Product Description
NFN-IP-SIPGW2 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a total solution for networks providing VoIP services. The IP-SIPGW2 VoIP functions are available via a regular analog telephone.

Product Feature
(1) Bridge/Router Protocols

• IEEE 802.1D (self learning transparent bridge)
• 128 MAC Addresses support
• Static IP routing (configurable route table)
• RIPv2 (backward compatible with RIPv1)
• DHCP server (configurable and supports up to 253 addresses)
• DHCP client (WAN)
• DHCP relay agent
• PPP auto reconnect and configurable timeouts
• PPP auto reconnect on WAN access
• PPP Auto, PAP, and CHAP
• PPTP
• 128 character support for PPPx username/passwords
• DNS proxy
• Dynamic DNS
• Port forwarding
• NAT
• NAPT
• Dynamic NAPT
• ALG support
• Wild Card DMZ
• Virtual server
• VPN pass through (IPSec - ESP Tunnel mode, L2TP, PPTP)
• Bridge filtering
• ICMP
• IGMP
• MAC Address Spoofing
• PPP Half Bridge

(2) VoIP Firmware Features

• Single voice port that supports legacy (analog) touch-tone telephones
• Connects legacy telephones to IP-based networks
• Advanced pre-processing to optimize full-duplex voice compression
• High performance line-echo cancellation eliminates noise and echo
• Dynamic network monitoring to reduce jitter artifacts such as packet loss
• Clear, natural-sounding voice quality

VoIP Specifications

• SIP 2.0 Protocol
• Digest Authentication MD5
• Record-Route Headers

Voice codecs

• G.729A
• G.711 A-Law
• G.711 u-Law

Dual-tone multi-frequency (DTMF)

• DTMF tone detection and generation
• Mid-Call DTMF for IVR support
• RFC 2833 AVT tones
• Configurable for two sets of frequencies and single set of on/off cadence (US, UK)

Line-echo cancellation

• G.165, G.168 compliant
• Echo canceller for each port
• 8 to128 msec configurable echo length
• Nonlinear echo suppression
• Convergence time = 250 ms
• ERLE = 10 to 20 dB
• Double-talk detection

Voice features

• Voice activity detection (VAD)
• Comfort noise generation (CNG)
• Dynamic jitter buffer (adaptive)
• Packet Loss Concealment
• Bell 202 and V.23 Caller ID

Modem & Fax modes

• G.711 fax pass-through
• Modem passthrough
• Modem detection

Web Interface

• SIP Enable
• SIP registrar address (accepts URL or IPv4 Address)
• Local SIP port (up to five digit port number)
• Registration port (up to five digit port number)
• SIP proxy address (accepts URL or IPv4 Address)
• SIP proxy port (up to five digit port number)
• Outbound proxy address (accepts URL or IPv4 Address)
• Outbound proxy port (up to five digit port number)
• Media base RTP port (up to five digit port number)
• SIP registration interval (time in seconds)
• Authentication method (selection of MD5 or TLS. MD5 Only)
• User name/number (alphanumeric sequence)
• Password (alphanumeric sequence)
• Authorization user ID (alphanumeric sequence)
• Display name (alphanumeric sequence)
• Codec preference (711U/711A/729A)
• Number plan ‘§C support for Pulver 5 and 6 digit dialing
• Number plan modification via config.reg
• Number plan modification via web GUI
• Regional call progress configuration (US, UK)

Control of Special Features

• Set Call Forward Number (alphanumeric sequence)
• Enable for all calls
• Enable for busy
• Enable for no answer
• Disable call forward
• Call waiting enable/disable
• Call return enable/disable
• 3-way conference call enable/disable£‘§Next Version Support£©
• Caller I.D. standard (v.23 / Bell 202)
• PSTN backup
• SIP registration LED
• STUN
• 1FXS support

Parameters to set up regional characteristics

• VoIP dial tone
• Peer2Peer dial tone
• VMI dial tone
• Ringback tone
• Busy tone
• Fast busy tone
• Alerting tone
• Call waiting tone
• Congestion tone
• Stutter dial tone
• Recall dial tone
• Warble tone
• Unobtainable tone
• Stutter dial tone

Call Protocol Mediator (CPM)

• Basic Call Establishment and Termination
• Call on Hold (using SDP, RFC3264)
• Consultation Hold
• Caller ID (type I, type II) support
• Call Forwarding (ATAA decides whether to forward: Unconditional, On Busy, or No Answer)
• Call Forwarding (Dial pad support e.g. *72)
• Blind Call Transfer (#90)
• Attended Call Transfer (a combination of Consultation Hold and a Transfer)
• Call Waiting (CPM notifies app on new call)
• SDP: Add usage of rtpmap, fmtp, a= for RFC2833 and Call on Hold

(3) Ethernet Interface

• IEEE 802.3
• 10/100 Mbps

Product Specification/Models
Model NFN-IP-SIPGW2
LAN interface 1XRJ45 10/100M Base-T
WAN interface 1XRJ45 10/100M Base-T
FXS telephone port 1 FXS
FXO PSTN port 1 FXO
LED Status, PSTN, VoIP
Universal Switching
Power Adaptor Input: 100-240VAC 50-60 Hz
Output: +5VDC, 1200mA
Dimension 130mm (W)
88mm (D)
30mm (H)
Weight 0.6lbs (0.3kg)
Temperature 32 - 130ԤH
0 ‘§C 45‘ζ
Humidity 10% - 95%

Payment
T/T

Delivery
FOB Shanghai

Origin
China

Minimum Order
100pcs