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.