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This gateway is an Internet based one port voice gateway. It adapts multi voice control protocols and voice compression codec to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.

It supports SIP protocols and also has Bridge and Router model. Therefore, the original analog telephone can dial both IP phone and PSTN, effectively using the existing broadband resources. It is compatible with various IPPBX and VoIP voice gateway to provide broadband IP voice service.

Key Features:

• Support two models: Bridge and Router(NAT&NAPT).
• Network Protocols: TCP/UDP/IP, ICMP, HTTP, DHCP Client (WAN Interface), DHCP Server(LAN Interface), DNS Client, DNS Relay, SNTP, PPPoE, FTP, TFTP.
• VoIP Protocols: SIP (RFC3261, RFC3262, RFC3264, RFC3265)
• Voice Codecs: G.711(A-law/U-law), G.723.1, G.729A/B,
• NAT penetration: Support STUN client,
• Support modify SIP register port, HTTP server port, Telnet server port and RTP port.
• Support two SIP server synchronously: Can register two different SIP server, and can make a call by either proxy.
• Support standard voice features such as numeric Caller ID Display, CDo-Not-disturb, Forward, ban outgoing.
• Support standard encryption and authentication
• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control).
• Provide easy configuration thru manual operation (Web interface and Telnet) or automated centralized configuration file via TFTP or HTTP.
• Support firmware upgrade via TFTP/FTP and HTTP.
• Support syslog, can send event of gateway to syslog server.
• Support reverse polarity