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DIGITAL VOIP MEDIA GATEWAY

SMG3008(SS7&SIP)

SMG3008

8 E1/T1 240 Ports Digital VoIP Media Gateway With Telco-grade Reliability & Versatility

  • DSP Voice Processing
  • Compact 1U Footprint
  • Telecom Reliability
  • Carrier-Grade Resilience
Full Compliance with Carrier Networks

With a more efficient manner to interconnect and deliver Telco-grade VoIP systems, SMG3008 enables service providers and carriers to maximize value of their networks and services. Adopted by over 50 well-known carriers worldwide, these hi-density SS7 VoIP gateways could convert digital PSTN message into SIP formats, and connects/secures sessions across IP and mixed network boundaries to support the seamless delivery of services.

Telco-Grade Reliability and Rich Functionality

Compared to rival products on marketplaces, SMG3000 features Telco’s high reliability and unparalleled cost efficiency, providing a perfect alternative for telecom operators and SPs. With complete signaling packets and full suite of media processing capability, SMG3000 series could deliver a range of values for your differentiation.

Telco-Grade Reliability and Rich Functionality

Compared to rival products on marketplaces, SMG3000 features Telco’s high reliability and unparalleled cost efficiency, providing a perfect alternative for telecom operators and SPs. With complete signaling packets and full suite of media processing capability, SMG3000 series could deliver a range of values for your differentiation.

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Features
  • Build-in SS7, SIGTRAN, SIP, Mecaco
Complimentary standardized SS7 Packets(Up to 64 SS7 links, ISUP, MTP1~3, TCAP), SIGTRAN, Megaco and SIP for any carrier networks worldwide; Field-proven by over 50 largest operators from European, China, USA, India, Brazil, South America;

  • 99.9999% Availability in Full Load
Support 4/8/16/64E1 (120/240/480/1,920Chs) per unit, with higher system responsiveness capability in extreme network for better operational results; High performance and stability in cases of unstable (low) bandwidth and high capacity;

  • Dedicated DSP-Empowered Capability
Telecom-style DSP algorithm has been optimized for over decades, assuring seamless compliance with any network environment. Plentiful DSP resources are allocated for signaling, media processing, bandwidth optimization, Telco redundancy;  

  • Highly Adjustable For Diverse SoftSwitch
Homegrown core technologies to assure seamless compliance with diverse softswitch platform, including Mitel, Avaya, Broadsoft, Yate, OpenSIP, Asterisk, VECTRA, VSC, SIPPULSE, Tropico, FreeSwitch and more other softswitch;

  • Integrated Transcoding
Eliminates the need to add separate hardware to support transcoding requirements helping to reduce CAPEX and number of platforms deployed, and support a range of Codecs, including G.723, G.729, G.711, iLBC, SIPINFO, RFC2833, RF3261, INBOUND and more.
Capabilities
  • Rich TDM/SIP Signaling  
Provide any-to-any network connectivity through its ability to interwork multiple protocols to deliver services. It can provide interworking between SS1, R2, ISDN, SS7, SIGTRAN, SIP formats and any-to-any media transcoding for popular voice codecs, T.38 and G.711 fax interworking, RTP, INBAND and SIPINFO;

  • Flexible and efficient Gateway Solution
Provide a clear migration path to an all-IP network. It can scale up to 1,920 simultaneous IP sessions and provide impressive sessions per second performance; Support SS7 signaling, call routing, call automated failover from IP to TDM for outbound routing;

  • Robust Telco-Grade Architecture
Dual power supply, 1U compact structure, industrial heating sink system, dual 1,000M Ethernet interfaces, and Telco’ grade hardware architecture ensure 200,000Hr MTBF and 0-defective rate in long lifecycle ;

  • User-friendly WEB Management

Adopt straight-forward configuration to achieve SPs and Operators’ sophisticated objectives. Over 1,000 units of SMG3000 can run together; The Web graphical user interface (WebUI) toolkit performs real-time monitoring and maintenance, and helps configure SIP, SIP trunking, SIP Mediation, PCM, SS7, ISDN, Routing and more;


  • Flexible Scalability
From 30 to 1,920simultaneous SIP sessions with multimedia transcoding provides high performance in a small footprint to help lower ownership cost and operational cost; a broad range of gateway toolkits help gateway’s maintenance and software upgrading for Web UI, gateway services and firmware;
Core Specifications
Product models SMG3008  8 E1/T1 and 240 SIP channels, VoIP Multimedia Trunking Gateway
Routing: Call routing and translation (from PCM to IP or reversely)
Processing Capability Eight-core CPU, 1GB RAM, 4GB ROM
To process multimedia and signaling
I/O Interfaces Rear I/O — T1/E1
Telephony: Fiber Optical 1~2 T1/E1 for timing (BITS clock),  
T1 and E1: signaling and bearer traffic, (T1 - 100 ohms and E1 - 120 ohms)
Clock Sync: Stratum
Physical Interface RJ48(Impedance 120Ω)
1* RS232, 115200bps
SDN PRI 23B+D(T1),30B+D(E1),NT or TE ITU-T Q.921, ITU-T Q.931, Q.Sig
IP Interfaces: Dual redundant 2 *100 Base-T Ethernet for VoIP payload and signaling
IP & PSTN Bearer

Compliant with TLS/SRTP, TCP/UDP, HTTP, ARP/RARP, DNS, NTP, TFTP, TELNET, STUN and more IP protocols
Echo cancellation: G.168 128 ms tail length
VLAN 802.1p/q  
Digit transmission via RFC 2833 (SIP)
Hosted NAT
Voice gain output from IP
R2 MFC and other 22 variant standards
E1 Frame Type : DF,CRC-4,CRC_ITU
T1 Frame Type :4-Frame Multi-frame (F4,FT);2-Frame Multi-frame (F12, D3/4);Extended Super- frame (F24, ESF) ;Remote Switch Mode (F72, SLC96); Line Codes: E1:NRZ,CMI,AMI,HDB3;T1:NRZ,CMI,AMI,B8ZS

Voice & Faxing Capability

Default codec:G.711a/μ law, AMR, G.723.1, G.729AB, iLBC (Licensable)
Silence Suppression
Comfort Noise
Voice Activity Detection and packet loss concealment
Echo Cancellation (G.168),with up to 128ms
Adaptive Dynamic Buffer and Static buffer
Voice ,Fax Gain Control
FAX:T.38 and Pass-through
DTMF Mode: RFC2833/SIP Info/In-band
Busy Tone Detection
Provide Ringback Tone
PCM Voice Gain
PSTN Call Forwarding        
Hot Back-up for E1
UUI Protocol Discriminator

SDK Features

Local/Transparent Ring Back Tone(self define or upload)
Dialing Rules, with up to 200
Voice Codecs Group
Access Rule Lists      T
ELNET, WEB, FTP,SSH
128 SIP Trunks   
Route direction: PSTN-IP, IP-PSTN, PSTN-PSTN , and IP-IP(SBC-enabled)
Working Period define
CDR
DHCP SERVER

Maintenance (OAM&P) Web GUI Configuration, HTTP/HTTPS
NTP Synchronization
Data Backup/Restore
PSTN Call Statistics
SIP Trunk Call Statistics
Firmware Upgrade via TFTP/Web
SNMP v1/v2/v3
Network Capture
Syslog: Debug, Info, Error, Warning , Notice
Call History Records via Syslog
Centralized Management System
IDS, DDOS( only ip to ip feature enable)
Call test
Ping, tracert.
User management
Radius
Firewall(IP tables)
Resiliency Smart IP probing
Automated failover (Ethernet and Fiber Optical links)
Failover via automatic protection switching
VoIP Protocols Core SIP Specifications and Notable Extensions
Notable SIP Extensions
RFC 3398 ISUP/SIP Mapping
RFC 3711 SRTP (for SIP)
Tel URI – RFC 3966
IP and ISUP interworking and more
SIP v2.0 (UDP/TCP), RFC3261 SDP, RTP(RFC2833), RFC3262, 3263, 3264, 3265, 3515, 2976, 3311
RTP/RTCP, RFC2198, 1889
TLS/SRTP
SIP Trunk Work Mode :Peer/Access
SIP/IMS Registration :with up to 256 SIP Accounts
NAT: Dynamic NAT, Rport
Different method to obtain callerid and calleeid
PAI PPI
RTP Self-adaption, RTP TIMEOUT
TOS DSCP
SIP Trunk Heart
Self definition To Field in INVITE Message
Whilelist blacklist, number pool filtering rule self define
PSTN Signaling Protocols ISDN PRI
MF R2
SS7 ISUP(Optional)
SS7 MTP1~3(Optional)
SS7 SIGTRAN(Optional)
SS7 TCAP(Optional)
QoS Adaptive jitter buffer
Packet loss compensation
Configurable Type of Service (ToS) fields for packet prioritization and routing
Environmental Requirement Operating Temperature:0 ℃ ~ 55 ℃
Storage Temperature: -20 ℃ ~85 ℃
Humidity:10%-90% Non-Condensing
Power Requirements AC Power Supply Range 100 – 240 VAC
The power supply will operate at frequencies between 47 Hz and 63 Hz
Power Consumption: About 15W(Normal Conditions)
Compliant Power Adapter: 100-240VAC@DC12V1A
Physical Dimensions High: 1.72 in (44 mm)
Wide: 17.32in (440 mm)
Deep: 10.51in (267 mm)
Weight 6.8 lb (Approx. 3.1kg)
Approvals & Compliance About RoHS compliance and other approvals, please contact Synway directly
CE, FCC or Any other Certificates Customizable
EMC/EMI: Compliant with most international standards
Safety: Compliant with most international standards
Telecom Approvals: Compliant with most international standards
Reliability & Warranty Estimated MTBF per Telcordia Method 1
Rear I/O Type 1 — T1/E1
NO PSTN Interface: 150,000 hours
3-year warranty: The first year exchange for free. 2nd & 3rd cost-based repair
Broadsoft, Elastix, Asterisk, Teams and other UC platforms compliant
Resources
Related Products
SMG3000-B2L
SMG3000-B1L
SMG3000-B1
Success Stories & Applications
Synway VoIP Gateways Enable Smooth Integration between Legacy TDM and IP Contact Center Platforms
ChinaUnionPay selected Synway' SMG series VoIP Gateway to connect its TDM and IP-based contact center operations to the PSTN.
Global BPO Provider Huayun Data Chose Synway SMG Gateway for Multiple Offshore Service Center
Founded in 1998 and headquartered in USA, Huayun Data, a leading provider of global Business Process Outsourcing (BPO) services, deployed Synway SMG2000 media gateways at its locations in the US to enable a one-stop, cost-effective and reliable VoIP connections between its affiliated agencies.......
ChinaUnionPay selected Synway's VoIP Gateway to connect its TDM and IP-based contact center operations
Asterisk based IP-PBXs are becoming more available and popular on the PBX business market, threatening the traditional proprietary PBX manufacturers position continuously.
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