Real-time live broadcast method based on sip+rtp and rtmp protocol interworking

Abstract

The invention discloses a real time live streaming method based on SIP+RTP and RTMP protocol intercommunication, and relates to the computer video live streaming technology; the method specifically comprises the following steps: a send terminal selects a SIP+RTP or RTMP mode according to network conditions and the application scene, and pushes audio and video data to an input adaptation layer of a media server; the media server authorizes the send terminal and builds connection so as to receive data; the media server parses signaling and media data pushed by the send terminal, and submits data to a media processing layer; an output adaptation layer of the media server draws a data request according to receive terminal application processing; the data request is authorized, and media data is sent down according to a receive terminal selected mode. The real time live streaming method can improve the real time live streaming system interactivity and high effectiveness.

Description

technical field [0001] The invention relates to the technical field of computer video live broadcast, in particular to a real-time live broadcast method based on SIP+RTP and RTMP protocol intercommunication. Background technique [0002] The existing mature multimedia communication solutions are mainly divided into two categories, SIP+RTP communication solutions and RTMP communication solutions: [0003] SIP+RTP communication solution: Widely used in instant messaging and video conferencing, with high real-time performance, but the quality of audio and video cannot be guaranteed in the case of large network jitter, and the cost of large-scale load is high. [0004] RTMP communication solution: Widely used in online live broadcasting and security monitoring fields, the audio and video quality is good, but the audio and video delay is relatively large. When the real-time requirements are high, the real-time performance of both parties and multiple parties cannot be guaranteed....

AI Technical Summary

Problems solved by technology

[0003] SIP+RTP communication solution: widely used in instant messaging and video conferencing, with high real-time performance, but cannot guarantee audio and video quality in the case of large network jitter, and the cost of large-scale load is high

[0004] RTMP communication solution: Widely used in online live broadcasting, security monitoring fields, audio and video quality is good, but the audio and video delay is relatively large, when the real-time requirements are high, the real-time performance of both parties and multiple parties cannot be guaranteed

[0005] The two solutions use different protocols and different encapsulation methods, and cannot interoperate natively

[0006] At present, the commonly used solution is that the server assembles the audio and video data into FLV format files and stores them on the hard disk. When the receiving terminal pulls the audio and video data, it reads the audio and video data from the hard disk and sends it to the sending terminal, which is suitable for on-demand, recording and broadcasting. Application scenarios with low real-time performance

[0007] Due to the need to store files on the hard disk and then read them from the hard disk, the encapsulation and analysis of files, the hard disk IO read and write process is time-consuming, and it is not suitable for high-real-time video live broadcast, video conferencing, and video calls.

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