Professional VoIP Phone Based on SIP/IAX2

Selling Points:

1. Stylish and functional in design; 2. Support 2 SIP lines

Item No.:SM-SKP0016
Order QTY(PCS) Unit Price(USD)
1 - 5 US$69
6 - 10 US$65
11 - 20 US$61
Quantity:

Product Description

Professional VoIP Phone Based on SIP/IAX2

 

Instruction:

 

Stylish and functional in design, the VoIP phone , broadly interoperable
with SIP/IAX2 platforms and VoIP hardware from major third party vendors, is
ideal for a residence or business using a hosted IP telephony service, an IP PBX,
or a large scale IP centrex deployment.

The VoIP phone features a full-duplex speakerphone with advanced
acoustic echo cancellation, dot-matrix graphic backlit LCD, additional features
including 3-way conferencing, call transfer (blind/attended), call forward, call
waiting, DND, Voicemail, SMS, customized dial peer, 3 soft keys, as well as DHCP
(client/server), NAT traversal (STUN), VLAN (voice VLAN/data VLAN), QoS with
diffserv, VPN (L2TP).

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

Key Features


- Support 2 SIP lines
- SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of
server, Peer to Peer/ IP call
- Compatible with IAX2 protocol
- 3-line dot-matrix graphic backlit LCD
- Dual 10/100Mbps Ethernet ports (switched/routed)
- DHCP (client/server), Static IP, PPPoE for xDSL
- Full-duplex speakerphone with advanced acoustic echo cancellation
- Support codec: G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.711(A-law/µ-law), G.722
- Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
- Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise
Generation), Line Echo Cancellation (G.168/165), and AGC (Automatic Gain Control)
- Call features: voicemail, SMS, caller ID display or block, conference call, call
Forward, call Transfer (blind or attended), call hold, call waiting, paging and
intercom, call park/pickup, join call, click to dial, DND, black list, limited list, call history
- 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

 

Hardware Specification

 

WAN Port
(for connecting to Internet)
1 X 10/100Mpbs RJ45 port 
LAN Port
(for connecting to PC)
1 X 10/100Mpbs RJ45 port
Speaker Full-duplex Speakerphone with advanced acoustic echo cancellation
LCD Display 3 lines dot-matrix graphic backlit LCD
Memory SDRAM: 8M
Flash Memory: 2M
Function Keys 14 dedicated function keys (MWI, Phonebook, Hold, Transfer, Speakerphone, Redial, Mute, Call history, Menu, volume control, etc.)
3 soft keys
4 navigation keys
Features & Benefits
Standard SIP v1 (RFC2543), v2 (RFC 3261) & correlative RFCs
Support 2 SIP lines
Support IAX2 protocol
SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
Compatible with Asterisk, Trixbox and other SIP/IAX platforms
Voice Codec G.711(A-law/ µ -law), G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.722 (wideband)
Voice Standard DTMF relay: RFC2833, SIP info
Auto Gain Control (AGC)
G.168/165 compliant line echo cancellation (LEC)
Acoustic Echo Cancellation (AEC)
Voice Activity Detection (VAD)
Comfort Noise Generation (CNG)
Adaptive Jitter Buffer
Call Features Call waiting, call forward, call transfer (blind / attended), call hold, 3-way conference call, paging and intercom, call park/pickup, auto-answer, join call, click to dial
Customized dial peer
Caller ID display / block
DND (do not disturb), Black List, Limited List
Support Voicemail, SMS
Call Logs: Incoming call, Outgoing call, Missed call (100 entries each)
Phonebook:500 entries
MWI: Message Waiting Indicator
Network and Management
Access Mode DHCP (client/server), Static IP, PPPoE for xDSL
Management Web, Keypad, Telnet management Management with different account right
Auto-provisioning through TFTP/FTP/HTTP
Firmware upgrade through TFTP/ FTP
Configuration file download/upload
Support Syslog
Protocols TCP/IP/UDP, DHCP, PPPoE, SNTP, STUN, MD5, DNS, RTP, RTCP, Telnet, HTTP, FTP, TFTP
Applications NAT Traversal (STUN); VLAN; QoS with diffserv; VPN (L2TP) ; SRTP security protocol; SNTP Client; DMZ; Firewall; DNS relay; Main DNS and secondary DNS server.
Operating Requirements
Operating Temp. 0~40 degree C
Storage Temp. -25~60 degree C
Operating Humidity 10~90% Non-condensing
Storage Humidity 10~90% Non-condensing
Power Requirement Input 100~240V AC, Output 5V DC 1A
Power Consumption Idle: 1.5W Active: 1.8W
Regulatory Compliance CE, FCC part 15 class B, RoHS
Packages Contents
UTP1400 IP Phone unit 1
Power Adapter   1  
RJ45 Ethernet Cable 1
CD with User Manual 1

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