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VOIP TRUNK GATEWAY SMG-1016M WITH IP-PBX SUPPORT
  • Scalable platform 1U
  • IP-PBX for 2 000 subscribers with VAS support 
  • High-quality voice processing
  • Carrier class reliability
  • up to 768 VoIP channels 
  • up to 16 Е1 flows (RJ-48) 
  • Support of 2 built-in SD 8Gb 
  • Hardware redudancy
 
SMG-1016M is used as a trunk gateway for interfacing of signal and media streams of TDM and VoIP networks. The gateway also might be used as an IP PBX with VAS support and a universal solution for infocommunication new generation networks (NGN). The wide function-set, strict compliance with requirements and standards and carrier class reliability allow service providers to solve most part of their objectives using SMG-1016M. 
 
Scalability 
SMG-1016M is a beneficial investment in the future of your project due to its scalability. The gateway supports up to 16 E1 streams (SS7, PRI, V5.2) and up to 768 VoIP channels. 
 
Carrier class reliability 
SMG-1016M provides high level of fault tolerance due to embedded state-of-the-art Marvell chip, uniform load distribution among submodules, power modules redundancy
and usage of up-to-date technologies based on parallel computing. The gateway will switch to a backup submodule in case of a primary submodule fault.  
 
Functional compatibility 
Strict compliance with up-to-date protocols’ requirements, recommendations and standards provides functional compatibility with a variety of equipment: digital PBX, IP PBX, Softswitches, VoIP gateways, SIP phones, programmable SIP clients, etc.
  

Media streams transcoding
The hardware transcoding based on MediaCodecs Mindspeed Technologies helps negotiate media streams with different VoIP codecs which are used in up-to-date networks.
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IP-АТS with VAS and LAES support 
Additional options for SMG-1016M gateway allow using it as a full-function IP-PBX up to 2 000 SIP users with supporting a wide set of VAS, as well as full compliance with regulatory documents according to LAES. The IP-PBX ECSS-10 program module is intended for fast deployment of VoIP communications node with minimal capital expenses (CAPEX). Availability of all types of certificates for a family of products ECSS-10 makes it possible to use IP-АТС ECSS-10 on the basis of trunk gateway SMG-2016 as an PBX of any level with subsequent acceptance for operation by the authorities of Russia's Communications Oversight Agency and Federal Security Service of Russia. 
 
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-1016M. In order to provide additional protection when connecting to public IP networks, a compatibility with the session border controllers (e.g., SBC-1000) performing the functions of internetwork screens for VoIP networks. 

 
RADIUS routing 
Intellectual call routing based on the billing system responses according to RADIUS protocol will help build flexible rules for calls processing.
 

Intellectual protection of IP networks
The intellectual protection against unauthorized external SIP subscribers connection and connections via http/https/telnet/ssh is realized on the SMG-1016M
(Dynamic Firewall, Static Firewall, black and white lists of IP addresses and subnetworks, etc.). For additional defense, SMG-1016M is compatible with session border controllers (e.g. SBC-1000) that are used as a firewall for VoIP networks.  
 

IP PBX with VAS support
Additional options for SMG-1016M gateway allows using it as a full-featured IP PBX with up to 2000 SIP subscribers connection and support for a wide range of value added services. A programmable IP PBX module ECSS-10 is dedicated to fast deployment of a VoIP node with a minimum of capital expenses. ECSS-10 and SMG-1016M might be used as a PBX of any level.

VOIP TRUNK GATEWAY SMG-1016M WITH IP-PBX SUPPORT

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  • Calls management

    • Interaction with STUN-server on the SIP interface
    • Routing based on called number (CdPN) or calling number (CgPN)
    • Number modifications before and after routing
    • Call recording according to number mask and dialplan1
    • Use of multiple dialplans
    • Subscriber lines restriction
    • Subscriber service mode
    • Trunk group cut-off
    • Call management via RADIUS1
    • Direct connection of trunk groups
    • Prefix for few trunk groups
    • Interactive voice menu (IVR)1
    • Uploading/downloading of configuration as a single file
    • Lines limiting for SIP interface
    • Egress and ingress lines restrictions for a subscriber
    • Ingress load limiting (calls per seconds) for a trunk group



    Voice codecs

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



    Fax transmission

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



    Voice standards

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


    Quality of service (QoS)

    • Diffserv and 802.1p priorities assignment for SIP and RTP
    • Dynamic and Static jitter buffer
    • Ingress/egress traffic rate limiting



    DTMF

    • INBAND, RFC 2833, SIP INFO, SIP NOTIFY transmission methods



    Billing

    • Billing data is recorded in CDR file. CDR files are kept on a local HDD and
    • remote FTP server.
    • RADIUS Accounting
    • Supported billing systems: Hydra Billing, LANBilling, PortaBilling, NetUP,
    • BGBilling (there is an opportunity of integration with other systems)



    Flexibility

    • Multiple network interfaces creation for telephony (SIP, RTP) with
    • different IP addresses
    • Operation with multiple numbering plans
    • Signal SS7 channel redundancy
    • Voice activity control (by the presence of RTP or RTCP)
    • Individual routing for streams from a single SS7 linkset



    TDM protocols

    • SS7
    • PRI (Q.931)
    • Q.699 (PRI and SS7 interaction)
    • V5.2 LE1
    • V5.2 AN1



    VoIP protocols

    • SIP, SIP-T/SIP-I, SIP-Q
    • H.3231
    • SIGTRAN (M2UA, IUA)1
    • H.2481



    Capacity and perfomance

    • up to 768 VoIP channels 
    • up to 16 Е1 streams (RJ-48)
    • Maximum load intensity 14 cps

     

    Interfaces

    • 2 x 1000Base-X ports (2 slots for SFP modules)
    • 3 x 10/100/1000Base-T (RJ-45) ports
    • E1 (2 x CENTRONICS-36 connectors)
    • 2 SATA ports (for SSD modules installation)  



    Management and monitoring

    • E1 and VoIP channels monitoring in web interface 
    • Management of channels and SS7 links in web interface
    • Alarm logging with the opportunity to save entries to syslog server
    • Tracings are stored on HDD and USB storages 
    • Emergency notification through SNMP  



    Security

    • Black and white IP addresses lists
    • Attempts of access to device are logged
    • Automatic blocking by IP address after unsuccessful login attempts or/and access via http/https/telnet/ssh
    • List of permitted IP addresses for access to control of the device 
    • Access rights delimitation – admin/user
    • Delimitation of rights to access calls records
    • Control of opposite RTP stream’s source IP address
    • Authentication of subscribers on RADIUS server and SIP registar
    • Digest authentication (RFC 5090, Draft-Sterman)
    • Digest authentication in RADIUS (RFC 5090, Draft-Sterman)  



    Redudancy

    • Operation in warm redundancy mode 1+1
    • The system switches the redundant part on automatically
    •  Automatic synchronization of main redundant module settings



    Advanced SIP/SIP-T/SIP-I functionality 

    • Registration and authentication of up to 3000 SIP subscribers 1
    • VAS support for up to 3000 SIP subscribers 1
    • SIP and SIP-T/SIP-I interaction
    • Trunking and subscriber registration of SIP trunks
    • Transit registration of subscribers on SIP trunk with switching to local service mode in case of server unavailability 
       

    Value added services1 

    • Call Forwarding
      • Call forwarding out of service (CFOS)
      • Call forwarding on no reply (CFNR)
      • Call forwarding unconditional (CFU)
      • Call forwarding on busy (CFB)
    • Call Transfer
    • Music on Hold (MOH)
    • Call Hold
    • Call Hunt
    • Call Pickup
    • Busy Lamp Field
    • Conference add-on (CONF)
    • Conference for a list of subscribers
    • 3-Way conference
    • Intercom
    • Paging
    • Outgoing calls restrictions
    • Egress communication by password (RBP)
    • Password activation (PWD ACT)
    • Password reset (PWD)
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