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  • up to 4 Е1 flows (RJ-48) 
  • up to 128 VoIP channels
  • high quality voice
  • carrier-class reliability
  • desktop case

SMG-4 platform can be used as a trunk gateway for conjugating signal flows and media flows of TDM and VoIP networks.


Optimal Solution for Digital PBX 

SMG-4 gateways make a smooth transition from TDM- infrastructure to advanced VoIP - networks possible, providing the full compatibility with the existing Devices. SMG-2  gives a splendid opportunity to connect the digital PBX with NGN.

Functional compatibility 

Strict conformity to the requirements of up-to-date protocols, recommendations and standards ensure 100% functional compatibility of SMG-4 with different equipment: digital АТS, IP-PBX, Softswitches, VoIP gateways, SIP telephones, program SIP clients, etc. 


Intellectual call routing based on billing system responses according to RADIUS protocol enables to create flexible methods of call processing.


Intellectual protection of IP networks  

The intellectual protection against unauthorized external connections of SIP users (fail2ban, iptables, white/blacklists, etc.) has been implemented in trunk gateway SMG-4. 

Media Flows Transcoding 

The hardware transcoding based on MediaCodecs Mindspeed Technologies helps coordinate media flows with different VoIP codecs.  

High Quality of Voice proccessing 

Modern hardware platform, support of all basic audio codecs used in VoIP- networks (G.711, G.723.1, G.726, G.729), echo elimination functions, silence detector, comfort noise generator, DTMF signal reception and generation, and traffic prioritization mechanism (QoS) provide high quality of voice processing. 


SMG-4 converter has 4 RJ-48 ports on board for E1 flows connection , 1 LAN 10/100/1000 Base-T (RJ-45) port to connect IP-network.


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  • Call management

    • Called party (CdPN) or calling party (CgPN) number routing
    • Before and after routing number modification
    • Use of multiple number plans
    • Trank-group cutoff
    • Call Control via RADIUS¹
    • Trank-groups direct connection
    • Prefix for few Trank-groups

    Voice codecs

    • G.711 (a-law, µ-law), G.729 (A/B), G.723.1, G.726 (32 Kbps)

    Fax support

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

    Voice standard

    • VAD (Voice Activity Detection)
    • CNG (Comfort Noise Generation
    • AEC (echo cancellation, G.168 recommendation)

    Quality of Service (QoS)

    • Diffserv and 802.1р priority assignment for SIP and RTP
    • Dynamic and static jitter buffer


    • Outband (RFC 2833, SIP INFO)
    • Inband (RFC 2833, SIP INFO)


    • RADIUS Accounting
    • Different billing systems support:Hydra Billing, LANBilling, PortaBilling, NetUP, BGBilling (integration with other systems is possible)
    • Billing information recordering to CDR file and sending to FTP-remote server

    TDM protocols

    • SS7
    • PRI (Q.931)

    VoIP protocols

    • SIP, SIP-T/SIP-I

    Capacity and Performance

    • 128 VoIP channels
    • E1 flows (RJ-48)
    • Maximum load intensity - 40 cps


    • Single file download-upload of configuration
    • Multiple network interfaces creating for (SIP, RTP) telephony with different IP-addresses
    • Number plans multiple operating
    • Signal channel SS7 redundancy
    • Talking connection monitoring (by RTP or RTCP availability)
    • Trunk registration of SIP-interfaces

    Control and monitoring

    • Channel flows Е1 and VoIP monitoring on web-interface 
    • Emergency logging with opportunity to save logs on syslog-server
    • Emergency notification by SNMP


    • Access attempts to device output in syslog
    • List of permitted IP addresses for device control access
    • Access rights delimitation admin / user
    • IP address control of RTP- counter flow source


    • 1 x 10/100/1000 Base-T(RJ-45) port
    • 4 x E1 (RJ-48) ports
    • 1 x Console (RJ-45) port
    • 1 x USB 2.0 port
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