JP4644813B2 - Multi-party call system, call terminal and call server in multi-party call system, multi-party call method - Google Patents

Description

  The present invention relates to a multi-party call system that performs multi-party calls with a plurality of call terminals, a call terminal and a call server in a multi-party call system, and a multi-party call method.

One of the application services in IP (Internet Protocol) telephone is multi-party call. A multi-party call is a service in which a large number of people make a call simultaneously. In circuit-switched networks, it was possible to make multiparty calls such as conference calls using dedicated devices called Multipoint Conference Units (MCUs). Because it is not, it was a service for the specific community that owns the device. In contrast, IP telephones can provide services on a software basis, and therefore can provide services to general users.

There are mainly two types of service offerings in the multi-party call service for IP phones.
One is a mode in which a function for multi-party calls is newly added to a user terminal, and a service is received by cooperation between users. This form has poor versatility because only the user who owns the terminal having the function can receive the service.
The other is a form in which services are provided by a server. In this form, although the load is concentrated on the server, the convenience of the user is improved and the billing management of the service provider becomes easy.

In a multi-party call service, in the case of an IP phone, a "signaling" function for making a call to the target and a "full duplex voice transmission" function for transmitting voice after it is determined that a call can be made with the other party through signaling are required. Become. These functions are usually for one-to-one communication.

  In the signaling function, each terminal (A, B, C) makes a call to the multi-party call server. By doing so, apparently A only has a one-to-one session with the multi-party call server, but the server also has a session with other users (B, C), so A, B , C virtually establishes a multi-session via the server.

“Full duplex” in the full-duplex audio transmission function means that the user's own voice and the other party's voice are handled simultaneously. In the full-duplex audio transmission function, audio transmitted from a user is appropriately mixed, and a plurality of audio streams are compressed into one stream and transmitted. A plurality of audio streams can be exchanged using this full-duplex audio transmission function.

As a conventional digital audio mixing system using a call server, there is “all participant mixing”. In this method, audio streams from a plurality of terminals are received by a call server, and mixing is performed by a method such as superimposing audio samples from each terminal. Next, a multi-party call is realized by transmitting the audio stream of the mixing result to all terminals. At this time, among the audio streams transmitted from the call server to all the terminals, when the sound transmitted from the terminal i to the call server is returned to the terminal i, the sound is prevented from being heard as an echo. It is known to subtract sound (see Non-Patent Document 1).

Each terminal continues to send an audio stream to the call server regardless of the presence or absence of audio input from the microphone. However, in the case of a terminal having a silence compression function, only a clock is counted without transmitting a voice packet during a silence interval.
Internet Telephony Workshop 2001 `` Centralized Conferencing using SIP '' (2001.4)

In the service via the multi-party call server, processing delay in the call server is added. In mixing, when superimposing waveforms, if the audio is 8kHz sampling, processing of 8000 times / second is required when mixing two streams. Also, the amount of processing increases in proportion to the number of terminals to be mixed.
In the all-participant mixing method, since the load is concentrated on the call server, the maximum number of streams that can be mixed by the call server is limited. In the call server, mixing processing is performed on the assumption that an audio stream is received regardless of whether audio data is transmitted from the terminal. For this reason, when the number of terminals increases, there is a problem that the processing amount of the call server increases. Further, even if there is a silent section, it is not possible to predict when a voice packet will come next. Even if the silent compression function is provided, the number of voice streams to be mixed by the call server is not reduced.

The present invention is intended to solve such problems of the conventional method, and a multi-party call system and multi-party call that mix audio streams from a plurality of call terminals with high efficiency. It is an object of the present invention to realize a call terminal, a call server, and a multi-party call method in a system.

  In order to achieve the above object, the multi-party call system according to claim 1 is configured such that a call server and a plurality of call terminals are connected via a network, and a voice of a specific call terminal establishes a session with the call server. In the multi-party call system transmitted to each call terminal, the call terminal has a voice input means for inputting voice, and the volume of the voice input to the voice input means is greater than a set value. An input voice judging means for judging whether or not the voice is loud, and when the voice inputted by the input voice judging means is larger than a set value, a mixing request is made to the call server, a response is received from the call server, and the mixing is performed Mixing state establishment means for establishing a state, and after the mixing state with the call server is established by the mixing state establishment means, the voice input means inputs the state. First voice packet transmitting means for packetizing the transmitted voice and transmitting it to the call server, wherein the call server transmits a response to the call terminal that has made the mixing request, and records the mixing as a target for mixing control Control target recording means, and second voice packet transmission means for transmitting voice packets transmitted from the call terminal recorded by the mixing control target recording means to other call terminals that have established a session. It is characterized by.

  According to a second aspect of the present invention, in the multi-party call system according to the first aspect, when the voice input by the call terminal is smaller than a set value by the input voice judging means, the voice is smaller than the set value. Number of times storage means for memorizing how many times the input is continuous, and when the number of voice input times smaller than the set value stored in the number of times storage means has continued for a predetermined number of times or more, a request for disconnecting mixing to the call server And a mixing state disconnecting unit that receives the response from the call server and disconnects the mixing state, and the call server responds to the communication terminal that has issued the mixing disconnection request by the mixing state disconnecting unit. And a mixing control object deletion means for deleting the mixing control object from the mixing control object.

  According to a third aspect of the present invention, in the multi-party call system according to the first or second aspect, the call terminal transmits a session start request message to the call server, and the call server performs a session for the session start request message. A session between the call terminal and the call server is established by transmitting a start response message.

  According to a fourth aspect of the present invention, in the multi-party call system according to the first or second aspect, the call terminal transmits a session disconnection request message to the call server, and the call server responds to a session corresponding to the session disconnection request message. A session between the call terminal and the call server is disconnected by transmitting a disconnect response message.

  The call terminal according to claim 5 is a multi-party in which a call server and a plurality of call terminals are connected via a network, and the voice of a specific call terminal is transmitted to each call terminal that has established a session with the call server. In a call terminal in the intercommunication system, a voice input means for inputting voice, an input voice judgment means for judging whether or not the volume of the voice inputted to the voice input means is larger than a set value, A mixing state establishing means for making a mixing request to the call server and receiving a response from the call server to establish a mixing state when the sound input by the input sound determining means is larger than a set value; and the mixing state After the mixing state with the call server is established by the establishing means, the voice input by the voice input means is packetized and the call server And having a first voice packet transmission means for transmitting.

  According to a sixth aspect of the present invention, in the call terminal according to the fifth aspect, when the voice inputted by the input voice judgment means is smaller than a set value, how many times the voice input smaller than the set value continues. When the number of voice input times smaller than a set value stored in the number storage unit is stored for a predetermined number of times or more, a disconnection request is made to the call server, and from the call server It further comprises mixing state cutting means for receiving the response and cutting the mixing state.

  The call server according to claim 7 is a multi-party in which a call server and a plurality of call terminals are connected by a network, and the voice of a specific call terminal is transmitted to each call terminal that has established a session with the call server. In the call server in the intercommunication system, a response is transmitted to the call terminal that has made the mixing request, and is recorded from the mixing control target recording means for recording as a mixing control target, and from the call terminal recorded by the mixing control target recording means. And a second voice packet transmitting means for transmitting the voice packet to another telephone terminal that has established a session.

  According to an eighth aspect of the present invention, in the call server according to the seventh aspect of the present invention, the communication server further includes a mixing control target deleting unit that deletes the call terminal that has issued the mixing disconnection request from the mixing control target.

  In the multi-party call method according to claim 9, a call server and a plurality of call terminals are connected via a network, and a voice of a specific call terminal is transmitted to each call terminal that has established a session with the call server. In the multi-party call method, the call terminal is configured to input voice, determine whether the input voice is larger than a set value, and set the input voice. If greater than the value, a request for mixing is made to the call server, a response is received from the call server and a mixing state is established; and after the mixing state with the call server is established, the voice input means Packetizing the input voice and transmitting it to the call server, wherein the call server sends a response to the call terminal that made the mixing request, And recording as a control target, the voice packet transmitted from the recorded the call terminal, and having a sending to the other call terminals that have established a session.

  According to a tenth aspect of the present invention, in the multi-party call method according to the ninth aspect, when the input voice is smaller than a set value, the call terminal continuously receives a number of voice inputs smaller than the set value. And when the number of voice inputs smaller than the stored setting value continues for a predetermined number of times or more, a request for mixing disconnection is made to the call server and a response from the call server is received to change the mixing state. Disconnecting, and further comprising the step of deleting, from the mixing control target, the call terminal from which the call server has transmitted a response to the mixing disconnection request.

As is apparent from the above description, according to the present invention, when the volume of the voice input to the call terminal is larger than the set value, the mixing state is established and the voice is transmitted to the call server. Transmits the voice to the other call terminal. Therefore, since only the voice from the call terminal that has actually input the voice needs to be the target of the mixing process, the mixing process of the call server can be performed efficiently. As a result, the number of terminals can be increased.
Also, by processing only the voice of the call terminal that is currently speaking by the call server, the number of streams that require mixing can be greatly reduced. In the call server, only the voice from the input call terminal is mixed, and the obtained voice packet is transmitted to another call terminal, thereby realizing a multi-party call.
A call terminal can start a multi-party call simply by starting a session with a call server. Furthermore, after establishing the session, the call terminal can make a call without requiring an additional operation by simply using an input device such as a microphone.

  According to the second aspect of the present invention, when the voice input smaller than the set value is continued for a predetermined number of times or more, the mixing state is disconnected, so that the network load when there is no voice input continuously is also reduced. Can be reduced.

  According to the third aspect of the present invention, the session is established by requesting the session start from the call terminal to the call server and responding from the call server to the call terminal. The terminal can be specified.

  According to the present invention, the session is disconnected by requesting the session disconnection from the call terminal to the call server and responding from the call server to the call terminal. The terminal can be specified.

  According to the telephone terminal of the fifth aspect, when the volume of the input voice is larger than the set value, the mixing state is established and the voice is transmitted to the call server. Therefore, the call terminal can start a multi-party call only by inputting voice.

  According to the call terminal of the sixth aspect, when the voice input smaller than the set value is continuously performed for a predetermined number of times or more, the disconnection request is made to the call server, so that the number of voice packets that need to be mixed is greatly reduced. be able to.

  According to the call server according to claim 7, since the voice packet from the call terminal that has made the mixing request is transmitted to another call terminal that has established a session, the call server can be efficiently mixed. Can do.

  According to the call server according to the eighth aspect, since the call terminal that has requested mixing disconnection is deleted from the mixing control target, the number of packets that need to be mixed in the call server can be significantly reduced.

According to the multi-party call method according to claim 9, when the volume of the voice input to the call terminal is larger than the set value, the mixing state is established, and the voice is transmitted to the call server. The voice is transmitted to another call terminal. Therefore, since only the voice from the call terminal that has actually input the voice needs to be the target of the mixing process, the mixing process of the call server can be performed efficiently. As a result, the number of terminals can be increased.
Also, by processing only the voice of the call terminal that is currently speaking by the call server, the number of streams that require mixing can be greatly reduced. In the call server, only the voice from the input call terminal is mixed, and the obtained voice packet is transmitted to another call terminal, thereby realizing a multi-party call.
A call terminal can start a multi-party call simply by starting a session with a call server. Furthermore, after establishing the session, the call terminal can make a call without requiring an additional operation by simply using an input device such as a microphone.

  According to the multi-party call method according to claim 10, since the mixing state is disconnected when a voice input smaller than the set value has been continued for a predetermined number of times or more, the load on the network when there is no continuous voice input Can also be reduced.

  Embodiments of the present invention will be described below with reference to the drawings. The “limited mixing” method proposed in the present invention is a method that pays attention to the fact that there are few terminals that actually speak at the same time even if there are many terminals participating.

  FIG. 1 is a block diagram showing a schematic configuration of a multi-party call server and a call terminal to which a multi-party call system according to an embodiment of the present invention is applied. In FIG. 1, reference numeral 1 denotes a multi-party call server, which provides a multi-party call service to a plurality of call terminals 2 (IP telephones). 1 and 2 are both connected by network 3. The network 3 also includes a dedicated network that transmits only IP telephone traffic and a shared line that transmits traffic other than telephone like the Internet.

The call server 1 has a voice mixing function. Each call terminal 2 establishes a session with the call server 1 in advance. A mixing state is established between the terminal (referred to as 2 </ b> A) that makes a statement with the call server 1. Only the voice from the terminal that has established the mixing state is transmitted from the call server 1 to the terminal that has established the session but has not yet established the mixing state. Next, when the call terminal 2A speaks, if the number of other terminals that speak at the same time exceeds a certain number (N), the other terminals cannot speak. That is, priority is given to the call terminal 2A that has made a statement (establishing the mixing state) first. The terminal ends the mixing state with the call server 1 when no voice is input for a certain period of time.

The signaling protocol used in the present embodiment uses SIP (Session Initiation Protocol) defined as a standard.
FIG. 2 shows the exchange of SIP messages when the session is established and terminated between the call server 1 and the call terminal 2. When the call terminal 2 starts a call, the call terminal 2 transmits an INVITE message as a session start request to the call server 1 (step S201). If the call is successful, the call server 1 returns a 180 Ringing, 200 OK message as a response (steps S101 and 103), and the call terminal 2 transmits an ACK in response thereto (step S203) to establish a session.
When disconnecting the session, the call terminal 2 that wishes to request disconnection transmits a BYE message (step S211), and the call server 1 returns a 200 OK message as a response (step S111).

FIG. 3 shows message exchange for establishing a mixing state after the call server 1 and the call terminal 2 establish a session. Here, SPEAK and MUTE are defined as messages conforming to SIP.
The call terminal 2 establishes a session with the call server 1 and requests the call server 1 to mix using the SPEAK message before transmitting voice data (step S221). The call server 1 that has received SPEAK returns 180 Ringing, 200 OK as in the case of receiving INVITE in FIG. 2 (steps S121 and 123). The call terminal 2 receives this and transmits ACK to the call server 1 (step S223), and establishes a mixing state with the call server 1.
When the call terminal 2 stops transmitting voice data, a MUTE message is transmitted from the call terminal 2 (step S231). MUTE corresponds to the BYE of the SIP message in FIG. The call server 1 that has received MUTE returns 200 OK to the call terminal 2 (step S131), and the mixing state ends.

  Here, the configuration of each of the call server 1 and the call terminal 2 and the detailed operation of each of the SIP message exchanges described with reference to FIGS. 2 and 3 will be described below.

  First, functional blocks of the call server 1 are shown in FIG. The call server 1 in this embodiment needs to process various SIP messages and RTP voice packets received from the connected call terminal 2. The call server 1 includes a control unit 10 (mixing control target recording unit, mixing control target deletion unit), a SIP protocol stack unit 11, a mixing unit 12, a terminal connection state management table 13, a mixing control table 14, a packet transmission / reception unit 15 (first). 2 transmission means), an encoding unit 16, a decoding unit 17, an RTP (Real-Time Transport Protocol) packet generation unit 18, and an RTP packet analysis unit 19.

The packet transmitter / receiver 15 is a module for receiving voice packets from the call terminal 2 and transmitting to the call terminals 2, and uses RTP (Real-Time Transport Protocol) to receive and receive voice packets. Check the validity of the packet. Then, packet loss and packet delay are detected, and control and discarding are performed.

The SIP protocol stack unit 11 performs a process of returning a response to the call terminal 2 that has transmitted the SIP message.

Next, detailed operation of the call server 1 will be described.
FIG. 5 shows an operation flow of the call server 1 when the call server 1 and the call terminal 2 establish / disconnect a session. The packet transmitting / receiving unit 15 of the call server 1 receives the SIP message from the specific call terminal 2A (step Sa101). When the received SIP message is an INVITE message (“Yes” in step Sa105), the SIP protocol stack unit 11 returns the response shown in steps S101 and 103 in FIG. 2 (step Sa103) and establishes a session with the call terminal 2A. (Step Sa107). Further, the control unit 10 adds that the call terminal 2A is connected to the terminal connection state management table 13 (step Sa109). Then, the processing proceeds to FIG.
When the SIP message received by the packet transmitting / receiving unit 15 from the call terminal 2A is a BYE message (“No” in step Sa105), the SIP protocol stack unit 11 returns the response shown in step S111 in FIG. 2 (step Sa121). The session with the call terminal 2A is disconnected (step Sa123). Then, the control unit 10 deletes the information on the call terminal 2A from the terminal connection state management table 13 (step Sa125).

FIG. 6 shows an operation flow of the call server 1 when the call server 1 and the call terminal 2 establish and disconnect the mixing state. The call server 1 establishes a session with the call terminal 2A in step Sa107 and waits for a mixing request from the call terminal 2A. The packet transmitting / receiving unit 15 receives a message from the call terminal 2A (step Sa151). The message received here conforms to SIP. If the received message is a SPEAK message (“Yes” in step Sa155), the SIP protocol stack unit 11 returns the responses shown in steps S121 and 123 of FIG. 3 (step Sa153). Then, the control unit 10 adds the call terminal 2A to the mixing control table 14 (step Sa157). As a result, the voice of the call terminal 2A is mixed. Then, the process proceeds to FIG.
When the message received by the packet transmitting / receiving unit 15 from the call terminal 2A is a MUTE message (“No” in step Sa155), the SIP protocol stack unit 11 returns the response shown in step S131 of FIG. 3 (step Sa159), and the control unit 10 deletes the call terminal 2A from the mixing control table 14 (step Sa161). That is, the voice of the call terminal 2A is prevented from being mixed.

FIG. 7 shows an operation flow of RTP transmission / reception processing of the call server 1 in the mixing established state. When the packet transmitting / receiving unit 15 receives the RTP packet from the call terminal 2A (step Sa171), the RTP packet analyzing unit 19 analyzes the received RTP packet. If there is no error in the RTP packet, the decoding unit 17 decodes the voice data (step Sa173) and sends it to the mixing unit 12. The mixing unit 12 calls the information of the call terminal 2A added in Step Sa157 from the mixing control table 14, and mixes only the voice of the call terminal 2A in which the mixing state is established (Step Sa175). After the mixed packet is encoded by the encoding unit 16 (step Sa177), the packet is generated so as to conform to the RTP, and the session is established within the network 3 using the packet transmitting / receiving unit 15 It transmits to all the terminals (step Sa179).

FIG. 8 shows the data structure of the terminal connection state management table 13 in the call server 1, and FIG. 9 shows the data structure of the mixing control table 14. The maximum number of elements in the terminal connection state management table 13 is the maximum number of participants, and the maximum number of elements in the mixing control table 14 is the maximum number of simultaneous speakers.
For example, when SPEAK is received from the call terminal 2, the control unit 10 stores various information (SIP exchange information and the like) in the mixing control table 14. Conversely, when receiving MUTE from the call terminal 2, the control unit 10 discards various information of the call terminal 2 from the mixing control table 14.

  Next, functional blocks of the call terminal 2 are shown in FIG. The call terminal 2 needs to process SIP transmission / reception and RTP transmission / reception. The call terminal 2 includes a memory 20 (number of times storage means), a SIP protocol stack section 21, a connection control section 22 (input voice determination means, a mixing state establishment means, a mixing state disconnection means), and a packet transmission / reception section 23 (second voice packet). Transmission means), voice input unit 24 (voice input means), voice output unit 25, encoding unit 26, decoding unit 27, RTP packet generation unit 28, RTP packet analysis unit 29, and mixing state management unit 30.

The packet transmission / reception unit 23 controls transmission / reception of packets with the call server 1. RTP is used for sending and receiving voice packets, and the validity of received packets is checked. Then, packet loss and packet delay are detected, and control and discarding are performed.

The SIP protocol stack unit 21 performs processing for transmitting a SIP message and receiving a response from the call server 1 in response thereto. The memory 20 stores a minimum value (threshold value) of the volume of sound determined to be input sound. The mixing state management unit 30 manages information indicating whether or not the call terminal 2 has established a mixing state with the call server 1.

Next, detailed operation of the call terminal 2 will be described.
FIG. 11 shows an operation flow of the call terminal 2B when a session is established / disconnected between the specific call terminal 2B and the call server 1. In FIG. 11a, when the user makes a call using the call terminal 2B and tries to establish a session with the call server 1, the SIP protocol stack unit 21 of the call terminal 2B sends an INVITE message via the packet transmission / reception unit 23. It transmits to the call server 1 (step Sb201). When the packet transmitting / receiving unit 23 receives the response shown in steps S101 and S103 in FIG. 2 from the call server 1 (step Sb203), the SIP protocol stack unit 21 returns ACK to the call server 1 (step Sb205), and establishes a session. To do.
In FIG. 11b, when there is a session disconnection request from the user, the SIP protocol stack unit 21 transmits a BYE message to the call server 1 via the packet transmission / reception unit 23 (step Sb207). The packet transmitting / receiving unit 23 receives the response shown in step S111 of FIG. 2 from the call server 1 (step Sb209).

FIG. 12 shows an operation flow of the call terminal 2 when the call terminal 2B and the call server 1 establish and disconnect the mixing state.
When voice is input from the voice input unit 24 of the call terminal 2B (step Sb251), the connection control unit 22 compares the volume of the voice with a voice threshold value stored in advance in the memory 20 (step Sb253). If the input voice is larger than the threshold (“Yes” in step Sb253), it is determined that the voice is input. Next, the connection control unit 22 acquires information on whether or not the call terminal 2 </ b> B and the call server 1 are in the mixing state from the mixing state management unit 30. When the mixing state is not established (“No” in step Sb255), the SIP protocol stack unit 21 transmits SPEAK to the call server 1 via the packet transmission / reception unit 23 (step Sb257). Then, when the packet transmitting / receiving unit 23 receives the response from the call server 1 shown in steps S121 and S123 of FIG. 3, the SIP protocol stack unit 21 returns ACK and establishes a mixing state (step Sb258). The connection control unit 22 records in the mixing state management unit 30 that the call terminal 2 is in the mixing state (step Sb259).
When the mixing state is established in step Sb255 (“Yes” in step Sb255), the process proceeds to step Sb275 in FIG.

When the volume of the input voice is equal to or less than the threshold value of the voice stored in the memory 20 (“No” in step Sb253), the connection control unit 22 records the number of voice inputs equal to or less than the threshold value in the memory 20 (step Sb260). ). Further, the connection control unit 22 acquires information on whether or not the call terminal 2B and the call server 1 are in the mixing state from the mixing state management unit 30. When the mixing state is established (“Yes” in step Sb261), the connection control unit 22 acquires the number of times that the voice input is continuously lower than the threshold, which is recorded in the memory 20.
If the voice input is continuously lower than the threshold value for a certain number of times (“Yes” in step Sb263), the connection control unit 22 determines that there is no voice input for a certain time. And in order to cut | disconnect a mixing state, the SIP protocol stack part 21 transmits MUTE to the telephone call server 1 via the packet transmission / reception part 23 (step Sb265). When the packet transmitter 23 receives the response from the call server 1 shown in step S131 of FIG. 3, the mixing state is disconnected (step Sb267). The connection control unit 22 records in the mixing state management unit 30 that the call terminal 2 is no longer in the mixing state (step Sb269).
If the number of times the voice input has fallen below the threshold is not continuous for a certain number of times (“No” in step Sb263), the next voice input is again waited.

FIG. 13a shows an operation flow of the RTP transmission process of the call terminal 2 in the mixing established state. The voice (step Sb271) input from the voice input unit 24 establishes a mixing state by the operation shown in FIG. 12 (step Sb273), and is encoded by the encoding unit 26 (step Sb275). The encoded voice data is generated as an RTP packet by the RTP packet generator 28 and transmitted from the packet transmitter / receiver 23 to the call server 1 (step Sb277).

FIG. 13 b shows an operation flow of the RTP reception process of the call terminal 2. When the packet transmitting / receiving unit 23 receives an RTP packet from the call server 1 (step Sb281), the RTP packet analyzing unit 29 analyzes the received RTP packet. If there is no error, the decoding unit decodes the audio data (step Sb283), sends the audio data to the audio output unit 25, and outputs the audio (step Sb285).

  As described above, the method called “limited mixing” of the present invention significantly reduces the number of streams that need to be mixed by processing only the voice of the terminal that is currently speaking by the multi-party call server. . A multi-party call is realized by mixing only the voice of the speaker in the multi-party call server and transmitting the obtained voice to the terminal.

When the voice is input, it is checked whether the mixing state has already been established with the call server, and if not established, a SPEAK message is transmitted to request the call server to establish the mixing state. When the mixing state is established, the input voice is encoded by the encoding unit. Next, the RTP packet generation unit shapes the packet as an RTP packet, and then passes the packet to the packet transmission / reception unit. If there is no voice input for a certain period of time, a MUTE message is transmitted to the server to end the mixing state. When the RTP packet analysis unit receives the RTP packet, the RTP packet analysis unit checks whether there is an error. If there is no error, the RTP packet analysis unit decodes the audio data and passes the data to the audio output unit.

The call terminal can start a multi-party call simply by starting a session with the multi-party call server. Furthermore, after establishing a session, the call terminal can make a call without using an additional operation such as pressing a floor request button by simply using an input device such as a microphone. Since the speaker can speak by voice input from the microphone, there is an effect that the consciousness of the speaking right as in the chairman-based speech method can be made unnecessary.

  As mentioned above, although embodiment of this invention was explained in full detail with reference to drawings, the specific structure is not restricted to this embodiment, The design change etc. of the range which does not deviate from the summary of this invention are included.

It is a block diagram which shows schematic structure of the multi-party call server and user terminal to which the call system which concerns on one Embodiment of this invention is applied. It is a figure which shows the exchange of the SIP message in the session establishment and disconnection of the call server 1 and the call terminal 2 in embodiment same as the above. It is a figure which shows the exchange of the message in establishment and disconnection of the mixing state of the call server 1 and the call terminal 2 in embodiment same as the above. It is a figure which shows the functional block of the telephone call server 1 in embodiment same as the above. It is a flowchart which shows operation | movement of the telephone call server 1 at the time of performing session establishment / disconnection. It is a flowchart which shows operation | movement of the telephone call server 1 at the time of establishing / disconnecting a mixing state. It is a flowchart which shows the operation | movement of the RTP transmission / reception process of the call server 1. 4 is a table showing a data structure of a terminal connection state management table 13 in the call server 1. It is a table which shows the data structure of the mixing control table 14 in the telephone call server 1. It is a figure which shows the functional block of the telephone terminal 2 in embodiment same as the above. It is a flowchart which shows operation | movement of the communication terminal 2 at the time of establishing a session. It is a flowchart which shows operation | movement of the telephone call terminal 2 at the time of disconnecting a session. It is a flowchart which shows operation | movement of the communication terminal 2 at the time of establishing / disconnecting a mixing state. 4 is a flowchart showing an operation of RTP transmission processing of the call terminal 2. 4 is a flowchart showing an operation of RTP reception processing of the call terminal 2.

Explanation of symbols

1 ... Call server, 2 ... Call terminal (IP phone), 3 ... Internet,
DESCRIPTION OF SYMBOLS 10 ... Control part, 11 ... SIP protocol stack part of call server, 12 ... Mixing part, 13 ... Terminal connection state management table, 14 ... Mixing control table, 15 ... Packet transmission / reception part, 16 ... Coding part, 17 ... Decoding , 18 ... RTP packet generator, 19 ... RTP packet analyzer,
DESCRIPTION OF SYMBOLS 20 ... Memory of terminal, 21 ... SIP protocol stack part, 22 ... Connection control part, 23 ... Packet transmission / reception part, 24 ... Voice input part, 25 ... Voice output part, 26 ... Encoding part, 27 ... Decoding part, 28 ... RTP packet generator, 29 ... RTP packet analyzer, 30 ... mixing state manager

Claims (8)

In a multi-party call system in which a call server and a plurality of call terminals are connected via a network, and audio from a specific call terminal is transmitted to each call terminal that has established a session with the call server.
The call terminal is
Voice input means for inputting voice;
Input voice determination means for determining whether or not the volume of the voice input to the voice input means is larger than a set value;
If the voice input by the input voice determination means is larger than a set value, a mixing request is made to the call server, a response is received from the call server, and a mixing state is established to establish a mixing state;
After the mixing state with the call server is established by the mixing state establishing means,
First voice packet transmitting means for packetizing the voice input by the voice input means and transmitting the packet to the call server;
The call server is
A mixing control target recording means for transmitting a response to the call terminal that has made the mixing request and recording the response as a mixing control target;
And a second voice packet transmitting means for transmitting the voice packet transmitted from the call terminal recorded by the mixing control target recording means to another call terminal that has established a session. Inter-call system The call terminal is
When the input voice is determined by the input voice determination means is smaller than the set value, the number of times storage means for storing how many times the voice input smaller than the set value is continued;
When the number of voice input times smaller than a set value stored in the number storage means continues for a predetermined number of times or more, a request to disconnect the mixing is made to the call server, a response from the call server is received, and the mixing state is changed. A mixing state cutting means for cutting;
The call server is
The multi-party call according to claim 1, further comprising: a mixing control target deleting unit that transmits a response to the communication terminal that has requested mixing disconnection by the mixing state disconnecting unit, and deletes the response from the mixing control target. system.   The call terminal transmits a session start request message to the call server, and the call server transmits a session start response message to the session start request message, thereby establishing a session between the call terminal and the call server. The multi-party call system according to claim 1 or 2.   The call terminal transmits a session disconnection request message to the call server, and the call server disconnects a session between the call terminal and the call server by transmitting a session disconnection response message in response to the session disconnection request message. The multi-party call system according to claim 1 or 2. In a call terminal in a multi-party call system in which a call server and a plurality of call terminals are connected via a network, and audio of a specific call terminal is transmitted to each call terminal that has established a session with the call server,
Voice input means for inputting voice;
Input voice determination means for determining whether or not the volume of the voice input to the voice input means is larger than a set value;
A mixing state establishing means for making a mixing request to the call server and receiving a response from the call server to establish a mixing state when the sound input by the input sound determining means is larger than a set value;
After the mixing state with the call server is established by the mixing state establishing means,
A call terminal comprising: a first voice packet transmission unit configured to packetize the voice input by the voice input unit and transmit the packet to the call server. When the input voice is determined by the input voice determination means is smaller than the set value, the number of times storage means for storing how many times the voice input smaller than the set value is continued;
When the number of voice input times smaller than a set value stored in the number storage means continues for a predetermined number of times or more, a request to disconnect the mixing is made to the call server, a response from the call server is received, and the mixing state is changed. 6. The call terminal according to claim 5, further comprising a mixing state cutting means for cutting. In a multi-party call method in which a call server and a plurality of call terminals are connected via a network, and audio from a specific call terminal is transmitted to each call terminal that has established a session with the call server.
  The call terminal is
  A step in which audio is input,
  Determining whether the volume of the input voice is greater than a set value;
  When the input voice is larger than the set value, a request for mixing is made to the call server,
Receiving a response from the call server and establishing a mixing state;
  The voice input by the voice input means after the mixing state with the call server is established
Packetizing and transmitting to the call server,
  The call server is
  Sends a response to the call terminal that made the mixing request and records it as a mixing control target
Steps,
  The recorded voice packet sent from the corresponding call terminal is sent to other established session.
And transmitting to the call terminal
  A multi-party call method characterized by the above. The call terminal is
  If the input audio is lower than the set value, the audio input lower than the set value will continue
A step of memorizing whether or not
  If the number of times of voice input that is smaller than the stored set value continues for a predetermined number of times,
Requests the disconnection to the talk server and receives a response from the call server
Cutting the state further,
  The call server is
  The call terminal that sent the response to the mixing disconnect request is deleted from the mixing control target.
And a step of removing
  The multi-party call method according to claim 9.

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