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What is a VoIP Gateway?
A VoIP gateway allows you to convert between a traditional telephony connection and a modern VoIP connection using SIP.
VoIP Analog gateways convert voice media between FXO and FXS connections and VoIP connections.
Conversions can go from SIP to traditional, traditional to SIP or even SIP to SIP for the purposes of failover or transcoding. In a traditional network to SIP deployment, a gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP to traditional network deployments use a gateway to connect a modern SIP communications system with digital or analog service from legacy carriers.
Niceuc Analog Gateway / IP PBX:
It can not only work as analog gateway that makes connection between FXO/FXS and VoIP SIP, but also can work as IP PBX that allows other SIP terminals to register into themselves.
There is no changes about these two work modes, only different firmware. As established in 2000, all hardware and software of this product are developed, manufactured, and produced by NICEUC ourselves.
By using carrier-class technology and mature DSP, all NICEUC Analog Gateway/ IP PBX serial product have excellent properties and stable performance. With scalable modular design, it is easy to expand according to customers' requirements.
·VoIP protocol support: -SIP
RFC3326 (Reason header in SIP messages), RFC3372 (SIGTRAN and SIP-T), RFC2327 (sdp), RFC3398 (ISUP-SIP Mapping), RFC3261 (sip), RFC5806 (Diversion Indication in SIP), RFC2833 (DTMF), RFC3362 (t.38), RFC 3261(SIP 2.0), RFC3204 (MIME media types for ISUP and QSIG Objects), RFC3578 (Mapping of ISUP overlap to SIP).
·Codec:G.711 U-Law and A-Law, G.711 Appendix 1, G.723.1 and G.723.1 Annex A, G.729 Annex A and Annex B, G.726,GSM,ARM, ILBC.
·Echo Cancellation: G.168 128 ms Network Echo Canceller.
·FAX: T30, T38 and pass through.
·Advanced Voice Processing Technology: Voice Activity Detection (VAD), Generic Comfort Noise Generation (CNG), Calling control: Called/Calling party number translation; second stage dialing, Auto dialing with DTMF, Ring Back tone, DTMF, Caller ID, and Flexible input/output Gain Control.
·Network Protocol: IP, NAT, ICMP, ARP, HTTP, BOOTP, FTP, TFTP, DHCP, PPPOE, SNMP, Diff-Serv
Carrier Grade Reliability
·With a new generation of processor, a separate high-end DSP processing chip and powerful hardware processing, ensure that the equipment at full load conditions remained stable and reliable;
·99.99% availability, support continuous long talk at least 48 hours;
·Support over-current protection function on POTS interfaces ;
·Support the protection of lightning, surges, power lines and other protective lap;
·Fully compatible;
Excellent voice communication quality
·Voice Activity Detection (VAD), to effectively save bandwidth resources;
·Comfort Noise Generation (CNG), to provide the same experience as PSTN in calls;
·Adaptive dynamic buffer technology, to meet the application under bad network;
·G.168 128ms echo cancellation technology. This indicator is far higher than the same kind of PBX, eliminating the echo interference caused by long distance call;
·DTMF detection / generation technology, to effectively support the business of the fax, callback, second dial.
Plentiful business functions
·Modification between calling number and called number;
·As a terminal, support the register trunk function to other GK(Gatekeeper) or SIP Server.
·As a gateway, support the control from superior GK or SIP Server to complete the trunking calls;
·Support any switch among SIP, FXO, FXS;
·Support automatic routing function;
·Support high capacity of black and white lists;
·Simultaneously support the register of 4 SIP Server;
·Support output of CDR;
·Fax support T.30, T38 and pass through;
·Support reelection function;
·Support detection of calling number and display of call ID;
·Support generation of dial tone, ring back tone, etc.;
·Port supports gain and comfort adjustment of noise;
·Multi-router and multi-bureau management, group, cluster and exchange management, intelligent routing distribution and equal traffic sharing;
·Support function of dividing groups, for the incoming calls, the subscriber line can be set to line up or take turns to choose;
·Flexibly control the incoming and outgoing of lines, the lock of long-distance calls can be controlled by the phone;
·Complete call functions,it can provide several kinds of call functions and special call way by customer's request;
·The voice processing capacity support voice play with built-in computer operator;
·High processing capacity in conference;
·Monitor and handle real-time telephone traffic;
Simple and convenient management/maintenance
·The way of management on software:
Local and remote WEB; Local and remote Telnet; Console port-Super terminal;
·On hardware, with rack design, they can be installed in the standard 19 inches rack, also can be separately installed in a small machinery room. Here this model NC-MG232 is rack-mountable 1U height.
Specification of VoIP Gateway NC-MG232's cards:
Card Name | Specification | Description |
Power supply Unit | AC 100-240V or DC -48V | /// |
MCU DSP 30 card | SIP and 200 SIP Subscribers, 32 G.711channels, 16 G.729,G.723,G.726 channels | Management card |
MCU DSP 16 card | SIP and 200 SIP Subscribers, 16 G.711channels, 8 G.729,G.723,G.726 channels | Management card |
Analog user card | 16 ports FXO/FXS | Modular design, to connect PSTN CO line and analog phone/extension |
Specification of VoIP Gateway NC-MG232 itself:
Item | NC-MG232 |
Slot for PSU | No (Built-in) |
Slot of MCU Card | No (Built-in) |
Slot for Analog User Card | 2 |
VoIP Channels / SIP Trunks | 16 or 32 |
SIP Subscribers | 30 to 200 |
Analog Port (CO line/Extension) | 0 to 32 |
Telephone Connector | RJ45 (One RJ45 to Four RJ11) |
Ethernet/LAN Port | 1 (10/100/1000M Base-T) |
Console port | 1 (RS232) |
SIP Compatibility | CISCO, Siemens, AVAYA, Huawei, ZTE, etc. |
Power Supply Unit | DC -48V or AC 110-240V |
Power | 40W |
Telephone Cable Transmission Distance | 5 KM |
Dimension | 480mm*325mm*44mm (1U) |
Weight | 4.5 KGS |
Working Environment | 0℃~50℃, Less 80% |
About Us
Be a Manufacturer for 18 Years-One of the first movers in VoIP products in China
We are a professional Manufacturer and national high-tech enterprise in VoIP products from 2000 year.
We can open our registeration number to every customer. Register No.:91440300723031804W, if you like, you can use this number to trace and check our company and history in Chinese goverment official system.
Location | Size | |
Office | Nanshan, Shenzhen, China | 456.02 Square Meters |
Factory | Baoan, Shenzhen, China | 2520.00 Square Meters |
Strong Independent R&D Capabilities
With strong independent R&D capabilities, we can make different kinds of development works according to customer's special requirements. Both OEM and ODM services are available.
Current Situation of R&D Department:
R&D Engineers | Experience | Average Time For New Products Launched | R&D Devices | ||||||
Over 30 Years | 21-30 Years | 11-20 Years | 6-10 Years | 2-5 Years | Less Than 2 Years | ||||
Office | 9 | 0 | 5 | 4 | 0 | 0 | 0 | 3 months | Oscilloscope, Digital Transmission Analyzer, Vibration Tester, Eletric Parameter Test, etc. |
Factory | 10 | 1 | 1 | 1 | 6 | 1 | 0 | 3 months |
So Many Big Customers
So many big customers from all over the world have been buying our products because of stable and reliable quality.
Registration Date | 2018/07/30(Year/Month/Date) |
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Buyer / Seller in EC21 | Seller |
Business Type | Manufacturer |
Year established | 2000 |
Employees total | 11 - 50 |
Annual revenue | USD 2,000,001 - 5,000,000 |
Company | Niceuc Communication Co., Ltd |
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Address | Unit 1101-1102, Yizhe Building, Yuquan Road, Nanshan DistrictShenzhenGuangdong518051China |
Phone | 86 - 755 - 26722761 |
Fax | 86 - 755 - 26526970 |
Homepage | www.niceuc.com |
Contact | Timmy Huang |