top of page
VOIP GATEWAYS TAU-4.IP
Key features
  • Office PBX functionality
  • 3G/4G channel reservation
  • IPsec encryption
  • TR-069/DHCP-based autoprovision
  • Maximum line length—6 km
  • Measurement of subscriber's line parameters
TAU-4.IP VoIP gateway is an optimal solution for provision of advanced VoIP services to corporate clients via analogue phone devices. 

Business solution
Due to the wide function set, TAU-4.IP can be used as an independent mini office PBX with internal switching and basic set of value added services as well as in interaction with IP PBX.

High-Quality Sound
A high-performance hardware platform based on an state-of-the-art Mindspeed Technologies chip, support for all popular audio codecs used in VoIP networks (G.711, G.723.1, G.726, G.729), echo cancellation, silence detector, comfort noise generator, DTMF signals reception and generation and traffic prioritization (QoS) ensure the high quality of the voice data.

Reservation
In case of loss of the main connection to the Internet, there is a possibility of automatic switching to a backup 3G/4G channel. If there is no a backup channel, the connection between gateway subscribers is saved.

Easy-to-Use
Centralized configuration downloading, intellectual firmware update and collection of data on subscriber gateway status can be realized through Eltex.ACS system via TR-069. Eltex.ACS provides simplicity of Eltex CPE management and reduces operating expenses.

VOIP GATEWAYS TAU-4.IP

₹0.00Price
Out of Stock

  • Interfaces

    • 4 FXS ports 
    • 1 WAN 10/100Base-T
    • 1 USB port

    VoIP protocols

    • SIP

    Voice codecs

    • G.711 (a-law, µ-law)
    • G.723.1
    • G.726 (24 Kbps and 32 Kbps)
    • G.729 (a/b)

    Fax

    • T.38 UDP Real-Time Fax
    • G.711 (a-law, µ-law) pass-through

    Voice standards

    • VAD (Voice Activity Detector)
    • CNG (Comfort Noise Generation)
    • AEC (Acoustic Echo Cancellation, G.168 recommendation)

    Functional features

    • Payphones connection (polarity reversal, meter impulses 12/16 KHz)
    • Subscriber line`s physical parameters measurement 
    • Internal switching is saved in case of a SIP-server connection failure
    • VAS management via a phone
    • Distinctive Ring
    • Keep-alive during operation behind NAT
    • Calling Party Control (CPC)

    DTMF

    • Detection and generation of signals
    • Transmission by INBAND, RFC 2833, SIP INFO methods

    Value Added services

    • Call Hold
    • Call Transfer
    • Call Waiting
    • Calls forwarding when busy
    • Calls forwarding on no reply
    • Calls forwarding unconditional
    • Number identification (FSK Type I, FSK Type II, DTMF)
    • Calling Line Identification Restriction (CLIR)
    • Hotline/Warmline
    • Call Group
    • 3-Way Conference
    • Pickup Group
    • Hunt group
    • 3-Way Conference gathered on a server (RFC 4579)
    • Pickup Group
    • Hunt groups
    • A dedicated server for 3-Way Conferencing (RFC-4579)  

    VoIP functions

    • Internal connection switching
    • Operation without a SIP server
    • Flexible dial plan
    • Management profiles for FXS ports
    • SIP profiles (support for up to 8 profiles)
    • Geographical redundancy of a SIP server (up to 4 redundant SIP servers)
    • Application of settings without reboot
    • Voice transmission through a protected channel (encryption through IPsec)
    • IMS (3GPP TS 24.623) for Call Hold, Call Waiting, 3-Way Conference, Hotline management
    • Using SIP servers from DHCP option 120
    • Support for operation behind NAT (STUN and Public IP)
    • Setting custom call-control signals

    Quality of service (QoS)

    • DSCP and 802.1p assignment for SIP and RTP packets
    • Bandwidth redundancy

    Network functions  

    • Different protocols for connection to a service provider network (Static, DHCP, PPPoE, PPTP, L2TP)
    • Local DNS server
    • Dynamic/static routing
    • VLAN per service (VLAN for each service: Internet, VoIP, Management)
    • Firewall
    • Operation via 3G/4G USB modems
    • Print server
    • IPsec (for voice transmission and remote control)

    Management

    • WEB (multilingual*)
    • SNMP (phone parameters configuration, monitoring and statistics gathering) 
    • Telnet
    • Syslog
    • Tcpdump
    • SSH
    • TR-069 (Eltex.ACS server is recommended)
    • DHCP-based autoprovision (43, 66, 67 DHCP options)
    • Management via IPsec encrypted channel 

    Security

    • Username and password authentication 
    • Firewall
    • Access rights differentiation for admin/user
    • Password encryption
    • Digest authentication

    USB port

    • USB-storage connection with FAT/FAT32/EXT2/EXT3/NTFS file systems – files exchange via a network according to FTP protocol
    • USB 3G/4G modem connection –  3G/4G channels reservation
    • Printer connection – setting up of a print server

    Technical characteristics

    • CPU Mindspeed
    • SDRAM 256 MB
    • Flash 32 MB
    • OS Linux

    Physical characteristics 

    • Power adapter: 12 VDC, 2A
    • Power consumption: up to 11 W
    • Temperature: +5°С to +40°С
    • Humidity: up to 80%
    • Dimensions: 218х120х49 mm, desktop case
    • Weight: no more than 0.3 kg

    Specifications

    • RFC 3261 SIP 2.0
    • RFC 3262 SIP PRACK
    • RFC 4566 Session Description Protocol (SDP)
    • RFC 3263 Locating SIP servers for DNS lookup SRV and A records
    • RFC 3264 SDP Offer/Answer Model
    • RFC 3311 SIP Update
    • RFC 3515 SIP REFER
    • RFC 3891 SIP Replaces Header
    • RFC 3892 SIP Referred-By Mechanism
    • RFC 4028 SIP Session Timer
    • RFC 2976 SIP INFO Method
    • RFC 2833 RTP Payload for DTMF Digits, Flash event
    • RFC 3108 Attributes ecan and silenceSupp in SDP
    • RFC 4579 SIP Call Control - Conferencing for User Agents
    • RFC 3361 DHCP Option 120
    • RFC 3550 RTP A Transport Protocol for Real-Time Applications
bottom of page