Contact
IDM Voice365-12 Audio IP Transmission Device

IDM Voice365-12 Audio IP Transmission Device

📅May 12, 2026
Model:IDM Voice365|Category:IP Access Gateway

The IDM Voice365-12 is an IP audio transmission device capable of transmitting high-fidelity audio and control information over IP networks. It can be registered to an IoT server or used in a local area network (LAN) for networking.

IP audio transmission equipmentIP audio access deviceaudio transmission equipment

Gallery

1 Overview

The IDM Voice365-12 is an IP audio transmission device capable of transmitting high-fidelity audio and control information over IP networks. It can be registered to an IoT server for use, or deployed in a local area network (LAN) for networking.

The IDM Voice365-12 audio IP transmission device can be used in point-to-point mode, extending high-fidelity audio terminal signals over IP networks. It can also be used in networked mode to achieve point-to-multipoint transmission and control.

The IDM Voice365-12 audio IP transmission device features 12 bidirectional audio signal interfaces, supporting connection to 12 E1 timeslot circuits (64K) or high-fidelity audio sources, providing 12-channel audio input and output.

@@IMG0@@

@@IMG1@@

Front and rear panel of the chassis

@@IMG2@@

IDM Voice365-12 Audio IP Transmission Device Application Diagram

2 Product Features

(1) Supports G.711A, G.729A, and G.722 audio encoding/decoding, or 16-bit high-fidelity lossless transmission;

(2) Each channel supports in-band signaling transmission;

(3) Supports SNMP network management protocol;

(4) The audio compression encoding format and transmission direction of each channel can be flexibly controlled by the gateway;

(5) Modular design with redundant AC/DC power supply; dual power supplies can operate simultaneously or one can be selected;

(6) Real-time fault detection with LED indicators displaying fault status.

3 Technical Specifications

  1. Transmission protocol: TCP/IP suite;

High-bandwidth audio encoding/decoding: G.711A and high-fidelity (suitable for high-speed links such as fiber optics and wireless bridges). Lossless PCM encoding/decoding: sampling frequency range 0kHz–8kHz, sampling rate ≥16kHz, 16-bit quantization; when the number of audio channels is less than 10, 24-bit quantization is supported.

  1. Low-bandwidth encoding/decoding methods:

Ø Low-speed line encoding: G.722 and G.729A encoding/decoding;

Ø G.729A and G.722 encoding (suitable for low-rate links such as satellite and 4G networks).

  1. The audio encoding/decoding method for each channel can be set via network management or automatically adjusted based on available bandwidth.

  2. System delay: Network encoding/decoding delay is adjustable. Excluding network latency, the system encoding/decoding delay ≤100ms.

  3. Network management agent protocol: SNMP V3.

  4. Bandwidth:

Ø ≤300 kbit/s under lossless PCM voice encoding/decoding;

Ø ≤100 kbit/s under G.722 voice encoding/decoding;

Ø ≤90 kbit/s under G.711A voice encoding/decoding;

Ø ≤30 kbit/s under G.729A voice encoding/decoding.

  1. Interface: Four-wire analog audio interface.