- Product Details
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Quick Details
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Place of Origin:
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Guangdong, China
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Concurrent Calls:
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Up to 10,000 SIP sessions
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Transcoding calls:
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Supports 5,000 transcoding calls
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SIP registrations:
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Up to 100,000 SIP registrations
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Calls Per Second (cps):
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800 Registration per second
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SIP:
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Unlimited SIP Trunks
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Firewall:
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Embedded VoIP Firewall
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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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Model Number:
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SBC8000
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Place of Origin:
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Guangdong, China
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Concurrent Calls:
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Up to 10,000 SIP sessions
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Transcoding calls:
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Supports 5,000 transcoding calls
-
SIP registrations:
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Up to 100,000 SIP registrations
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Calls Per Second (cps):
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800 Registration per second
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SIP:
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Unlimited SIP Trunks
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Firewall:
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Embedded VoIP Firewall
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Advantage:
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Bandwidth limitation & traffic control
Products Description
A Session Border Controller(SBC) is a SIP/IP network core element that deployed for next generation networks (NGN) to support peer/access connection by SIP/IMS trunk and provide interworking between incompatible signaling messages or media flow from endpoint devices and SIP applications. In particular SBC supports SIP/IMS network connection, sip endpoint remote registration, NAT traversal, signaling modification, media control, flexible call routing, codec transcoding, billing and QoS management.
It supports 500 concurrent SIP sessions and transcodes up to 200 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.
DINSTAR SBC8000 is a new software-based SBC. It provides very rich features for medium and large enterprise SIP network connection and supports various security mechanisms, call routing based on control policies, carrier-grade high availability(HA) and debug tools for service providers and telecom operators.
SBC8000 can be installed on VM/cloud-based system or third-party hardware appliances. Users can install SBC8000 in X86, ARM architecture and any cloud-based platforms for many different SIP implementations with advantages of VM and cloud platforms.
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SBC Series
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Model Options
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Description
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SBC300
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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 |
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SBC1000
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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 |
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SBC3000
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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 |
key features
Up to 10,000 concurrent call sessions, 5,000 media transcoding and 100,000 SIP registrations Operate on physical server, virtual machine and public cloud Intelligent bandwidth limit and dynamic blacklist Cross-network, NAT traversal and high availability(HA) SIP over TLS, SRTP Compatible with different codecs: G.711A/U, G.723.1, G.729A/B, iLBC, AMR, OPUS Flexible call routing Broad compatibility with IMS network VoIP firewall, anti-attacks and core network protection Call recording
Solution
Product Paramenters
Capabilities
Concurrent Calls
Up to 10,000 SIP sessions
Transcoding
Supports 5,000 transcoding calls
CPS for Call
800 calls per second at maximum
Registrations
Up to 100,000 SIP registrations
CPS for Registration
800 Registration per second
VoIP
SIP 2.0 Compliant, UDP, TCP, TLS,
SIP Trunk (Peer to peer)
SIP Trunk (Access)
SIP Proxy Registrations: Up to 3,000
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 Header Manipulation
SIP Malformed Packet Protection
Multiple Soft-switches Supported
QoS (ToS, DSCP)
NAT Traversal
Media Capabilities
Codecs：G.711a/μ, G.723, G.729A/B, iLBC, G.726, AMR, OPUS
Silence Suppression
Voice Activity Detection(VAD)
Comfort Noise Generator(CNG)
Echo Cancellation: G.168 with up to 128ms
RTP/RTCP
Voice Interrupt Protection
Adaptive Dynamic Buffer
Adjustable Gain Control
Automatic Gain Control (AGC)
FAX: T.38, Pass-through
DTMF: RFC2833/Signal/Inband
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
NTP
Multiple languages support
SNMP
Remote Web and Telnet
Security
Prevention of DoS and DDoS Attacks
Control of Access Policies
Policy-based Anti-attacks
Message format detection and processing
UDP-Flood Anti-attacks
TCP-Flood Anti-attacks
Call Security with TLS/SRTP
Whitelist & Blacklist
Access Control List
Built-in VoIP Firewall
Call Control
Dynamic Load Balancing and Call Routing
Flexible Routing Engine
Routing Based on Caller/Called Prefixes
Regular Express
Call Routing Base on Time Profile
Call Routing Base on SIP URI
Call Routing Base on SIP Method
Caller/ Called Number Manipulation
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