PSTN/VOIP/FXS/ Router

PSTN/VOIP
1 support ITU H.323V4 and IETF SIPV2.
2 ensure the high quality speech
3 support VLAN and QoS
4 NAT Transversal
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