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QOS improving for voice streaming over wireless ad-hoc networks using an adaptive playout adjustment algorithm

机译:使用自适应播出调整算法改进无线自组织网络上语音流的QOS

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Providing a service with good quality for transmission and playing real-time voice conversations (voice streaming) over wireless ad-hoc networks is always a big challenge. Buffering and adjusting the playout time of packets is one of the methods used to overcome this challenge which can be deployed in receiver side. In this paper, a new adaptive playout adjustment algorithm to stream the voice conversations over wireless ad-hoc networks is proposed. This algorithm always tries to be aware of network's conditions, adapts itself with these conditions and adjusts the playout time of voice packets as good as possible. So, not only most of the packets must be received before their playout time which is scheduled in receiver, but also the playout time must not be too long as much that has a bad effect on the interactivity between sender and receiver. The main features of the presented method are: First, adjusting the threshold adaptively with respect to variant conditions of networks, in order to determine the state of system. Second, calculating the mean network jitter and especially alpha parameter dynamically based on the current conditions of networks, in order to calculate the playout delay of current packet. Third, being optimistic about the future state of networks and not using the history of delay, in order to calculate the mean network delay. The results of simulation show that the proposed algorithm adapts itself with the network's dynamic conditions and adjusts the playout delay of voice packets better than the other algorithms. other algorithms.
机译:在无线自组织网络上提供高质量的服务以进行传输和播放实时语音对话(语音流)始终是一个很大的挑战。缓冲和调整数据包的播出时间是用于克服此挑战的方法之一,可以在接收器端进行部署。本文提出了一种新的自适应播放调整算法,用于在无线自组织网络上流式传输语音对话。该算法始终尝试了解网络状况,使其适应这些状况,并尽可能地调整语音数据包的播放时间。因此,不仅大多数分组必须在其在接收器中安排的播出时间之前被接收,而且播出时间也不能太长,以免影响发送器和接收器之间的交互性。所提出的方法的主要特征是:首先,针对网络的变化条件自适应地调整阈值,以便确定系统的状态。其次,根据网络的当前状况动态计算平均网络抖动,尤其是alpha参数,以计算当前数据包的播放延迟。第三,对网络的未来状态持乐观态度,不使用延迟历史来计算平均网络延迟。仿真结果表明,与其他算法相比,该算法能够适应网络的动态条件,并能更好地调节语音包的播放时延。其他算法。

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