Specifications

1,2 fxs and 2 fxo channels voip gateway.
2,support ITU H.323 V4 and IETF SIP V2.
3,peer to peer to IP calls
4,high quality.

Product Introduction:

The VoIP gateway HTM222 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)

Two 10/100 Ethernet for WAN / LAN connections

Peer-to-Peer IP Calls

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

 

Support 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