A rate control method for data transmission

Abstract

The invention provides a data transmission rate control method and system. The method provided by the invention comprises the following steps that: a transmitting end transmits data packets to a receiving end at a preset rate; the receiving end caches the data packets; and after the data packets are started to be played, the receiving end compares the number of the cached data packets with an early-warning value of a buffer region 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 region, the receiving end acquires the network delay time of the data packets; when the network delay time of the data packets is longer than the playing time of a single data packet, the receiving end calculates the transmission rate of the transmitting end according to the transmission rate, the playing time of the single data packet and the network delay time of the data packets, transmits a notification to the transmitting end and adjusts the transmission rate according to the notification.

Description

technical field [0001] The invention belongs to the field of network communication, and in particular relates to a data transmission 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 v...

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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