VoIP Trunk Gateway SMG-3016 with IP-PBX support

Mass production

Key benefits

Scalable platform 1U
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 streams (RJ-48)
Support for 2 HDD SATA 2.5"
Hardware redundancy

Hybrid platform SMG-3016 is used as a trunk gateway for reciprocation of signal and media flows of TDM and VoIP networks. The gateway also might be used as an IP PBX with value added services (VAS) support and a multipurpose solution for infocommunication new generation networks (NGN). The wide function-set, strict compliance with require-ments and standards and carrier class reliability allow service providers and carriers to solve most part of their objectives on the basis of SMG-3016.

Scalability
SMG-3016 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.

IP PBX with VAS support
Additional options for SMG-3016 gateway allow using it as a full-featured IP PBX for up to 3000 SIP subscribers with support for a wide range of value added services. A programmable IP PBX module ECSS-10 is designed for fast deployment of a VoIP node with a minimum of capital expenses. ECSS-10 and SMG-3016 might be used as a PBX of any level.

Functional compatibility
The strict compliance with up-to-date protocols’ requirements, recommendations and standards provides functional compatibility with a variety of equipment: digital PBX, IP PBX, Softswitch, VoIP gateways, SIP phones, software SIP clients, etc.

Carrier class reliability
SMG-3016 provides high level of fault tolerance, uniform load distribution among submodules, usage of up-to-date technologies based on parallel computing and power modules redundancy. In case of a primary submodule fault, the gateway switches to a backup submodule.

Media streams transcoding
The hardware transcoding helps to negotiate media streams with different VoIP codecs which are used in up-to-date networks.

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-3016 (Dynamic Firewall, Static Firewall, black and white lists of IP addresses and subnetworks, etc.). For additional defense, SMG-3016 may be used in coupling with with session border controller (e.g. SBC-1000) that is used as a firewall for VoIP networks.

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

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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 settings
Trunk group cut-off
Call management via RADIUS¹
Direct forwarding for trunk groups
Prefix for several trunk groups
Interactive voice response (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

DTMF

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

Billing

Billing data is recorded as a CDR file that is kept on a local HDD and remote FTP server simultaneously.
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 LE²
V5.2 AN²

VoIP protocols

SIP, SIP-T/SIP-I, SIP-Q
H.323¹
SIGTRAN (M2UA, IUA)²
H.248²

Capacity and performance

Up to 768 VoIP channels
Up to 16 E1 streams (RJ-48)
Maximum load intensity — 120 cps
RAM2GB

Interfaces

16 E1 ports (RJ-48)
2 ports of 10/100/1000Base-T (RJ-45) / 1000Base-X (SFP)
2 ports of 10/100/1000Base-T (RJ-45) ports
2 USB2.0 ports
2 slots for SATA HDD 2,5'’

Management and monitoring

E1 and VoIP channels monitoring via web interface
Channels and SS7 links management via 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
Console port RS-232 (RJ-45)
Allocated management port (OOB) 10/100/1000BASE-T (RJ-45)

Security

Black and white IP addresses lists
Attempts of accessing the device are logged
Automatic blocking by an IP address after unsuccessful login or/and access attempts via http/https/telnet/ssh
List of permitted IP addresses for access to control the device
Access rights delimitation – admin/user
Delimitation of rights to access calls records
Control for 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)

Advanced SIP/SIP-T/SIP-I functionality

Registration and authentication of up to 3000 SIP subscribers¹
VAS support for up to 3000 SIP subscribers¹
SIP and SIP-T/SIP-I interaction
Trunking and subscriber registration of SIP trunks
Transit registration of subscribers on SIP trunk with switching to a local servicing in case of server unavailability

Redundancy²

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

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

¹Оptional

²Not supported in the current firmware version — 3.15.0

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Descriptions

Datasheet

Downloads

Operational lifetime of the ELTEX equipment

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.