Method and device for transmitting medium streams in video conference

Abstract

The embodiments of the present invention relate to a transmission method for a media stream in a video conference. The method comprises: receiving a first media stream sent by a conference server, the first media stream comprising video and audio information about all conference participants in a video conference; receiving a selection instruction, the selection instruction specifically indicating to select conference participants corresponding to required video and audio information from the video and audio information about all the conference participants; according to the selection instruction, sending a transmission request message to the conference server via a WebRTC application, the transmission request message comprising identification information about the selected conference participants; and receiving a second media stream sent by the conference server according to the identification information, the second media stream comprising the video and audio information about the selected conference participants.

Description

technical field [0001] The invention relates to the field of communication technology, in particular to a method and device for transmitting media streams in a video conference. Background technique [0002] Web Real Time Communication (Web Real Time Communication, referred to as: WebRTC) technology can realize video and audio communication between different browsers or between browsers and terminals, multi-party conferences and other functions. like figure 1 shown. [0003] exist figure 1 On the left side, user 1 visits the WeRTC application website to open the WebRTC application webpage, and establishes a communication connection with the WeRTC application website through the JavaScript code in the webpage; figure 1 On the right side, user 2 also visits the WeRTC application website to open the WebRTC application webpage, and establishes a communication connection with the WeRTC application website through the JavaScript code in the webpage. At this time, user 1 and us...

AI Technical Summary

Problems solved by technology

[0004] At present, in the prior art, when user 1 uses terminal 1 (for example: mobile phone) to participate in a WebRTC video conference, the conference server will mix / mix the video / audio of all participants in the video conference, and then send to the terminal 1, or the user 1 uploads the video and audio information in the terminal 1 to the conference server through the terminal 1; however, the following disadvantages are also exposed in the prior art: 1) the conference server can only upload all participants in the video conference The video / audio of the participant is mixed / mixed and then distributed to each user, but the video and audio of a specific participant cannot be collected and combined, which reduces the user experience, and also increases the bandwidth used by the terminal and the local resource consumption of the terminal.

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