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Networked SIP Intercom Module 2101T Audio Module

SIP2101T module supports a variety of network protocols and audio codec protocols, and can be used in applications such as VoIP and IP paging and high-quality music streaming media playback. At the same time, SIP2101T also provides a serial port, allowing users to control through commands.
  • SIP2101T

  • SINREY

Availability:

SIP2101T module supports a variety of network protocols and audio codec protocols, and can be used in applications such as VoIP and IP paging and high-quality music streaming media playback. At the same time, SIP2101T also provides a serial port, allowing users to control through commands.


Features


  • Standard RJ45 network interface, providing firmware online remote upgrade;

  • Based on ARM + DSP architecture, high-speed industrial chip is adopted, with startup time ≤ 1s;

  • Provide multiple IO interfaces and enrich module functions;

  • With MIC input interface, realize two-way voice intercom;

  • It has the function of controlling serial port and can control the work of the module;

  • It has the general serial port function and can realize the transparent data transmission function;

  • It has an audio line output port, which can be externally connected to an active speaker for sound amplification;

  • With audio line input port, external active microphone can be connected for intercom.

  • 2 * 15W audio power amplifier output, can be connected to 8 ohm constant resistance horn


Parameters


  • power input: Voltage DC 4.7~16V Maximum working current, 200mA

  • temperature: Operating temperature range Industrial grade: -40~85℃,storage temperature range -40~85℃

  • Network : 10/100M Base adaptive Ethernet interface

  • Mic input: Typical amplitude 50mVrms, signal-to-noise ratio 95dB

  • LineIn input: Typical amplitude 1000mVrms, SNR 95dB

  • Line Out output:Load 10KΩ, typical 1000Vrms, signal-to-noise ratio 95dB (in playback mode)

  • Decoding : Provide stereo playback, up to 48kHz, 320kbps audio stream, support MP3.WAV (PCM +IMA ADPCM), G.711 a/u, G.722 and other formats..The minimum delay is50ms

  • Encoding : Support G.711a/u, G.722 and RTP mode.The minimum delay is 30ms

  • Bidirectional : Two-way intercom, with high-performance echo suppression algorithm, support G.711 a/u, G.722 encoding

  • The minimum delay is 80ms

  • Intercom control port:There are three control interfaces of Call1, Call2 and Call3 in total, and

  • this interface has two working modes of [Pulse Mode] and [Hold Mode].

  • control serial interface: Baud rate 115.2kbps

  • module size: Length x width x height: 5 x 5 x 1.7cm


Application


  • Various commercial network audio streaming applications

  • Network alarm, network player

  • Universal two-way full-duplex VoIP intercom for paging and intercom applications

  • Source Encoder for IP Audio Distribution Applications

  • Background Music and Music On Hold Encoder for VoIP Applications

  • Compatible with various low-latency IP audio encoders

  • Unmanned parking lot, self-service charging station intercom terminal

  • Expressw tunnel intercom, toll station intercom terminal


Downloads


SIP2101T Network Audio Module Specification.pdf

Manager(V1.3.7).zip

Manager Manual.pdf

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