JPH06350725A - Video conference system - Google Patents

Abstract

PURPOSE:To prevent any unpleasant feeling from being given to a user by preventing noise from being generated from a speaker or the like by inhibiting the input/output of voices until a voice CODEC is normally operated. CONSTITUTION:When establishing communication in a prescribed procedure and requesting the changes of voice CODECs 18a-18d from the call originating side, an audio command BAS is sent and voice data are sent on the new encoding side. When the change of the BAS is identified on the call incoming side, an SW 50 is switched to the confirmed voice CODEC 18. At such a time, the voice CODEC 18 under using at present is stopped, the requested CODEC is started, the CODEC 18 under using at present is disconnected from amplifiers 53 and 54 by an SW 55, and a microphone 19 and a speaker 20 are disconnected. At the same time, a timer is started, the SW 55 is operated corresponding to time-out after the lapse of prescribed time, and the microphone and speaker are connected to the switched CODEC. Thus, no unpleasant noise is applied at the time of usual switching.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a video conference system, and more particularly to a video telephone and a video conference system, as well as to AV equipment in general, especially when switching a voice codec. The present invention relates to a video conference system that can prevent noise from being generated in a microphone or a speaker.

[0002]

2. Description of the Related Art Recently, information exchange of voice, image, etc. in real time has been carried out in a video telephone or a video conference system.
A communication terminal that operates via an SDN digital line has been developed and put on the market. These terminals are multimedia communications mainly composed of a combination of audio (audio) media and image (visual) media. The communication of audio includes communication with a handset (telephone type) and communication with a microphone and a speaker that can be communicated by a plurality of people. The former is used for a terminal such as a videophone that performs one-to-one communication, and the latter is used for a terminal such as a videoconference that performs a plural-to-plural communication.

In addition, during such a video conference, a voice codec is transmitted according to a predetermined CCITT recommendation, for example, G.264. 711
And G. By switching in accordance with 722 or the like, it is possible to change the quality or the like of the sound input / output to / from the microphone or the speaker.

[0004]

However, in the conventional video conference system, after switching the voice codec, a predetermined time is required until the newly switched voice codec operates normally. That is, for example, in advanced voice codecs, etc., it takes time from startup to normal operation, so noise is generated in the speaker etc. between startup and normal operation, which causes user discomfort. was there.

Further, since there is no means for switching the voice codec at the will of the user during voice communication,
There is a problem that it is not possible to input and output a voice with a voice codec desired by the user. Therefore, in the invention according to claim 1, when the voice codec is switched during voice communication, by disconnecting the input / output unit from the voice codec for a predetermined time, it is possible to prevent noise from occurring in the speaker or the like. It is an object of the present invention to provide a video conferencing system that can prevent the user from feeling uncomfortable.

According to the second aspect of the present invention, when the voice codec is switched during voice communication, noise is generated in the speaker or the like by lowering the voice level of the input / output unit for a predetermined time. It is an object of the present invention to provide a video conference system that can be prevented and can prevent the user from feeling uncomfortable. According to the invention of claim 3, among the voice codecs, the one that requires time from the start of activation to the normal operation is separated from the voice codec for a long time, and the time until the normal operation is achieved. To provide a video conferencing system that can optimize the switching time of the voice codec by separating the unnecessary ones for a short time and optimize the noise suppression according to the voice codec. It is an object.

According to the fourth aspect of the present invention, by outputting a melody from the input / output unit within a predetermined time to notify that the system is not abnormal, the user can be surely notified that the system is not abnormal. , Is intended to provide a video conferencing system with high performance. According to the invention of claim 5, by displaying a message or an image on a television screen or a display panel within a predetermined time to notify that the system is not abnormal, the user can be surely notified that the system is not abnormal.
It is intended to provide a video conference system with high usage performance.

According to a sixth aspect of the present invention, an operating section having a switch for switching the voice codec and a switching section for switching the voice codec when it is detected that the switch is operated during voice communication are provided. Accordingly, it is an object of the present invention to provide a video operability conference system with high operability in which the voice codec can be easily switched by the user's intention during the voice communication and the voice communication can be performed by the voice codec desired by the user.

[0009]

[Means for Solving the Problems] Claim 1
In a video conference system connected to an ISDN (Integrated Services Digital Network) line for exchanging data such as voice, moving images, and facsimile still images, CCITT Recommendation H.264. A media multiplexing / demultiplexing unit that multiplexes / demultiplexes media according to the 221 framing format, and audio data multiplexed or demultiplexed by the media multiplexing / demultiplexing unit are encoded / combined according to a predetermined CCITT recommendation or the like. A voice codec unit having a plurality of voice codecs, an input / output unit connected to the voice codec unit for inputting / outputting voice data to / from the codec unit, and detecting that the voice codec is switched during voice communication Then, the control unit controls the input / output unit to be disconnected from the voice codec only for a predetermined time.

The invention of claim 2 is the ISDN (Integr
CCTVT Recommendation H.264 in a video conferencing system that is connected to a line and exchanges data such as voice, moving images, and facsimile still images. A media multiplexing / demultiplexing unit that multiplexes / demultiplexes media according to the 221 framing format, and audio data multiplexed or demultiplexed by the media multiplexing / demultiplexing unit are encoded / combined according to a predetermined CCITT recommendation or the like. A voice codec unit having a plurality of voice codecs, an input / output unit connected to the voice codec unit for inputting / outputting voice data to / from the codec unit, and detecting that the voice codec is switched during voice communication Then, a voice level control unit that controls to reduce the voice level of the input / output unit only for a predetermined time,
It is characterized by having.

The invention of claim 3 is the same as claim 1 or 2.
In the invention described above, the predetermined time is a voice codec that takes a long time to start operating normally after being started, and is set to a long time, and does not take time to operate normally. Is characterized by being set to a short time. Claim 4 The invention of claim 4
In the invention described in any one of 3 above, the melody is output from the input / output unit within the predetermined time to notify that the system is not abnormal.

According to a fifth aspect of the invention, in the invention according to any one of the first to third aspects, the system is not abnormal by displaying a message or video on the television screen or the display panel within the predetermined time. The feature is to inform. The invention of claim 6 is the ISDN (In
CCITT in a video conferencing system that is connected to an integrated services digital network) and exchanges data of voice, moving images and facsimile still images.
Recommendation H. 221, a media demultiplexing / demultiplexing unit that demultiplexes and demultiplexes multimedia, and audio data that is multiplexed or demultiplexed by the multimedia demultiplexing / demultiplexing unit is encoded / combined according to a predetermined CCTI recommendation or the like. A voice codec unit having a plurality of voice codecs to be converted, an operation unit having a switch for switching the voice codec, and a switching unit for switching the voice codec when detecting that the switch is operated during voice communication, It is characterized by having.

[0013]

According to the first aspect of the invention, when the voice codec is switched during voice communication, the input / output unit is disconnected from the voice codec only for a predetermined time. Therefore, input / output of voice is not performed until the voice codec operates normally, and noise is not generated in the speaker or the like. As a result, the user does not feel uncomfortable.

According to the second aspect of the invention, when the voice codec is switched during voice communication, the voice level of the input / output unit is lowered only for a predetermined time.
Therefore, input / output of voice is not performed until the voice codec operates normally, and noise is not generated in the speaker or the like. As a result, the user does not feel uncomfortable.

According to the third aspect of the present invention, it is necessary for the input / output unit to be disconnected from the voice codec for a long time and to operate normally when it takes time from the start of activation of the voice codec to the normal operation. Things that don't take a long time are designed to be separated for a short time.
Therefore, the switching time of the voice codec is optimized, and the noise generated in the speaker or the like according to the voice codec is optimally suppressed.

According to the invention of claim 4, within a predetermined time,
A melody is output from the input / output unit to notify that the system is not abnormal. Therefore,
The user can be surely notified that the system is not abnormal, and the usage performance of the conference system is improved. According to the fifth aspect of the invention, the message or the image is displayed on the television screen or the display panel within the predetermined time to notify that the system is not abnormal. Therefore, the user can be surely notified that the system is not abnormal, and the usage performance of the conference system is improved.

According to a sixth aspect of the present invention, an operating section having a switch for switching the voice codec and a switching section for switching the voice codec when it is detected that the switch is operated during voice communication are provided. . Therefore, the voice codec can be easily switched at will of the user during voice communication, and the sound quality and the like can be easily switched. As a result, the operability of the video conference system is improved.

[0018]

Embodiments First, the frame structure will be described. A communication terminal such as a video telephone or a video conference system for exchanging information such as voice and images in real time via an ISDN digital line has been developed and put on the market. These terminals are multimedia communications mainly composed of a combination of audio (audio) media and image (visual) media. The communication of audio includes communication with a handset (telephone type) and communication with a microphone and a speaker that can be communicated by a plurality of people. The former is used for a terminal such as a videophone that performs one-to-one communication, and the latter is used for a terminal such as a videoconference that performs a plural-to-plural communication. In addition, as a document exchange necessary for such a video conference, data transmission such as an electronic blackboard function, a conference material transmission function, or telewriting is performed as in the facsimile function. A multimedia terminal needs to be divided and multiplexed into various signals such as voice, data, video and system control.

The rule for this is called frame structure. When viewed in the bit direction, each sub-channel of 8 kbps can be defined, and two frames can be collected to form one sub-multi-frame. The time length of each unit is: frame: 10 ms, sub-multiframe: 20
msec, multiframe: 160 msec. FIG. 1 shows Recommendation H.264. FIG. 22 is a diagram illustrating details of a frame configuration when 64 kbps is used for an AV terminal according to 221. The FAS (frame synchronization signal) assigned to the eighth sub-multiframe synchronizes the frame and the sub-multiframe, and the BA
By controlling S (bit rate allocation signal), the multiplexing method can be switched for each sub-multiframe (20 msec). The FAS (frame synchronization signal) includes a control signal and alarm information in addition to the framing information and the multi-framing information. Furthermore, error detection information for controlling the end-to-end error characteristics and checking the effectiveness of frame synchronization is also included. And
The capacity of the terminal is exchanged by BAS (bit rate allocation signal), and as a result of this capacity exchange, the command is output with the maximum capacity and the exchange mode is determined. This signal can also be used for control and notification signals (C & I).

The types of voice codecs (described later) that can be used in the above-mentioned multimedia terminals are G. 711 (64 kbit / s 3.4 KH
z-band audio PCM coding system 0U OF), G.I.
722 (64 Kbit / s or less 7 KHz band audio encoding system), G.I. 728 (16kbit / s 3.4K
Hz band audio encoding method) and G. 721, or audio encoding unique to each company.

FIG. 2 is a block diagram showing the configuration of a video conference system according to an embodiment of the invention described in any one of claims 1 to 6. In FIG. 2, reference numeral 1 is a CPU unit, and this CPU unit 1 manages and controls this system according to the data stored in the ROM of the memory unit 2. The memory unit 2 has a working area other than the ROM. RAM
Exists.

Further, 3 is a timer section, and this timer section 3 provides a timer necessary for the system,
4 is F according to the communication protocol defined by CCITT
A communication control unit for exchanging AX, video, or control information with a communication partner terminal, 5 is CCITT recommendation H.264. 22
1 is a media multiplexing / separating unit that multiplexes / separates media according to the framing format 1.

Reference numeral 7 is a network interface S / IF6.
Is an ISDN network connected to the BCH and the data on the Bch is connected to the media multiplexing / separating unit 5, and 9 is a user who inputs the telephone number and operates the video on / off switch to display the telephone number of the other party. An operation / display unit providing an interface portion, 8 is an input control unit having an interface between the operation / display unit 9 and the system, and 10 is CCITT Recommendation H.264. An image codec unit that encodes and decodes a moving image according to H.261.

Reference numeral 11 is an image input / output I / F for interfacing the image codec unit 10 with an image input device 12 such as a camera and an image output device 13 such as a television in accordance with a standard such as NTSC, and 14 is a hard disk. The I / O unit serves as a digital interface unit such as SCSI, serial, and parallel for interfacing with 15, the still image device 16, and the other telematic device 17 (telewriting, etc.).

Further, 18 is CCITT recommendation G. 711, G.I. Reference numeral 722 denotes a voice codec unit that performs voice coding and decoding, 19 is a microphone that is a voice input device, and 20 is a speaker that is a voice output device. Further, 22 is a printer that is a document output device, 23 is a scanner that is a document reading device, and 21 is a document control unit that controls the document by interfacing the printer 22 with the scanner 23 and the system. is there.

FIG. 3 is a block diagram showing the configuration around the voice codec unit 18. As shown in the figure, the voice codec 18 has a plurality of voice codecs 18a to 18d. 711, G712, or audio data that is multiplexed or demultiplexed by the media demultiplexing / demultiplexing unit 5 according to each company's own audio encoding or the like.

The voice codec unit 18 is connected to the media multiplex / separation unit 5 via a switch 50, and the switch 50 can switch the media multiplex / separation unit 5 and each of the voice codecs 18a to 18d. There is. The microphone 19 and the speaker 20 are connected to the external microphone 51 and the external speaker 52, respectively, and the audio codec 18a is connected via the amplifiers 53 and 54.
.About.18d, and the audio level can be varied by the amplifiers 53 and 54. The microphone 19, the speaker 20, and the amplifiers 53 and 54 form an input / output unit that inputs / outputs audio data to / from the audio codecs 18a to 18d.

Further, the amplifiers 53 and 54 and each voice codec
A switch 55 is interposed between 18a to 18d, and the microphone 19 and the speaker 20 are switched and connected to the voice codecs 18a to 18d by the switch 55. Further, the switch 50 is switched by the CPU unit 1.
When the CPU section 1 determines which encoding rule the currently received audio data is in accordance with a command BAS signal which will be described later, the CPU section 1 selects the switch 50 according to the type of the audio codec. Voice codecs 18a to 18 for switching the media and designating the media multiplexing / separating unit 5
Connected to d, the designated voice codecs 18a-18d are activated, and the voice data from the media multiplex / separation unit 5 can be transmitted to the designated voice codecs 18a-18d.

Further, the CPU unit 1 has a voice codec 18
When a change in the audio BAS command is detected when a to 18d are switched, the timer section 3 is activated for a predetermined time, and the amplifiers 53 and 54 are disconnected from the voice codecs 18a to 18d only during this period. Therefore, the CPU unit 1 and the switch 55 form a control unit.

The start-up time of the timer unit 3 is set to a long time among the voice codecs 18a to 18d, which takes a long time from the start of the start-up to the normal operation until the normal operation is completed. Things that don't take a long time are set to be short. Also, the CPU unit 1
Is designed to output a melody from the speaker 20 within the activation time of the timer 3, and when switching the voice codecs 18a to 18d, each codec 18a to 18d.
When the 18d, the microphone 19 and the speaker 20 are disconnected from each other, the interruption of the voice informs that the system is not abnormal.

Next, the communication sequence and operation procedure of the video conference system according to the first embodiment of the present invention will be described with reference to FIGS. Here, the AV terminal is the calling side terminal (X side terminal), the main conference system is the receiving side terminal, and that terminal is the Y side terminal. First, the X-side terminal sets the circuit switching mode as the transfer mode, and at the same time, the call-setting message SE that specifies the Y-side terminal as the destination.
TUP is sent to ISDN to request call setup with the Y-side terminal, and ISDN sends a call setup message SETUP to the designated Y-side terminal to make a call. ISDN is
Call setup acceptance message CA for notifying the call setup status
LL-PROC is sent to the X-side terminal.

Next, when the Y-side terminal detects an incoming call, it starts ringing a ringing sound to call the user of the terminal itself, and also sends ALERT a message notifying the ISDN that the called party is calling. The ISDN sends a call message ALERT to the X-side terminal to notify that the call has started. When the user performs an incoming call response such as picking up the handset on the Y side terminal, the response message C
The ONN is sent to the ISDN, and the ISDN sends the response message CONN to the X side terminal to notify the Y side terminal that the call has been accepted.

The ISDN sends a response confirmation message CONN-ACK to the Y-side terminal to confirm the response from the Y-side terminal, and at that time, a data transfer (call) is made between the X-side terminal and the Y-side terminal. Bch is established for this purpose, and the call operation using Bch is appropriately performed between the users of the X-side terminal and the Y-side terminal. On Bch, H.264 shown in FIG. Initialize the first channel in the framing format 211. In initialization, after synchronization is established by FAS, B
We will exchange capabilities with each other. Until the capability exchange sequence is completed, the voice communication is G.264. 711 voice codec (OF 56 kbit / s).

If the X-side terminal requests a change in the voice codecs 18a-18d, the audio command BA
Simultaneously with the transmission of S, the voice data is started to be transmitted by the new coding law. When the change in the audio command BAS is identified by the received BAS, the Y-side terminal switches the switch 50 to the voice codecs 18a to 18d confirmed by the BAS.

Here, voice communication is referred to as G. From 711 onward, G.
The procedure for switching to 722 will be described with reference to the flowchart of FIG. The flowchart shows a control program executed by the CPU unit 1 and stored in the ROM of the memory unit 2. First, it is determined whether or not the received audio command BAS has changed during the ongoing conference (during voice data communication, step S1).
(Step S2), for example, G.I. When there is a request for mode switching in 722, the voice codecs 18a-18d currently used are stopped and the voice codecs 18a-18d for which switching is requested are activated. In addition to this, the switch 55 is operated to disconnect the currently used voice codec from the amplifiers 53 and 54, disconnect the codec from the microphone 19 and the speaker 20, and activate the timer unit 3 (step S3).

Next, it is determined whether or not the timer section 3 has timed out (step S4). If it has timed out, the switch 55 is operated to connect the microphone 19 and the speaker 20 to the voice codec after switching. (Step S5). Also, while the timer unit 3 is operating, a melody is sent from the speaker 20 to inform the user that the system is not abnormal.

Further, the activation time of the timer unit 3 is set to a long time if it takes time from the start of activation of the voice codec until it normally operates, and it does not take time to operate normally. Is set for a short time. FIG. 6 shows a specific example. In FIG. 6, the horizontal line indicates the currently used voice codec, and the vertical line indicates the newly activated codec. 711 is used and G. When switching to 722, the activation time of the timer unit 3 is set to 0, 5 seconds.

On the other hand, when the voice communication ends, the line is disconnected according to the following procedure. For example, when the user of the X-side terminal first returns the handset and goes on hook, the X-side terminal sends a disconnection message DISC to ISDN and Bc
Request the release of h. ISDN is a disconnection message DI
The SC is sent to the Y side terminal of the conference system. In response to the disconnection message DISC, the Y-side terminal responds to the ISDN with a release message REL notifying the disconnection completion.
The release message REL is sent from the DN to the X-side terminal.

When the release of Bch is completed, the Y-side terminal sends a release completion message REL-COMP to ISDN to notify it, and ISDN releases the release completion message RE.
L-COMP is sent to the Y-side terminal. As a result, the Bch set between the X-side terminal and the Y-side terminal is completely released. As described above, in this embodiment, when the voice codecs 18a to 18d are switched during voice communication, the timer unit 3 is activated to disconnect the microphone 19 and the speaker 20 from the voice codecs 18a to 18d for a predetermined time. Therefore, it is possible to prevent the voice input / output from being performed until the voice codecs 18a to 18d operate normally, and it is possible to prevent noise from being generated from the microphone 19 and the speaker 20. As a result, it is possible to prevent the user from feeling uncomfortable.

Among the voice codecs 18a to 18d, the ones which require time from the start of activation to the normal operation of the voice codecs 18a to 18d.
It is separated from 18d for a long time, and those that do not require time to operate normally are separated for a short time. Therefore, the switching time of the voice codecs 18a to 18d can be optimized, and the voice codec 18a can be switched.
It is possible to optimally suppress the noise generated in the microphone 19 and the speaker 20 in accordance with ~ 18d.

When the timer unit 3 is activated, the speaker
Since the melody is output from 20 to notify that the system is not abnormal, the user can be surely notified that the system is not abnormal, and the usage performance of the conference system can be improved. In the present embodiment, the system 20 is informed that the system is not abnormal by sending out a melody from the speaker 20, but the present invention is not limited to this, and the image output device 13 and the operation / display as in the invention of claim 4 are performed. A message may be displayed on the section 9 to notify that the system is not abnormal.

The operation according to the invention of claim 6 is also possible.
A switch for switching the voice codecs 18a to 18d is provided on the display unit 9, and when the user manually operates this switch during voice communication, the CPU unit 1 detects this and switches the voice codecs 18a to 18d. Is also good. By doing so, the voice codecs 18a to 18d can be easily switched by the user's will during the voice communication to easily switch the sound quality and the like, and the operability of the video conference system can be improved. it can. In this case, the CPU section 1 constitutes the switching section referred to in claim 6.

FIG. 7 is a flow chart showing an embodiment of the video conference system according to the present invention.
The feature of the present embodiment is that when the voice codec is switched during voice communication in the above embodiment, the timer unit 3 is activated to disconnect the microphone 19 and the speaker 20 from the voice codecs 18a to 18d for a predetermined time. On the other hand, C
When the PU unit 1 detects that the voice codecs 18a to 18d are switched during voice communication, the timer unit 3 is activated to lower the voice level of the microphone 19 and the speaker 20 for a predetermined time. 2 and 3 of the above-described embodiment, since other configurations are similar to those of the above-described embodiment.
Follow the instructions below.

In this embodiment, as shown in FIG. 7, it is determined whether or not the received audio command BAS has changed during the ongoing conference (during voice data communication, step S11) (step S12). , G.G. When there is a request for mode switching in 722, the voice codecs 18a-18d currently used are stopped and the voice codecs 18a-18d for which switching is requested are activated. In addition to this, at the same time that the levels of the amplifiers 53 and 54 are attenuated (decreased), the timer unit 3 is activated (step S13).

Next, it is determined whether or not the timer section 3 has timed out (step S14), and if it has timed out, the levels of the amplifiers 53 and 54 are restored (step S15). Then, while the timer unit 3 is operating, a melody is sent from the speaker 20 to inform the user that the system is not abnormal. In this embodiment, the CPU
The unit 1 constitutes a voice level control unit.

Also in this embodiment, it is possible to prevent the voice input / output from being performed until the voice codecs 18a to 18d operate normally as in the above-described embodiment.
It is possible to prevent noise from being generated from the speaker 20. As a result, it is possible to prevent the user from feeling uncomfortable.

[0047]

According to the first and second aspects of the present invention, it is possible to prevent the voice input / output from being performed until the voice codec operates normally, and prevent noise from being generated from the speaker or the like. be able to. As a result, it is possible to prevent the user from feeling uncomfortable.

According to the invention described in claim 3, the switching time of the voice codec can be optimized.
It is possible to optimally suppress the noise generated in the speaker or the like according to the voice codec. According to the fourth and fifth aspects of the present invention, the user can be surely notified that the system is not abnormal, and the usage performance of the conference system can be improved.

According to the invention described in claim 6, the voice codec can be easily switched by the user's will during the voice communication so that the voice codec can be easily switched. Can be improved.

[Brief description of drawings]

1 Recommendation H. FIG. 22 is a diagram illustrating details of a frame configuration when 64 kbps is used for an AV terminal according to 221.

FIG. 2 is a block diagram showing a system configuration of a video conference system according to an embodiment of the invention described in any one of claims 1 to 6.

FIG. 3 is a block diagram showing a configuration around a voice codec unit according to an embodiment.

FIG. 4 is a diagram showing an example of a communication sequence of the video conference system according to the embodiment.

FIG. 5 is a diagram showing a procedure when switching a voice codec according to an embodiment.

FIG. 6 is a diagram showing a timer activation time according to a voice codec according to an embodiment.

FIG. 7 is a diagram showing a procedure at the time of switching a voice codec according to an embodiment of the invention described in claim 2;

[Explanation of symbols]

1 CPU unit (control unit, audio level control unit, switching unit) 2 Memory unit 3 Timer unit 4 Communication control unit 5 Multimedia multiplexing / separating unit 6 S / IF 7 ISDN network 8 Input control unit 9 Operation / display unit 10 image Codec unit 11 Image input / output I / F 12 Image input device 13 Image output device 14 I / O unit 15 Hard disk 16 Still image device 17 Telematic device 18 Voice codec unit 18a-18d Voice codec 19 Microphone (input unit) 20 Speaker ( Output section) 21 Document control section 22 Printer 23 Scanner 53, 54 Amplifier (input / output section) 55 Switch (control section)

Claims (6)

[Claims] 1. ISDN (Integrated Services Digital)
Network) A video conference system connected to a line for exchanging data such as voice, moving images and facsimile still images. 22. Media demultiplexing that demultiplexes and demultiplexes media according to the 221 framing format.
A demultiplexing unit, a voice codec unit having a plurality of voice codecs for encoding / decoding voice data multiplexed or demultiplexed by the media multiplexing / demultiplexing unit according to a predetermined CCITT recommendation, and a voice codec unit, respectively. When it is detected that the input / output unit is connected and that inputs / outputs voice data to / from the codec unit and that the voice codec is switched during voice communication, the input / output unit is disconnected from the voice codec for a predetermined time. A video conferencing system, comprising: 2. ISDN (Integrated Services Digital)
Network) A video conference system connected to a line for exchanging data such as voice, moving images and facsimile still images. 22. Media demultiplexing that demultiplexes and demultiplexes media according to the 221 framing format.
A demultiplexing unit, a voice codec unit having a plurality of voice codecs for encoding / decoding voice data multiplexed or demultiplexed by the media multiplexing / demultiplexing unit according to a predetermined CCITT recommendation, and a voice codec unit, respectively. When the input / output unit that is connected and inputs / outputs audio data to / from the codec unit is detected, and the voice codec is switched during voice communication, the audio level of the input / output unit is lowered for a predetermined time. A video conferencing system, comprising: 3. The predetermined time is a voice codec that requires a long time from the start of activation to a normal operation and is set to a long time, and does not require a long time to operate normally. The video conference system according to claim 1 or 2, wherein is set to a short time. 4. The video conferencing system according to claim 1, wherein the input / output unit outputs a melody within the predetermined time to notify that the system is not abnormal. 5. The video conference according to claim 1, wherein a message or an image is displayed on a television screen or a display panel within the predetermined time to inform that the system is not abnormal. system. 6. ISDN (Integrated Services Digital)
Network) A video conferencing system connected to a line for exchanging data of voice, moving images, and facsimile still images. 221, a media demultiplexing / demultiplexing unit for demultiplexing and demultiplexing multimedia according to the 221 framing format, and audio data multiplexed or demultiplexed by the multimedia demultiplexing / demultiplexing unit are encoded / combined according to a predetermined CCTI recommendation or the like A voice codec unit having a plurality of voice codecs to be converted, an operation unit having a switch for switching the voice codec, and a switching unit for switching the voice codec when detecting that the switch is operated during voice communication, A video conferencing system having.