Product Main

Specifications

Overview

The HT-342 is designed as a compact, high performance, and low cost FXO Gateway.  The FXO detection is optimized to avoid the hold up of the PSTN line when the other party is disconnected.  This has been one of the key issue in the design of FXO gateway.  The incoming PSTN Caller ID is also
transmitted to the VoIP user for more user friendly operation. The HT-342 is a full featured FXO gateway and is designed for easy installation and configuration. It is an ideal solution for VoIP to PSTN termination in both SME and SOHO environment.
Key Features

  • Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
  • Single or Multiple Server Registrations
  • 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 10/100 Ethernet circuits connect to the LAN and an additional device
  • Four RJ-11 FXO ports for PSTN terminations (PSTN lines or PBX's extensions)
  • LEDs for Power, Ready, Status, WAN, PC, FXO ports 
  • Call forward from PSTN to VoIP and VoIP to PSTN
  • Dial in mode or dial out mode only
  • Dial Plan
  • Password protection for both PSTN dial in or dial out
  • Retransmit PSTN Caller ID to VoIP terminal

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