Specifications

1)1 channel voip sip phone
2)three lines Alphanumeric LCD
3)NAT and SIP
4)Relay encryption
5)Can be customized.

Overview
   The EP-636 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. 

 

Key Features
--Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
--All standard PBX functions
--Four call appearances support two simultaneous calls
--Two 10/100 Ethernet circuits connect to the LAN and an additional device
--3-Line LCD (Icons, Alphabets/Numbers, Numbers)
--Buttons and keys for all commonly used functions
--Message waiting indicator
--Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
--Full duplex speaker phone
--VLAN and QoS support
--NAT Transversal and router functions
--Power over Ethernet (PoE) or AC/DC adapter
--Menu, HTTP Web, Auto Provision support for configuration and updates
--Highly stable embedded Linux operating system in high performance ARM 9 Processor
 
Basic Phone Features
--Call forward
--Call transfer
--Call hold
--Mute
--Redial
--Display caller ID
--Display call duration
--Display date and time
--Access voice mail
--Send DTMF tones
--Message waiting indication (MWI)
--100 phone book entries
--30 most recent call records for dialed, 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 
 

Application:

Saving Cost on International and Long Distance Calls,when you are in the other countries& you have this gateway at your Local office,you could dial & receive phone call from your customers as you are there without Long Distance fee.
Very excellent solution for internal company with branch offices oversea.

 

 

 

If you have any doubt or want to know further information,please feel free to contact me.