2 FXS and PSTN VoIP Gateway HT-522 with Open Standard VoIP Protocols, Retail and Wholesale

 2 FXS and PSTN VoIP Gateway

Overview

Clear Voice 2 FXS+PSTN  VoIP Gateway with Open Standard VoIP Protocols 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

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