Eacome VoIP conference phone IP500 is equipped with the international advanced SonicClear IV HD-voice processing technology. IP500 is designed for SIP VoIP, USB VoIP and analog PBX environments, with unique features that take advantage of the benefits provided by the different types of telephony environments. You can combine and switch easily between among these connectivity to make a multi-party conference call with kinds of telephony platform. With highly stable and reliable acoustic performance, these products create a more realistic and joyful experience in conference phone calls for users.
With the VoIP conference phone dedicated so many connectivity, you can easily hold a 3-way SIP conference call in IP network, make a conference call between video conferencing system via USB connection with IP PBX system or audio teleconferencing system with PSTN network. It can replace multiple conference phones and simplify the conference room layout. It is undoubtedly a broadest telephony platform for remote collaboration.
IP500 is equipped with four internal uni-directional microphones and two external microphones, it’s resulting in microphones that can gather cover 5 meters with more high-definition voice quality. They offer an auxiliary audio port to connect to external speaker devices so that it will enlarge the speaker coverage range. Therefore, these products are suitable for different types of environment such as larger conference rooms, medium meeting rooms.
● Wide-band acoustic echo cancellation with HD-voice dynamic noise suppression, automatic gain control, voice band equalization and intelligent microphone mixing.
● Best for conference room within 20-60㎡ and supports audio teleconferencing within 15 attendees.
● 4 inner uni-directional microphones gather over 3m, 360 degree range.
● Support two extension microphones gather over 5 meters.
● Support VoIP communication network, SIP2.0 protocol.
● Support 3 independent SIP accounts.
● Support USB, designed for video conferencing system, multimedia communication system, UC unified communication platform, VPN platform etc.
● Support analog telephone, PABX and PSTN.
● Built-in 3-way VoIP multi-party conference call
● Collaboration: bridge SIP VoIP, USB VoIP and analog PBX at same time to make multi-party conference call.
● Support audio output to the amplifier and mixing consoles, expand the room-coverage options to meet the requirements of larger meeting rooms.
● Speaker volume up to 95db,16 levels adjustable.
● 128*64 full dot matrix graphic LCD display with ivory white backlight.
● Multiple languages display, support Chinese and English.
● Built-in Web GUI for configuration and provisioning.
● Practical and powerful phone function: call forwarding, call hold, call out, call pickup, call history, redial/rapid-dial, mute, hold, do not disturb function, flexible setting functions for many users.
● Powerful phonebook function: store 140 entries including name, number and any message.
● DSP based deep echo cancellation>-65dB,500ms.
● Bi-directional noise suppression>-18dB.
● LEC>-25dB,length 32ms.
● Microphone frequency response : 100Hz - 24 KHz.
● Speaker frequency response: 100Hz - 24 KHz.
● Microphone coverage: built-in 3 meters, extended 5 meters, 360 degree range.
● Speaker Volume: up to 90dB.
● Voice frequency response: 100Hz-24000Hz.
● Voice Codecs: G.711a, G.711u, G.729a, iLBC, G.726.
● Wideband voice Codec: G.722.
● Support DTMF pass through, RFC2833 and DTMF SIP info to transfer DTMF signal tone.
● Adaptive jitter buffer.
● Silence suppression and comfort noise generation.
● Packet loss concealment.
One integrated 10/100Mbps Ethernet port, support 802.1 p/q
USB B type
Two 3.5mm jack IN/OUT
Two RJ9 connector
SIP RFC3261 fully compliant
HTTP / FTP / TFTP for centralized automatic provisioning
RTP RFC3550/3551, RTCP RFC1889
SNTP for time synchronization
Built-in NAT traversal and DHCP server
Firewalls and NAT traversal with periodical refreshing SIP registration
128*64 full dot matrix graphic LCD display with ivory white backlight.
22 keypads: Menu, Left/Right moving, Selection, Hands-free, Redial, Mute, Transfer, Volume up/down, Alphanumerical 0–9
LCD menu feature: Multiple menu functions.
Built-in Web GUI for configuration and provisioning.
Time/Date, Call Status, Call history.
call forwarding, call hold, call out, call pickup, call transfer.
Built-in 3-way calling.
Support calls directly from the phonebook.
Do not disturb function and private key.
Define a variety of cases of call forward, busy forward, no answer forward, forced forward.
Environment and specification
Reverberation Time: < 0.5s.
Noise Degree: < - 48 dBA.
Power Requirement: AC input: 100/240V，50/60Hz，12W.
Working Temperature: 0°C-50°C (running).
Safety/certification specifications: 3C, FCC, CE, RoHS.