A method and system for audio and video forwarding control

Abstract

The present invention provides an audio and video forwarding control method and system, the method comprising: the sending end sends RTP packets, and after sending a set number of RTP packets, generates and sends SR packets according to the number of currently sent RTP packets; the receiving end Receive the RTP packet and SR packet, and after receiving the SR packet, count the number of RTP packets currently received, calculate the packet loss rate and generate RR according to the number of currently sent RTP packets carried in the SR packet The packet is returned to the sending end; the sending end receives the RR packet, obtains the packet loss rate; adjusts the rate of sending RTP packets according to the packet loss rate. Compared with the prior art, the present invention can dynamically adjust and control the transmission rate of audio and video data according to the network congestion state.

Description

technical field [0001] The invention relates to the technical field of multimedia network transmission, in particular to an audio and video forwarding control method and system. Background technique [0002] In recent years, with the rapid development of Internet and multimedia technology, network video conferencing with good real-time and interactive features has been widely used. However, the currently used video conferencing mostly uses the TCP / IP protocol to transmit real-time audio and video data, the real-time effect is poor and cascading cannot be performed between servers. [0003] Due to the connection-oriented retransmission mechanism of the traditional TCP / IP protocol, it can no longer meet the real-time transmission of multimedia video data, and its congestion mechanism is likely to cause network congestion, which is not conducive to the transmission of large amounts of multimedia audio and video data. Although the UDP protocol uses a best-effort delivery method...

AI Technical Summary

Problems solved by technology

[0013] Existing flow control algorithms based on constant increase and constant decrease will cause flow control. If the two increase and decrease constants are set too large, the jitter of the RTP flow will be large, which will aggravate network congestion and if the setting is too large Small, the speed of the RTP stream will increase too slowly or decrease too fast, which is not conducive to the improvement of network bandwidth utilization

[0014] In the existing technology, the use of TCP retransmission mechanism is likely to cause network congestion, while UDP's best-effort delivery transmission method avoids the retransmission mechanism, but because it does not provide a congestion mechanism, it cannot be adjusted according to network conditions

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