SIP or IP Network Paging Microphone with Built-in Speaker Call Station for Monitoring and Intercom

• Type: Multipoint Control Unit (MCU)
• Protocol: TCP/IP
• Wire or Not: Wire
• Color: Black
• Communication Protocol: SIP
• Network Interface: 10/100base-Tx, RJ45
• Model NO.: SIPM-10R
• Speaker Spl (Max): 96dB
• Decoding: MP3/WMA/Wav
• Microphone Frequency Range: 70Hz-12.5kHz
• Installation: Desktop
• Power Supply: DC 12V-24V or Poe
• Transport Package: Gift Box/Export Carton/Pallet
• Specification: 150 x 140 x 55 mm
• Trademark: Ben & Fellows
• Origin: China
• HS Code: 85182900

Product Description

SIP PA system Paging Microphone with 2 Buttons for Intercom


 

Professional SIP Call Station to control and make announcement to 
all devices in the system.
Adopt high-speed industrial grade chip, start-up in less than 1s
Support zone paging, and the red indicator will light up when talking
Built-in speaker for monitoring and intercom
With audio output for external active speaker
Support offline talkback (without server)

Specifications
 

Communication Protocol	SIP
Network Interface	10/100Base-TX,RJ45
Microphone Frequency Range	70Hz~12.5KHz
Decoding	MPEG,ADPC
Frequency and Sampling Rate	8kHz-32 kHz, 16 bit
Installation	Desktop
Enclosure material	Metal
Power Supply	DC 12V-24V or POE
Operating Temp.	0~65°C
Net Weight	0.6Kg
Dimension	150 x 140 x 55 mm

As a host of the IP broadcast intercom system, SIPM-10 can be 
used in places where intercom, monitor and emergency alarms are 
needed, such as control centers of self-service banks, interrogation 
rooms, classrooms, hospitals and parking lots.

As an intercom terminal, SIPM-10 supports full-duplex two-way 
intercom and has good echo suppression. We offer standard sip 
mode, which can be used in other standard SIP systems and 
compatible with standard SIP terminal devices

 

With a 10/100M Ethernet interface, line input and output, SIPM-10 
is an SIP desktop intercom host that is used with SIP server. 
SIPM-10 network paging microphone can perform point-to-point IP 
intercom and multicast to the IP terminal through a microphone or 
local line input.

Q1. Does the SIP public address system provide playback software or server software?
A1: Neither, we only provide you with a configuration software called SIPManager to setup all the parameters. However, the third party softwares can be provided for use.
But our SIP system also supports our own IP protocol, you can use our Eco IP system manager software to route audio / scheduled message and control of the entire system.

Q2. Which SIP phone manufacturers are supported?
A2: All of our SIP products (speakers, terminals, microphones, etc.) support the standard SIP protocol, so technically all the SIP phones on the market are supported.

Q3. Does the SIP speaker support multicast function?
A3: Yes, the multicast address and port number can be set in the SIPManager. After setting, you can directly call the multicast address on the SIP phone.

Q4. Can the SIP phone call the SIP speaker directly by its IP address?
A4: Yes, the SIP phones can call the SIP speaker by its IP address as well as its account and multicast address.

Q5. Can the SIP speaker automatically obtain IP address?
A5: Yes, SIP speaker can automatically obtain IP address when the network supports DHCP and the SIP speaker is set to DHCP mode.

Q6. Does the SIP system support play music in the IP broadcast mode?
A6. Yes, SIP speakers need to be configured with relevant parameters in the IP software.

Q7. Does the SIP speaker support the prompt tone modification?
A7. No, it doesn't.

Q8. Can SIP intercom terminal be set to call multiple terminals?
A8. Yes, it can be set to call one of the terminals and can also call multiple terminals by calling the multicast monitor address.

Q9. Can a SIP account be numbers plus English letters?
A9. Yes.

Q10. Open the SIPManager software and many texts are displayed as "???"
A10: Change the language to English.