Specifications

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

Product Introduction:

The 2 FXO ports SIP VoIP gateway is  designed as a compact, high performance, and low cost FXO Gateway.  The FXO detection is optimized to avoid the hold up of the PSTN line when the other party is disconnected.  This has been one of the key issue in the design of FXO gateway.  The incoming PSTN Caller ID is also transmitted to the VoIP user for more user friendly operation. The HT-322 is a full featured FXO gateway and is designed for easy installation and configuration. It is an ideal solution for VoIP to PSTN termination in both SME and SOHO environment.

 

 

 Product Feartures:

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 .

 

 

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 .