A rate control method

Abstract

The invention provides a speed control method and system. The speed control method comprises the following steps that: a sending end sends data packets to a receiving end by a server at a preset speed; and after the receiving end caches the data packets and starts to play the data packets, the receiving end compares the number of the cached data packets with an early warning value of a buffer area of the receiving end. When the number of the data packets cached by the receiving end is smaller than or equal to the early warning value of the buffer area, the receiving end acquires network delay time of the data packets and sends the network delay time to a server. The server compares the network delay time of the data packets with single data packet playing time, and when the network delay time of the data packets is more than the data packet play time, the server calculates the sending speed according to a transmission speed, the single data packet playing time and the network delay time of the data packets, and sends the calculated sending speed to the sending end; and the sending end regulates the sending speed according to a notice.

Description

technical field [0001] The invention belongs to the field of network communication, and in particular relates to a rate control method. Background technique [0002] With the rapid development of the information society, various communication tools such as Fetion, QQ, etc. are widely used by people. In addition to personal applications, some enterprise instant messaging methods are gradually becoming familiar to the public, such as video conferencing. Whether audio and video data can be played continuously and smoothly has become a key factor affecting the quality of video conferences. However, in practical applications, jitter delays often exist in the transmission of audio and video data. When the delay time is too long, it is easy to cause the audio and video playback at the receiving end to be interrupted. However, in the prior art, when the amount of data buffered locally at the receiving end is not enough to support continuous playback of audio and video, the sending ...

AI Technical Summary

Problems solved by technology

However, in practical applications, jitter delays often exist in the transmission of audio and video data. When the delay time is too long, it is easy to cause the audio and video playback at the receiving end to be interrupted.

However, in the prior art, when the amount of data buffered locally at the receiving end is not enough to support continuous playback of audio and video, the sending end will send a large amount of audio and video data to the receiving end in a short period of time, which will easily cause data packet congestion and even cause other network applications to Stuck

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