Digital E1/T1 VoIP Media Gateway

SMG3000 OVERVIEW

E1/T1 Ports Digital VoIP Media Gateway For Enterprises, Service Providers and Operators

E1 Gateway Overview

SMG3000-C1LS

SMG3000-B2L

SMG3000-B1

SMG3000-B2

SMG3000-B4

SMG3008(SS7&ISDN)

SMG3016(SS7&ISDN)

SMG3032(SS7&ISDN)

SMG3064(SS7&ISDN)

HomeE1 Gateway Overview

Carrier-Grade Resilience
SS7/ISDN/R2/SIP Signaling
Telecom Reliability
SBC Add-on Functionality

E1/T1 Digital VoIP Gateway Maximizes Efficiency for Carriers and SPs

With a more efficient architecture designed for E1/T1 Digital VoIP Gateway solutions, the SMG3000 empowers carrier and service providers to interconnect and deliver small-to-high scalability VoIP systems, which have been adopted by over 1,000 enterprises and service providers and carriers' worldwide.As a wide range of digital VoIP gateways, the SMG3000 converts digital PSTN messages into SIP formats, ensuring seamless session connectivity and security across IP and mixed network boundaries and making it an ideal choice for organizations that want to bridge traditional and modern VoIP networks.

Unparalleled Cost Efficiency and Robust Telco' Architecture For SPs

Compared to rival E1/T1 Digital VoIP Gateway products on marketplaces, the SMG3000 stands out with Telco-grade high reliability and unparalleled cost efficiency—making it a perfect alternative for telecom operators and service providers (SPs). Ensuring stable performance even in high-traffic environments, addressing the critical needs of businesses relying on robust VoIP connectivity, and equipped with complete signaling packets and a full suite of media processing capabilities, The E1/T1 Digital VoIP Gateway SMG3000 delivers a range of differentiated values while maintaining cost-effectiveness that rivals can’t match.

Unparalleled Cost Efficiency and Robust Telco' Architecture For SPs

Compared to rival E1/T1 Digital VoIP Gateway products on marketplaces, the SMG3000 stands out with Telco-grade high reliability and unparalleled cost efficiency—making it a perfect alternative for telecom operators and service providers (SPs). Ensuring stable performance even in high-traffic environments, addressing the critical needs of businesses relying on robust VoIP connectivity, and equipped with complete signaling packets and a full suite of media processing capabilities, The E1/T1 Digital VoIP Gateway SMG3000 delivers a range of differentiated values while maintaining cost-effectiveness that rivals can’t match.

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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	SMG3000 Series 1~64E1/T1 and 30~1,920 SIP channels, VoIP Multimedia Trunking Gateway Routing: Call routing and translation (from PCM to IP or reversely)
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)
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
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

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Digital E1/T1 VoIP Media Gateway

SMG3000 OVERVIEW

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