PSTN/VOIP/FXS/ Router

Specifications

PSTN/VOIP
Product Introduction:

The HT-822P VoIP gateway 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 environments where both VoIP and PSTN services are required.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.

Product Features:

• Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
• Single or Multiple Server Registrations
• Two 10/100 Ethernet circuits connect to the LAN and an additional device
• Two FXO ports for PSTN terminations
• 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 .

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
• DTMF: RFC 2833, In-band DTMF, SIP INFO

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