Product Details

Specifications

voip ata adapter
1,support ITU H.323V4 and IETF SIPV2
2,ensure high quality speech
3,support VLAN and QoS
4,NAT

2 FXS and 2 PSTN VoIP Gateway

The 2-FXS SIP Voip Gateway (ATA) with 2 PSTN Live Line is designed as a compact, high performance, and low cost VoIP Analog Terminal Adapter (FXS Gateway). It comes with a FXS port to interface with a traditional analog phone set or a PBX trunk line for VoIP communications. By connecting a PSTN line to the Bypass port, the phone set connected to the FXS port can also access the PSTN line for traditional telephone service. The HT-522 is a full featured FXS gateway and is designed for easy installation and configuration. It is an ideal low cost solution for travelers and SOHO users

Key Features

Static IP support
Switch mold support
support DHCP,PPPOE
Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
Two 10/100 Ethernet for WAN / LAN connections
Peer-to-Peer IP Calls
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
Line Echo Cancellation
VLAN and QoS support
NAT Transversal and Router functions
Voice prompts, HTTP Web, Auto Provision support for configuration and updates
Highly stable embedded Linux operating system in high performance ARM 9 Processor

Basic Features

Four RJ-11 FXS port for traditional phone set or PBX's trunk line
LEDs for Power, Ready, Status, WAN, PC, FXS
Call Forward, Call Hold, Call Transfer
Dial Plan
Caller ID
3 times re-password
Billing support
Keyboard setting
IP voice enrollment

Enhanced Features

Support call forward/transfer/hold, phone book
Support G.711 A/μ law, G.729A/B, G.723.1 Codecs
Support QoS, NAT transversal and router function
Support VAD, CNG, EC
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), G.729A/B, G.723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO