Dinstar SBC300 Session Border Controller
Dinstar SBC300 provides rich SIP-based services such as safe network access, robust security, system interworking, flexible session routing & policy management, QoS, media transcoding and media processing for small-and-medium telecommunication operators. With distributed multi-core processor, rear panel for non-blocking gigabit switching data network, embedded Linux operating system, SBC300 delivers high capability while achieves low power dissipation. It supports 50 concurrent SIP sessions and transcodes up to 50 concurrent calls, and allows encrypted sessions via TLS and SRTP. The session border controller supports transcoding of G.729, G.723, G.711, G.726, iLBC, AMR and OPUS. Besides, it also supports WebRTC, allowing desktop and mobile browsers to initiate and receive calls to/from your SIP service over websocket and WebRTC completely transparently.
- Supports up to 50 SIP sessions and 50 transcoding sessions
- Support WebRTC2SIP to turn WebRTC client into a phone with audio capability
- SIP trunks & flexible routing rules for accessing IMS
- Embedded VoIP firewall, prevention of DoS and DDoS attacks
- Bandwidth limitation and dynamic white list & black list
- QoS, static route, NAT traversal
- Import & export of remote upgrade and configuration data
- User-friendly web interface, multiple management ways
Physical Interfaces Ethernet Ports:
4x 10/100/1000M Base-T Ethernet ports
1x 10/100/1000M Base-T (Admin)
1x USB
1x TF Card Slot
Serial Console
1x RS232, 115200bps, RJ45
Capabilities Concurrent Calls
Supports up to 50 SIP sessions
Transcoding
Supports up to 50 transcoding calls
Registrations
Maximum SIP registrations: 1000
CPS for Registration
20 Registration per second
SIP Trunks
Unlimited SIP Trunks
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,AMR, OPUS
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: DC 12V
Power Consumption: 10W
Operating Temperature. 0 ℃ - 45 ℃
Storage Temperature: -20 ℃ -80 ℃
Humidity: 10%-90% Non-Condensing
Dimensions (W/D/H): 227x147x39mm
Unit Weight: 1.5 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
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
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