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Designer and manufacturer of communication equipment
Russia (GMT +7)

VoIP Trunk Gateway SMG-2016 with IP-PBX support

  • Scalable platform 1U
  • Quad-Core ARMv7 Marvell Armada-XP Processor 
  • IP-PBX for 3 000 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 HDD SATA 2.5" 
  • Hardware redudancy

Hybrid gateway SMG-2016  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 on the basis of SMG-2016.

Scalability  
SMG-2016  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 768 VoIP channels.

Carrier class reliability   
SMG-2016 provides high level of fault tolerance due to embedded state-of-the-art Quad-Core ARMv7 Marvell Armada-XP processor, uniform load distribution among
submodules, power modules redundancy and usage of up-todate 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 to negotiate media streams with different VoIP codecs which are used in up-to-date networks.

IP-PBX with VAS support 
Additional options for SMG-2016 gateway allow using it as a full-featured IP PBX with up to 3000 SIP subscribers connection and support of 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-2016 might be used as a PBX of any level.

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-2016 (Dynamic Firewall, Static Firewall, black and white lists of IP addresses and subnetworks, etc.). For additional defense, SMG-2016 is compatible with session border controllers (e.g. SBC-1000) that are used as a firewall for VoIP networks.

RADIUS routing 
Intellectual call routing based on the billing system responses via the RADIUS protocol allows customers to create flexible methods of call processing. 

Specifications
Documents and files
Warranty
Calls management 
  • Interaction with STUN-server on the SIP interface 
  • Routing based on called number (CdPN) or calling number (CgPN) based
    routing
  • Number modifications before and after routing 
  • Call recording according to number mask and dialplan
  • Use of multiple dialplans  
  • Subscriber lines restriction 
  • Subscriber service mode settings
  • Trunk group cut-off
  • Call management via RADIUS
  • Direct connection of trunk groups 
  • Prefix for several trunk groups 
  • Interactive voice response (IVR)
  • 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 dialplans 
  • Signal SS7 channel redundancy
  • Voice activity control (by the presence of RTP or RTCP) 
  • Individual routing for streams of a single SS7 linkset 

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

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 - 120 cps
  • Quad-Core ARMv7 based Marvell Armada-XP 1.6 GHz
  • RAM 4 GB 

Interfaces
  • 16 x E1 ports (RJ-48)
  • 2 x 10/100/1000Base-T (RJ-45) / 1000Base-X (SFP) ports 
  • 2 x 10/100/1000Base-T (RJ-45) ports
  • 2 slots for SATA HDD 2,5'’  

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)
  • Do not disturb
  • Blacklist


Оptional
  Current firmware version 3.14.0

Operational lifetime of the ELTEX equipment
In development
1
Pre-production
2
Mass production
3
Mass production is over
4
Sold out
5
Support is over
6
Regardless of the operational lifetime stage, Eltex provides a 12 months warranty on all its telecommunication equipment.
During the warranty period the manufacturer ensures technical support and free-of-charge repair at the Enterprise which is situated in Novosibirsk.
As part of the warranty service, technical support is provided on the first-in first-out principle.
The priority support packages of 8/5 and 27/7 types are subjects to additional charges.