HT912T(SIP and H.323),1 fxs port ata voip gateway

Specifications

Overview

The HT-912T is designed as a compact, high performance, and low cost VoIP Analog Terminal Adapter (FXS Gateway). It comes with one FXS port to interface with a traditional analog phone set or a PBX trunk line for VoIP communications.  It bundles with lots of features to meet the demand in various network environment.  It is an ideal low cost VoIP solution for travelers and SOHO users.
 

Key Features
--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
--One 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
 

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:

SKYLINE International Communication Co., ltd is a high-tech enterprise integrated with scientific research, development, production and sales. The company has been engaged in the research, development and production of communication products since its establishment. Our  products are listed as follows.

 FXS series:HT-912(one fxs port),HT-922(2 fxs ports),HT-842R(4 fxs ports).HT-882(8 fxs ports)

 FXS+PSTN series:HT-812P(1 fxs port with a PSTN live line),HT-822P(2 fxs ports with a PSTN live line),HT-522(2 fxs ports with 2 PSTN live lines),HT-544(4 fxs ports with 4 PSTN live lines)

 FXO series:HT-322(2 fxo ports),HT-342(4 fxo ports)

 FXO+FXS series:HT-112(1 fxo port and 1 fxs port),HT-222(2 fxo ports and 2 fxs ports),HT-442(4 fxo ports and 4 fxs ports)

 GSM series:GoIP 1 (1 channel), GoIP 4 (4 channels), GoIP 8 (8 channels)

 RoIP series: RoIP302, RoIP302M, RoIP102

 IP Phone: EP-636 (1 lines), EP-8201 (4 lines)

 Recorder Box:TYH8200 (1 line), TYH8201 (2 lines), TYH636 (8 lines)

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