Product Main

Specifications

Overview The HT-522 is designed as a compact, high performance, and low cost VoIP Analog Terminal Adapter (FXS Gateway).  It comes with a FXS port to interface with a traditional analog phone set or a PBX trunk line for VoIP communications. By connecting a PSTN line to the Bypass port, the phone
set connected to the FXS port can also access the PSTN line for traditional telephone service. The HT-522 is a full featured FXS gateway and is designed for easy installation and configuration. It is an ideal low cost 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
  • Two RJ-11 FXS port for traditional phone set or PBX's trunk line
  • Two RJ-11 port for connection to PSTN line (Bypass function only)
  • 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