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SIP2701T
SINREY
Description
SINREY SIP2701T network audio module is a general-purpose independent SIP audio function module, which can be easily embedded into OEM products. This module encodes and decodes SIP protocol and rtp audio stream from the network.
SIP2701T audio 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, SIP2701T also provides a serial port, allowing users to control through serial port commands.
Encode: Support G.711a/u, G.722 and RTP multicast mode.
Decode: That is audio playback. Receive the audio stream from the network, and output high-quality audio signal through the line after decoding by the module. Currently supports the following audio formats:MP3.WAV (PCM +IMA ADPCM), G.711, G.722, etc. Can play audio streams up to 48khz sampling rate 320kps bit rate. Receive the audio stream sent by TCP, UDP, RTP, multicast, broadcast, etc. in the network, and play it after local decoding.
Button control: Directly through the IO port on the module, control the module SIP intercom or initiate RTP multicast Serial port control: Through the control serial port on the module, control the module SIP intercom or initiate RTP multicast Network Control: Through network commands on other devices in the network, control the control module SIP intercom or initiate RTP multicast
Audio input (M-ST), output status (P-ST): You can connect indicator lights or control relays. server connection indication
Real Time Clock: The module is powered off, or the time is not adjusted by the server to provide accurate time, which can be obtained through the serial port or network, or the audio file can be played regularly
Parameters
power input: Voltage DC 4.7～5.5V
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: 52 x 4 x 15mm
Description
SINREY SIP2701T network audio module is a general-purpose independent SIP audio function module, which can be easily embedded into OEM products. This module encodes and decodes SIP protocol and rtp audio stream from the network.
SIP2701T audio 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, SIP2701T also provides a serial port, allowing users to control through serial port commands.
Encode: Support G.711a/u, G.722 and RTP multicast mode.
Decode: That is audio playback. Receive the audio stream from the network, and output high-quality audio signal through the line after decoding by the module. Currently supports the following audio formats:MP3.WAV (PCM +IMA ADPCM), G.711, G.722, etc. Can play audio streams up to 48khz sampling rate 320kps bit rate. Receive the audio stream sent by TCP, UDP, RTP, multicast, broadcast, etc. in the network, and play it after local decoding.
Button control: Directly through the IO port on the module, control the module SIP intercom or initiate RTP multicast Serial port control: Through the control serial port on the module, control the module SIP intercom or initiate RTP multicast Network Control: Through network commands on other devices in the network, control the control module SIP intercom or initiate RTP multicast
Audio input (M-ST), output status (P-ST): You can connect indicator lights or control relays. server connection indication
Real Time Clock: The module is powered off, or the time is not adjusted by the server to provide accurate time, which can be obtained through the serial port or network, or the audio file can be played regularly
Parameters
power input: Voltage DC 4.7～5.5V
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: 52 x 4 x 15mm
