Jitter buffer regulation method and device

Abstract

The embodiment of the invention relates to the field of electronic technology, and discloses a jitter buffer regulation method and device. The method comprises the following steps: acquiring a target statistical characteristic value for representing the weighted statistical characteristic of the network delay of history received voice packets, wherein the history received voice packets include a current time unit and voice packets received before the current time unit; computing the predicted network delay of voice packets to be received according to the target statistical characteristic value; and regulating the length of a jitter buffer according to the predicted network delay. Through implementation of the embodiment of the invention, the call smoothness can be ensured; the call delay can be reduced; and high timeliness is achieved.

Description

technical field [0001] The invention relates to the field of electronic technology, in particular to a jitter buffer adjustment method and device. Background technique [0002] With the increasing popularity of the Internet, a new type of voice over Internet (Voice Over Internet Protocol, VOIP) implemented through the Internet has attracted more and more attention because of its low cost and easy expansion. However, compared with the stability of traditional telephones, due to the network jitter of the Internet itself, voice packets arrive later than the expected playback time in Internet telephony, causing the call to freeze, thereby reducing the call quality. [0003] In order to combat the instability of the Internet, the receiving end usually creates a jitter buffer, and voice packets start playing after accumulating in the jitter buffer for a period of time. By adjusting the size of the jitter buffer, the voice packets can always Playing ensures the fluency of the voic...

AI Technical Summary

Problems solved by technology

However, compared with the stability of traditional telephones, due to the network jitter of the Internet itself, voice packets arrive later than the expected playback time, causing the call to freeze, thereby reducing the quality of the call

[0003] In order to combat the instability of the Internet, the receiving end usually creates a jitter buffer, and voice packets start to play after accumulating in the jitter buffer for a period of time. By adjusting the size of the jitter buffer, the voice packets can always Playing to ensure the fluency of the voice, but usually in order to ensure the fluency of the voice, the jitter buffer is set too large, although this ensures the smoothness of the call, but the delay of the call is significantly increased, and the real-time performance is not strong

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