KASDA KT102, the VoIP Phone applies:

a SIP-based phone newly launched by Technologies. Each KT102 has two Ethernet interfaces: the LAN interface for access to a LAN or the Internet and the PC interface for connection to a PC. The built-in network address translation (NAT) and dynamic host configuration protocol (DHCP) functions enable a KT102 and the PCs connect to it to share the same connection to a LAN or the Internet. 

The comprehensive software features and superior voice coding performance enable KT102 to provide you with cost-effective voice communication solutions. The LCD-and Web-based configuration allows you to configure KT102 easily and conveniently.

   Voice Processing Features

¨ Codec options:G.711A/µ,G.729A/B,G.723.1

¨ Voice activity detection (VAD)

¨ Comfort noise generation (CNG)

¨ Dynamic jitter buffer (adaptive)

   Ethernet Features

¨ Two integrated 10/100BaseT Ethernet MACs for LAN and PC connection

¨ 802.1p & Q standard compliant, and support two VLAN tags

¨ Provide NAT/Bridge forwarding route

   PSTN line

¨ One PSTN line port

   VoIP Protocol Stacks

¨ SIP/MGCP/H.323 signaling stacks

¨ RTP/RTCP/SDP media processing stacks

   Management

¨ SNMP,TFTP,HTTP

¨ ManageCenter

   Input/ Output Units

¨ 128*65 Dot Matrix LCD

¨ Key board

¨ 3.5mm microphone port

   Networks

¨ Multiple ways to access the Internet, such as asymmetric digital subscriber line (ADSL), digital data network (DDN), cable Modem, LAN, and WAN

¨ Two Ethernet interfaces (RJ-45), which are 10/100 Mbps autosensing and support MDI/MDIX

¨ NAT and Bridge route forwarding

¨ DHCP client/ server

¨ DMZ and static port mapping

¨ Supports QoS via 802.1p/Q ensuring the reliable VoIP service in multi-service environment

   VoIP Services

¨ Compatible with domestic and foreign common softswitch platforms, including ZTE, Huawei, SONUS, Cisco, Lucent, Nortel, Siemens, UT Starcom, etc.

¨ Supplementary Services-Call transfer(Blind transfer and Attended transfer), Call forward (Unconditional, When busy, When no answer),Call wait, Call hold, Call pickup, Multi-way conference, DnD, Call block, etc

¨ Primary and backup server connection switching

¨ Firmware upgrading automatically

   Basic Functions

¨ Up to nine number slots for speed dial provided, along with other related functions such as redial, and mute

¨ Automatic vocal answering

¨ Caller ID

¨ Hands-free mode, adjustable volume, and multiple ringing tones

¨ Simple and flexible configuration through LCD, web and ManageCenter

¨ Plug and play (PnP), identical to legacy phones on placing and answering a call

¨ Voice quality almost the same as that of a legacy phone

¨ Ring files select and upload

¨ Phone book

¨ Call history

   Protocols supported

¨ SIP: RFC3261~RFC3266

¨ MGCP: RFC3435 (Compatible RFC2705),

¨ H.323: H.225,H.235,H.245,Q.931

¨ RTP/RTCP:  RFC3550(Compatible RFC1889/RFC1990)

¨ SDP: RFC2327

¨ TCP/IP, PPPoE, DHCP, NAT, Bridge, IEEE802.1p/Q

¨ SNMP,TFTP

    Voice Codecs:

¨ G.711A/µ

¨ G.729 A/B

¨ G.723.1

   Voice Feature

¨ Voice activity detection (VAD)

¨ Comfort noise generation (CNG)

¨ Dynamic jitter buffer (adaptive)

¨ Echo Cancellation

   DTMF Relay

¨ In-band DTMF Relay

¨ Out-band DTMF(RFC 2833)

   Configuration

¨ Voice director

¨ Web

¨ SNMP

¨ ManageCenter ( Management System)

   Upgrading Firmware:

¨ ManageCenter (Management System)

¨ TFTP

¨ Web

¨ FLASH: 2M

¨ SDRAM: 8M

¨ Ethernet: two RJ-45 ports

¨ Power: DC 12V,600mA

¨ Operating temperature:   0 to 45 (32°F to 113°F)

¨ Storage temperature:    -10 to 70(14°F to 158°F)

¨ Relative humidity:       10% to 90

VOIP Phone KT102