- Product Details
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Quick Details
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Concurrent Calls:
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5 to 50 Simultaneous calls
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Transcoding calls:
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5 to 50 transcoding calls
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SIP registrations:
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1000
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Calls Per Second (cps):
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20
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SIP:
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UDP/TCP/TLS
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SIP trunks:
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Unlimited SIP trunks
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Prevention attacks:
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Prevention of DoS and DDos attacks
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Firewall:
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Embedded VoIP Firewall
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Security:
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TLS / SRTP security
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Advantage:
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Bandwidth limitation & traffic control
Quick Details
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Type:
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VoIP Gateway
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Brand Name:
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DINSTAR
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Place of Origin:
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Guangdong, China
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Concurrent Calls:
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5 to 50 Simultaneous calls
-
Transcoding calls:
-
5 to 50 transcoding calls
-
SIP registrations:
-
1000
-
Calls Per Second (cps):
-
20
-
SIP:
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UDP/TCP/TLS
-
SIP trunks:
-
Unlimited SIP trunks
-
Prevention attacks:
-
Prevention of DoS and DDos attacks
-
Firewall:
-
Embedded VoIP Firewall
-
Security:
-
TLS / SRTP security
-
Advantage:
-
Bandwidth limitation & traffic control
Products Description
SBC300
SESSION BORDER CONTROLLER
Dinstar SBC300 are designed to deliver security, interoperability and
transcoding between SMB and service providers' VoIP networks. SBC300
help SME easily access to service providers' SIP trunks / telecom operators
IMS with high-level security,meanwhile perform SIP mediation and audio
transcoding. Scaling from 5 to 50 SIP sessions, SBC300 always meet SME
demands today and in the future with only small investment.
SBC Series
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Model Options
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Description
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SBC300
|
SBC300-05
SBC300-10 SBC300-20 SBC300-50 |
5 calls, 5 transcoding, Registered Users 1000
10 calls, 10 transcoding, Registered Users 1000 20 calls, 20 transcoding, Registered Users 1000 50 calls, 50 transcoding, Registered Users 1000 |
SBC1000
|
SBC1000-50
SBC1000-100 SBC1000-200 SBC1000-300 SBC1000-500 |
50 calls, 50 transcoding, Registered Users 5000
100 calls, 100 transcoding, Registered Users 5000 200calls, 200 transcoding, Registered Users 5000 300calls, 200 transcoding, Registered Users 5000 500 calls, 200 transcoding, Registered Users 5000 |
SBC3000
|
SBC3000-500
SBC3000-1000 SBC3000-1500 SBC3000-2000 |
500 calls, 300 transcoding, Registered Users 10000
1000 calls, 600 transcoding, Registered Users 10000 1500 calls, 900 transcoding, Registered Users 10000 2000 calls, 1200 transcoding, Registered Users 10000 |
Front
Back
key features
* Supports up to 50 SIP sessions and 50 transcoding sessions
* 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
* Encrypted sessions
* User-friendly web interface, multiple management ways
Solution
Product Paramenters
Capabilities
4* 10/100/1000M Base-T
Ethernet ports
1*10/100/1000M Base-T (Admin)
1* USB
1* TF Card Slot
Serial Console:1* RS232, 115200bps, RJ45
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
1* USB
1* TF Card Slot
Serial Console:1* RS232, 115200bps, RJ45
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)
* 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
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
Codecs: G.729, G.723, G.711, iLBC
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
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
* 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
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
* 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
* 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
Why Choose Us
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