voip phone sip ip phone EP-636

voip phone sip , Relay encryption, SIP2.0,CE&FCC certification LCD, Can be customized.

 

 

 

voip phone sip ip phone EP-636

 Product Overview

It is compatible with SIP V2.

Single file of English letters and single file of digital LCD displayer

Achieve full dulplex hands-free call.

It also has perfect H.450 protocol to complete all telecom value-added services such as call transfer

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 Feature

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

voip phone sip ip phone EP-636