Products
DINSTAR  VoIP converter media Gateway SBC1000 Session Border Controller
DINSTAR  VoIP converter media Gateway SBC1000 Session Border Controller
DINSTAR  VoIP converter media Gateway SBC1000 Session Border Controller
DINSTAR  VoIP converter media Gateway SBC1000 Session Border Controller
DINSTAR  VoIP converter media Gateway SBC1000 Session Border Controller
DINSTAR  VoIP converter media Gateway SBC1000 Session Border Controller

DINSTAR VoIP converter media Gateway SBC1000 Session Border Controller

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US $2600
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Quick Details

Place of Origin:
Guangdong, China
Concurrent Calls:
Supports up to 500 SIP sessions
Transcoding calls:
Supports up to 200 transcoding calls
SIP registrations:
5000
Calls Per Second (cps):
25 Registration per second
SIP:
Unlimited SIP Trunks
Firewall:
Embedded VoIP Firewall
Advantage:
Bandwidth limitation & traffic control

Quick Details

Type:
VoIP Gateway
Brand Name:
DINSTAR
Model Number:
SBC1000
Place of Origin:
Guangdong, China
Concurrent Calls:
Supports up to 500 SIP sessions
Transcoding calls:
Supports up to 200 transcoding calls
SIP registrations:
5000
Calls Per Second (cps):
25 Registration per second
SIP:
Unlimited SIP Trunks
Firewall:
Embedded VoIP Firewall
Advantage:
Bandwidth limitation & traffic control
Products Description

Dinstar SBC1000 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, SBC1000 delivers high capability while achieves low power dissipation.

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.

SBC Series
Model Options
Description
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

key features

Supports up to 500 SIP sessions and 200 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

Solution
Product Paramenters

Capabilities

Concurrent Calls Supports up to 500 SIP sessions Transcoding Supports up to 200 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

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

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

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

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
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