Dinstar SBC3000 Pro Session Border Controller
SBC3000 Pro delivers secure SIP services, including flexible session routing, QoS, media transcoding, and robust security for small-to-medium telecom operators. Featuring a multi-core processor and non-blocking gigabit switch, it ensures high performance with low power usage. It supports dual hot-swappable power supplies, TLS/SRTP encryption, and a wide range of codecs, including G.729, AMR, and OPUS, offering unmatched reliability and flexibility.
- Supports up to 5000 concurrent SIP sessions
- Support WebRTC2SIP to turn WebRTC client into a phone with audio capability
- Support transcoding of 1500 calls on media and fax
- 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
Physical Interfaces MCU (Main Control Unit): 1
MFU (Main Function Unit): 4
Ethernet Ports:
8x 10/100/1000M Base-T Ethernet ports
Serial Console
1x USB Port
Capabilities Concurrent Calls:
Supports 5000 SIP sessions at maximum
Transcoding:
Supports 1500 transcoding calls
CPS for Call:
500 calls per second at maximum
Registrations:
Maximum SIP registrations: 20000
CPS for Registration:
500 Registration per second
SIP Trunk:
Maximum of 1024 SIP trunks
SIPREC (Concurrency):
Supports up to 2000 recording calls
SRTP (Concurrency):
Supports Up to 3000 encrypted calls
WebRTC (Voice):
Up to 2000 concurrent calls
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
Media Capabilities Voice, FAX support
Codecs: G.729, G.723, G.711, iLBC, OPUS, G.722, 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
SIP Recording (SIPREC)
Comfort Noise
Voice Activity Detection
Echo Cancellation
Adaptive Dynamic Buffer
Environmental Dual Power Supply: AC 100-240V, 50/60Hz
Power Consumption: 70W
Operating Temperature: 0 C - 45 C
Storage Temperature: -20 C -80 C
Humidity: 10%-90% Non-Condensing
Dimensions (W/D/H): 437x320x44mm
Unit Weight: 6 kg
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
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
Port Mapping
Network port binding
Static Routing
Remote capture
Signaling tracking
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
TLS mutual authentication
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