Session Border Controller Dinstar SBC300-20

- Supports up to 20 SIP sessions and 20 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 4 x 10/100/1000M Base-T Ethernet ports

1 x 10/100/1000M Base-T (Admin)

1x USB

1x TF Card Slot

Serial Console

1x RS232, 115200bps, RJ45

Media Capabilities Voice, FAX support

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

iLBC,AMR, 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: DC 12V

Power Consumption: 10W

Operating Temperature. 0 ℃ ~ 45 ℃

Storage Temperature: -20 ℃ ~80 ℃

Humidity: 10%-90% Non-Condensing

Dimensions (W/D/H): 227×147×39mm

Unit Weight: 1.5 kg

Capabilities Concurrent Calls: Supports up to 20 SIP sessions

Transcoding: Supports up to 20 transcoding calls

Registrations: Maximum SIP registrations: 1000

CPS for Registration: 20 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,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.