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IP Phone AT-620
 
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IP Phone AT-620
AT-620 IP SIP Phone is feature-rich yet cost competitive Ethernet Phone, which provides the voice data convergence solution with state- of-the-art technology. With high-quality voice, the IP SIP Phone can work perfectly with standard-based IP telephony system of SIP Protocol Stack.

The IP SIP Phone Users can experience both the familiar operation of legacy hi-end PSTN Phone and intelligence of an IP network device with the LCD display, which can provide all the necessary information when proceeding a calling activity.

The IP SIP Phone is the most powerful network device for next generation networks.

Key Features

Phone Faceplate
- 12 Dialing Buttons (0-9, *, #)
  9 Fixed Function Buttons,
  20 DSS-Key with 2 colors LED Buttons,
  2 Volume Buttons.

Applications
- Instant Messenger
- Voice Mail Integration
- NAT Traversal
- 500 Entries Address Book

Audio Codec
- G.711 (A-Law, U-Law)
- G.723.1
- G.729A/B
- Adaptive Jitter Buffer Management
- Voice Activity Detection
- Comfort Noise Generation
- Acoustic Echo Cancellation

Call Features
- 3-way Conference
- Auto Answer
- Auto Redial
- Call History
- Call Hold
- Call Forwarding
- Call Mute
- Call Park
- Caller ID
- Distinctive Ring
- Do Not Disturb
- Full-duplex Speakerphone
- Phone Locked

 

Protocols
- SIP (RFC3261)
  Backward Compatible with RFC2543
- Session Timer (RFC4028)
- SDP (RFC2327)
- RTP/RTCP (RFC1889 and RFC1890)
- NAPTR for SIP URI Lookup (RFC2915)
- STUN (RFC3489)
- ARP/RARP (RFC826/903)
- SNTP (RFC2030)
- DHCP/PPPoE
- HTTP Server for Web Management
- TFTP/HTTP for Auto Provisioning
- Message Waiting Indicator (RFC3842)
- DHCP Option Codes for SIP (RFC3361)
- 802.1P TOS
- 802.1Q VLAN

Management
- Firmware Upgradeable
- Web Management Interface
- LCD Menu-driven Management Interface
- Telnet Management Interface
- SNMP v2
- Auto Provisioning
- 192-bit AES Key Encryption

I/O Interfaces
- 2 RJ-45
  Dual 10/100 Switched Ethernet Ports
- 2 RJ-22
  Handset and Headset
- Embedded MIC/SPEAKER