Abstract: A voice communication arrangement having a voice communication program installed in a computer loaded with a versatile OS thereon includes a voice data input/output function for passing through the OS voice data produced from voice sound captured by a microphone of the arrangement or received from another voice communication device, when connected to the arrangement, without being processed by the OS as audio data.

Abstract: There are provided with an IP telephone system having both convenience of the softphone and durability of the hardphone, and a speech processing peripheral device (10) for the IP telephone system. The speech processing peripheral device as a peripheral device of an information processing device (20) is connected to the information processing device (20) for executing a call control program and functioning as a call controller. This speech processing peripheral device (10) includes a host connection means for sending/receiving an audio signal to/from the information processing device (20) and a speech processing means for executing speech processing in IP telephone communication. The IP telephone system according to the present invention includes the information processing device (20) for executing the call control program and functioning as the call controller and the speech processing peripheral device (10) according to the present invention, connected to the information processing device (20).

Abstract: A telephone terminal executes voice communication with a partner telephone terminal by using a selected codec program to code and decode voice signals. Both telephone terminals have facilities for storing codec programs and transferring codec programs over a communication network. During a voice call, one of the two telephone terminals may transfer a codec program to the other telephone terminal, and the call may then proceed by use of the transferred codec program. This enables voice communication to take place with the best codec program that either telephone terminal can provide, even if the two telephone terminals are associated with different communication systems.

Abstract: A voice mixing apparatus decodes input encoded narrowband voice data and encoded voice data for narrowband region of input encoded wideband voice data, and detects a speaker in accordance with the decoded voice signals of the entire narrowband. When encoded voice data from a speaker is included in the narrowband, a signal in a region outside the narrowband of the expanded data is encoded. When the data is included in the wideband, encoded voice data of the region outside the narrowband is extracted for output. When the destination terminal is compatible with the encoded narrowband voice data, the narrowband voice signal mixed is encoded and output. When the destination terminal is compatible with wideband, the narrowband voice signal mixed is encoded for the narrowband region, and the voice data of the speaker is used as the encoded voice data for the region outside the narrowband.

Abstract: Voice communications apparatus is connected to a general subscriber telephone set or a broadband telephone set, to communicate over the public switched telephone network. The apparatus includes a filter converting signals such as to satisfy signal conditions prescribed for the telephone network. The apparatus also includes a terminal class determiner for determining the class of the telephone set connected, and a circuit for changing at least the sampling frequency at which the analog signals from the telephone set are sampled. This establishes high quality in broadband voice communications. The terminal class determiner may automatically determine the class of the telephone set at any timing of a call sequence by detecting a band component of the signals from the telephone set, a predetermined frequency of a signal intermittently transmitted from the telephone set or the characteristics of the telephone set.

Abstract: Speech telephones are incorporated in a LAN, and, for example, when a voice telephone 5 communicates with another voice telephone 8, call control is performed through exchange of information between a PC 1 and a PC 4, juxtaposed with the voice telephones, respectively, via a LAN hub 10, a LAN switching unit 50, and a LAN hub 20. Speech data of the respective telephones 5, and 8 are packetized in the data block of a MAC frame by concentrators 30, and 40 for voice telephones, respectively, and relayed to LAN-SW interfaces 30a, and 40a. Frames having a function of absorbing fluctuation are applied to the data block of the interfaces 30a and 40a. Respective voice data packetized according to the MAC address of the MAC frame are delivered to the voice telephones 5, and 8, respectively, via the LAN switching unit 50, and the concentrators 30, and 40 for voice telephones.

Abstract: The present invention concerns a voice response system in which a voice response unit sends a monaural voice signal corresponding to a request signal from a communication terminal to the communication terminal. The communication terminal comprises a monaural/stereo conversion portion for converting a monaural voice signal from the voice response unit to a stereo voice signal and a stereo voice outputting portion for outputting a converted stereo voice signal.

Abstract: A voice communication arrangement having a voice communication program installed in a computer loaded with a versatile OS thereon includes a voice data input/output function for passing through the OS voice data produced from voice sound captured by a microphone of the arrangement or received from another voice communication device, when connected to the arrangement, without being processed by the OS as audio data.

Abstract: A voice communication system transfers voice information stored in a protocol data unit under a predetermined communication protocol over a network, and includes end voice terminals for transferring and receiving the voice information, and a management server for managing said end voice terminal. The management server includes a manager for storing subscriber information indicative of whether or not the end voice terminals are provided with the capability of predetermined broadband voice communication. A user is informed, at the latest before the user initiates a call from the end voice terminal, of whether or not a telephone set of a called party is provided with the capability of the broadband voice communication through the end voice terminal based on the subscriber information. running in a network under a predetermined communication. This increases the satisfaction of the user and allows a business model justifying charging customers to be built.

Abstract: A voice mixing apparatus decodes input encoded narrowband voice data and encoded voice data for narrowband region of input encoded wideband voice data, and detects a speaker in accordance with the decoded voice signals of the entire narrowband. When encoded voice data from a speaker is included in the narrowband, a signal in a region outside the narrowband of the expanded data is encoded. When the data is included in the wideband, encoded voice data of the region outside the narrowband is extracted for output. When the destination terminal is compatible with the encoded narrowband voice data, the narrowband voice signal mixed is encoded and output. When the destination terminal is compatible with wideband, the narrowband voice signal mixed is encoded for the narrowband region, and the voice data of the speaker is used as the encoded voice data for the region outside the narrowband.

Abstract: Voice communications apparatus is connected to a general subscriber telephone set or a broadband telephone set, to communicate over the public switched telephone network. The apparatus includes a filter converting signals such as to satisfy signal conditions prescribed for the telephone network. The apparatus also includes a terminal class determiner for determining the class of the telephone set connected, and a circuit for changing at least the sampling frequency at which the analog signals from the telephone set are sampled. This establishes high quality in broadband voice communications. The terminal class determiner may automatically determine the class of the telephone set at any timing of a call sequence by detecting a band component of the signals from the telephone set, a predetermined frequency of a signal intermittently transmitted from the telephone set or the characteristics of the telephone set.

Abstract: Voice communications apparatus is connected to a general subscriber telephone set or a broadband telephone set, to communicate over the public switched telephone network. The apparatus includes a filter converting signals such as to satisfy signal conditions prescribed for the telephone network. The apparatus also includes a terminal class determiner for determining the class of the telephone set connected, and a circuit for changing at least the sampling frequency at which the analog signals from the telephone set are sampled. This establishes high quality in broadband voice communications. The terminal class determiner may automatically determine the class of the telephone set at any timing of a call sequence by detecting a band component of the signals from the telephone set, a predetermined frequency of a signal intermittently transmitted from the telephone set or the characteristics of the telephone set.

Abstract: A telephone switching system includes line packages and a switch unit allowing a path to be established between the line packages. At least one line package is interconnected to, e.g. a broadband-adaptive unit, and is adaptive to both of a first coding rule expected of the switch unit and a second coding rule higher in compression ratio than the first coding rule. The broadband-adaptive line package, monitoring the coding rule of another line package connected thereto, adapts itself to the first coding rule if the other line package is adaptive only to the first coding rule, or adapts itself to the second coding rule if the other line package is adaptive to both of the first and second coding rules.

Abstract: A VoIP system has plural CPE units connected to an IP network as parties to be connected with each other. Each CPE unit is composed of a VoIP-IF circuit and an audio codec circuit. The VoIP-IF circuit executes at least one of the coding process to the first audio frequency signal and the restoration process to the media information to transmit the media information generated by the process to the party connected and restore the first audio frequency signal from the media information supplied from the party connected to output it. The audio codec circuit performs at least one of the sending of the received music data to the party and the conversion of the music data supplied from the party into the second audio frequency signal.

Abstract: Speech telephones are incorporated in a LAN, and, for example, when a voice telephone 5 communicates with another voice telephone 8, call control is performed through exchange of information between a PC 1 and a PC 4, juxtaposed with the voice telephones, respectively, via a LAN hub 10, a LAN switching unit 50, and a LAN hub 20. Speech data of the respective telephones 5, and 8 are packetized in the data block of a MAC frame by concentrators 30, and 40 for voice telephones, respectively, and relayed to LAN-SW interfaces 30a, and 40a. Frames having a function of absorbing fluctuation are applied to the data block of the interfaces 30a and 40a. Respective voice data packetized according to the MAC address of the MAC frame are delivered to the voice telephones 5, and 8, respectively, via the LAN switching unit 50, and the concentrators 30, and 40 for voice telephones.

Abstract: A telephone communications apparatus for mixing a music signal with a speech signal, in which the gamut of music that may be enjoyed, such as a calling/called party, or the sorts of telephone terminal, can be expanded as compared to a conventional system. The apparatus includes a reproducing circuit for reading out a music signal recorded in a recording medium loaded thereon from the recording medium to reproduce the read out music signal, a setting circuit responsive to an operation by an operator for setting whether or not the music signal is to be mixed with a transmitting signal and/or a received signal, a mixer for mixing the music signal reproduced with the transmitting signal and/or the received signal, and a control circuit operative in response to the setting circuit for determining whether or not to enable the mixer to mix.

Abstract: Speech telephones are incorporated in a LAN, and, for example, when a voice telephone 5 communicates with another voice telephone 8, call control is performed through exchange of information between a PC 1 and a PC 4, juxtaposed with the voice telephones, respectively, via a LAN hub 10, a LAN switching unit 50, and a LAN hub 20. Speech data of the respective telephones 5, and 8 are packetized in the data block of a MAC frame by concentrators 30, and 40 for voice telephones, respectively, and relayed to LAN-SW interfaces 30a, and 40a. Frames having a function of absorbing fluctuation are applied to the data block of the interfaces 30a and 40a. Respective voice data packetized according to the MAC address of the MAC frame are delivered to the voice telephones 5, and 8, respectively, via the LAN switching unit 50, and the concentrators 30, and 40 for voice telephones.

Abstract: There are provided with an IP telephone system having both convenience of the softphone and durability of the hardphone, and a speech processing peripheral device (10) for the IP telephone system. The speech processing peripheral device as a peripheral device of an information processing device (20) is connected to the information processing device (20) for executing a call control program and functioning as a call controller. This speech processing peripheral device (10) includes a host connection means for sending/receiving an audio signal to/from the information processing device (20) and a speech processing means for executing speech processing in IP telephone communication. The IP telephone system according to the present invention includes the information processing device (20) for executing the call control program and functioning as the call controller and the speech processing peripheral device (10) according to the present invention, connected to the information processing device (20).

Abstract: A telephone communication system in which a call-originating side and a call-incoming side communicate by telephone via a communication network capable of wideband communication and narrowband communication. When an apparatus exclusively used for narrowband communication, an apparatus capable of wideband communication and narrowband communication, and an apparatus exclusively used for wideband communication exist mixedly in between one end and another end, at least one of phone-call path parts between adjacent ones of the apparatuses is set to be different in transmission capability on a per phone-call path part basis, by sending and receiving information about transmission capabilities of the apparatuses in between the one end and the another end.

Abstract: Speech telephones are incorporated in a LAN, and, for example, when a voice telephone 5 communicates with another voice telephone 8, call control is performed through exchange of information between a PC 1 and a. PC 4, juxtaposed with the voice telephones, respectively, via a LAN hub 10, a LAN switching unit 50, and a LAN hub 20. Speech data of the respective telephones 5, and 8 are packetized in the data block of a MAC frame by concentrators 30, and 40 for voice telephones, respectively, and relayed to LAN-SW interfaces 30a, and 40a. Frames having a function of absorbing fluctuation are applied to the data block of the interfaces 30a and 40a. Respective voice data packetized according to the MAC address of the MAC frame are delivered to the voice telephones 5, and 8, respectively, via the LAN switching unit 50, and the concentrators 30, and 40 for voice telephones.