IDM MAP4E1-30 Hardware Voice Conference Equipment

The IDM MAP4E1-30 hardware voice conference equipment is an integrated platform combining 64K narrowband service access, PCM multiplexing, 64K timeslot cross-connection, and a 30-party voice conference system.

      The IDM MAP4E1-30 hardware voice conference equipment is primarily used for service access and cross-connection at the access layer in data communication networks. It features full synchronous timing functionality, providing non-blocking 64K timeslot cross-connection across 4 E1 channels, and supports up to 30 64K service paths, including two-wire/four-wire audio, telephone, magneto telephone, common battery telephone, and V.24 data services.

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IDM MAP4E1-30 Hardware Voice Conference Equipment

    The local end can support up to 30 audio terminals, including the following types:

(1) FXS interface for connecting telephones;

(2) FXO interface for connecting to switches;

(3) EMVOICE24 interface for connecting radios, trunked two-way radios, access control intercoms, conference microphones, headset earphones, and similar devices;

(4) MicPhone interface for connecting microphones and speakers, enabling monitoring and dedicated line intercom functionality;

(5) NKT five-party intercom interface for transmitting Otis brand elevator five-party intercom systems over IP networks;

(6) VoiceSL five-party intercom module for connecting Mitsubishi/Deling brand elevator five-party intercom systems to IP-based IoT networks for transmission and dispatching;

(7) The conference room supports optional voice-triggered communication activation, enabling integration of half-duplex communication devices (such as trunked radios, wireless radios, network intercoms) into full-duplex telephone networks.

The conference room can not only access local terminals but also receive audio signals via E1 interfaces from remote locations. Multiple devices can be cascaded directly to form a larger conference room.

The device features gateway configuration capabilities, allowing any audio interface to be added to or removed from the conference room as needed.

By default, all participants can speak, but individual mute modes can be configured for specific users.

Hardware-based voice conference terminals offer zero-latency performance, clear and stable audio quality, and convenient control. Compared to server-based IP voice conference systems, they provide lower cost and superior performance. PCM-based voice conference terminals support a wide variety of terminal types and enable large-scale conference coverage.