Specifications

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

Product Introduction:

The HTM-222 VoIP gateway(with 2 FXS and 2FXO ports)is  the performance Gateway of FXO plus FXS. HTM-222 built-in H.323 and SIP protocols. Highly reliable line detection prevents the line hanging death in the largest tent.

 

It can forward outside Caller ID Number under the SIP protocol, which is an important function of PBX application.set connected to the FXS port can also access the PSTN line for traditional telephone service. The HTM-222 is a full featured FXS gateway and is designed for easy installation and configuration. It is an ideal low cost solution for travelers and SOHO users

 

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 .

 

Basic Features

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

Two RJ-11 port for connection to PSTN line (Bypass function only)

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

 DTMF: RFC 2833, In