Specifications

1,support ITU H.323V4 and IETF SIPV2
2,ensure high quality speech
3,support VLAN and QoS
4,NAT Transversal
5,Router power

Product Introduction:

The HT-812p VoIP gateway (with 1 FXS and 1 PSTN ports)is a PSTN Bypass Port is designed as a compact, high performance, and low cost Analog Terminal Adaptor (FXS PSTN Bypass Gateway).  It comes with 1 FXS ports to interface with traditional analog phone sets or PBX trunk lines for VoIP communications.  The HT-812P is a full featured PSTN Bypass gateway and is designed for easy installation and configuration.  It supports the two most widely used Open VoIP Standards (SIP and H.323). This allows the HT-812P to interoperate seamlessly with softswitches or IP PBXs made by various vendors. Its high performance offers toll quality voice, flexible networking, and feature-rich call functions. It is an ideal low cost solution for SME environment where multiple lines are required 

 

Product Feature:

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

GSM module for making GSM calls

Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer

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

Codec: G.711 (A/µ law), G.729A/B, G.723.1

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