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Method and device for WebRTC P2P (web real-time communication peer-to-peer) audio and video call

An audio, video and audio technology, applied in the fields of WebRTC P2P audio and video calls, WebRTC servers and WebRTC clients, it can solve the problems of small screen windows, difficulty, and inability to save subtitles, and achieve the effect of convenient calls

Active Publication Date: 2015-07-15
ZTE CORP
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  • Summary
  • Abstract
  • Description
  • Claims
  • Application Information

AI Technical Summary

Problems solved by technology

However, the effect and user experience of the audio-video multi-party call conference in the browser developed using the webrtc standard interface still needs further improvement. Records can only be saved in the form of audio recording, but subtitles cannot be saved. For example, when the participants in the meeting use different languages, the language barrier of communication needs to assist the display of subtitles to better improve the user experience

Method used

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  • Method and device for WebRTC P2P (web real-time communication peer-to-peer) audio and video call
  • Method and device for WebRTC P2P (web real-time communication peer-to-peer) audio and video call
  • Method and device for WebRTC P2P (web real-time communication peer-to-peer) audio and video call

Examples

Experimental program
Comparison scheme
Effect test

Embodiment 1

[0085] image 3 It is an operation diagram for requesting subtitles during a P2P call between webrtc parties. Suppose user A and user B have followed the figure 2 The process of the WEBRTC application or the process of the WEBRTC application itself has established a media channel, and the media channel can be used for normal P2P video calls. This embodiment describes a flow chart of user A requesting user B's subtitle during a P2P video call.

[0086]Step 301, user A sends a subtitle request message to the signaling server of webrtc server;

[0087] Step 302, the webrtc signaling server sends a subtitle request message to user B;

[0088] Step 303, after receiving the subtitle request, user B sends his own audio to the speech analysis subtitle server;

[0089] Step 304, the voice analysis subtitle server parses the audio into subtitles, and returns the subtitles to user B;

[0090] Step 305, user B returns the subtitle to the webrtc signaling server;

[0091] Step 306, ...

Embodiment 2

[0094] Embodiment 2 is the process of requesting subtitle translation. Compared with embodiment 1, the process in embodiment 2 has one more step after the subtitles are parsed by speech analysis. This step is to send each sentence of the parsed words to the external translation server, and the external translation server will translate the subtitles and Returns text subtitles or speech audio in the translated language. Figure 4 It is the step diagram of requesting translation of subtitles in Embodiment 2. in,

[0095] Step 401, user A sends a subtitle translation request message to the signaling server of the webrtc server, and specifies the target language for translation, assuming that the language used by B is English, and A hopes that B's subtitles will be translated into Chinese and displayed;

[0096] Step 402, the webrtc signaling server sends a subtitle request message to user B, and the request message includes translation source language, translation target langua...

Embodiment 3

[0110] Example 3, Figure 6 It shows the process of implementing subtitle parsing after webrtc has completed the P2P media channel connection.

[0111] Preconditions: User A, user B and user C have logged in using the WEBRTC video conferencing system and established a three-party P2P call, and a media channel has been established between A, B and C. The signaling channel still performs command operations through the signaling server of webrtc.

[0112] In this embodiment, it is assumed that A requests subtitles for the speeches of B and C.

[0113] Step 601, user A requests the subtitles of user B and user C from the webrtc signaling server;

[0114] Step 602, the webrtc signaling server sends a subtitle request to user C;

[0115] Step 603, user C sends his speech audio to an external speech analysis subtitle server, requesting subtitle analysis;

[0116] Step 604, the speech analysis subtitle server returns the subtitle obtained by speech analysis to C;

[0117] Step 60...

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Abstract

The invention provides a method for WebRTC (web real-time communication) peer-to-peer audio and video call as well as a WebRTC server and a WebRTC client, and the method enables a user to stride over language obstacles and make a call more conveniently. In a multi-user video conference, a spokesman automatically analyzes and displays subtitles, translates the subtitles or translates audios, and the user can easily judge who is speaking and identify speech content without searching the spokesman in a plurality of video windows.

Description

technical field [0001] The invention relates to the communication field, in particular to a WebRTC P2P audio and video call method, a WebRTC server and a WebRTC client. Background technique [0002] With the development of the www World Wide Web and the mobile Internet, HTML5 (Hyper Text Mark-up Language 5, Hypertext Mark-up Language 5) has gradually become a hot spot in both the market and standards in recent years. As a new development direction of network technology, an important core technical content of HTML5 is WEBRTC technology. WebRTC (Web Real-Time Communication, webpage real-time communication) implements webpage-based video conferencing, and aims to achieve real-time communication (Real-Time Communications) capabilities by providing simple javascript through the browser. [0003] The ultimate goal of the WebRTC project is to enable web developers to easily and quickly develop rich real-time multimedia applications based on browsers (such as Chrome, FireFox...) wi...

Claims

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Application Information

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IPC IPC(8): H04N7/14H04N7/15G06F17/30
CPCH04N7/14
Inventor 巫妍
Owner ZTE CORP
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