Session Border Controller Dinstar SBC1000-100

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Session Border Controller Dinstar SBC1000-100

Session Border Controller Dinstar SBC1000-100

- Supports up to 100 SIP sessions and 100 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.

Specifications

Physical Interfaces Ethernet Ports:
+ 4 x 10/100/1000M Base-T Ethernet ports
+ 1 x 10/100/100M Base-T (Admin)
1 x USB 2.0
Serial Console
+ 1 x RS232, 115200bps, RJ45
Media Capabilities Voice, FAX support
Codecs: G.729, G.723, G.711, iLBC,
G.726, 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: 1+1,100-240VAC, 50-60 Hz
Power Consumption: 15W
Operating Temperature. 0 ℃ ~ 45 ℃
Storage Temperature: -20 ℃ ~80 ℃
Humidity: 10%-90% Non-Condensing
Dimensions (W/D/H): 437*300*44mm
Unit Weight: 4.5 kg
Capabilities Concurrent Calls: Supports up to 100 SIP sessions
Transcoding: Supports up to 100 transcoding calls
CPS for Call: 25 calls per second at maximum
Registrations: Maximum SIP registrations: 5000
CPS for Registration: 25 Registration per second
SIP Trunks: Unlimited SIP Trunks
Security Prevention of DoS and DDoS attacks
Control of access policies
Policy-based anti-attacks
Call Security with TLS
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.

Giá: Vui lòng gọi

Số lượng :

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