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| WOIP IP Phone SIP |
| WOIP's IP Phone provides a cost-saving solution for small businesses on their telecommunication needs. This desktop IP phone delivers outstanding voice quality while achieving major savings on long distance or international phone calls by simply accessing your existing Internet and Intranet (LAN) telephony network. This not only reduces your telecommunication costs, but also improves your current phone service capabilities with many built-in features of the unit. Enjoy WOIP's state-of-the-art Internet telephony for your today's business needs. |
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Features : - SIP support STUN protocol (RFC 3489) for use in virtual IP environment (optional) - Call with or without SIP server (direct IP dialing) - Hands free dialing - 10 sets speed dial records - 40 Call records each for Answered, Missed, Dialed calls - Configuration via LCD & keypad or Telnet commands - Pre-paid account remaining time display (Optional) - Provide prepaid call service |
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Specifications : Hardware specification: LCD display: 2 lines * 24 character mode for alphanumeric display Keypad: 12 alphanumeric keypads, Speaker, Mute, Volume up, Volume down Redial, Speed Num, Balance, Call Record, Voice Message LED: 6 LEDs - Call Record: Flash indicates new voice record - Voice Msg: Flash indicates new miss calls - Power Status: On indicates power is ready - Ethernet Status: Flash indicates activated of Ethernet physical layer - Spk: ON indicates speaker is activated - Mute: ON indicates microphone is muted Ethernet: IEEE802.3 10 Base-T, RJ45 connector Physical: - Power adaptor: 100~240 Vac(input) +5Vdc(output) - Size: 245*217*65mm - Weight: 775g Protocols: SIP Protocol - IETF RFC3261 (SIP :Session Initiation Protocol) - Backward compatible with RFC 2543 will be released in the future - UDT translation only - IETF RFC 2976 (SIP INFO Method) - IETF RFC 2327 (SDP: Session Description Protocol) - IETF RFC 3264 (An Offer/Answer Model with the Session Description Protocol) - IETF RFC 1889 (RTP: Real-time Transport Protocol) - IETF RFC 1890 (RTP Profile for Audio and Video Conferences with Minimal Control) - IETF RFC 2833 (RTP Payload for DTMF digits, Telephone Tones and Telephone Signals) - Support inband DTMF tone send/receive with RTP - Supports out of band DTMF signaling based on RFC2833 - IETF RFC 3489 (Simple Traversal of User Datagram Protocol (UDP) through Network Address Translators (NATs)) - Draft-ietf-sip-refer-06.txt (SIP Refer Method) - IETF RFC 2617 (HTTP Authentication) - Supplementary Services I. Call Hold II. Call Waiting III. Call Transfer IV. Call Waiting V. Call Forwarding TCP/IP protocols PPPoE: PAP (Password Authentication Protocol), CHAP/MD5 Telnet server DHCP client TCP/UDP/RTP/IP SNTP HTTP web configuration TFTP/FTP file transfer protocol for remote firmware upgrade |
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1. Handset 2. Speaker 3. LCD display 4. Dailing keypad 5. 4 context sensitive soft keys 6. Ethernet Status 7. Voice Msg with LED 8. Call Record with LED 9. Power Status 10. Balance key 11. Redial key 12. Speed Num key 13. Mute key with LED 14. Spk key with LED 15. -Volume+ key |
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