1 fxs port voip ata gateway,HT-912T
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 PBXs 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
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 PBXs 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
Main Products
voip gateway,voip phone