Method and system for controlling transmission rate

Abstract

The invention provides a method and a system for controlling transmission rate. The method provided by the invention comprises the following steps that: a transmitting end transmits data packets to a receiving end according to a preset rate, and the receiving end buffers the data packets, and further compares the number of the buffered data packets with an early warning value of a receiving end buffer zone after starting to play the data packets; when the number of the data packets buffered by the receiving end is not more than the early warning value of the buffer zone, the receiving end obtains the data packet network delay time and transmits the obtained time to the transmitting end; and the transmitting end compares the received data packet network delay time with the single data packet play time, when the data packet network delay time is greater than the single data packet play time, the transmitting end calculates and regulates the transmitting rate according to the transmission rate, the single data packet play time and the data packet network delay time.

Description

technical field [0001] The invention belongs to the field of network communication, in particular to a transmission rate control method and system. 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, ...

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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