2 FXS and 1 PSTN VOIP Gateway HT-822P, Retail and Wholesale

 2 FXS and 1 PSTN VOIP Gateway

Overview

2-FXS Port with a PSTN Bypass Port
Support dual protocols: ITU-H.323 V4 and IETF SIP V2
Provide 2 RJ-11 ports for POTS phones or PBX's trunk lines
Provide 2 RJ-45 Ethernet Ports
Provide 1 RJ-11 port for PSTN Life Line (Bypass only)
Support single or multiple registrations
Support hunt line, 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

Highlights

Built-in H.323 V4 and SIP V2 Protocols
Support for both VOIP and PSTN calls
High performance router
Caller ID for both VOIP and PSTN calls
Auto provisioning and firmware updates
Encryption transversal
Multiple languages

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
Two RJ-11 FXS ports for traditional phone sets or PBX's trunk lines
One RJ-11 port for connection to PSTN line (Bypass function only)
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

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