Product Main

Specifications

Overview The HT-822P is designed as a compact, high performance, and low cost VoIP Analog Terminal Adapter (FXS Gateway). It offers 2 FXS ports to interface with traditional analog phone sets or PBX trunk lines for VoIP communications.  By connecting a PSTN line to the Bypass port, the phone sets
connected to the FXS ports can also access traditional telephone service via the bypass port. The HT-822P is a full featured FXS gateway and is designed for easy installation and configuration. It is an ideal low cost solution for SME and SOHO environments where both VoIP and PSTN services 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
  • 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