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在以包为单位进行数据传输合、语音应用程序(VOIP,Voice Over Internet Protocol)中,为了补偿数据包在网络传输中不可预知的网络传输延迟,在接收端首先必须把接收到的数据包缓存起来,缓存一定的时间再播放出来,以减少通话的抖动,得到比较满意的通话质量。文章主要研究动态缓出时延算法,力求使这个缓出时延尽可能小,同时尽可能减少包的丢失率。文章提出了一个有效动态缓出时延算法,该算法主要跟踪最近到达的数据包的网络传输时延求出其近似分布函数,并利用这些信息和延迟峰的侦测算法预测下一个语音峰的缓出时延。实验结果表明利用该算法可以在缓出时延和包丢失率之间达到最佳平衡,是一种理想、有效的算法。
In a packet-based voice over Internet protocol (VOIP), in order to compensate for unpredictable network transmission delay of data packets in network transmission, the receiving end first needs to buffer the received data packets Up, cache a certain amount of time and then play out to reduce the jitter call, get a more satisfactory call quality. This paper mainly studies the dynamic delay delay algorithm, trying to make this delay delay as small as possible, while reducing packet loss rate as much as possible. In this paper, an effective dynamic delay mitigation algorithm is proposed. The algorithm mainly tracks the network transmission delay of the most recently arrived packets to obtain the approximate distribution function and uses these information and the delay peak detection algorithm to predict the next speech peak Slow down the delay. Experimental results show that this algorithm can achieve the best balance between delay and packet loss rate, which is an ideal and effective algorithm.