VoIP Phone Based on SIP/IAX2: To meet the demands of next generation internet telephony communications, Uni-Ta introduces the award-winning VoIP Phone model UTP1600, based on industry open standards and broadly interoperable with SIP/IAX2 platforms and VoIP hardware from major third party vendors. The VoIP Phone UTP1600 is a feature-rich, toll-quality voice over internet protocol solution at affordable cost.

The VoIP Phone UTP1600 features a full-duplex speakerphone with advanced acoustic echo cancellation, additional features including 3-way conferencing, call transfer (blind/attended), call forward, call waiting, caller ID display or block, DND, customized dial plan, 9 programmable speed-dial keys, as well as DHCP (client/server), NAT traversal (STUN), VLAN (voice VLAN/data VLAN), QoS with diffserv, VPN (L2TP/UDP Tunnel).

By utilizing the cutting-edge quality of service, echo cancellation, comfort noisy generation and voice compensation technology, the VoIP Phone UTP1600 can effortlessly provides the excellent voice quality. Meanwhile, the dual 10M/100Mbps auto-sensing Ethernet ports on the VoIP Phone allow users to install in an existing network location without interfering with desktop PC network connections. The VoIP Phone UTP1600 also provides easy configuration thru manual operation (phone keypad and web interfaces) or personalized automated provisioning via central configuration file for mass deployment.

Note: Model UTP1606 has same features as UTP1600, but with additional PoE function.

Key Features of the VoIP Phone UTP1600
- Supports 2 SIP lines registering simultaneously; Compatible with IAX2 protocol
- SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
- Dual 10/100Mbps Ethernet ports (switched/routed)
- Support DHCP (client/server), Static IP, PPPoE for xDSL
- Full-duplex speakerphone with advanced acoustic echo cancellation
- Support RJ9 headset connector
- Support codec: G.711 (A-law/u-law), G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.722
- Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
- DTMF relay: RFC2833, SIP info
- Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
- Call features: caller ID display or block, conference call, call transfer (blind or attended), Call hold, Call waiting, DND, Black List, Limited List, Call history, Voicemail
- Support comprehensive customized dial peer
- Support NAT Traversal (STUN); VLAN (voice VLAN/ data VLAN); QoS with diffserv; VPN (L2TP); DMZ; Firewall; DNS relay
- Support automated provisioning through TFTP/TFP/HTTP for mass deployment
- Support management via web interfaces, keypad and telnet
- 2 X 16 characters dot-matrix LCD display


Specifications of the VoIP Phone UTP1600