英文,是Stanford一个research center在1998年做的一些实验,叹一句,人家做的果然是很早啊。 Website: http://www.slac.stanford.edu/comp/net/wan-mon/tutorial.html Jitter, see also Jitter,The short term variability or "jitter" of the response time is very important for real-time applications such as telephony. Web browsing and mail are fairly resistent to jitter, but any kind of streaming media (voice, video, music) is quite suceptible to jitter. Jitter is a symptom that there is congestion, or not enough bandwidt to handle the traffic. The jitter specifies the length of the VoIP codec de-jitter buffer to prevent over- or under-flow. An objective could be to specify that say 95% of packet delay variations should be within the interval [-30msec, +30msec].The ITU has a Proposed Method for Measuring Packet Delay Variation. This requires in |