- up to 2 Е1 flows (RJ-48)
- up to 64 VoIP channels
- high quality voice
- carrier-class reliability
- desktop case
The SMG-2 platform can be used as a trunk gateway for reciprocation of signal and media flows of TDM and VoIP networks.
Optimal Solution for Digital PBX
SMG-2 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.
Strict conformity to the requirements of up-to-date protocols, recommendations and standards ensure 100% functional compatibility of SMG-2 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, IP-tables, white/blacklists, etc.) has been implemented in trunk gateway SMG-2.
Media Flows Transcoding
The hardware transcoding based on Media Codecs Mindspeed Technologies helps coordinate media flows with different VoIP codecs.
High Quality of Voice processing
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.
The SMG-2 trunk gateway has 2 ports RJ-48 for E1 stream connection, 1 10/100/1000Base-T LAN port (RJ-45) for connection to an IP network.
VOIP TRUNK GATEWAY SMG-2
- Called party (CdPN) or calling party (CgPN) number routing
- Before and after routing number modification
- Use of multiple numbering plans
- Trunk-group cutoff
- Call Control via RADIUS¹
- Trunk-groups direct connection
- Prefix for few Trank-groups
- G.711 (a-law, µ-law), G.729 (A/B), G.723.1, G.726 (32 Kbps)
- T.38 Real-Time Fax, G.711 (a-law, µ-law) pass-through
- 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 record to CDR file and sending to FTP-remote server
- PRI (Q.931)
- SIP, SIP-T/SIP-I
Capacity and Performance
- 64 VoIP channels
up to 2 E1 flows (RJ-48)
Maximum load intensity - 40 cps
- Single file download-upload of configuration
- Multiple network interfaces creation for (SIP, RTP) telephony with different IP-addresses
- Numbering 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
- 1 x E1 (RJ-48) port
- 1 x add-on port Е1 (RJ-48)¹
- 1 x Console (RJ-45) port
- 1 x USB 2.0 port