The VoIP IP Phone 4 SIP VoIP Phone Network network device with H.323 &PoE (EP-8201) is a quality phone with lots of features for both business and residential users. Its slim and upright design makes it an ideal desktop phone. The phone is based on ITU-H.323 V4 and IETF SIP V2 open standards. The two protocols approach makes the phone to be compatible to most VoIP systems in deployment today. The Phone is designed for the ease of installation and setup. The PoE option simplify the installation in a PoE LAN environment. In addition, the second Ethernet Port allows the existing PC to be connected to the phone directly without addition an additional Ethernet Hub or Switch. Various configuration modes allow the user / system administrator to configure the phone automatically or quickly.

Basic Phone Features Call forward Call transfer Call hold Mute Redial Display caller ID Display call duration Display date and time SMS Capable Access voice mail Send DTMF tones Message waiting indication (MWI) 100 phone book entries 30 most recent call records for dialled, incoming, and missed calls Adjustment of LCD contrast (4 levels) Adjustment of handset volume (6 levels) Adjustment of speaker phone volume (6 levels) Enhanced Features Dynamic selection of codec Advanced jitter buffer Automatic traversal of NAT and firewall VLAN / Qos Router Echo cancellation for Speakerphone Comfort noise generation (CNG) Voice activity detection (VAD) Auto provisioning (requires auto provisioning server) On line firmware upgrade Multi-language support: English and Chinese Supported Standards ITU: H.323 V4, H.225, H.235, H.245, H.450 RFC 1889 - RTP/RTCP RFC 2327 – SDP RFC 2833 – RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals RFC 2976 – SIP INFO Method RFC 3261 – SIP RFC 3264 – Offer/Answer model with SDP RFC 3515 – SIP REFER Method RFC 3842 – A Message Summary and Message Waiting Indicator RFC 3489 – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs) RFC 3891 – SIP “Replaces” Header RFC 3892 – SIP Referred-By Mechanism draft-ietf-sipping-cc-transfer-04 – Session Initiation Protocol Call Control - Transfer Codec: G.711 (A/µ law), GSM, G.729A/B, G.723.1 DTMF: RFC 2833, In-band DTMF, SIP INFO

Supported Standards

ITU: H.323 V4, H.225, H.235, H.245, H.450 RFC 1889 - RTP/RTCP RFC 2327 – SDP RFC 2833 – RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals RFC 2976 – SIP INFO Method RFC 3261 – SIP RFC 3264 – Offer/Answer model with SDP RFC 3515 – SIP REFER Method RFC 3842 – A Message Summary and Message Waiting Indicator RFC 3489 – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs) RFC 3891 – SIP “Replaces” Header RFC 3892 – SIP Referred-By Mechanism draft-ietf-sipping-cc-transfer-04 – Session Initiation Protocol Call Control - Transfer Codec: G.711 (A/µ law), GSM, G.729A/B, G.723.1 DTMF: RFC 2833, In-band DTMF, SIP INFO