4 lines VoIP Phone,support H.323 and SIP,voip payphone

Specifications

Product Introduction:

The Sip Phone with H.323 and Sip (EP-8201) (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