CN111432075B - Voice call real-time monitoring method and device based on VOIP network - Google Patents
Voice call real-time monitoring method and device based on VOIP network Download PDFInfo
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- CN111432075B CN111432075B CN202010171001.8A CN202010171001A CN111432075B CN 111432075 B CN111432075 B CN 111432075B CN 202010171001 A CN202010171001 A CN 202010171001A CN 111432075 B CN111432075 B CN 111432075B
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/22—Arrangements for supervision, monitoring or testing
- H04M3/2281—Call monitoring, e.g. for law enforcement purposes; Call tracing; Detection or prevention of malicious calls
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
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Abstract
The application relates to a voice call real-time monitoring method and a voice call real-time monitoring device based on a VOIP network, wherein the method comprises the following steps: the device is arranged on a communication transceiving link of a first terminal and a second terminal, is respectively connected to a mixer through a first RSP and a second RSP, and is also arranged on the mixer to be connected to an FXS user through an LSP; when the media stream of the user needs to be monitored, controlling the media streams transmitted and received by the first terminal and the second terminal to respectively pass through the first RSP and the second RSP to send the mixer; the mixer superposes the first RSP and the second RSP voice streams and then sends the superposed voice streams to the LSP; and networking to a monitored FXS user through the time slot of the LSP user, wherein the FXS user realizes voice monitoring on both the first terminal and the second terminal. The invention can realize bidirectional voice monitoring, and has simple structure, convenient operation and high monitoring efficiency.
Description
Technical Field
The present application relates to the field of VoIP systems, and in particular, to a method, an apparatus, a device, and a readable storage medium for monitoring a voice call in real time based on a VoIP network.
Background
Currently, voice monitoring is performed in PSTN (public switched telephone network) networks by using time slot networking methods, such as user a and user B talking, time slot networking of user a and user B, and recording as
C monitors A talking, then A is unidirectionally networked to C and recorded as A->C, C can hear A say like this, and one control user of this kind of control mode can only monitor the pronunciation of unidirectional, can't realize two-way control.
Later, in a VoIP (voice over IP) network, voice monitoring is realized by copying a media stream, for example, when a user a and a user B talk, a monitoring server D that monitors the talk between a and B and can send the media stream received and transmitted by the a at the a end needs to have a media stream decoding function, the monitoring operation is time-consuming and labor-consuming, and the decoded voice file is also a file of 2 unidirectional voices, and bidirectional monitoring cannot be realized.
Therefore, the prior art is in need of improvement.
Disclosure of Invention
The invention aims to solve the technical problem of providing a method, a device, equipment and a readable storage medium for monitoring voice call in real time based on a VOIP network, which can realize bidirectional voice monitoring and have simple structure, convenient operation and high monitoring efficiency.
The technical scheme of the invention is as follows:
a voice call real-time monitoring method based on a VOIP network is disclosed, wherein the method comprises the following steps:
the device is arranged on a communication transceiving link of a first terminal and a second terminal, is respectively connected to a mixer through a first RSP and a second RSP, and is also arranged on the mixer to be connected to an FXS user through an LSP;
when the media stream of the user needs to be monitored, controlling the media streams transmitted and received by the first terminal and the second terminal to respectively pass through the first RSP and the second RSP to send the mixer;
the mixer superposes the first RSP and the second RSP voice streams and then sends the superposed voice streams to the LSP;
and networking to a monitored FXS user through the time slot of the LSP user, wherein the FXS user realizes voice monitoring on both the first terminal and the second terminal.
The method for monitoring the voice call in real time based on the VOIP network comprises the following steps that the communication transceiving links arranged on the first terminal and the second terminal are respectively connected to a mixer through the first RSP and the second RSP, and the mixer is connected to the FXS user through the LSP:
setting a first terminal and a second terminal to be in communication connection with a media controller respectively; and the media controller is used for sending the media stream control of the first terminal and the second terminal to the media controller for forwarding when the signaling negotiation of the media controller is carried out.
The method for monitoring the voice call in real time based on the VOIP network comprises the following steps that the communication transceiving links arranged on the first terminal and the second terminal are respectively connected to a mixer through the first RSP and the second RSP, and the mixer is connected to the FXS user through the LSP:
setting a mixer with 2 RSP users and 1 LSP user in a media controller;
the mixer is arranged on a communication transceiving link of the first terminal and the second terminal and is connected to a mixer through the first RSP and the second RSP respectively, and the mixer is arranged to be connected to the FXS user through the LSP.
The method for monitoring the voice call in real time based on the VOIP network comprises the following steps of controlling media streams transmitted and received by a first terminal and a second terminal to respectively pass through a first RSP and a second RSP to send mixers when the media streams of a user need to be monitored comprise:
when the media stream of a user needs to be monitored, the media stream of the first terminal and the media stream of the second terminal are sent to a media controller for forwarding during the signaling negotiation of the media controller;
enabling a mixer of a conference mixing room, comprising 2 RSPs and 1 LSP;
the media controller copies the media stream transmitted and received by the first terminal to the first RSP and the second RSP of the mixer respectively;
the media streams are transmitted to the mixer by passing through the first RSP and the second RSP, respectively.
The method for monitoring the voice call in real time based on the VOIP network, wherein the step that the mixer superposes the voice flows of the first RSP and the second RSP and then sends the superposed voice flows to the LSP comprises the following steps:
the voice streams of the first RSP and the second RSP are transmitted in a one-way mode; the mixer superposes the voice streams of the first RSP and the second RSP and then sends the superposed voice streams to the LSP.
The voice call real-time monitoring method based on the VOIP network, wherein the FXS user is networked to a monitored FXS user through the time slot of the LSP user, and the FXS user realizes the voice monitoring of the first terminal and the second terminal, comprises the following steps:
and monitoring the voice stream after the first RSP and the second RSP are superposed through the time slot of the LSP user and the FXS user connected with the LSP, so as to realize the voice monitoring of the first terminal and the second terminal.
The VOIP network-based real-time monitoring voice call method is characterized in that the RSP is a participant of a remote user and is responsible for accessing an IP-based voice transmission user;
the LSP is a participant of a near-end user and is responsible for accessing a public switched telephone network user;
the FXS is an foreign exchange station.
A voice call device for real-time monitoring based on VOIP network, wherein the device comprises:
the first terminal and the second terminal are respectively in communication connection with the media controller; when the signaling negotiation of the media controller is carried out, the media stream of the first terminal and the media stream of the second terminal are controlled and sent to the media controller for forwarding;
a mixer with 2 RSPs which are respectively a first RSP and a second RSP and an LSP user is arranged in the media controller;
the communication transceiving links of the first terminal and the second terminal are respectively connected to a mixer through a first RSP and a second RSP, and the mixer is set to be connected to the FXS user through an LSP;
the mixer is used for sending the sound superposition of the first RSP and the second RSP to the LSP, and the FXS user realizes the voice monitoring of the first terminal and the second terminal through the LSP.
An apparatus comprising a memory and a processor, the memory storing a computer program, wherein the processor implements the steps of any of the methods when executing the computer program.
A computer-readable storage medium, on which a computer program is stored, wherein the computer program realizes the steps of any of the methods when executed by a processor.
Compared with the prior art, the embodiment of the invention has the following advantages:
the invention provides a voice call method, a device, equipment and a readable storage medium based on VOIP network real-time monitoring, wherein a conference mixer is adopted to realize the function of real-time monitoring, a conference room is established in monitoring equipment (a media controller MGC), 2 RSP members and 1 LSP member are arranged in the conference room, the voice of a receiving party and a transmitting party of a voice terminal A is respectively copied to the 2 RSP members, and the time slots of the LSP members are networked to the time slot of a monitoring FXS user C, so that the C can listen to the call voice of A and B on line; the invention can realize bidirectional voice monitoring, and has simple structure, convenient operation and high monitoring efficiency.
Drawings
In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the drawings used in the description of the embodiments or the prior art will be briefly described below, it is obvious that the drawings in the following description are only some embodiments described in the present invention, and for those skilled in the art, other drawings can be obtained according to the drawings without creative efforts.
Fig. 1 is a schematic block diagram of a prior art call model architecture.
Fig. 2 is a schematic block diagram of a call model architecture.
Fig. 3 is a schematic block diagram of a call model structure according to an embodiment of the method for monitoring a voice call in real time based on a VOIP network of the present invention.
Fig. 4 is a flowchart illustrating a method for monitoring a voice call in real time based on a VOIP network in an embodiment of the present invention.
Fig. 5 is a functional schematic block diagram of an apparatus according to an embodiment of the present invention.
Detailed Description
In order to make the technical solutions of the present invention better understood, the technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are only a part of the embodiments of the present invention, and not all of the embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.
The inventor finds that in a VoIP (voice over IP) network in the prior art, voice monitoring is implemented in a media stream copying manner, for example, when a user a and a user B talk, a monitoring server D that monitors the talk of the user a and the user B and sends the media stream received and sent by the user a at the terminal a needs to monitor, so that in the prior art, the monitoring server needs to have a media stream decoding function, the monitoring operation is time-consuming and labor-consuming, and the decoded voice file is also 2 unidirectional voice files, so that the technical problem of bidirectional monitoring cannot be implemented.
In order to solve the above problems, an embodiment of the present invention provides a voice call method for real-time monitoring based on a VOIP network, which uses a conference mixer to implement a real-time monitoring function, and creates a conference room in a monitoring device (media controller MGC), wherein the conference room has 2 RSP (far-end user participant) members and 1 LSP (near-end user participant) member, and copies the voice in the transmitting and receiving directions of a voice terminal a to 2 RSPs, and networks the time slots of the LSP members to the time slot of a monitoring FXS user C, so that the C can listen to the call voice of a and B online. The invention can realize bidirectional voice monitoring, and has simple structure, convenient operation and high monitoring efficiency.
Among them, RSP is a participant of a remote user, and is responsible for accessing VoIP (voice over IP) users, such as SIP (session initiation protocol) users, which are accessed through RTP (real-time transport protocol) streams.
LSPs are participants of near-end users, responsible for accessing PSTN (public switched telephone network) users, such as FXS (foreign exchange), PRI (base group rate interface) users, which are accessed via TDM (time slot switching) systems.
SIP Session Initiation Protocol, Session Initiation Protocol;
FXS English full name Foreign Exchange Station
PRI primary rate interface base group rate interface.
A normal call model is that the media controller controls the terminal 1 and the terminal 2 to establish a media stream to the other party, and the media stream may not pass through the media controller (see fig. 1). The media stream may need to be controlled through the media controller in order to be monitored for voice at the media controller.
As shown in fig. 2, to perform voice monitoring, it is necessary to control the media stream to pass through the media controller, so that the media controller can perform a copy operation on the media stream, where RTP in fig. 2 is a real-time transport protocol stream. If only copying requires additional decoding software to decode the media stream into the media stream, the monitoring operation is time-consuming and labor-consuming, and is not on-line monitoring. On the basis of reserving the signaling stream in fig. 1, the media stream is mutually transmitted and transferred from the terminal 1 and the terminal 2 to a media controller for forwarding.
As shown in fig. 2 and 3, a voice call device for real-time monitoring based on a VOIP network according to an embodiment of the present invention includes:
a first terminal (terminal 1) and a second terminal (terminal 2) are respectively in communication connection with a media controller (MGC); and when the signaling negotiation of the media controller is carried out, the media stream of the first terminal and the media stream of the second terminal are controlled and sent to the media controller for forwarding.
As shown in fig. 3, a Mixer (MIX) having 2 RSPs, which are a first RSP and a second RSP, respectively, and an LSP user is provided in the media controller.
The embodiment of the invention specifically comprises that a first terminal (terminal 1) and a second terminal (terminal 2) are respectively connected to a mixer through a first RSP and a second RSP on communication transceiving links, and the mixer is set to be connected to an FXS user through an LSP. That is, as shown in fig. 3, since the terminal 1 is in communication connection with the terminal through the media channel 1 of the access terminal 1 and the media channel 2 of the access terminal 2, the media controller is in communication connection with the terminal 1 connected to the far end through the media channel 1 of the access terminal 1; the media controller is connected to the remote terminal 2 via the media channel 2 of the access terminal 2. The mixer is connected to the transceiving links of media channel 1 and media channel 2 through the first RSP and the second RSP, respectively. The media controller may control the copying of packets for media channel 1 and media channel 2.
The mixer is used for sending the sound superposition of the first RSP and the second RSP to the LSP, and the FXS user realizes the voice monitoring of the first terminal and the second terminal through the LSP.
As shown in fig. 3, the method of the present invention is implemented based on the block diagram shown in fig. 3. The invention connects to a Mixer (MIX) through a first RSP and a second RSP respectively on the communication transceiving links of a first terminal (terminal 1) and a second terminal (terminal 2), and sets the mixer to be connected to FXS (foreign exchange station) users through LSP.
The invention adopts the technical scheme that a mixer (also called a mixing chamber) with 2 RSP users and 1 LSP user is established in a media controller, the media stream transmitted and received by a terminal 1 can be selected in the media controller to be respectively transmitted to the 2 RSP users, and the time slot of the LSP user is networked to a monitored FXS user, so that the FXS user can hear the voices of both the terminal 1 and the terminal 2.
The mixer (mixing chamber MIX) in the embodiment of the present invention functions to superpose the voice streams of the first RSP1+ the second RSP2 to be sent to the LSP. The flow copied to RSP is also unidirectional, so only the tone RSP1+ RSP2 is sent to the LSP.
Among them, RSP is a participant of a remote user, and is responsible for accessing VoIP (voice over IP) users, such as SIP (session initiation protocol) users, which are accessed through RTP (real-time transport protocol) streams.
LSPs are participants of near-end users, responsible for accessing PSTN (public switched telephone network) users, such as FXS (foreign exchange), PRI (base group Rate interface) users, which are accessed via TDM (time Slot switching) systems
Various non-limiting embodiments of the present invention are described in detail below with reference to the accompanying drawings.
Referring to fig. 1, fig. 1 shows a method for monitoring a voice call in real time based on a VOIP network in an embodiment of the present invention, where the method includes:
step S1, connecting the communication transceiving links of the first terminal and the second terminal to a mixer through a first RSP and a second RSP respectively, and setting the mixer to be connected to the FXS user through the LSP;
in the embodiment of the present invention, specifically, as shown in fig. 3, a first terminal (terminal 1) and a second terminal (terminal 2) are respectively configured to be in communication connection with a media controller; and the media controller is used for sending the media stream control of the first terminal and the second terminal to the media controller for forwarding when the signaling negotiation of the media controller is carried out.
In the invention, a Mixer (MIX) with 2 RSP users and 1 LSP user is arranged in a media controller; in the method, a mixer is connected to a communication transceiving link of a first terminal and a second terminal through a first RSP and a second RSP respectively, and the mixer is connected to an FXS user through an LSP.
Where RSP is a participant of the remote user. LSPs are participants of near-end users.
Step S2, when the media stream of the user needs to be monitored, controlling the media streams transmitted and received by the first terminal and the second terminal to respectively pass through the first RSP and the second RSP to send the mixer;
in the embodiment of the method, when the media stream of a user needs to be monitored, the media stream of a first terminal and the media stream of a second terminal are controlled and sent to a media controller for forwarding when the signaling of the media controller is negotiated;
then starting a mixer for the conference mixing room in the media controller, wherein the mixer comprises 2 RSPs and 1 LSP;
the media controller copies the media stream transmitted and received by the first terminal to the first RSP and the second RSP of the mixer respectively; the media streams are transmitted to the mixer by passing through the first RSP and the second RSP, respectively.
Step S3, the mixer sends the first RSP and the second RSP voice stream to the LSP after overlapping;
in the embodiment of the invention, the voice streams of the first RSP and the second RSP are transmitted in a unidirectional way; the mixer superposes the voice streams of the first RSP and the second RSP and then sends the superposed voice streams to the LSP so as to send the monitored sound data to the FXS user side.
And step S4, networking to a monitored FXS user through the time slot of the LSP user, wherein the FXS user realizes voice monitoring on both the first terminal and the second terminal.
The invention monitors the voice stream after the first RSP and the second RSP are superposed through the time slot of the LSP user and the FXS user connected with the LSP, thereby realizing the voice monitoring of the first terminal and the second terminal.
The RSP is a participant of a remote user and is responsible for accessing an IP-based voice transmission user;
the LSP is a participant of a near-end user and is responsible for accessing a public switched telephone network user;
the FXS is an foreign exchange station.
As can be seen from the above, in the specific embodiment of the present invention, if a voice stream of a certain user needs to be monitored, when a signaling negotiation of an MGC (media controller), media stream control of a terminal 1 and a terminal 2 needs to be sent to the MGC (media controller) for forwarding; and a mixing room of a conference is started at the media controller, which comprises 2 RSPs and 1 LSP, the MGC (media controller) copies the voice streams transmitted and received by the terminal 1 to the RSP1 and the RSP2 of the mixer respectively, and the voice streams of the RSP1 and the RSP2 are superposed and transmitted to the LSP. The LSP time slot is unidirectionally networked to a monitored FXS user, the monitored FXS user does not do other operations except unidirectional networking operation, so that the FXS (foreign exchange station) user can monitor the conversation of the terminal 1 and the terminal 2 in real time. It can be seen that the embodiment of the present invention uses a mixer (audio mixer) to monitor the voice in the transceiving direction of the terminal. The invention can realize bidirectional voice monitoring, and has simple structure, convenient operation and high monitoring efficiency.
In one embodiment, the present invention provides an apparatus, which may be a terminal, having an internal structure as shown in fig. 5. The apparatus includes a processor, a memory, a network interface, a display screen, and an input device connected by a system bus. Wherein the processor of the device is configured to provide computing and control capabilities. The memory of the device comprises a nonvolatile storage medium and an internal memory. The non-volatile storage medium stores an operating system and a computer program. The internal memory provides an environment for the operation of an operating system and computer programs in the non-volatile storage medium. The network interface of the device is used for communicating with an external terminal through a network connection. The computer program is executed by a processor to implement a method of generating a natural language model. The display screen of the equipment can be a liquid crystal display screen or an electronic ink display screen, and the input device of the equipment can be a touch layer covered on the display screen, a key, a track ball or a touch pad arranged on the shell of the equipment, an external keyboard, a touch pad or a mouse and the like.
Those skilled in the art will appreciate that fig. 5 is a block diagram of only a portion of the structure associated with the disclosed aspects and is not intended to limit the devices to which the disclosed aspects apply, and that a particular device may include more or less components than those shown, or may combine certain components, or have a different arrangement of components.
The embodiment of the invention provides equipment, which comprises a memory and a processor, wherein the memory stores a computer program, and the processor executes the computer program to realize the following steps:
the device is arranged on a communication transceiving link of a first terminal and a second terminal, is respectively connected to a mixer through a first RSP and a second RSP, and is also arranged on the mixer to be connected to an FXS user through an LSP;
when the media stream of the user needs to be monitored, controlling the media streams transmitted and received by the first terminal and the second terminal to respectively pass through the first RSP and the second RSP to send the mixer;
the mixer superposes the first RSP and the second RSP voice streams and then sends the superposed voice streams to the LSP;
the FXS user is networked to a monitored FXS user through the time slot of the LSP user, and the FXS user implements voice monitoring of both the first terminal and the second terminal, as described above.
In summary, compared with the prior art, the embodiment of the invention has the following advantages:
the invention provides a method, a device and a readable storage medium for monitoring voice call in real time based on a VOIP network.A conference mixer is adopted to realize the function of real-time monitoring, a conference room is established in monitoring equipment (a media controller MGC), 2 RSP members and 1 LSP member are arranged in the conference room, the voice of a receiving party and a transmitting party of a voice terminal A are respectively copied to the 2 RSPs, and the time slots of the LSP members are networked to a time slot of a monitoring FXS user C, so that the C can listen to the call voice of A and B on line; the invention can realize bidirectional voice monitoring, and has simple structure, convenient operation and high monitoring efficiency.
The technical features of the above embodiments can be arbitrarily combined, and for the sake of brevity, all possible combinations of the technical features in the above embodiments are not described, but should be considered as the scope of the present specification as long as there is no contradiction between the combinations of the technical features.
The above-mentioned embodiments only express several embodiments of the present application, and the description thereof is more specific and detailed, but not construed as limiting the scope of the invention. It should be noted that, for a person skilled in the art, several variations and modifications can be made without departing from the concept of the present application, which falls within the scope of protection of the present application. Therefore, the protection scope of the present patent shall be subject to the appended claims.
Claims (8)
1. A voice call real-time monitoring method based on a VOIP network is characterized by comprising the following steps:
the method comprises the steps that a mixer is arranged on a communication transceiving link of a first terminal and a second terminal and is connected to a foreign exchange station (FXS) user through a participant RSP of a first far-end user and a participant RSP of a second far-end user respectively, and the mixer is arranged to be connected to the FXS user through a participant LSP of a near-end user;
when the media stream of the user needs to be monitored, controlling the media streams transmitted and received by the first terminal and the second terminal to respectively transmit to the mixer through the first RSP and the second RSP;
the mixer superposes the first RSP and the second RSP voice streams and then sends the superposed voice streams to the LSP;
the method comprises the steps that a monitored FXS user is networked through a time slot of an LSP user, and the FXS user realizes voice monitoring on a first terminal and a second terminal;
wherein, when the media stream of the user needs to be monitored, the step of controlling the media streams sent and received by the first terminal and the second terminal to respectively send the mixer through the first RSP and the second RSP comprises:
when the media stream of a user needs to be monitored, the media stream of the first terminal and the media stream of the second terminal are sent to a media controller for forwarding during the signaling negotiation of the media controller;
enabling a mixer of a conference mixing room, comprising 2 RSPs and 1 LSP;
a media controller (MGC) copies media streams transmitted and received by a first terminal to a first RSP and a second RSP of the mixer respectively;
the media stream is sent to the mixer through the first RSP and the second RSP respectively;
wherein, the networking is to a monitored FXS user through the time slot of the LSP user, and the FXS user implements the voice monitoring of both the first terminal and the second terminal including:
and monitoring the voice stream after the first RSP and the second RSP are superposed through the time slot of the LSP user and the FXS user connected with the LSP, so as to realize the voice monitoring of the first terminal and the second terminal.
2. The method for real-time monitoring voice call based on VOIP network as claimed in claim 1, wherein the step of setting up the mixer connected to the communication transceiving links of the first terminal and the second terminal through the first RSP and the second RSP respectively, and setting up the mixer connected to the FXS user through the LSP comprises:
setting a first terminal and a second terminal to be in communication connection with a media controller respectively; and the media controller is used for sending the media stream control of the first terminal and the second terminal to the media controller for forwarding when the signaling negotiation of the media controller is carried out.
3. The method of claim 1, wherein the step of setting up a mixer connected to the transceiving links of the first terminal and the second terminal via the first RSP and the second RSP, respectively, and setting up the mixer connected to the FXS user via the LSP further comprises:
setting a mixer with 2 RSP users and 1 LSP user in a media controller;
the mixer is arranged on a communication transceiving link of the first terminal and the second terminal and is connected to a mixer through the first RSP and the second RSP respectively, and the mixer is arranged to be connected to the FXS user through the LSP.
4. The method for real-time monitoring voice call based on VOIP network as claimed in claim 1, wherein the step of the mixer sending the superimposed voice streams of the first RSP and the second RSP to the LSP includes:
the voice streams of the first RSP and the second RSP are transmitted in a one-way mode; the mixer superposes the voice streams of the first RSP and the second RSP and then sends the superposed voice streams to the LSP.
5. The method for real-time monitoring voice call over VOIP network according to claim 1, wherein the RSP is a participant of a remote user and is responsible for accessing an IP-based voice transmission user;
the LSP is a participant of a near-end user and is responsible for accessing a public switched telephone network user;
the FXS is an foreign exchange station.
6. A voice call device based on VOIP network real-time monitoring, the device includes:
the first terminal and the second terminal are respectively in communication connection with the media controller; when the signaling negotiation of the media controller is carried out, the media stream of the first terminal and the media stream of the second terminal are controlled and sent to the media controller for forwarding;
a conference mixing room with a mixer of 2 RSPs which are respectively a participant RSP of a first far-end user, a participant RSP of a second far-end user and a participant LSP user of a near-end user is arranged in the media controller; wherein, the media controller (MGC) copies the media stream transmitted and received by the first terminal to the first RSP and the second RSP of the mixer respectively, and the media stream is transmitted to the mixer through the first RSP and the second RSP respectively;
the communication transceiving links of the first terminal and the second terminal are respectively connected to a mixer through a first RSP and a second RSP, and the mixer is set to be connected to an FXS user through an LSP;
the mixer is used for overlapping and sending the voice of the first RSP and the second RSP to the LSP, the FXS user is connected with the LSP through the time slot of the LSP user, the voice flow after the overlapping of the first RSP and the second RSP is monitored, and the voice monitoring of the first terminal and the voice monitoring of the second terminal are achieved.
7. A mobile terminal comprising a memory and a processor, the memory storing a computer program, characterized in that the processor, when executing the computer program, implements the steps of the method according to any of claims 1 to 5.
8. A computer-readable storage medium, on which a computer program is stored, which, when being executed by a processor, carries out the steps of the method of any one of claims 1 to 5.
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