DIGITAL VOIP MEDIA GATEWAY
SMG3000 OVERVIEW
E1/T1 Ports Digital VoIP Media Gateway For Enterprises, Service Providers and Operators
E1/T1 Ports Digital VoIP Media Gateway For Enterprises, Service Providers and Operators
With a more efficient manner to interconnect and deliver small-to-High scalability VoIP systems, SMG3000 enables small and medium enterprises to maximize value of their networks and services. Adopted by over 1,000 enterprises and service providers worldwide, these mini-sized VoIP digital 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.
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, SMG2000 series could deliver a range of values for your differentiation.
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, SMG2000 series could deliver a range of values for your differentiation.
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;
Product models | SMG3000-B1L 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 |
Voice & Faxing Capability |
Default codec:G.711a/μ law, AMR, G.723.1, G.729AB, iLBC (Licensable) |
SDK Features |
Local/Transparent Ring Back Tone(self define or upload) |
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 |