VoIP Gateway,support H323,SIP,VLAN And QoS;2FXS

Specifications

FXS:1 support ITU H323V4 and IETF SIPV2
2 ensure high quality speech
3 support VLAN and QoS
4 NAT Transversal
5 Router

VoIP gateway,support H323,SIP,VLAN and QoS;2FXS

Product Introduction:

The two channels FXS VoIP gateway is designed as a compact, high performance, and low cost Analog Terminal Adaptor (FXS Gateway).  It comes with 2 FXS ports to interface with traditional analog phone sets or PBX trunk lines for VoIP communications.  The HT-922 is a full featured FXS 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-922 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 Features:

Basic Features

Two RJ-11 FXS ports for traditional phone set or PBX's trunk line

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