TitleProduct

1 Fxs Port VoIP Gateway (HT-912) , SIP&H. 323

  • Price:

    Negotiable

  • minimum:

  • Total supply:

  • Delivery term:

    The date of payment from buyers deliver within days

  • seat:

    Guangdong

  • Validity to:

    Long-term effective

  • Last update:

    2017-12-21 09:47

  • Browse the number:

    277

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Shenzhen DBL Technology Limited
Contactaixin:

Mr. Sasa(Mr.)  

Email:

telephone:

phone:

Arrea:

Guangdong

Address:

Room 5c, Aozhihao Integrated Building, Xinzhou 4th Street, Futian District, Shenzhen, China 518048

Website:

http://dbltek.hdsshfm.com/

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The Supply 1 FXS Port VoIP Gateway, ATA, SIP&H. 323 HT-912 is designed as a compact, high performance, and low cost Analog Terminal Adaptor (FXS Gateway). It comes with 4 FXS ports to interface with traditional analog phone sets or PBX trunk lines for VoIP communications. The HT-912 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-912 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.

Key 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
One RJ-11 FXS port 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