Specifications

1)8 fxs port voip ata gateway,HT882
2)Support SIP and H.323
3)NAT transversal and route function
4)Vlan and QoS

Overview 

The HT-882 is designed as a compact, high performance, and low cost VoIP Analog Terminal Adaptor (FXS Gateway).  It comes with 8 FXS ports to interface with traditional analog phone sets or PBX trunk lines for VoIP communications.  The HT-882 is a full featured FXS gateway and is designed for easy installation and configuration. It supports the two most widely used Open VoIP Standards (SIP and H.323).  This allows the HT-882 to interoperate seamlessly with softswitches or IP PBXs made by various vendors.  Its high performance offers toll quality voice, flexible networking, and feature-rich call functions.  It is an ideal low cost solution for SME environment where multiple lines are required.

 

Key Features
--Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
--Single or Multiple Server Registrations
--Peer-to-Peer IP Calls
--Two 10/100 Ethernet for WAN / LAN connections
--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 ports for traditional phone sets or PBX's trunk lines
--LEDs for Power, Ready, Status, WAN, PC, FXS ports 
--Call Forward, Call Hold, Call Transfer
--Dial Plan
--Caller ID
 

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), 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.

 

Example:
peer to peer:it is a new function that you can use our voip without platform,what is more,it is free roaming for international call.you just pay the local call

Model 1:

 

Model 2:

 

 

Model 3:

 

 

 

 

 

 

 

 

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