Session Border Controller Dinstar SBC3000-2000
– Supports up to 2000 concurrent SIP sessions, 1200 transcoding.
– Support WebRTC2SIP to turn WebRTC client into a phone with audio capability.
– Triple anti-attacking, embedded VoIP firewall, prevention of DoS and DDoS attacks.
– Bandwidth limitation and dynamic white list & black list.
– VLAN, IPSec, QoS, static route, NAT traversal.
– Import & export of remote upgrade and configuration data.
– User-friendly web interface, multiple management ways.
– 1+1 high availability with two SBCs, dual hot-swappable power supplies.
Specifications
| Physical Interfaces | MCU (Main Control Unit): 1 MFU (Main Function Unit): 4 Ethernet Ports: 4* 10/100/1000M Base-T Ethernet ports 1* USB 2.0 Serial Console: 1* RS232, 115200bps, RJ45 each MCU Power Supply: 2 |
| Media Capabilities | Voice, FAX support Codecs:G.729, G.723, G.711, iLBC, OPUS, G.726, AMR RTP Transcoding Pass-through fax No RTP detection One-way audio detection RTP/RTCP, SRTP RTCP statistics reports DTMF: RFC2833, SIP Info, INBAND Silence Suppression Comfort Noise Voice Activity Detection Echo Cancellation Adaptive Dynamic Buffer |
| Environmental | Power Supply: AC 100-240V, 50-60Hz Power Consumption: 70W Operating Temperature. 0 ℃ ~ 45 ℃ Storage Temperature: -20 ℃ ~80 ℃ Humidity: 10%-90% Non-Condensing Dimensions (W/D/H): 437×320×44mm Unit Weight: 6 kg |
| Capabilities | Concurrent Calls: Supports 2000 SIP sessions at maximum Transcoding: Supports 1200 transcoding calls CPS for Call: 200 calls per second at maximum Registrations: Maximum SIP registrations: 10000 CPS for Registration: 200 Registration per second SIP Trunk: Unlimited SIP Trunks |
| Security | Prevention of DoS and DDoS attacks Control of access policies Policy-based anti-attacks Call Security with TLS/SRTP White List & Black List Access Rule List Embedded VoIP Firewall |
| Call Control | Dynamic load balancing and call routing Flexible Routing Engine Call routing base on prefixes Call routing base on caller/called number regular express Call routing base on time profile Call routing base on SIP URI Call routing base on SIP method Call routing base on endpoint Caller/ Called number Manipulation |
| VoIP | SIP 2.0 compliant, UDP, TCP, TLS SIP trunk (Peer to peer) SIP trunk (Access) SIP Registrations B2BUA (Back-to-Back User Agent) SIP Request rate limiting SIP registration rate limiting SIP registration scan attack detection SIP call scan attack detection SIP anti-attack SIP Header manipulation SIP malformed packet protection Multiple Soft-switches supported QoS (ToS, DSCP) NAT Traversal |
| Maintenance | Web-bases GUI for Configurations Configuration Restore/Backup HTTP Firmware Upgrade CDR Report and Export Ping and Tracert Network Capture System log Statistics and Reports Multiple language support Centralized management system Remote Web and Telnet |
– Bảo hành: 12 tháng.
