Session Border Controller Dinstar SBC3000-500

- Supports up to 500 concurrent SIP sessions, 300 transcoding.

- Support WebRTC2SIP to turn WebRTC client into a phone with audio capability.

- Triple anti-attacking, embedded VoIP firewall, prevention of DoS and DDoS attacks.

- Bandwidth limitation and dynamic white list & black list.

- VLAN, IPSec, QoS, static route, NAT traversal.

- Import & export of remote upgrade and configuration data.

- User-friendly web interface, multiple management ways.

- 1+1 high availability with two SBCs, dual hot-swappable power supplies.

Specifications

Physical Interfaces MCU (Main Control Unit): 1

MFU (Main Function Unit): 4

Ethernet Ports: 4* 10/100/1000M Base-T Ethernet ports

1* USB 2.0

Serial Console: 1* RS232, 115200bps, RJ45 each MCU

Power Supply: 2

Media Capabilities Voice, FAX support

Codecs:G.729, G.723, G.711, iLBC,

OPUS, G.726, AMR

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: AC 100-240V, 50-60Hz

Power Consumption: 70W

Operating Temperature. 0 ℃ ~ 45 ℃

Storage Temperature: -20 ℃ ~80 ℃

Humidity: 10%-90% Non-Condensing

Dimensions (W/D/H): 437×320×44mm

Unit Weight: 6 kg

Capabilities Concurrent Calls: Supports 500 SIP sessions at maximum

Transcoding: Supports 300 transcoding calls

CPS for Call: 200 calls per second at maximum

Registrations: Maximum SIP registrations: 10000

CPS for Registration: 200 Registration per second

SIP Trunk: Unlimited SIP Trunks

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

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.