WO2015139610A1 - Video conferencing method, device and system - Google Patents

Video conferencing method, device and system Download PDF

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Publication number
WO2015139610A1
WO2015139610A1 PCT/CN2015/074382 CN2015074382W WO2015139610A1 WO 2015139610 A1 WO2015139610 A1 WO 2015139610A1 CN 2015074382 W CN2015074382 W CN 2015074382W WO 2015139610 A1 WO2015139610 A1 WO 2015139610A1
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WO
WIPO (PCT)
Prior art keywords
telephone terminal
participant
host
data
conference call
Prior art date
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PCT/CN2015/074382
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French (fr)
Chinese (zh)
Inventor
周新中
黄绍彰
储成
Original Assignee
华为技术有限公司
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Publication date
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Publication of WO2015139610A1 publication Critical patent/WO2015139610A1/en

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1083In-session procedures
    • H04L65/1093In-session procedures by adding participants; by removing participants
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/40Support for services or applications
    • H04L65/403Arrangements for multi-party communication, e.g. for conferences

Definitions

  • the present invention relates to the field of communications technologies, and in particular, to a method for processing a conference call, a host telephone terminal, and a conference call system.
  • local conference call is a very common function. Unlike server conference call, local conference call does not require server support, and the site is established on the host phone terminal. Moreover, during the local conference call, the host telephone terminal can add a new conference party telephone terminal to the conference call at any time.
  • the present invention solves the problem of the prior party meeting when a new conference party telephone terminal is added to a conference call in the host telephone terminal in the prior art by providing a conference call processing method, a host telephone terminal, and a conference call system.
  • the telephone terminal cannot continue the technical problem of the conference call.
  • a method for processing a conference call including:
  • the host telephone terminal When the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, the host telephone terminal acquires a user operation, and the user operation is used to add a third participant telephone terminal to the office Call conference call;
  • the host telephone terminal responds to the user operation to add the third participant telephone terminal to the conference call, and while adding the third participant telephone terminal to the conference call, maintain the conference a first real-time transport protocol RTP channel established between the first participant-side telephone terminal and a second RTP channel established between the second-party participant telephone terminal, so that the first participant telephone terminal and the The second participant telephone terminal can continue the conference call.
  • the host telephone terminal responds to the user operation to add the third participant telephone terminal to the conference call, including:
  • the host telephone terminal responds to the user operation to establish a third RTP channel with the third participant telephone terminal;
  • the host telephone terminal acquires third party participant voice data of the third participant telephone terminal through the third RTP channel;
  • the host telephone terminal adds the third participant voice data to the conference site of the conference call, thereby implementing adding the third participant party terminal to the conference call.
  • the host phone maintains a first real-time transport protocol RTP channel established between the first party video terminal and the a second RTP channel established between the second participant telephone terminal to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call, including:
  • the host telephone terminal maintains the first RTP channel and the second RTP channel;
  • the host telephone terminal performs a mixing process on the current sound data in the conference site of the conference call to obtain first mix data; wherein the first mix data includes: a voice passing through the host phone terminal
  • the host phone terminal filters out the host sound data in the first mix data to obtain second mix data;
  • the host telephone terminal transmits the second mix data to the first RTP channel and the second RTP channel to receive the first participant telephone terminal and the second participant telephone terminal To the second mix data.
  • the host voice is filtered out in the first mix data at the host phone terminal Data, after obtaining the second mix data, the method further includes:
  • the host telephone terminal controls the second mix data to be in a forbidden output state, so that the second mix data cannot be output through the host voice output device of the host telephone terminal itself.
  • the host telephone terminal adds the third participant telephone terminal to the conference call,
  • the method also includes:
  • the host phone terminal performs a mixing process on the current voice data in the site to obtain a third mix data.
  • the third mix data includes: the host voice data, the first participant. Square sound data, the second participant sound data, and the third participant sound data;
  • the host phone terminal performs a mixing process on the current voice data in the conference site to obtain the first After the three mixing data, the method further includes:
  • the host telephone terminal filters out the host sound data in the third mix data to obtain fourth mix data;
  • the host telephone terminal outputs the fourth mix data through the host sound output device.
  • a host telephone terminal including:
  • An obtaining unit configured to acquire a user operation when the host phone terminal performs a conference call with the first participant phone terminal and the second participant phone terminal, where the user operation is used to add the third participant phone terminal to The conference call;
  • a holding unit configured to maintain a first real-time transport protocol RTP channel established between the first participant telephone terminal and the first party while adding the third participant telephone terminal to the conference call And a second RTP channel established between the two party telephone terminals to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call.
  • the adding unit include:
  • An acquiring module configured to acquire third party audio data of the third participant telephone terminal by using the third RTP channel
  • a adding module configured to add the third participant voice data to the conference site of the conference call, thereby implementing adding the third participant party terminal to the conference call.
  • the holding unit includes:
  • a holding module for holding the first RTP channel and the second RTP channel
  • a first mixing module configured to perform mixing processing on current sound data in a conference site of the conference call to obtain first mixing data; wherein the first mixing data includes: passing the host telephone terminal The host sound data collected by the sound collecting device, the first party sound data of the first participant telephone terminal obtained through the first RTP channel, and the obtained by the second RTP channel Second participant party voice data of the second participant party telephone terminal;
  • a first filtering module configured to filter the host sound data in the first mix data to obtain second mix data
  • a first sending module configured to send the second mix data to the first RTP channel and the second RTP channel, so that the first participant phone terminal and the second participant phone terminal Receiving the second mix data.
  • the holding unit further includes:
  • a prohibiting module configured to filter the host sound data in the first mix data to obtain the second mix data, and then control the second mix data to be in a forbidden output state, so as to The second mix data cannot be output through the host sound output device of the host phone terminal itself.
  • the host telephone terminal further includes:
  • the third mix data includes: the host sound data, the first participant sound data, the second participant sound data, and the third participant sound data;
  • a second sending module configured to send the third mix data to the first RTP channel, the second RTP channel, and the third RTP channel, so that the first participant telephone terminal, The second participant telephone terminal is connected, and the third participant telephone terminal receives the third mix data.
  • the host telephone terminal further includes:
  • a second filtering module configured to perform mixing processing on the current sound data in the conference site, and after obtaining the third mixing data, filtering the host sound data in the third mixing data , obtaining the fourth mix data;
  • an output module configured to output the fourth mix data through the host sound output device.
  • a host telephone terminal including:
  • a memory for storing program code
  • a processor coupled to the memory for reading the program code to perform:
  • the processor is further configured to:
  • the processor is further configured to:
  • the processor is further configured to:
  • the processor is further configured to:
  • the third mix data includes : the host sound data, the first participant sound data, the second participant sound data, and the third participant sound data; transmitting the third mix data to the first RTP a channel, the second RTP channel, and the third RTP channel to enable the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal to receive the Third mix data.
  • the processor is further configured to:
  • the fourth mix data is output by the host sound output device.
  • a teleconferencing system including:
  • a plurality of participant telephone terminals including: a first participant telephone terminal, a second participant telephone terminal, and a third participant telephone terminal;
  • the host telephone terminal is used to:
  • Obtaining a user operation when the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, and the user operation is used to add a third participant telephone terminal to the conference Determining a conference call; responding to the user operation to add the third participant telephone terminal to the conference call, and while adding the third participant telephone terminal to the conference call, maintaining the a first real-time transport protocol RTP channel established between the participant's telephone terminal and a second RTP channel established with the second participant's telephone terminal to enable the first participant telephone terminal and the first The second party telephone terminal can continue the conference call.
  • the host telephone terminal is further configured to:
  • the host telephone terminal is further configured to:
  • the host telephone terminal is further configured to:
  • the host telephone terminal is further configured to:
  • the third mix data includes : the host sound data, the first participant sound data, the second participant sound data, and the third participant sound data; transmitting the third mix data to the first RTP a channel, the second RTP channel, and the third RTP channel to enable the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal to receive the Third mix data.
  • the host telephone terminal is further configured to:
  • the fourth mix data is output by the host sound output device.
  • the host telephone terminal Since the host telephone terminal adds the third participant telephone terminal to the conference call, the host telephone terminal maintains the first RTP channel established between the first participant party telephone terminal and the second participant party telephone terminal. A second RTP channel is established to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call. Therefore, when the new conference party telephone terminal is added to the conference call by the host telephone terminal, the original conference party telephone terminal can continue the conference call through the RTP channel maintained by the host telephone terminal, thereby improving the original conference party telephone terminal. The technical effect of the user's experience.
  • FIG. 1 is a schematic diagram of a conference call system according to an embodiment of the present invention.
  • FIG. 2 is a flowchart of a method for processing a conference call according to an embodiment of the present invention
  • FIG. 3 is a detailed flowchart of step S102 in the embodiment of the present invention.
  • step S103 is a detailed flowchart of step S103 in the embodiment of the present invention.
  • FIG. 5 is a detailed flowchart of step S104 in the embodiment of the present invention.
  • FIG. 6 is a schematic structural diagram of a host telephone terminal according to an embodiment of the present invention.
  • FIG. 7 is a schematic structural diagram of a host telephone terminal according to an embodiment of the present invention.
  • the embodiment of the present invention provides a method for processing a conference call, a host telephone terminal, and a conference call system, which solves the problem in the prior art when a new conference party telephone terminal is added to the conference call by the host telephone terminal.
  • the telephone terminal cannot continue the technical problem of the conference call.
  • a method of processing a conference call comprising:
  • the host telephone terminal When the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, the host telephone terminal acquires a user operation, and the user operation is used to add a third participant telephone terminal to the office a conference call; the host telephone terminal responds to the user operation to add the third participant telephone terminal to the conference call, and while adding the third participant telephone terminal to the conference call Maintaining a first real-time transport protocol RTP channel established between the first participant party telephone terminal and a second RTP channel established with the second participant party telephone terminal to enable the first participant party The telephone terminal and the second participant telephone terminal can continue the conference call.
  • FIG. 1 is a conference call system for implementing a method for processing a conference call of the present invention, where
  • the utility model comprises: a host telephone terminal and a plurality of participant conference call terminals (for example: a first participant telephone terminal, a second participant party telephone terminal, and a third participant party telephone terminal, etc.).
  • the host telephone terminal and the conference call terminals
  • the host telephone terminal may be: an IP (Internet Protocol) telephone or other telephone terminal supporting the local conference call.
  • This embodiment provides a method for processing a conference call, as shown in FIG. 2, including:
  • Step S101 When the host telephone terminal makes a conference call with the first participant telephone terminal and the second participant telephone terminal, the host telephone terminal acquires a user operation, and the user operation is used to add the third participant telephone terminal to the conference call. ;
  • Step S102 the host telephone terminal responds to the user operation to add the third participant telephone terminal to the conference call;
  • Step S103 The host phone terminal maintains a first RTP (Real-time Transport Protocol) channel established between the first participant party and the first participant phone terminal while adding the third participant phone terminal to the conference call, and a second RTP channel established with the second participant telephone terminal to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call;
  • RTP Real-time Transport Protocol
  • Step S104 The host telephone terminal performs a conference call with the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal.
  • the host telephone terminal establishes a first RTP channel with the first participant telephone terminal, and establishes a second RTP channel with the second participant telephone terminal to interact with the first participant telephone terminal and the second participant party.
  • Conducting (three-way) conference call of the telephone terminal when the host phone terminal acquires a user operation for adding the third participant terminal to the conference call, the host phone terminal adds the third participant phone terminal to the conference call, and the third party attends the conference at the host phone terminal.
  • Square electricity While the terminal is in the conference call the host telephone terminal maintains a first RTP channel established between the first participant party telephone terminal and a second RTP channel established with the second participant party telephone terminal to make the first The participant telephone terminal and the second participant telephone terminal can continue the conference call.
  • the host telephone terminal can be performed with the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal (quartet). telephone conference.
  • the specific implementation of the host phone terminal acquiring the user operation may be:
  • an input device such as a function button (or a touch screen) is provided on the host phone terminal, when it is detected that the function button is pressed (or a specific area of the touch screen is touched), the user operation can be confirmed.
  • the host telephone terminal when the host party telephone terminal adds a new conference party telephone terminal to the conference call, the host telephone terminal releases the RTP channel established with the original conference party, and completes adding the new conference party telephone terminal to the conference call. At the same time, the RTP channel established with the original party is restored.
  • the original conference party telephone terminal for example, the first participant party telephone terminal and the second participant party telephone terminal
  • the host telephone terminal while the host telephone terminal adds the third participant telephone terminal to the conference call, the host telephone terminal maintains the first RTP channel established between the first participant telephone terminal and the first A second RTP channel established between the two party telephone terminals to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call. Therefore, the prior art solves the technical problem that the original conference party telephone terminal cannot continue the conference call when the host party telephone terminal adds a new conference party telephone terminal to the conference call.
  • the new conference party telephone terminal is added to the conference call by the host telephone terminal, the original conference party telephone terminal can continue the conference call through the RTP channel maintained by the host telephone terminal, thereby improving the user of the original conference party telephone terminal.
  • the technical effect of the experience is described in the user of the original conference party telephone terminal.
  • step S102 includes:
  • Step S201 the host phone terminal responds to the user operation to establish a third RTP channel with the third party phone terminal;
  • Step S202 The host telephone terminal acquires third party participant voice data of the third participant party terminal through the third RTP channel;
  • Step S203 The host telephone terminal adds the third party voice data to the conference site of the conference call, thereby implementing the third conference party telephone terminal to be added to the conference call.
  • the host telephone terminal can obtain the telephone number of the third participant telephone terminal, and call the third participant telephone terminal based on the telephone number of the third participant telephone terminal. After the third party phone terminal answers, the host phone terminal establishes a third RTP channel with the third party phone terminal. At this time, the host user can interact with the third party user through the third RTP channel for voice data. .
  • the host telephone terminal acquires another user operation of the user (the user operation is used to determine to add the third participant telephone terminal to the conference call), and the host telephone terminal acquires the third participant telephone terminal through the third RTP channel.
  • the third party party voice data is added to the conference site of the conference call to perform the mixing process, thereby adding the third party phone terminal to the conference call.
  • the host phone terminal establishes a third RTP channel with the third party phone terminal, and adds the third party voice data from the third RTP channel to the conference site of the conference call for mixing processing.
  • the host phone terminal establishes a third RTP channel with the third party phone terminal, and adds the third party voice data from the third RTP channel to the conference site of the conference call for mixing processing.
  • step S103 includes:
  • Step S301 the host phone maintains the first RTP channel and the second RTP channel
  • Step S302 The host telephone terminal performs a mixing process on the current voice data in the conference site of the conference call to obtain the first mix data.
  • the first mix data includes: the host voice data collected by the voice collection device of the host phone terminal, the first participant voice data of the first participant phone terminal obtained through the first RTP channel, and Second party audio data of the second participant telephone terminal obtained by the second RTP channel;
  • Step S303 The host phone terminal filters out the host voice data in the first mix data to obtain the second mix data.
  • Step S304 The host telephone terminal sends the second mix data to the first RTP channel and the second RTP channel, so that the first participant phone terminal and the second participant phone terminal receive the second mix data.
  • the host telephone terminal includes a call control module and a media engine module
  • the call control module notifies the media engine module to maintain the first RTP channel and the second RTP channel, and the media engine module responds to the notification of the call control module to maintain the first An RTP channel and a second RTP channel
  • the call control module notifies the body engine module to stop transmitting the moderator voice data to the first participant phone terminal and the second participant phone terminal, and the media engine module responds to the notification of the call control module
  • the current sound data in the venue is subjected to mixing processing to obtain first mixed data (wherein the first mixed sound data includes: host sound data, first party sound data, and second participant sound data), and filtering
  • the second mix data is obtained, and the second mix data is sent to the first RTP channel and the second RTP channel.
  • the host phone maintains the first RTP channel and the second RTP channel, and performs sound mixing processing on the current voice data in the conference site of the conference call, and passes the first RTP channel and the first
  • the second RTP channel sends the obtained mixing data to the first participant telephone terminal and the second participant telephone terminal, thereby realizing the addition of the third participant telephone terminal to the conference call at the host telephone terminal, and the first participant party The technical effect of the telephone conference and the second participant telephone terminal continuing the conference call.
  • the host side voice data is not included in the mix data received by the first participant party terminal and the second participant party terminal, the host party is eliminated from adding the third party.
  • the operation sound (or other noise) generated by the telephone terminal to the conference call interferes with the user of the first participant's telephone terminal and the user of the second participant's telephone terminal, thereby ensuring the first participant's telephone terminal and the second participant The quality of the call between the party's telephone terminals.
  • step S303 after step S303 (or while performing step S304), the method further includes:
  • Step S305 The host telephone terminal controls the second mix data to be in a forbidden output state, so that the second mix data cannot be output through the host voice output device of the host telephone terminal itself.
  • the call control module in the host phone terminal notifies the media engine module of the host phone terminal to prohibit playing the second mix data, and the media engine module controls the second mix data to be in the forbidden output state in response to the notification of the control module. So that the second mix data cannot be output through the host sound output device of the host telephone terminal itself.
  • the user of the host telephone terminal can make a call with the user of the third participant telephone terminal.
  • the host telephone terminal The self-host sound output device is prohibited from outputting the second mix data, and the second mix data can be prevented from affecting the call between the user of the host telephone terminal and the user of the third participant telephone terminal.
  • step S104 includes:
  • Step S401 The host phone terminal performs a mixing process on the current voice data in the site to obtain a third mix data.
  • the third mix data includes: the host voice data, the first participant voice data, and the second participant. Square sound data, and third party sound data;
  • Step S402 The host telephone terminal sends the third mix data to the first RTP channel, the RTP channel, and the third RTP channel, so that the first participant telephone terminal, the second participant telephone terminal, and the third participant The party telephone terminal receives the third mix data.
  • the call control module notifies the media engine module to resume sending the host voice data to the first participant phone terminal and the second participant phone terminal, and the media engine module responds to the notification of the call control module to the current voice data in the conference site.
  • the media engine module transmits the third mix data to the first RTP channel, the second RTP channel, and the third RTP channel.
  • the method further includes:
  • Step S403 The host phone terminal filters out the host sound data in the third mix data to obtain the fourth mix data;
  • Step S404 The host telephone terminal outputs the fourth mix data through the host sound output device.
  • the host phone terminal when playing the third mix data, the host phone terminal first filters out the host sound data in the third mix data, obtains the fourth mix data, and outputs the sound through the host sound output device. Fourth mix data.
  • the host voice data is not filtered, the host party's telephone terminal output host sound data will be superimposed with the voice of the user's mouth of the host telephone terminal, appearing The echo phenomenon, so that the user of the host telephone terminal cannot hear the voice information in the conference call.
  • the host phone terminal when playing the third mix data, the host phone terminal first filters the host sound data in the third mix data to obtain the fourth mix data, and then outputs the sound through the host mode.
  • the device outputs the fourth mix data.
  • This embodiment provides a host phone terminal, as shown in FIG. 6, including:
  • the obtaining unit 501 is configured to acquire a user operation when the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, and the user operates to add the third participant telephone terminal to the conference call;
  • An adding unit 502 configured to respond to a user operation to add a third party phone terminal to the conference call;
  • the maintaining unit 503 is configured to maintain a first real-time transport protocol RTP channel established between the first participant party telephone terminal and the second participant party telephone terminal while adding the third participant party terminal to the conference call Establishing a second RTP channel to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call;
  • the performing unit 504 is configured to perform a conference call with the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal.
  • the host telephone terminal Since the host telephone terminal adds the third participant telephone terminal to the conference call, the host telephone terminal maintains the first RTP channel established between the first participant party telephone terminal and the second participant party telephone terminal. Establishing a second RTP channel to enable the first party phone terminal and The second participant telephone terminal can continue the conference call. Therefore, when the new conference party telephone terminal is added to the conference call by the host telephone terminal, the original conference party telephone terminal can continue the conference call through the RTP channel maintained by the host telephone terminal, thereby improving the original conference party telephone terminal. The technical effect of the user's experience.
  • the adding unit 502 includes:
  • An acquiring module configured to acquire, by using the third RTP channel, third party audio data of the third party telephone terminal;
  • the adding module is configured to add the third party voice data to the conference site of the conference call, thereby adding the third party phone terminal to the conference call.
  • the holding unit 503 includes:
  • a holding module for maintaining the first RTP channel and the second RTP channel
  • a first mixing module configured to perform a mixing process on the current sound data in the conference site of the conference call to obtain the first sound mixing data; wherein the first sound mixing data includes: collected by the sound collecting device of the host telephone terminal The host sound data to the first participant party voice data of the first participant party terminal obtained through the first RTP channel, and the second participant party voice data of the second participant party terminal obtained through the second RTP channel;
  • a first filtering module configured to filter the host sound data in the first mix data to obtain the second mix data
  • the first sending module is configured to send the second mix data to the first RTP channel and the second RTP channel, so that the first participant phone terminal and the second participant phone terminal receive the second mix data.
  • the holding unit 503 further includes:
  • a prohibiting module configured to filter the host sound data in the first mix data, obtain the second mix data, and control the second mix data to be in a forbidden output state, so that the second mix data cannot pass the host The host terminal's own voice output device output.
  • the performing unit 504 includes:
  • a second mixing module configured to: after adding the third participant telephone terminal to the conference call, perform mixing processing on the current sound data in the conference site to obtain third mixing data; wherein the third mixing data includes: a host Sound data, first party sound data, second party sound data, and third party sound data;
  • a second sending module configured to send the third mixing data to the first RTP channel, the second RTP channel, and the third RTP channel, so that the first participant telephone terminal, the second participant telephone terminal, and the The third party telephone terminal receives the third mix data.
  • the performing unit 504 further includes:
  • a second filtering module configured to perform mixing processing on the current sound data in the conference site, and after obtaining the third mixing data, filtering the host sound data in the third mixing data to obtain the fourth mixing data;
  • an output module configured to output the fourth mix data through the host sound output device.
  • the embodiment provides a host telephone terminal, as shown in FIG. 7, including:
  • a memory 601 configured to store program code
  • the processor 602 is coupled to the memory 601 via a bus 605 for reading program code to perform:
  • the host telephone terminal Since the host telephone terminal adds the third participant telephone terminal to the conference call, the host telephone terminal maintains the first RTP channel established between the first participant party telephone terminal and the second participant party telephone terminal. A second RTP channel is established to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call. Therefore, when the new conference party telephone terminal is added to the conference call by the host telephone terminal, the original conference party telephone terminal can continue the conference call through the RTP channel maintained by the host telephone terminal, thereby improving the original conference party telephone terminal. The technical effect of the user's experience.
  • the processor 602 is further configured to:
  • the processor 602 is further configured to:
  • the first sound mixing data includes: sound through the host telephone terminal The host sound data collected by the collecting device, the first party sound data of the first participant party terminal obtained through the first RTP channel, and the second RTP pass The second participant party voice data of the second participant party telephone terminal obtained by the track; filtering the host party voice data in the first mix data to obtain the second mix data; and transmitting the second mix data to the first RTP channel and In the second RTP channel, the first participant telephone terminal and the second participant telephone terminal receive the second mix data.
  • the processor 602 is further configured to:
  • the processor 602 is further configured to:
  • the third voice data includes: the host voice data, and the first party. Sound data, second party sound data, and third party sound data; transmitting the third sound data to the first RTP channel, the second RTP channel, and the third RTP channel to make the first participant telephone terminal
  • the second party telephone terminal is connected, and the third party telephone terminal receives the third mixing data.
  • the processor 602 is further configured to:
  • the embodiment provides a conference call system, as shown in FIG. 1 , including:
  • a plurality of participant telephone terminals including: a first participant telephone terminal, a second participant telephone terminal, and a third participant telephone terminal;
  • the host telephone terminal is used to:
  • the host telephone terminal Since the host telephone terminal adds the third participant telephone terminal to the conference call, the host telephone terminal maintains the first RTP channel established between the first participant party telephone terminal and the second participant party telephone terminal. A second RTP channel is established to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call. Therefore, when the new conference party telephone terminal is added to the conference call by the host telephone terminal, the original conference party telephone terminal can continue the conference call through the RTP channel maintained by the host telephone terminal, thereby improving the original conference party telephone terminal. The technical effect of the user's experience.
  • the host phone terminal is further configured to:
  • the host phone terminal is further configured to:
  • the first sound mixing data includes: sound through the host telephone terminal The host sound data collected by the collecting device, the first participant party voice data of the first participant party terminal obtained through the first RTP channel, and the second participant meeting of the second participant party terminal obtained through the second RTP channel Square sound data; filter the host sound data in the first mix data to obtain the second mix data; send the second mix data to the first RTP channel and the second RTP channel to make the first party call
  • the terminal and the second participant telephone terminal receive the second mix data.
  • the host phone terminal is further configured to:
  • the host phone terminal is further configured to:
  • the third voice data includes: the host voice data, and the first party. Sound data, second party sound data, and third party sound data; transmitting the third sound data to the first RTP channel, the second RTP channel, and the third RTP channel to make the first participant telephone terminal
  • the second party telephone terminal is connected, and the third party telephone terminal receives the third mixing data.
  • the host phone terminal is further configured to:
  • the host sound data in the third sound mixing data is filtered out to obtain the fourth sound mixing data;
  • the device outputs the fourth mix data.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
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  • General Business, Economics & Management (AREA)
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Abstract

Disclosed are a teleconferencing method, host telephone terminal, and teleconferencing system. The teleconferencing method comprises: when a host telephone terminal conducts teleconferencing with a first participant telephone terminal and a second participant telephone terminal, the host telephone terminal acquires a user operation used for adding a third participant telephone terminal to the teleconference; the host telephone terminal adds the third participant telephone terminal to the teleconference in response to the user operation, while maintaining a first RTP channel with the first participant telephone terminal, and a second RTP channel with the second participant telephone terminal, so that the first participant telephone terminal and the second participant telephone terminal can continue the teleconference.

Description

视频会议方法、装置及系统Video conference method, device and system
本申请要求于2014年3月20日提交中国专利局、申请号为201410105068.6、发明名称为“一种电话会议的处理方法、主持方电话终端、及电话会议系统”的中国专利申请的优先权,其全部内容通过引用结合在本申请中。This application claims the priority of the Chinese patent application filed on March 20, 2014, the Chinese Patent Office, the application number is 201410105068.6, and the invention name is “a method of processing a conference call, a telephone terminal of a host, and a conference call system”. The entire contents of this application are incorporated herein by reference.
技术领域Technical field
本发明涉及通信技术领域,尤其涉及一种电话会议的处理方法、主持方电话终端、及电话会议系统。The present invention relates to the field of communications technologies, and in particular, to a method for processing a conference call, a host telephone terminal, and a conference call system.
背景技术Background technique
随着现代社会的发展,交流沟通所涉及的范围越来越广。各类商业、政治活动需要在更广阔的区域进行沟通交流,例如:召开会议、协同工作等等。这些分布式沟通是完成各类商业、政治活动的重要工作。为了顺应这种市场潮流的发展,电话会议作为一个新兴的产品为分布式沟通提供了很大的便利。With the development of modern society, the scope of communication is becoming wider and wider. All kinds of commercial and political activities need to communicate in a wider area, such as: holding meetings, working together, and so on. These distributed communications are important tasks in completing various commercial and political activities. In order to comply with the development of this market trend, the conference call as a new product provides great convenience for distributed communication.
在企业通信中,本地电话会议是非常常用的功能,与服务器电话会议不同,本地电话会议不需要服务器的支持,会场建立在主持方电话终端上。且,在进行本地电话会议时,主持方电话终端可以随时添加新的与会方电话终端到电话会议中。In enterprise communication, local conference call is a very common function. Unlike server conference call, local conference call does not require server support, and the site is established on the host phone terminal. Moreover, during the local conference call, the host telephone terminal can add a new conference party telephone terminal to the conference call at any time.
但在现有技术中,在主持方电话终端添加新的与会方电话终端到电话会议时,会出现原与会方电话终端无法继续进行电话会议的问题。However, in the prior art, when a new participant's telephone terminal is added to the conference call at the host telephone terminal, there is a problem that the original conference party telephone terminal cannot continue the conference call.
发明内容Summary of the invention
本发明通过提供一种电话会议的处理方法、主持方电话终端、及电话会议系统,解决了现有技术中在主持方电话终端添加新的与会方电话终端到电话会议时,会出现原与会方电话终端无法继续进行电话会议的技术问题。 The present invention solves the problem of the prior party meeting when a new conference party telephone terminal is added to a conference call in the host telephone terminal in the prior art by providing a conference call processing method, a host telephone terminal, and a conference call system. The telephone terminal cannot continue the technical problem of the conference call.
第一方面,提供一种电话会议的处理方法,包括:In a first aspect, a method for processing a conference call is provided, including:
在主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,所述主持方电话终端获取用户操作,所述用户操作用于添加第三与会方电话终端到所述电话会议;When the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, the host telephone terminal acquires a user operation, and the user operation is used to add a third participant telephone terminal to the office Call conference call;
所述主持方电话终端响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议,并在添加所述第三与会方电话终端到所述电话会议的同时,保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议。The host telephone terminal responds to the user operation to add the third participant telephone terminal to the conference call, and while adding the third participant telephone terminal to the conference call, maintain the conference a first real-time transport protocol RTP channel established between the first participant-side telephone terminal and a second RTP channel established between the second-party participant telephone terminal, so that the first participant telephone terminal and the The second participant telephone terminal can continue the conference call.
结合第一方面,在第一方面的第一种可能的实施方式中,所述主持方电话终端响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议,包括:In conjunction with the first aspect, in a first possible implementation manner of the first aspect, the host telephone terminal responds to the user operation to add the third participant telephone terminal to the conference call, including:
所述主持方电话终端响应所述用户操作,以与所述第三与会方电话终端建立第三RTP通道;The host telephone terminal responds to the user operation to establish a third RTP channel with the third participant telephone terminal;
所述主持电话终端通过所述第三RTP通道获取所述第三与会方电话终端的第三与会方声音数据;The host telephone terminal acquires third party participant voice data of the third participant telephone terminal through the third RTP channel;
所述主持电话终端将所述第三与会方声音数据添加到所述电话会议的会场中,进而实现将所述第三与会方电话终端添加到所述电话会议。The host telephone terminal adds the third participant voice data to the conference site of the conference call, thereby implementing adding the third participant party terminal to the conference call.
结合第一方面,在第一方面的第二种可能的实施方式中,所述主持方电话保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议,包括: In conjunction with the first aspect, in a second possible implementation of the first aspect, the host phone maintains a first real-time transport protocol RTP channel established between the first party video terminal and the a second RTP channel established between the second participant telephone terminal to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call, including:
所述主持方电话终端保持所述第一RTP通道、以及所述第二RTP通道;The host telephone terminal maintains the first RTP channel and the second RTP channel;
所述主持方电话终端对所述电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,所述第一混音数据包括:通过所述主持方电话终端的声音采集装置所采集到的主持方声音数据,通过所述第一RTP通道获得的所述第一与会方电话终端的第一与会方声音数据,以及通过所述第二RTP通道获得的所述第二与会方电话终端的第二与会方声音数据;The host telephone terminal performs a mixing process on the current sound data in the conference site of the conference call to obtain first mix data; wherein the first mix data includes: a voice passing through the host phone terminal The host sound data collected by the collecting device, the first participant sound data of the first participant telephone terminal obtained through the first RTP channel, and the second obtained through the second RTP channel Second party audio data of the participant's telephone terminal;
所述主持方电话终端滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据;The host phone terminal filters out the host sound data in the first mix data to obtain second mix data;
所述主持方电话终端发送所述第二混音数据到所述第一RTP通道和所述第二RTP通道中,以使所述第一与会方电话终端和所述第二与会方电话终端接收到所述第二混音数据。The host telephone terminal transmits the second mix data to the first RTP channel and the second RTP channel to receive the first participant telephone terminal and the second participant telephone terminal To the second mix data.
结合第一方面的第二种可能的实施方式,在第一方面的第三种可能的实施方式中,在所述主持方电话终端滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据之后,所述方法还包括:In conjunction with the second possible implementation of the first aspect, in a third possible implementation of the first aspect, the host voice is filtered out in the first mix data at the host phone terminal Data, after obtaining the second mix data, the method further includes:
所述主持方电话终端控制所述第二混音数据处于禁止输出状态,以使所述第二混音数据不能通过所述主持方电话终端自身的主持方声音输出装置输出。The host telephone terminal controls the second mix data to be in a forbidden output state, so that the second mix data cannot be output through the host voice output device of the host telephone terminal itself.
结合第一方面的第三种可能的实施方式,在第一方面的第四种可能的实施方式中,所述主持方电话终端添加所述第三与会方电话终端到所述电话会议之后,所述方法还包括:In conjunction with the third possible implementation manner of the first aspect, in a fourth possible implementation manner of the first aspect, the host telephone terminal adds the third participant telephone terminal to the conference call, The method also includes:
所述主持方电话终端对所述会场中的当前声音数据进行混音处理,获得第三混音数据;其中,所述第三混音数据包括:所述主持方声音数据、所述第一与会方声音数据、所述第二与会方声音数据、以及所述第三与会方声音数据; The host phone terminal performs a mixing process on the current voice data in the site to obtain a third mix data. The third mix data includes: the host voice data, the first participant. Square sound data, the second participant sound data, and the third participant sound data;
所述主持方电话终端发送所述第三混音数据到所述第一RTP通道、所述第二RTP通道、以及所述第三RTP通道中,以使所述第一与会方电话终端、第二与会方电话终端接、以及所述第三与会方电话终端收到所述第三混音数据。Transmitting, by the host telephone terminal, the third mix data to the first RTP channel, the second RTP channel, and the third RTP channel, so that the first participant telephone terminal, The second party phone terminal is connected, and the third party phone terminal receives the third mix data.
结合第一方面的第四种可能的实施方式,在第一方面的第五种可能的实施方式中,在所述主持方电话终端对所述会场中的当前声音数据进行混音处理,获得第三混音数据之后,所述方法还包括:With reference to the fourth possible implementation manner of the first aspect, in a fifth possible implementation manner of the first aspect, the host phone terminal performs a mixing process on the current voice data in the conference site to obtain the first After the three mixing data, the method further includes:
所述主持方电话终端滤除所述第三混音数据中的所述主持方声音数据,获得第四混音数据;The host telephone terminal filters out the host sound data in the third mix data to obtain fourth mix data;
所述主持方电话终端通过所述主持方声音输出装置输出所述第四混音数据。The host telephone terminal outputs the fourth mix data through the host sound output device.
第二方面,基于同一发明构思,提供一种主持方电话终端,包括:In a second aspect, based on the same inventive concept, a host telephone terminal is provided, including:
获取单元,用于在所述主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,获取用户操作,所述用户操作用于添加第三与会方电话终端到所述电话会议;An obtaining unit, configured to acquire a user operation when the host phone terminal performs a conference call with the first participant phone terminal and the second participant phone terminal, where the user operation is used to add the third participant phone terminal to The conference call;
添加单元,用于响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议;Adding a unit for responding to the user operation to add the third participant telephone terminal to the conference call;
保持单元,用于在添加所述第三与会方电话终端到所述电话会议的同时,保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议。a holding unit, configured to maintain a first real-time transport protocol RTP channel established between the first participant telephone terminal and the first party while adding the third participant telephone terminal to the conference call And a second RTP channel established between the two party telephone terminals to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call.
结合第二方面,在第二方面的第一种可能的实施方式中,所述添加单元, 包括:With reference to the second aspect, in a first possible implementation manner of the second aspect, the adding unit, include:
建立模块,用于响应所述用户操作,以与所述第三与会方电话终端建立第三RTP通道;Establishing a module, configured to respond to the user operation to establish a third RTP channel with the third participant telephone terminal;
获取模块,用于通过所述第三RTP通道获取所述第三与会方电话终端的第三与会方声音数据;An acquiring module, configured to acquire third party audio data of the third participant telephone terminal by using the third RTP channel;
添加模块,用于将所述第三与会方声音数据添加到所述电话会议的会场中,进而实现将所述第三与会方电话终端添加到所述电话会议。And a adding module, configured to add the third participant voice data to the conference site of the conference call, thereby implementing adding the third participant party terminal to the conference call.
结合第二方面,在第二方面的第二种可能的实施方式中,所述保持单元,包括:With reference to the second aspect, in a second possible implementation manner of the second aspect, the holding unit includes:
保持模块,用于保持所述第一RTP通道、以及所述第二RTP通道;a holding module for holding the first RTP channel and the second RTP channel;
第一混音模块,用于对所述电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,所述第一混音数据包括:通过所述主持方电话终端的声音采集装置所采集到的主持方声音数据,通过所述第一RTP通道获得的所述第一与会方电话终端的第一与会方声音数据,以及通过所述第二RTP通道获得的所述第二与会方电话终端的第二与会方声音数据;a first mixing module, configured to perform mixing processing on current sound data in a conference site of the conference call to obtain first mixing data; wherein the first mixing data includes: passing the host telephone terminal The host sound data collected by the sound collecting device, the first party sound data of the first participant telephone terminal obtained through the first RTP channel, and the obtained by the second RTP channel Second participant party voice data of the second participant party telephone terminal;
第一滤除模块,用于滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据;a first filtering module, configured to filter the host sound data in the first mix data to obtain second mix data;
第一发送模块,用于发送所述第二混音数据到所述第一RTP通道和所述第二RTP通道中,以使所述第一与会方电话终端和所述第二与会方电话终端接收到所述第二混音数据。a first sending module, configured to send the second mix data to the first RTP channel and the second RTP channel, so that the first participant phone terminal and the second participant phone terminal Receiving the second mix data.
结合第二方面的第二种可能的实施方式,在第二方面的第三种可能的实施方式中,所述保持单元,还包括: In conjunction with the second possible implementation of the second aspect, in the third possible implementation of the second aspect, the holding unit further includes:
禁止模块,用于在所述滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据之后,控制所述第二混音数据处于禁止输出状态,以使所述第二混音数据不能通过所述主持方电话终端自身的主持方声音输出装置输出。a prohibiting module, configured to filter the host sound data in the first mix data to obtain the second mix data, and then control the second mix data to be in a forbidden output state, so as to The second mix data cannot be output through the host sound output device of the host phone terminal itself.
结合第二方面的第三种可能的实施方式,在第二方面的第四种可能的实施方式中,所述主持方电话终端,还包括:In conjunction with the third possible implementation of the second aspect, in a fourth possible implementation of the second aspect, the host telephone terminal further includes:
第二混音模块,用于在所述添加所述第三与会方电话终端到所述电话会议之后,对所述会场中的当前声音数据进行混音处理,获得第三混音数据;其中,所述第三混音数据包括:所述主持方声音数据、所述第一与会方声音数据、所述第二与会方声音数据、以及所述第三与会方声音数据;a second mixing module, configured to perform a mixing process on the current sound data in the site after the adding the third party phone terminal to the conference call, to obtain a third mixing data; The third mix data includes: the host sound data, the first participant sound data, the second participant sound data, and the third participant sound data;
第二发送模块,用于发送所述第三混音数据到所述第一RTP通道、所述第二RTP通道、以及所述第三RTP通道中,以使所述第一与会方电话终端、第二与会方电话终端接、以及所述第三与会方电话终端收到所述第三混音数据。a second sending module, configured to send the third mix data to the first RTP channel, the second RTP channel, and the third RTP channel, so that the first participant telephone terminal, The second participant telephone terminal is connected, and the third participant telephone terminal receives the third mix data.
结合第二方面的第四种可能的实施方式,在第二方面的第五种可能的实施方式中,所述主持方电话终端,还包括:In conjunction with the fourth possible implementation of the second aspect, in a fifth possible implementation manner of the second aspect, the host telephone terminal further includes:
第二滤除模块,用于在所述对所述会场中的当前声音数据进行混音处理,获得第三混音数据之后,滤除所述第三混音数据中的所述主持方声音数据,获得第四混音数据;a second filtering module, configured to perform mixing processing on the current sound data in the conference site, and after obtaining the third mixing data, filtering the host sound data in the third mixing data , obtaining the fourth mix data;
输出模块,用于通过所述主持方声音输出装置输出所述第四混音数据。And an output module, configured to output the fourth mix data through the host sound output device.
第三方面,基于同一发明构思,提供一种主持方电话终端,包括:In a third aspect, based on the same inventive concept, a host telephone terminal is provided, including:
存储器,用于存储程序代码;a memory for storing program code;
处理器,与所述存储器连接,用于读取所述程序代码,以执行: a processor coupled to the memory for reading the program code to perform:
在所述主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,获取用户操作,所述用户操作用于添加第三与会方电话终端到所述电话会议;响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议,并在添加所述第三与会方电话终端到所述电话会议的同时,保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议。Obtaining a user operation when the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, and the user operation is used to add a third participant telephone terminal to the conference call; Responding to the user operation to add the third party phone terminal to the conference call, and to maintain the first party call while adding the third party phone terminal to the conference call a first real-time transport protocol RTP channel established between the terminals, and a second RTP channel established between the second participant party and the second participant telephone terminal, so that the first participant telephone terminal and the second participant telephone The terminal can proceed with the conference call.
结合第三方面,在第三方面的第一种可能的实施方式中,所述处理器,还用于:In conjunction with the third aspect, in a first possible implementation manner of the third aspect, the processor is further configured to:
响应所述用户操作,以与所述第三与会方电话终端建立第三RTP通道;通过所述第三RTP通道获取所述第三与会方电话终端的第三与会方声音数据;将所述第三与会方声音数据添加到所述电话会议的会场中,进而实现将所述第三与会方电话终端添加到所述电话会议。Responding to the user operation to establish a third RTP channel with the third participant telephone terminal; acquiring third party audio data of the third participant telephone terminal through the third RTP channel; The third party voice data is added to the conference site of the conference call, thereby implementing adding the third party phone terminal to the conference call.
结合第三方面,在第三方面的第二种可能的实施方式中,所述处理器,还用于:In conjunction with the third aspect, in a second possible implementation manner of the third aspect, the processor is further configured to:
保持所述第一RTP通道、以及所述第二RTP通道;对所述电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,所述第一混音数据包括:通过所述主持方电话终端的声音采集装置所采集到的主持方声音数据,通过所述第一RTP通道获得的所述第一与会方电话终端的第一与会方声音数据,以及通过所述第二RTP通道获得的所述第二与会方电话终端的第二与会方声音数据;滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据;发送所述第二混音数据到所述第一RTP通道和所述第二RTP通道 中,以使所述第一与会方电话终端和所述第二与会方电话终端接收到所述第二混音数据。Maintaining the first RTP channel and the second RTP channel; performing mixing processing on current sound data in the conference site of the conference call to obtain first mix data; wherein the first mix data includes : the first party audio data of the first participant telephone terminal obtained through the first RTP channel by the moderator voice data collected by the voice collection device of the host phone terminal, and by the Second participant audio data of the second participant telephone terminal obtained by the second RTP channel; filtering the host voice data in the first mix data to obtain second mix data; Second mix data to the first RTP channel and the second RTP channel And wherein the first participant telephone terminal and the second participant telephone terminal receive the second mix data.
结合第三方面的第二种可能的实施方式,在第三方面的第三种可能的实施方式中,所述处理器,还用于:In conjunction with the second possible implementation of the third aspect, in a third possible implementation of the third aspect, the processor is further configured to:
在所述滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据之后,控制所述第二混音数据处于禁止输出状态,以使所述第二混音数据不能通过所述主持方电话终端自身的主持方声音输出装置输出。After filtering the host sound data in the first mix data to obtain the second mix data, controlling the second mix data to be in a forbidden output state, so that the second mix The data cannot be output through the host sound output device of the host telephone terminal itself.
结合第三方面的第三种可能的实施方式,在第三方面的第四种可能的实施方式中,所述处理器,还用于:In conjunction with the third possible implementation of the third aspect, in a fourth possible implementation manner of the third aspect, the processor is further configured to:
在所述添加所述第三与会方电话终端到所述电话会议之后,对所述会场中的当前声音数据进行混音处理,获得第三混音数据;其中,所述第三混音数据包括:所述主持方声音数据、所述第一与会方声音数据、所述第二与会方声音数据、以及所述第三与会方声音数据;发送所述第三混音数据到所述第一RTP通道、所述第二RTP通道、以及所述第三RTP通道中,以使所述第一与会方电话终端、第二与会方电话终端接、以及所述第三与会方电话终端收到所述第三混音数据。After the adding the third participant telephone terminal to the conference call, performing mixing processing on the current voice data in the conference site to obtain third mix data; wherein the third mix data includes : the host sound data, the first participant sound data, the second participant sound data, and the third participant sound data; transmitting the third mix data to the first RTP a channel, the second RTP channel, and the third RTP channel to enable the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal to receive the Third mix data.
结合第三方面的第四种可能的实施方式,在第三方面的第五种可能的实施方式中,所述处理器,还用于:In conjunction with the fourth possible implementation of the third aspect, in a fifth possible implementation manner of the third aspect, the processor is further configured to:
在所述对所述会场中的当前声音数据进行混音处理,获得第三混音数据之后,滤除所述第三混音数据中的所述主持方声音数据,获得第四混音数据;通过所述主持方声音输出装置输出所述第四混音数据。Performing a mixing process on the current sound data in the meeting site to obtain the third mixing data, filtering the host sound data in the third mixing data to obtain a fourth mixing data; The fourth mix data is output by the host sound output device.
第四方面,基于同一发明构思,提供一种电话会议系统,包括: In a fourth aspect, based on the same inventive concept, a teleconferencing system is provided, including:
主持方电话终端;Host telephone terminal;
多个与会方电话终端,包括:第一与会方电话终端、第二与会方电话终端、和第三与会方电话终端;a plurality of participant telephone terminals, including: a first participant telephone terminal, a second participant telephone terminal, and a third participant telephone terminal;
其中,所述主持方电话终端,用于:Wherein, the host telephone terminal is used to:
在所述主持方电话终端与所述第一与会方电话终端、以及所述第二与会方电话终端进行电话会议时,获取用户操作,所述用户操作用于添加第三与会方电话终端到所述电话会议;响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议,并在添加所述第三与会方电话终端到所述电话会议的同时,保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议。Obtaining a user operation when the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, and the user operation is used to add a third participant telephone terminal to the conference Determining a conference call; responding to the user operation to add the third participant telephone terminal to the conference call, and while adding the third participant telephone terminal to the conference call, maintaining the a first real-time transport protocol RTP channel established between the participant's telephone terminal and a second RTP channel established with the second participant's telephone terminal to enable the first participant telephone terminal and the first The second party telephone terminal can continue the conference call.
结合第四方面,在第四方面的第一种可能的实施方式中,所述主持方电话终端,还用于:In conjunction with the fourth aspect, in a first possible implementation manner of the fourth aspect, the host telephone terminal is further configured to:
响应所述用户操作,以与所述第三与会方电话终端建立第三RTP通道;通过所述第三RTP通道获取所述第三与会方电话终端的第三与会方声音数据;将所述第三与会方声音数据添加到所述电话会议的会场中,进而实现将所述第三与会方电话终端添加到所述电话会议。Responding to the user operation to establish a third RTP channel with the third participant telephone terminal; acquiring third party audio data of the third participant telephone terminal through the third RTP channel; The third party voice data is added to the conference site of the conference call, thereby implementing adding the third party phone terminal to the conference call.
结合第四方面,在第四方面的第二种可能的实施方式中,所述主持方电话终端,还用于:In conjunction with the fourth aspect, in a second possible implementation manner of the fourth aspect, the host telephone terminal is further configured to:
保持所述第一RTP通道、以及所述第二RTP通道;对所述电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,所述第一混音数据包括:通过所述主持方电话终端的声音采集装置所采集到的主持方声音数 据,通过所述第一RTP通道获得的所述第一与会方电话终端的第一与会方声音数据,以及通过所述第二RTP通道获得的所述第二与会方电话终端的第二与会方声音数据;滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据;发送所述第二混音数据到所述第一RTP通道和所述第二RTP通道中,以使所述第一与会方电话终端和所述第二与会方电话终端接收到所述第二混音数据。Maintaining the first RTP channel and the second RTP channel; performing mixing processing on current sound data in the conference site of the conference call to obtain first mix data; wherein the first mix data includes : the number of host sounds collected by the sound collection device of the host telephone terminal According to the first participant audio terminal data of the first participant party terminal obtained through the first RTP channel, and the second participant party of the second participant party terminal obtained through the second RTP channel Sound data; filtering the host sound data in the first mix data to obtain second mix data; and transmitting the second mix data to the first RTP channel and the second RTP channel And wherein the first participant telephone terminal and the second participant telephone terminal receive the second mix data.
结合第四方面的第二种可能的实施方式,在第四方面的第三种可能的实施方式中,所述主持方电话终端,还用于:In conjunction with the second possible implementation of the fourth aspect, in a third possible implementation manner of the fourth aspect, the host telephone terminal is further configured to:
在所述滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据之后,控制所述第二混音数据处于禁止输出状态,以使所述第二混音数据不能通过所述主持方电话终端自身的主持方声音输出装置输出。After filtering the host sound data in the first mix data to obtain the second mix data, controlling the second mix data to be in a forbidden output state, so that the second mix The data cannot be output through the host sound output device of the host telephone terminal itself.
结合第四方面的第三种可能的实施方式,在第四方面的第四种可能的实施方式中,所述主持方电话终端,还用于:In conjunction with the third possible implementation of the fourth aspect, in a fourth possible implementation manner of the fourth aspect, the host telephone terminal is further configured to:
在所述添加所述第三与会方电话终端到所述电话会议之后,对所述会场中的当前声音数据进行混音处理,获得第三混音数据;其中,所述第三混音数据包括:所述主持方声音数据、所述第一与会方声音数据、所述第二与会方声音数据、以及所述第三与会方声音数据;发送所述第三混音数据到所述第一RTP通道、所述第二RTP通道、以及所述第三RTP通道中,以使所述第一与会方电话终端、第二与会方电话终端接、以及所述第三与会方电话终端收到所述第三混音数据。After the adding the third participant telephone terminal to the conference call, performing mixing processing on the current voice data in the conference site to obtain third mix data; wherein the third mix data includes : the host sound data, the first participant sound data, the second participant sound data, and the third participant sound data; transmitting the third mix data to the first RTP a channel, the second RTP channel, and the third RTP channel to enable the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal to receive the Third mix data.
结合第四方面的第四种可能的实施方式,在第四方面的第五种可能的实施方式中,所述主持方电话终端,还用于: In conjunction with the fourth possible implementation of the fourth aspect, in a fifth possible implementation manner of the fourth aspect, the host telephone terminal is further configured to:
在所述对所述会场中的当前声音数据进行混音处理,获得第三混音数据之后,滤除所述第三混音数据中的所述主持方声音数据,获得第四混音数据;通过所述主持方声音输出装置输出所述第四混音数据。Performing a mixing process on the current sound data in the meeting site to obtain the third mixing data, filtering the host sound data in the third mixing data to obtain a fourth mixing data; The fourth mix data is output by the host sound output device.
本发明具有如下技术效果:The invention has the following technical effects:
由于在主持方电话终端在添加第三与会方电话终端到电话会议的同时,主持方电话终端保持与第一与会方电话终端之间建立的第一RTP通道、以及与第二与会方电话终端之间建立的第二RTP通道,以使第一与会方电话终端和第二与会方电话终端能够继续进行电话会议。所以实现了在主持方电话终端添加新的与会方电话终端到电话会议时,原与会方电话终端可以通过与主持方电话终端保持的RTP通道继续进行电话会议,从而提高了原与会方电话终端的用户的体验感的技术效果。Since the host telephone terminal adds the third participant telephone terminal to the conference call, the host telephone terminal maintains the first RTP channel established between the first participant party telephone terminal and the second participant party telephone terminal. A second RTP channel is established to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call. Therefore, when the new conference party telephone terminal is added to the conference call by the host telephone terminal, the original conference party telephone terminal can continue the conference call through the RTP channel maintained by the host telephone terminal, thereby improving the original conference party telephone terminal. The technical effect of the user's experience.
附图说明DRAWINGS
为了更清楚地说明本发明实施例或现有技术中的技术方案,下面将对实施例或现有技术描述中所需要使用的附图作一简单地介绍,显而易见地,下面描述中的附图是本发明的一些实施例,对于本领域普通技术人员来讲,在不付出创造性劳动的前提下,还可以根据这些附图获得其他的附图。In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, a brief description of the drawings used in the embodiments or the prior art description will be briefly described below. Obviously, the drawings in the following description It is a certain embodiment of the present invention, and other drawings can be obtained from those skilled in the art without any creative work.
图1为本发明实施例中电话会议系统的示意图;1 is a schematic diagram of a conference call system according to an embodiment of the present invention;
图2为本发明实施例中电话会议的处理方法的流程图;2 is a flowchart of a method for processing a conference call according to an embodiment of the present invention;
图3为本发明实施例中步骤S102的细化流程图;FIG. 3 is a detailed flowchart of step S102 in the embodiment of the present invention;
图4为本发明实施例中步骤S103的细化流程图;4 is a detailed flowchart of step S103 in the embodiment of the present invention;
图5为本发明实施例中步骤S104的细化流程图;FIG. 5 is a detailed flowchart of step S104 in the embodiment of the present invention;
图6为本发明实施例中主持方电话终端的结构示意图; 6 is a schematic structural diagram of a host telephone terminal according to an embodiment of the present invention;
图7为本发明实施例中主持方电话终端的结构示意图。FIG. 7 is a schematic structural diagram of a host telephone terminal according to an embodiment of the present invention.
具体实施方式detailed description
本发明实施例通过提供一种电话会议的处理方法、主持方电话终端、及电话会议系统,解决了现有技术中在主持方电话终端添加新的与会方电话终端到电话会议时,原与会方电话终端无法继续进行电话会议的技术问题。The embodiment of the present invention provides a method for processing a conference call, a host telephone terminal, and a conference call system, which solves the problem in the prior art when a new conference party telephone terminal is added to the conference call by the host telephone terminal. The telephone terminal cannot continue the technical problem of the conference call.
本发明实施例的技术方案为解决上述技术问题,总体思路如下:The technical solution of the embodiment of the present invention is to solve the above technical problem, and the general idea is as follows:
一种电话会议的处理方法,包括:A method of processing a conference call, comprising:
在主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,所述主持方电话终端获取用户操作,所述用户操作用于添加第三与会方电话终端到所述电话会议;所述主持方电话终端响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议,并在添加所述第三与会方电话终端到所述电话会议的同时,保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议。When the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, the host telephone terminal acquires a user operation, and the user operation is used to add a third participant telephone terminal to the office a conference call; the host telephone terminal responds to the user operation to add the third participant telephone terminal to the conference call, and while adding the third participant telephone terminal to the conference call Maintaining a first real-time transport protocol RTP channel established between the first participant party telephone terminal and a second RTP channel established with the second participant party telephone terminal to enable the first participant party The telephone terminal and the second participant telephone terminal can continue the conference call.
为使本发明实施例的目的、技术方案和优点更加清楚,下面将结合附图,对本发明实施例中的技术方案进行清楚、完整地描述,显然,所描述的实施例是本发明一部分实施例,而不是全部的实施例。基于本发明中的实施例,本领域普通技术人员在没有作出创造性劳动前提下所获得的所有其他实施例,都属于本发明保护的范围。The technical solutions in the embodiments of the present invention will be clearly and completely described in the following with reference to the accompanying drawings. Instead of all the embodiments. All other embodiments obtained by those skilled in the art based on the embodiments of the present invention without creative efforts are within the scope of the present invention.
在介绍本申请实施例之前,先对本发明所涉及到的电话会议系统进行如下介绍:如图1所示,图1为一种用于实施本发明电话会议的处理方法的电话会议系统,其中,包括:一个主持方电话终端和多个与会方电话会议终端(例如:第一与会方电话终、第二与会方电话终端、以及第三与会方电话终端等等)。 在该电话会议系统进行电话会议时,会场建立在主持方电话终端上。其中,主持方电话终端(以及各与会方电话会议终端),具体可以是:IP(Internet Protocol,网络协议)电话或其他支持本地电话会议的电话终端。Before introducing the embodiment of the present application, the following is a description of the conference call system according to the present invention. As shown in FIG. 1 , FIG. 1 is a conference call system for implementing a method for processing a conference call of the present invention, where The utility model comprises: a host telephone terminal and a plurality of participant conference call terminals (for example: a first participant telephone terminal, a second participant party telephone terminal, and a third participant party telephone terminal, etc.). When the conference call system conducts a conference call, the venue is established on the host telephone terminal. The host telephone terminal (and the conference call terminals) may be: an IP (Internet Protocol) telephone or other telephone terminal supporting the local conference call.
实施例一Embodiment 1
本实施例提供一种电话会议的处理方法,如图2所示,包括:This embodiment provides a method for processing a conference call, as shown in FIG. 2, including:
步骤S101:在主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,主持方电话终端获取用户操作,用户操作用于添加第三与会方电话终端到电话会议;Step S101: When the host telephone terminal makes a conference call with the first participant telephone terminal and the second participant telephone terminal, the host telephone terminal acquires a user operation, and the user operation is used to add the third participant telephone terminal to the conference call. ;
步骤S102:主持方电话终端响应用户操作,以添加第三与会方电话终端到电话会议;Step S102: the host telephone terminal responds to the user operation to add the third participant telephone terminal to the conference call;
步骤S103:主持方电话终端在添加第三与会方电话终端到电话会议的同时,保持与第一与会方电话终端之间建立的第一RTP(Real-time Transport Protocol,实时传输协议)通道、以及与第二与会方电话终端之间建立的第二RTP通道,以使第一与会方电话终端和第二与会方电话终端能够继续进行电话会议;Step S103: The host phone terminal maintains a first RTP (Real-time Transport Protocol) channel established between the first participant party and the first participant phone terminal while adding the third participant phone terminal to the conference call, and a second RTP channel established with the second participant telephone terminal to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call;
步骤S104:主持方电话终端与第一与会方电话终端、第二与会方电话终端以及第三与会方电话终端进行电话会议。Step S104: The host telephone terminal performs a conference call with the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal.
具体来讲,首先,主持方电话终端与第一与会方电话终端建立第一RTP通道,并且与第二与会方电话终端建立第二RTP通道,以与第一与会方电话终端和第二与会方电话终端的进行(三方)电话会议。然后,在主持方电话终端获取用于添加第三与会终端到电话会议的用户操作时,主持方电话终端添加第三与会方电话终端到电话会议,且,在主持方电话终端在添加第三与会方电 话终端到电话会议的同时,主持方电话终端保持与第一与会方电话终端之间建立的第一RTP通道、以及与第二与会方电话终端之间建立的第二RTP通道,以使第一与会方电话终端和第二与会方电话终端能够继续进行电话会议。最后,在主持方电话终端添加第三与会方电话终端到电话会议之后,主持方电话终端即可与第一与会方电话终端、第二与会方电话终端以及第三与会方电话终端进行(四方)电话会议。Specifically, first, the host telephone terminal establishes a first RTP channel with the first participant telephone terminal, and establishes a second RTP channel with the second participant telephone terminal to interact with the first participant telephone terminal and the second participant party. Conducting (three-way) conference call of the telephone terminal. Then, when the host phone terminal acquires a user operation for adding the third participant terminal to the conference call, the host phone terminal adds the third participant phone terminal to the conference call, and the third party attends the conference at the host phone terminal. Square electricity While the terminal is in the conference call, the host telephone terminal maintains a first RTP channel established between the first participant party telephone terminal and a second RTP channel established with the second participant party telephone terminal to make the first The participant telephone terminal and the second participant telephone terminal can continue the conference call. Finally, after the third party telephone terminal is added to the conference call at the host telephone terminal, the host telephone terminal can be performed with the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal (quartet). telephone conference.
具体来讲,所述主持方电话终端获取用户操作的具体实现可以是:Specifically, the specific implementation of the host phone terminal acquiring the user operation may be:
如果主持方电话终端上设置有功能按键(或触摸屏)等输入设备,在检测到该功能按键被按键(或该触摸屏某一特定区域被触摸)时,即可确认获取到该用户操作。If an input device such as a function button (or a touch screen) is provided on the host phone terminal, when it is detected that the function button is pressed (or a specific area of the touch screen is touched), the user operation can be confirmed.
由于,现有技术中,在主持方电话终端添加新的与会方电话终端到电话会议时,主持方电话终端会释放与原与会方建立的RTP通道,在完成添加新与会方电话终端到电话会议时,再恢复与原与会方建立的RTP通道。从而导致在主持方电话终端添加新的与会方电话终端到电话会议时,原与会方电话终端(例如:第一与会方电话终端和第二与会方电话终端)无法继续进行电话会议。Because, in the prior art, when the host party telephone terminal adds a new conference party telephone terminal to the conference call, the host telephone terminal releases the RTP channel established with the original conference party, and completes adding the new conference party telephone terminal to the conference call. At the same time, the RTP channel established with the original party is restored. As a result, when the host party telephone terminal adds a new conference party telephone terminal to the conference call, the original conference party telephone terminal (for example, the first participant party telephone terminal and the second participant party telephone terminal) cannot continue the conference call.
而在本实施例中,在主持方电话终端在添加第三与会方电话终端到电话会议的同时,主持方电话终端保持与第一与会方电话终端之间建立的第一RTP通道、以及与第二与会方电话终端之间建立的第二RTP通道,以使第一与会方电话终端和第二与会方电话终端能够继续进行电话会议。所以,解决了现有技术中在主持方电话终端添加新的与会方电话终端到电话会议时,会出现原与会方电话终端无法继续进行电话会议的技术问题。实现了在主持方电话终端添加新的与会方电话终端到电话会议时,原与会方电话终端可以通过与主持方电话终端保持的RTP通道继续进行电话会议,从而提高了原与会方电话终端的用户体验的技术效果。 In the embodiment, while the host telephone terminal adds the third participant telephone terminal to the conference call, the host telephone terminal maintains the first RTP channel established between the first participant telephone terminal and the first A second RTP channel established between the two party telephone terminals to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call. Therefore, the prior art solves the technical problem that the original conference party telephone terminal cannot continue the conference call when the host party telephone terminal adds a new conference party telephone terminal to the conference call. When the new conference party telephone terminal is added to the conference call by the host telephone terminal, the original conference party telephone terminal can continue the conference call through the RTP channel maintained by the host telephone terminal, thereby improving the user of the original conference party telephone terminal. The technical effect of the experience.
可选的,在本实施例中,如图3所示,步骤S102,包括:Optionally, in this embodiment, as shown in FIG. 3, step S102 includes:
步骤S201:主持方电话终端响应用户操作,以与第三与会方电话终端建立第三RTP通道;Step S201: the host phone terminal responds to the user operation to establish a third RTP channel with the third party phone terminal;
步骤S202:主持电话终端通过第三RTP通道获取第三与会方电话终端的第三与会方声音数据;Step S202: The host telephone terminal acquires third party participant voice data of the third participant party terminal through the third RTP channel;
步骤S203:主持电话终端将第三与会方声音数据添加到电话会议的会场中,进而实现将第三与会方电话终端添加到电话会议。Step S203: The host telephone terminal adds the third party voice data to the conference site of the conference call, thereby implementing the third conference party telephone terminal to be added to the conference call.
具体来讲,主持方电话终端可以获取第三与会方电话终端的电话号码,并基于第三与会方电话终端的电话号码呼叫第三与会方电话终端。待第三与会方电话终端应答后,主持方电话终端与第三与会方电话终端建立第三RTP通道,此时,主持方用户可以和第三与会方用户通过第三RTP通道进行声音数据的交互。Specifically, the host telephone terminal can obtain the telephone number of the third participant telephone terminal, and call the third participant telephone terminal based on the telephone number of the third participant telephone terminal. After the third party phone terminal answers, the host phone terminal establishes a third RTP channel with the third party phone terminal. At this time, the host user can interact with the third party user through the third RTP channel for voice data. .
进一步,主持方电话终端获取用户的另一用户操作(该用户操作用于确定将第三与会方电话终端添加到电话会议中),主持电话终端通过第三RTP通道获取第三与会方电话终端的第三与会方声音数据,并将第三与会方声音数据添加到电话会议的会场中进行混音处理,进而实现将第三与会方电话终端添加到电话会议。Further, the host telephone terminal acquires another user operation of the user (the user operation is used to determine to add the third participant telephone terminal to the conference call), and the host telephone terminal acquires the third participant telephone terminal through the third RTP channel. The third party party voice data is added to the conference site of the conference call to perform the mixing process, thereby adding the third party phone terminal to the conference call.
在本实施例中,主持方电话终端与第三与会方电话终端建立第三RTP通道,并将来自第三RTP通道中的第三与会方声音数据添加到电话会议的会场中进行混音处理,从而实现将第三与会方电话终端添加到电话会议的技术效果。In this embodiment, the host phone terminal establishes a third RTP channel with the third party phone terminal, and adds the third party voice data from the third RTP channel to the conference site of the conference call for mixing processing. Thereby achieving the technical effect of adding the third party phone terminal to the conference call.
可选的,在本实施例中,如图4所示,步骤S103,包括: Optionally, in this embodiment, as shown in FIG. 4, step S103 includes:
步骤S301:主持方电话保持第一RTP通道、以及第二RTP通道;Step S301: the host phone maintains the first RTP channel and the second RTP channel;
步骤S302:主持方电话终端对电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据。其中,第一混音数据包括:通过主持方电话终端的声音采集装置所采集到的主持方声音数据,通过第一RTP通道获得的第一与会方电话终端的第一与会方声音数据,以及通过第二RTP通道获得的第二与会方电话终端的第二与会方声音数据;Step S302: The host telephone terminal performs a mixing process on the current voice data in the conference site of the conference call to obtain the first mix data. The first mix data includes: the host voice data collected by the voice collection device of the host phone terminal, the first participant voice data of the first participant phone terminal obtained through the first RTP channel, and Second party audio data of the second participant telephone terminal obtained by the second RTP channel;
步骤S303:主持方电话终端滤除第一混音数据中的主持方声音数据,获得第二混音数据;Step S303: The host phone terminal filters out the host voice data in the first mix data to obtain the second mix data.
步骤S304:主持方电话终端发送第二混音数据到第一RTP通道和第二RTP通道中,以使第一与会方电话终端和第二与会方电话终端接收到第二混音数据。Step S304: The host telephone terminal sends the second mix data to the first RTP channel and the second RTP channel, so that the first participant phone terminal and the second participant phone terminal receive the second mix data.
具体来讲,主持方电话终端中包括有呼叫控制模块和媒体引擎模块,呼叫控制模块通知媒体引擎模块保持第一RTP通道和第二RTP通道,媒体引擎模块响应呼叫控制模块的通知,保持第一RTP通道和第二RTP通道;进一步,呼叫控制模块通知体引擎模块停止将主持方声音数据发送给第一与会方电话终端以及第二与会方电话终端,媒体引擎模块响应呼叫控制模块的通知,对会场中的当前声音数据进行混音处理,获得第一混音数据(其中,第一混音数据包括:主持方声音数据、第一与会方声音数据、以及第二与会方声音数据),并滤除第一混音数据中的主持方声音数据,获得第二混音数据,再将第二混音数据发送到第一RTP通道和第二RTP通道中。Specifically, the host telephone terminal includes a call control module and a media engine module, and the call control module notifies the media engine module to maintain the first RTP channel and the second RTP channel, and the media engine module responds to the notification of the call control module to maintain the first An RTP channel and a second RTP channel; further, the call control module notifies the body engine module to stop transmitting the moderator voice data to the first participant phone terminal and the second participant phone terminal, and the media engine module responds to the notification of the call control module, The current sound data in the venue is subjected to mixing processing to obtain first mixed data (wherein the first mixed sound data includes: host sound data, first party sound data, and second participant sound data), and filtering In addition to the host sound data in the first mix data, the second mix data is obtained, and the second mix data is sent to the first RTP channel and the second RTP channel.
在本实施例中,主持方电话保持第一RTP通道、以及第二RTP通道,对电话会议的会场中的当前声音数据进行混音处理,并通过第一RTP通道和第 二RTP通道将获得的混音数据发送给第一与会方电话终端以及第二与会方电话终端,从而实现了在主持方电话终端添加第三与会方电话终端到电话会议的同时,第一与会方电话终端和第二与会方电话终端继续进行电话会议的技术效果。In this embodiment, the host phone maintains the first RTP channel and the second RTP channel, and performs sound mixing processing on the current voice data in the conference site of the conference call, and passes the first RTP channel and the first The second RTP channel sends the obtained mixing data to the first participant telephone terminal and the second participant telephone terminal, thereby realizing the addition of the third participant telephone terminal to the conference call at the host telephone terminal, and the first participant party The technical effect of the telephone conference and the second participant telephone terminal continuing the conference call.
并且,在本实施例中,由于在第一与会方电话终端和第二与会方电话终端收到的混音数据中不包含主持方声音数据,从而消除了主持方电话终端在添加第三与会方电话终端到电话会议时所产生的操作音(或其他杂音)对第一与会方电话终端的用户和第二与会方电话终端的用户的干扰,进而保证了第一与会方电话终端和第二与会方电话终端之间的通话质量。Moreover, in the embodiment, since the host side voice data is not included in the mix data received by the first participant party terminal and the second participant party terminal, the host party is eliminated from adding the third party. The operation sound (or other noise) generated by the telephone terminal to the conference call interferes with the user of the first participant's telephone terminal and the user of the second participant's telephone terminal, thereby ensuring the first participant's telephone terminal and the second participant The quality of the call between the party's telephone terminals.
可选的,在本实施例中,在步骤S303之后(或在执行步骤S304的同时),该方法还包括:Optionally, in this embodiment, after step S303 (or while performing step S304), the method further includes:
步骤S305:主持方电话终端控制第二混音数据处于禁止输出状态,以使第二混音数据不能通过主持方电话终端自身的主持方声音输出装置输出。Step S305: The host telephone terminal controls the second mix data to be in a forbidden output state, so that the second mix data cannot be output through the host voice output device of the host telephone terminal itself.
具体来讲,主持方电话终端中的呼叫控制模块通知主持方电话终端中媒体引擎模块禁止播放第二混音数据,媒体引擎模块响应控制模块的通知,控制第二混音数据处于禁止输出状态,以使第二混音数据不能通过主持方电话终端自身的主持方声音输出装置输出。Specifically, the call control module in the host phone terminal notifies the media engine module of the host phone terminal to prohibit playing the second mix data, and the media engine module controls the second mix data to be in the forbidden output state in response to the notification of the control module. So that the second mix data cannot be output through the host sound output device of the host telephone terminal itself.
在本实施例中,由于在主持方电话终端添加第三与会方电话终端到电话会议时,主持方电话终端的用户可以与第三与会方电话终端的用户进行通话,此时,主持方电话终端禁止自身的主持方声音输出装置输出第二混音数据,能够防止第二混音数据对主持方电话终端的用户与第三与会方电话终端的用户之间的通话造成影响。 In this embodiment, since the third party telephone terminal is added to the conference call at the host telephone terminal, the user of the host telephone terminal can make a call with the user of the third participant telephone terminal. At this time, the host telephone terminal The self-host sound output device is prohibited from outputting the second mix data, and the second mix data can be prevented from affecting the call between the user of the host telephone terminal and the user of the third participant telephone terminal.
可选的,在本实施例中,如图5所示,步骤S104,包括:Optionally, in this embodiment, as shown in FIG. 5, step S104 includes:
步骤S401:主持方电话终端对会场中的当前声音数据进行混音处理,获得第三混音数据;其中,第三混音数据包括:主持方声音数据、第一与会方声音数据、第二与会方声音数据、以及第三与会方声音数据;Step S401: The host phone terminal performs a mixing process on the current voice data in the site to obtain a third mix data. The third mix data includes: the host voice data, the first participant voice data, and the second participant. Square sound data, and third party sound data;
步骤S402:主持方电话终端发送第三混音数据到第一RTP通道、RTP通道、以及第三RTP通道中,以使第一与会方电话终端、第二与会方电话终端接、以及第三与会方电话终端收到第三混音数据。Step S402: The host telephone terminal sends the third mix data to the first RTP channel, the RTP channel, and the third RTP channel, so that the first participant telephone terminal, the second participant telephone terminal, and the third participant The party telephone terminal receives the third mix data.
具体来讲,呼叫控制模块通知媒体引擎模块恢复将主持方声音数据发送给第一与会方电话终端以及第二与会方电话终端,媒体引擎模块响应呼叫控制模块的通知,对会场中的当前声音数据进行混音处理,获得第三混音数据(其中,第三混音数据包括:主持方声音数据、第一与会方声音数据、第二与会方声音数据以及第三与会方声音数据),进一步,媒体引擎模块将第三混音数据发送到第一RTP通道、第二RTP通道、以及第三RTP通道中。Specifically, the call control module notifies the media engine module to resume sending the host voice data to the first participant phone terminal and the second participant phone terminal, and the media engine module responds to the notification of the call control module to the current voice data in the conference site. Performing a mixing process to obtain a third mix data (wherein the third mix data includes: host sound data, first participant sound data, second participant sound data, and third participant sound data), and further, The media engine module transmits the third mix data to the first RTP channel, the second RTP channel, and the third RTP channel.
可选的,在本实施例中,如图5所示,在步骤S401之后(或在执行步骤S402的同时),该方法还包括:Optionally, in this embodiment, as shown in FIG. 5, after step S401 (or at the same time as step S402), the method further includes:
步骤S403:主持方电话终端滤除第三混音数据中的主持方声音数据,获得第四混音数据;Step S403: The host phone terminal filters out the host sound data in the third mix data to obtain the fourth mix data;
步骤S404:主持方电话终端通过主持方声音输出装置输出第四混音数据。Step S404: The host telephone terminal outputs the fourth mix data through the host sound output device.
具体来讲,主持方电话终端在播放第三混音数据时,会先滤除第三混音数据中的主持方声音数据,获得第四混音数据,再通过自身的主持方声音输出装置输出第四混音数据。此处,若不滤除主持方声音数据,主持方电话终端输出主持方声音数据将与主持方电话终端的用户的口腔发出的声音相互叠加,出现 回声现象,从而使主持方电话终端的用户无法听清楚电话会议中的声音信息。Specifically, when playing the third mix data, the host phone terminal first filters out the host sound data in the third mix data, obtains the fourth mix data, and outputs the sound through the host sound output device. Fourth mix data. Here, if the host voice data is not filtered, the host party's telephone terminal output host sound data will be superimposed with the voice of the user's mouth of the host telephone terminal, appearing The echo phenomenon, so that the user of the host telephone terminal cannot hear the voice information in the conference call.
在本实施例中,主持方电话终端在播放第三混音数据时,会先滤除第三混音数据中的主持方声音数据,获得第四混音数据,再通过自身的主持方声音输出装置输出第四混音数据。从而消除了主持方电话终端输出主持方声音数据与主持方电话终端的用户的口腔发出的声音相互叠加,出现回声现象的可能性,实现了主持方电话终端的用户可以听清楚电话会议中的声音信息的技术效果。In this embodiment, when playing the third mix data, the host phone terminal first filters the host sound data in the third mix data to obtain the fourth mix data, and then outputs the sound through the host mode. The device outputs the fourth mix data. Thereby, the possibility that the voice outputted by the host telephone terminal and the voice of the user of the host telephone terminal are superimposed on each other, and the echo phenomenon occurs, so that the user of the host telephone terminal can hear the voice in the conference call. The technical effect of the information.
实施例二Embodiment 2
本实施例提供一种主持方电话终端,如图6所示,包括:This embodiment provides a host phone terminal, as shown in FIG. 6, including:
获取单元501,用于在主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,获取用户操作,用户操作用于添加第三与会方电话终端到电话会议;The obtaining unit 501 is configured to acquire a user operation when the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, and the user operates to add the third participant telephone terminal to the conference call;
添加单元502,用于响应用户操作,以添加第三与会方电话终端到电话会议;An adding unit 502, configured to respond to a user operation to add a third party phone terminal to the conference call;
保持单元503,用于在添加第三与会方电话终端到电话会议的同时,保持与第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与第二与会方电话终端之间建立的第二RTP通道,以使第一与会方电话终端和第二与会方电话终端能够继续进行电话会议;The maintaining unit 503 is configured to maintain a first real-time transport protocol RTP channel established between the first participant party telephone terminal and the second participant party telephone terminal while adding the third participant party terminal to the conference call Establishing a second RTP channel to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call;
进行单元504,用于与第一与会方电话终端、第二与会方电话终端以及第三与会方电话终端进行电话会议。The performing unit 504 is configured to perform a conference call with the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal.
由于在主持方电话终端在添加第三与会方电话终端到电话会议的同时,主持方电话终端保持与第一与会方电话终端之间建立的第一RTP通道、以及与第二与会方电话终端之间建立的第二RTP通道,以使第一与会方电话终端和 第二与会方电话终端能够继续进行电话会议。所以实现了在主持方电话终端添加新的与会方电话终端到电话会议时,原与会方电话终端可以通过与主持方电话终端保持的RTP通道继续进行电话会议,从而提高了原与会方电话终端的用户的体验感的技术效果。Since the host telephone terminal adds the third participant telephone terminal to the conference call, the host telephone terminal maintains the first RTP channel established between the first participant party telephone terminal and the second participant party telephone terminal. Establishing a second RTP channel to enable the first party phone terminal and The second participant telephone terminal can continue the conference call. Therefore, when the new conference party telephone terminal is added to the conference call by the host telephone terminal, the original conference party telephone terminal can continue the conference call through the RTP channel maintained by the host telephone terminal, thereby improving the original conference party telephone terminal. The technical effect of the user's experience.
可选的,在本实施例中,添加单元502,包括:Optionally, in this embodiment, the adding unit 502 includes:
建立模块,用于响应用户操作,以与第三与会方电话终端建立第三RTP通道;Establishing a module for responding to user operations to establish a third RTP channel with the third party telephone terminal;
获取模块,用于通过第三RTP通道获取第三与会方电话终端的第三与会方声音数据;An acquiring module, configured to acquire, by using the third RTP channel, third party audio data of the third party telephone terminal;
添加模块,用于将第三与会方声音数据添加到电话会议的会场中,进而实现将第三与会方电话终端添加到电话会议。The adding module is configured to add the third party voice data to the conference site of the conference call, thereby adding the third party phone terminal to the conference call.
可选的,在本实施例中,保持单元503,包括:Optionally, in this embodiment, the holding unit 503 includes:
保持模块,用于保持第一RTP通道、以及第二RTP通道;a holding module for maintaining the first RTP channel and the second RTP channel;
第一混音模块,用于对电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,第一混音数据包括:通过主持方电话终端的声音采集装置所采集到的主持方声音数据,通过第一RTP通道获得的第一与会方电话终端的第一与会方声音数据,以及通过第二RTP通道获得的第二与会方电话终端的第二与会方声音数据;a first mixing module, configured to perform a mixing process on the current sound data in the conference site of the conference call to obtain the first sound mixing data; wherein the first sound mixing data includes: collected by the sound collecting device of the host telephone terminal The host sound data to the first participant party voice data of the first participant party terminal obtained through the first RTP channel, and the second participant party voice data of the second participant party terminal obtained through the second RTP channel;
第一滤除模块,用于滤除第一混音数据中的主持方声音数据,获得第二混音数据;a first filtering module, configured to filter the host sound data in the first mix data to obtain the second mix data;
第一发送模块,用于发送第二混音数据到第一RTP通道和第二RTP通道中,以使第一与会方电话终端和第二与会方电话终端接收到第二混音数据。The first sending module is configured to send the second mix data to the first RTP channel and the second RTP channel, so that the first participant phone terminal and the second participant phone terminal receive the second mix data.
可选的,在本实施例中,保持单元503,还包括: Optionally, in this embodiment, the holding unit 503 further includes:
禁止模块,用于在滤除第一混音数据中的主持方声音数据,获得第二混音数据之后,控制第二混音数据处于禁止输出状态,以使第二混音数据不能通过主持方电话终端自身的主持方声音输出装置输出。a prohibiting module, configured to filter the host sound data in the first mix data, obtain the second mix data, and control the second mix data to be in a forbidden output state, so that the second mix data cannot pass the host The host terminal's own voice output device output.
可选的,在本实施例中,进行单元504,包括:Optionally, in this embodiment, the performing unit 504 includes:
第二混音模块,用于添加第三与会方电话终端到电话会议之后,对会场中的当前声音数据进行混音处理,获得第三混音数据;其中,第三混音数据包括:主持方声音数据、第一与会方声音数据、第二与会方声音数据、以及第三与会方声音数据;a second mixing module, configured to: after adding the third participant telephone terminal to the conference call, perform mixing processing on the current sound data in the conference site to obtain third mixing data; wherein the third mixing data includes: a host Sound data, first party sound data, second party sound data, and third party sound data;
第二发送模块,用于发送第三混音数据到第一RTP通道、第二RTP通道、以及第三RTP通道中,以使第一与会方电话终端、第二与会方电话终端接、以及第三与会方电话终端收到第三混音数据。a second sending module, configured to send the third mixing data to the first RTP channel, the second RTP channel, and the third RTP channel, so that the first participant telephone terminal, the second participant telephone terminal, and the The third party telephone terminal receives the third mix data.
可选的,在本实施例中,进行单元504,还包括:Optionally, in this embodiment, the performing unit 504 further includes:
第二滤除模块,用于在对会场中的当前声音数据进行混音处理,获得第三混音数据之后,滤除第三混音数据中的主持方声音数据,获得第四混音数据;a second filtering module, configured to perform mixing processing on the current sound data in the conference site, and after obtaining the third mixing data, filtering the host sound data in the third mixing data to obtain the fourth mixing data;
输出模块,用于通过主持方声音输出装置输出第四混音数据。And an output module, configured to output the fourth mix data through the host sound output device.
实施例三Embodiment 3
基于同一发明构思,本实施例提供一种主持方电话终端,如图7所示,包括:Based on the same inventive concept, the embodiment provides a host telephone terminal, as shown in FIG. 7, including:
存储器601,用于存储程序代码;a memory 601, configured to store program code;
处理器602,与存储器601通过总线605连接,用于读取程序代码,以执行: The processor 602 is coupled to the memory 601 via a bus 605 for reading program code to perform:
在主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,获取用户操作,用户操作用于添加第三与会方电话终端到电话会议;响应用户操作,以添加第三与会方电话终端到电话会议,并在添加第三与会方电话终端到电话会议的同时,保持与第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与第二与会方电话终端之间建立的第二RTP通道,以使第一与会方电话终端和第二与会方电话终端能够继续进行电话会议。Obtaining a user operation when the host phone terminal performs a conference call with the first participant phone terminal and the second participant phone terminal, and the user operation is used to add the third participant phone terminal to the conference call; in response to the user operation, to add The third party telephone terminal to the conference call, and while adding the third party telephone terminal to the conference call, maintaining the first real-time transmission protocol RTP channel established with the first participant party telephone terminal, and the second conference A second RTP channel established between the party telephone terminals to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call.
由于在主持方电话终端在添加第三与会方电话终端到电话会议的同时,主持方电话终端保持与第一与会方电话终端之间建立的第一RTP通道、以及与第二与会方电话终端之间建立的第二RTP通道,以使第一与会方电话终端和第二与会方电话终端能够继续进行电话会议。所以实现了在主持方电话终端添加新的与会方电话终端到电话会议时,原与会方电话终端可以通过与主持方电话终端保持的RTP通道继续进行电话会议,从而提高了原与会方电话终端的用户的体验感的技术效果。Since the host telephone terminal adds the third participant telephone terminal to the conference call, the host telephone terminal maintains the first RTP channel established between the first participant party telephone terminal and the second participant party telephone terminal. A second RTP channel is established to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call. Therefore, when the new conference party telephone terminal is added to the conference call by the host telephone terminal, the original conference party telephone terminal can continue the conference call through the RTP channel maintained by the host telephone terminal, thereby improving the original conference party telephone terminal. The technical effect of the user's experience.
可选的,在本实施例中,处理器602,还用于:Optionally, in this embodiment, the processor 602 is further configured to:
响应用户操作,以与第三与会方电话终端建立第三RTP通道;通过第三RTP通道获取第三与会方电话终端的第三与会方声音数据;将第三与会方声音数据添加到电话会议的会场中,进而实现将第三与会方电话终端添加到电话会议。Responding to user operation to establish a third RTP channel with the third party phone terminal; acquiring third party party voice data of the third party phone terminal through the third RTP channel; adding the third party party voice data to the conference call In the venue, the third party telephone terminal is added to the conference call.
可选的,在本实施例中,处理器602,还用于:Optionally, in this embodiment, the processor 602 is further configured to:
保持第一RTP通道、以及第二RTP通道;对电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,第一混音数据包括:通过主持方电话终端的声音采集装置所采集到的主持方声音数据,通过第一RTP通道获得的第一与会方电话终端的第一与会方声音数据,以及通过第二RTP通 道获得的第二与会方电话终端的第二与会方声音数据;滤除第一混音数据中的主持方声音数据,获得第二混音数据;发送第二混音数据到第一RTP通道和第二RTP通道中,以使第一与会方电话终端和第二与会方电话终端接收到第二混音数据。Maintaining the first RTP channel and the second RTP channel; performing mixing processing on the current sound data in the conference site of the conference call to obtain the first sound mixing data; wherein the first sound mixing data includes: sound through the host telephone terminal The host sound data collected by the collecting device, the first party sound data of the first participant party terminal obtained through the first RTP channel, and the second RTP pass The second participant party voice data of the second participant party telephone terminal obtained by the track; filtering the host party voice data in the first mix data to obtain the second mix data; and transmitting the second mix data to the first RTP channel and In the second RTP channel, the first participant telephone terminal and the second participant telephone terminal receive the second mix data.
可选的,在本实施例中,处理器602,还用于:Optionally, in this embodiment, the processor 602 is further configured to:
在滤除第一混音数据中的主持方声音数据,获得第二混音数据之后,控制第二混音数据处于禁止输出状态,以使第二混音数据不能通过主持方电话终端自身的主持方声音输出装置输出。After filtering the host sound data in the first mix data to obtain the second mix data, controlling the second mix data to be in a forbidden output state, so that the second mix data cannot be hosted by the host phone terminal itself. Square sound output device output.
可选的,在本实施例中,处理器602,还用于:Optionally, in this embodiment, the processor 602 is further configured to:
在添加第三与会方电话终端到电话会议之后,对会场中的当前声音数据进行混音处理,获得第三混音数据;其中,第三混音数据包括:主持方声音数据、第一与会方声音数据、第二与会方声音数据、以及第三与会方声音数据;发送第三混音数据到第一RTP通道、第二RTP通道、以及第三RTP通道中,以使第一与会方电话终端、第二与会方电话终端接、以及第三与会方电话终端收到第三混音数据。After the third conference party telephone terminal is added to the conference call, the current voice data in the conference site is subjected to a mixing process to obtain a third voice data. The third voice data includes: the host voice data, and the first party. Sound data, second party sound data, and third party sound data; transmitting the third sound data to the first RTP channel, the second RTP channel, and the third RTP channel to make the first participant telephone terminal The second party telephone terminal is connected, and the third party telephone terminal receives the third mixing data.
可选的,在本实施例中,处理器602,还用于:Optionally, in this embodiment, the processor 602 is further configured to:
在对会场中的当前声音数据进行混音处理,获得第三混音数据之后,滤除第三混音数据中的主持方声音数据,获得第四混音数据;通过主持方声音输出装置输出第四混音数据。After mixing the current sound data in the conference field to obtain the third sound mixing data, filtering the host sound data in the third sound mixing data to obtain the fourth sound mixing data; and outputting the first sound output device through the host side Four mix data.
实施例四Embodiment 4
基于同一发明构思,本实施例提供一种电话会议系统,如图1所示,包括: Based on the same inventive concept, the embodiment provides a conference call system, as shown in FIG. 1 , including:
主持方电话终端;Host telephone terminal;
多个与会方电话终端,包括:第一与会方电话终端、第二与会方电话终端、和第三与会方电话终端;a plurality of participant telephone terminals, including: a first participant telephone terminal, a second participant telephone terminal, and a third participant telephone terminal;
其中,主持方电话终端,用于:Among them, the host telephone terminal is used to:
在主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,获取用户操作,用户操作用于添加第三与会方电话终端到电话会议;响应用户操作,以添加第三与会方电话终端到电话会议,并在添加第三与会方电话终端到电话会议的同时,保持与第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与第二与会方电话终端之间建立的第二RTP通道,以使第一与会方电话终端和第二与会方电话终端能够继续进行电话会议。Obtaining a user operation when the host phone terminal performs a conference call with the first participant phone terminal and the second participant phone terminal, and the user operation is used to add the third participant phone terminal to the conference call; in response to the user operation, to add The third party telephone terminal to the conference call, and while adding the third party telephone terminal to the conference call, maintaining the first real-time transmission protocol RTP channel established with the first participant party telephone terminal, and the second conference A second RTP channel established between the party telephone terminals to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call.
由于在主持方电话终端在添加第三与会方电话终端到电话会议的同时,主持方电话终端保持与第一与会方电话终端之间建立的第一RTP通道、以及与第二与会方电话终端之间建立的第二RTP通道,以使第一与会方电话终端和第二与会方电话终端能够继续进行电话会议。所以实现了在主持方电话终端添加新的与会方电话终端到电话会议时,原与会方电话终端可以通过与主持方电话终端保持的RTP通道继续进行电话会议,从而提高了原与会方电话终端的用户的体验感的技术效果。Since the host telephone terminal adds the third participant telephone terminal to the conference call, the host telephone terminal maintains the first RTP channel established between the first participant party telephone terminal and the second participant party telephone terminal. A second RTP channel is established to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call. Therefore, when the new conference party telephone terminal is added to the conference call by the host telephone terminal, the original conference party telephone terminal can continue the conference call through the RTP channel maintained by the host telephone terminal, thereby improving the original conference party telephone terminal. The technical effect of the user's experience.
可选的,在本实施例中,主持方电话终端,还用于:Optionally, in this embodiment, the host phone terminal is further configured to:
响应用户操作,以与第三与会方电话终端建立第三RTP通道;通过第三RTP通道获取第三与会方电话终端的第三与会方声音数据;将第三与会方声音数据添加到电话会议的会场中,进而实现将第三与会方电话终端添加到电话会议。Responding to user operation to establish a third RTP channel with the third party phone terminal; acquiring third party party voice data of the third party phone terminal through the third RTP channel; adding the third party party voice data to the conference call In the venue, the third party telephone terminal is added to the conference call.
可选的,在本实施例中,主持方电话终端,还用于: Optionally, in this embodiment, the host phone terminal is further configured to:
保持第一RTP通道、以及第二RTP通道;对电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,第一混音数据包括:通过主持方电话终端的声音采集装置所采集到的主持方声音数据,通过第一RTP通道获得的第一与会方电话终端的第一与会方声音数据,以及通过第二RTP通道获得的第二与会方电话终端的第二与会方声音数据;滤除第一混音数据中的主持方声音数据,获得第二混音数据;发送第二混音数据到第一RTP通道和第二RTP通道中,以使第一与会方电话终端和第二与会方电话终端接收到第二混音数据。Maintaining the first RTP channel and the second RTP channel; performing mixing processing on the current sound data in the conference site of the conference call to obtain the first sound mixing data; wherein the first sound mixing data includes: sound through the host telephone terminal The host sound data collected by the collecting device, the first participant party voice data of the first participant party terminal obtained through the first RTP channel, and the second participant meeting of the second participant party terminal obtained through the second RTP channel Square sound data; filter the host sound data in the first mix data to obtain the second mix data; send the second mix data to the first RTP channel and the second RTP channel to make the first party call The terminal and the second participant telephone terminal receive the second mix data.
可选的,在本实施例中,主持方电话终端,还用于:Optionally, in this embodiment, the host phone terminal is further configured to:
在滤除第一混音数据中的主持方声音数据,获得第二混音数据之后,控制第二混音数据处于禁止输出状态,以使第二混音数据不能通过主持方电话终端自身的主持方声音输出装置输出。After filtering the host sound data in the first mix data to obtain the second mix data, controlling the second mix data to be in a forbidden output state, so that the second mix data cannot be hosted by the host phone terminal itself. Square sound output device output.
可选的,在本实施例中,主持方电话终端,还用于:Optionally, in this embodiment, the host phone terminal is further configured to:
在添加第三与会方电话终端到电话会议之后,对会场中的当前声音数据进行混音处理,获得第三混音数据;其中,第三混音数据包括:主持方声音数据、第一与会方声音数据、第二与会方声音数据、以及第三与会方声音数据;发送第三混音数据到第一RTP通道、第二RTP通道、以及第三RTP通道中,以使第一与会方电话终端、第二与会方电话终端接、以及第三与会方电话终端收到第三混音数据。After the third conference party telephone terminal is added to the conference call, the current voice data in the conference site is subjected to a mixing process to obtain a third voice data. The third voice data includes: the host voice data, and the first party. Sound data, second party sound data, and third party sound data; transmitting the third sound data to the first RTP channel, the second RTP channel, and the third RTP channel to make the first participant telephone terminal The second party telephone terminal is connected, and the third party telephone terminal receives the third mixing data.
可选的,在本实施例中,主持方电话终端,还用于:Optionally, in this embodiment, the host phone terminal is further configured to:
在对会场中的当前声音数据进行混音处理,获得第三混音数据之后,滤除第三混音数据中的主持方声音数据,获得第四混音数据;通过主持方声音输出 装置输出第四混音数据。After the sound processing of the current sound data in the venue is performed to obtain the third sound mixing data, the host sound data in the third sound mixing data is filtered out to obtain the fourth sound mixing data; The device outputs the fourth mix data.
尽管已描述了本发明的优选实施例,但本领域内的技术人员一旦得知了基本创造性概念,则可对这些实施例作出另外的变更和修改。所以,所附权利要求意欲解释为包括优选实施例以及落入本发明范围的所有变更和修改。While the preferred embodiment of the invention has been described, it will be understood that Therefore, the appended claims are intended to be interpreted as including the preferred embodiments and the modifications and
以上所述,仅为本发明的具体实施方式,但本发明的保护范围并不局限于此,任何熟悉本技术领域的技术人员在本发明揭露的技术范围内,可轻易想到变化或替换,都应涵盖在本发明的保护范围之内。因此,本发明的保护范围应所述以权利要求的保护范围为准。 The above is only a specific embodiment of the present invention, but the scope of the present invention is not limited thereto, and any person skilled in the art can easily think of changes or substitutions within the technical scope of the present invention. It should be covered by the scope of the present invention. Therefore, the scope of the invention should be determined by the scope of the claims.

Claims (15)

  1. 一种电话会议的处理方法,其特征在于,包括:A method for processing a conference call, comprising:
    在主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,所述主持方电话终端获取用户操作,所述用户操作用于添加第三与会方电话终端到所述电话会议;When the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, the host telephone terminal acquires a user operation, and the user operation is used to add a third participant telephone terminal to the office Call conference call;
    所述主持方电话终端响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议,并在添加所述第三与会方电话终端到所述电话会议的同时,保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议。The host telephone terminal responds to the user operation to add the third participant telephone terminal to the conference call, and while adding the third participant telephone terminal to the conference call, maintain the conference a first real-time transport protocol RTP channel established between the first participant-side telephone terminal and a second RTP channel established between the second-party participant telephone terminal, so that the first participant telephone terminal and the The second participant telephone terminal can continue the conference call.
  2. 如权利要求1所述的方法,其特征在于,所述主持方电话终端响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议,包括:The method of claim 1, wherein the host telephone terminal responds to the user operation to add the third participant telephone terminal to the conference call, including:
    所述主持方电话终端响应所述用户操作,以与所述第三与会方电话终端建立第三RTP通道;The host telephone terminal responds to the user operation to establish a third RTP channel with the third participant telephone terminal;
    所述主持电话终端通过所述第三RTP通道获取所述第三与会方电话终端的第三与会方声音数据;The host telephone terminal acquires third party participant voice data of the third participant telephone terminal through the third RTP channel;
    所述主持电话终端将所述第三与会方声音数据添加到所述电话会议的会场中,进而实现将所述第三与会方电话终端添加到所述电话会议。The host telephone terminal adds the third participant voice data to the conference site of the conference call, thereby implementing adding the third participant party terminal to the conference call.
  3. 如权利要求1所述的方法,其特征在于,所述主持方电话保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议,包括: The method of claim 1 wherein said host phone maintains a first real time transport protocol RTP channel established with said first party phone terminal, and said second party phone terminal A second RTP channel is established between the first participant telephone terminal and the second participant telephone terminal to continue the conference call, including:
    所述主持方电话终端保持所述第一RTP通道、以及所述第二RTP通道;The host telephone terminal maintains the first RTP channel and the second RTP channel;
    所述主持方电话终端对所述电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,所述第一混音数据包括:通过所述主持方电话终端的声音采集装置所采集到的主持方声音数据,通过所述第一RTP通道获得的所述第一与会方电话终端的第一与会方声音数据,以及通过所述第二RTP通道获得的所述第二与会方电话终端的第二与会方声音数据;The host telephone terminal performs a mixing process on the current sound data in the conference site of the conference call to obtain first mix data; wherein the first mix data includes: a voice passing through the host phone terminal The host sound data collected by the collecting device, the first participant sound data of the first participant telephone terminal obtained through the first RTP channel, and the second obtained through the second RTP channel Second party audio data of the participant's telephone terminal;
    所述主持方电话终端滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据;The host phone terminal filters out the host sound data in the first mix data to obtain second mix data;
    所述主持方电话终端发送所述第二混音数据到所述第一RTP通道和所述第二RTP通道中,以使所述第一与会方电话终端和所述第二与会方电话终端接收到所述第二混音数据。The host telephone terminal transmits the second mix data to the first RTP channel and the second RTP channel to receive the first participant telephone terminal and the second participant telephone terminal To the second mix data.
  4. 如权利要求3所述的方法,其特征在于,在所述主持方电话终端滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据之后,所述方法还包括:The method according to claim 3, wherein after said host telephone terminal filters said host sound data in said first mix data to obtain second mix data, said method further include:
    所述主持方电话终端控制所述第二混音数据处于禁止输出状态,以使所述第二混音数据不能通过所述主持方电话终端自身的主持方声音输出装置输出。The host telephone terminal controls the second mix data to be in a forbidden output state, so that the second mix data cannot be output through the host voice output device of the host telephone terminal itself.
  5. 如权利要求4所述的方法,其特征在于,所述主持方电话终端添加所述第三与会方电话终端到所述电话会议之后,所述方法还包括:The method of claim 4, wherein after the host telephone terminal adds the third participant telephone terminal to the conference call, the method further includes:
    所述主持方电话终端对所述会场中的当前声音数据进行混音处理,获得第三混音数据;其中,所述第三混音数据包括:所述主持方声音数据、所述第一与会方声音数据、所述第二与会方声音数据、以及所述第三与会方声音数据;The host phone terminal performs a mixing process on the current voice data in the site to obtain a third mix data. The third mix data includes: the host voice data, the first participant. Square sound data, the second participant sound data, and the third participant sound data;
    所述主持方电话终端发送所述第三混音数据到所述第一RTP通道、所述 第二RTP通道、以及所述第三RTP通道中,以使所述第一与会方电话终端、第二与会方电话终端接、以及所述第三与会方电话终端收到所述第三混音数据。Transmitting, by the host telephone terminal, the third mix data to the first RTP channel, the a second RTP channel, and the third RTP channel, to enable the first participant telephone terminal, the second participant telephone terminal, and the third participant telephone terminal to receive the third mix data.
  6. 如权利要求5所述的方法,其特征在于,在所述主持方电话终端对所述会场中的当前声音数据进行混音处理,获得第三混音数据之后,所述方法还包括:The method of claim 5, wherein after the host telephone terminal performs a mixing process on the current sound data in the conference site to obtain the third sound mixing data, the method further includes:
    所述主持方电话终端滤除所述第三混音数据中的所述主持方声音数据,获得第四混音数据;The host telephone terminal filters out the host sound data in the third mix data to obtain fourth mix data;
    所述主持方电话终端通过所述主持方声音输出装置输出所述第四混音数据。The host telephone terminal outputs the fourth mix data through the host sound output device.
  7. 一种主持方电话终端,其特征在于,包括:A host telephone terminal, comprising:
    获取单元,用于在所述主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,获取用户操作,所述用户操作用于添加第三与会方电话终端到所述电话会议;An obtaining unit, configured to acquire a user operation when the host phone terminal performs a conference call with the first participant phone terminal and the second participant phone terminal, where the user operation is used to add the third participant phone terminal to The conference call;
    添加单元,用于响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议;Adding a unit for responding to the user operation to add the third participant telephone terminal to the conference call;
    保持单元,用于在所述添加所述第三与会方电话终端到所述电话会议的同时,保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议。a holding unit, configured to maintain a first real-time transport protocol RTP channel established between the first participant party telephone terminal and the conference party while adding the third participant party terminal to the conference call Determining a second RTP channel established between the second participant telephone terminal to enable the first participant telephone terminal and the second participant telephone terminal to continue the conference call.
  8. 如权利要求7所述的主持方电话终端,其特征在于,所述添加单元,包括: The host telephone terminal according to claim 7, wherein the adding unit comprises:
    建立模块,用于响应所述用户操作,以与所述第三与会方电话终端建立第三RTP通道;Establishing a module, configured to respond to the user operation to establish a third RTP channel with the third participant telephone terminal;
    获取模块,用于通过所述第三RTP通道获取所述第三与会方电话终端的第三与会方声音数据;An acquiring module, configured to acquire third party audio data of the third participant telephone terminal by using the third RTP channel;
    添加模块,用于将所述第三与会方声音数据添加到所述电话会议的会场中,进而实现将所述第三与会方电话终端添加到所述电话会议。And a adding module, configured to add the third participant voice data to the conference site of the conference call, thereby implementing adding the third participant party terminal to the conference call.
  9. 如权利要求7所述的主持方电话终端,其特征在于,所述保持单元,包括:The host telephone terminal according to claim 7, wherein the holding unit comprises:
    保持模块,用于保持所述第一RTP通道、以及所述第二RTP通道;a holding module for holding the first RTP channel and the second RTP channel;
    混音模块,用于对所述电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,所述第一混音数据包括:通过所述主持方电话终端的声音采集装置所采集到的主持方声音数据,通过所述第一RTP通道获得的所述第一与会方电话终端的第一与会方声音数据,以及通过所述第二RTP通道获得的所述第二与会方电话终端的第二与会方声音数据;a sound mixing module, configured to perform mixing processing on current sound data in a conference site of the conference call to obtain first sound mixing data; wherein the first sound mixing data includes: sound passing through the host telephone terminal The host sound data collected by the collecting device, the first participant sound data of the first participant telephone terminal obtained through the first RTP channel, and the second obtained through the second RTP channel Second party audio data of the participant's telephone terminal;
    滤除模块,用于滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据;a filtering module, configured to filter the host sound data in the first mix data to obtain second mix data;
    发送模块,用于发送所述第二混音数据到所述第一RTP通道和所述第二RTP通道中,以使所述第一与会方电话终端和所述第二与会方电话终端接收到所述第二混音数据。a sending module, configured to send the second mix data to the first RTP channel and the second RTP channel, so that the first participant phone terminal and the second participant phone terminal receive The second mix data.
  10. 一种主持方电话终端,其特征在于,包括:A host telephone terminal, comprising:
    存储器,用于存储程序代码;a memory for storing program code;
    处理器,与所述存储器连接,用于读取所述程序代码,以执行: a processor coupled to the memory for reading the program code to perform:
    在所述主持方电话终端与第一与会方电话终端、以及第二与会方电话终端进行电话会议时,获取用户操作,所述用户操作用于添加第三与会方电话终端到所述电话会议;响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议,并在添加所述第三与会方电话终端到所述电话会议的同时,保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议。Obtaining a user operation when the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, and the user operation is used to add a third participant telephone terminal to the conference call; Responding to the user operation to add the third party phone terminal to the conference call, and to maintain the first party call while adding the third party phone terminal to the conference call a first real-time transport protocol RTP channel established between the terminals, and a second RTP channel established between the second participant party and the second participant telephone terminal, so that the first participant telephone terminal and the second participant telephone The terminal can proceed with the conference call.
  11. 如权利要求10所述的主持方电话终端,其特征在于,所述处理器,还用于:The host telephone terminal according to claim 10, wherein the processor is further configured to:
    响应所述用户操作,以与所述第三与会方电话终端建立第三RTP通道;通过所述第三RTP通道获取所述第三与会方电话终端的第三与会方声音数据;将所述第三与会方声音数据添加到所述电话会议的会场中,进而实现将所述第三与会方电话终端添加到所述电话会议。Responding to the user operation to establish a third RTP channel with the third participant telephone terminal; acquiring third party audio data of the third participant telephone terminal through the third RTP channel; The third party voice data is added to the conference site of the conference call, thereby implementing adding the third party phone terminal to the conference call.
  12. 如权利要求10所述的主持方电话终端,其特征在于,所述处理器,还用于:The host telephone terminal according to claim 10, wherein the processor is further configured to:
    保持所述第一RTP通道、以及所述第二RTP通道;对所述电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,所述第一混音数据包括:通过所述主持方电话终端的声音采集装置所采集到的主持方声音数据,通过所述第一RTP通道获得的所述第一与会方电话终端的第一与会方声音数据,以及通过所述第二RTP通道获得的所述第二与会方电话终端的第二与会方声音数据;滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据;发送所述第二混音数据到所述第一RTP通道和所述第二RTP通道 中,以使所述第一与会方电话终端和所述第二与会方电话终端接收到所述第二混音数据。Maintaining the first RTP channel and the second RTP channel; performing mixing processing on current sound data in the conference site of the conference call to obtain first mix data; wherein the first mix data includes : the first party audio data of the first participant telephone terminal obtained through the first RTP channel by the moderator voice data collected by the voice collection device of the host phone terminal, and by the Second participant audio data of the second participant telephone terminal obtained by the second RTP channel; filtering the host voice data in the first mix data to obtain second mix data; Second mix data to the first RTP channel and the second RTP channel And wherein the first participant telephone terminal and the second participant telephone terminal receive the second mix data.
  13. 一种电话会议系统,其特征在于,包括:A teleconferencing system, comprising:
    主持方电话终端;Host telephone terminal;
    多个与会方电话终端,包括:第一与会方电话终端、第二与会方电话终端、和第三与会方电话终端;a plurality of participant telephone terminals, including: a first participant telephone terminal, a second participant telephone terminal, and a third participant telephone terminal;
    其中,所述主持方电话终端,用于:Wherein, the host telephone terminal is used to:
    在所述主持方电话终端与所述第一与会方电话终端、以及所述第二与会方电话终端进行电话会议时,获取用户操作,所述用户操作用于添加第三与会方电话终端到所述电话会议;响应所述用户操作,以添加所述第三与会方电话终端到所述电话会议,并在添加所述第三与会方电话终端到所述电话会议的同时,保持与所述第一与会方电话终端之间建立的第一实时传输协议RTP通道、以及与所述第二与会方电话终端之间建立的第二RTP通道,以使所述第一与会方电话终端和所述第二与会方电话终端能够继续进行所述电话会议。Obtaining a user operation when the host telephone terminal performs a conference call with the first participant telephone terminal and the second participant telephone terminal, and the user operation is used to add a third participant telephone terminal to the conference Determining a conference call; responding to the user operation to add the third participant telephone terminal to the conference call, and while adding the third participant telephone terminal to the conference call, maintaining the a first real-time transport protocol RTP channel established between the participant's telephone terminal and a second RTP channel established with the second participant's telephone terminal to enable the first participant telephone terminal and the first The second party telephone terminal can continue the conference call.
  14. 如权利要求13所述的电话会议系统,其特征在于,所述主持方电话终端,还用于:The conference call system of claim 13 wherein said host telephone terminal is further configured to:
    响应所述用户操作,以与所述第三与会方电话终端建立第三RTP通道;通过所述第三RTP通道获取所述第三与会方电话终端的第三与会方声音数据;将所述第三与会方声音数据添加到所述电话会议的会场中,进而实现将所述第三与会方电话终端添加到所述电话会议。Responding to the user operation to establish a third RTP channel with the third participant telephone terminal; acquiring third party audio data of the third participant telephone terminal through the third RTP channel; The third party voice data is added to the conference site of the conference call, thereby implementing adding the third party phone terminal to the conference call.
  15. 如权利要求13所述的电话会议系统,其特征在于,所述主持方电话终端,还用于: The conference call system of claim 13 wherein said host telephone terminal is further configured to:
    保持所述第一RTP通道、以及所述第二RTP通道;对所述电话会议的会场中的当前声音数据进行混音处理,获得第一混音数据;其中,所述第一混音数据包括:通过所述主持方电话终端的声音采集装置所采集到的主持方声音数据,通过所述第一RTP通道获得的所述第一与会方电话终端的第一与会方声音数据,以及通过所述第二RTP通道获得的所述第二与会方电话终端的第二与会方声音数据;滤除所述第一混音数据中的所述主持方声音数据,获得第二混音数据;发送所述第二混音数据到所述第一RTP通道和所述第二RTP通道中,以使所述第一与会方电话终端和所述第二与会方电话终端接收到所述第二混音数据。 Maintaining the first RTP channel and the second RTP channel; performing mixing processing on current sound data in the conference site of the conference call to obtain first mix data; wherein the first mix data includes : the first party audio data of the first participant telephone terminal obtained through the first RTP channel by the moderator voice data collected by the voice collection device of the host phone terminal, and by the Second participant audio data of the second participant telephone terminal obtained by the second RTP channel; filtering the host voice data in the first mix data to obtain second mix data; Second mixing data into the first RTP channel and the second RTP channel to cause the first participant telephone terminal and the second participant telephone terminal to receive the second mix data.
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Families Citing this family (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN108540678A (en) * 2017-03-03 2018-09-14 展讯通信(上海)有限公司 Conference telephone implementation method, device and mostly logical terminal
CN110299144B (en) * 2018-03-21 2021-05-28 腾讯科技(深圳)有限公司 Audio mixing method, server and client
CN112543253A (en) * 2020-11-24 2021-03-23 厦门亿联网络技术股份有限公司 Method for realizing local multi-party teleconference and telephone
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Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN101175243A (en) * 2007-09-17 2008-05-07 华为技术有限公司 Method, system and conference server for providing song-ordering service
EP2202934A1 (en) * 2007-11-05 2010-06-30 Huawei Technologies Co., Ltd. A multimedia session call control method and the application server thereof
US20100272245A1 (en) * 2009-04-22 2010-10-28 Avaya Inc. Join-us call-log and call-answer messages

Family Cites Families (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US7552175B2 (en) * 2004-04-30 2009-06-23 Microsoft Corporation Mechanism for controlling communication paths between conference members
CN100492978C (en) * 2006-02-24 2009-05-27 腾讯科技(深圳)有限公司 Multi-party communication connection establishing method and connection processing system
CN101119533A (en) * 2006-08-02 2008-02-06 中兴通讯股份有限公司 Method for organizing telephone conference
CN201726453U (en) * 2010-06-03 2011-01-26 郭佳 Heterogeneous synergetic switching system

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN101175243A (en) * 2007-09-17 2008-05-07 华为技术有限公司 Method, system and conference server for providing song-ordering service
EP2202934A1 (en) * 2007-11-05 2010-06-30 Huawei Technologies Co., Ltd. A multimedia session call control method and the application server thereof
US20100272245A1 (en) * 2009-04-22 2010-10-28 Avaya Inc. Join-us call-log and call-answer messages

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