Specifications

1,supported H.323V4 and SIP V2.
2,high quality speech can be ensured
3,supported VLAN and QoS.
4,NAT Transversal
5,Router

Product Introduction:

The 1FXS and 1 PSTN ports SIP VoIP gateway 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 requ.

 

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
• 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