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bekIOT-SBC3000 Session Border Controller

bekIOT-SBC3000

科能融合SBC3000会话边界控制器为中小电信业务运营商的SIP网络提供丰富的安全,接入,互连,路由/策略管理,信令流控,QoS和媒体处理等业务。设备采用多核处理器,无阻塞千兆交换网,嵌入式Linux操作系统,在实现高性能的同时具有极低的功耗, 同时支持双热插拔电源、双机热备(HA)和WebRTC,电信级高可靠。SBC3000单机支持2000并发会话和1500路语音媒体转码处理,并且支持拓扑隐藏、......

Product Introduction

bekIOT SBC3000 Session Border Controller (SBC) provides small and medium-sized telecom service operators with comprehensive services for their SIP networks, including security, access, interconnection, routing/policy management, signaling flow control, QoS, and media processing. Leveraging a multi-core processor, non-blocking Gigabit switching fabric, and embedded Linux OS, the device delivers high performance with extremely low power consumption. It also supports dual hot-swappable power supplies, active/standby high availability (HA) clustering, and WebRTC, ensuring carrier-grade reliability.

The SBC3000 supports 2,000 concurrent sessions and 1,500 channels of voice media transcoding per standalone unit. It integrates advanced security features such as topology hiding, NAT/PAT traversal, DoS/DDoS mitigation, SRTP/DTLS-SRTP encryption, and TLS. Additionally, it supports a wide range of media codecs, including G.729, G.723, G.711a/u, G.726, AMR, OPUS, and iLBC. The device offers load balancing capabilities and enables application integration via SDKs for four major platforms (Windows/macOS/iOS/Android), along with network voice coding optimization. Furthermore, the SBC3000 is equipped with E1 interfaces to facilitate seamless integration with PBX systems and PSTN lines, addressing diverse networking requirements for customers.

High Capacity Digital VoIP Gateway for Carriers & ITSPs

  • 16 to 63 ports E1/T1 in 2U chassis, STM-1 interface
  • Up to 1890 simultaneous calls
  • Redundancy Dual MCU units
  • Dual Power Supplies
  • Flexible routing
  • Multiple SIP trunks
  • Fully compatible with mainstream VoIP platforms

Rich Experiences on PSTN Protocols

  • ISDN PRI
  • ISDN SS7, SS7 links redundancy
  • R2 MFC
  • T.38 and Pass-through fax
  • Support modem and POS machines
  • More than 10-year experiences to integrate with a wide range of Legacy PBXs / Service providers' PSTN networks

Easy Management 

  • Intuitive Web interface
  • Support SNMP
  • Automated provisioning
  • Dinstar Cloud Management System
  • Configuration Backup & Restore
  • Advanced Debug tools

Product parameters

Single Machine Performance:

Call Concurrency: Maximum support for 2000 concurrent calls
Transcoding Concurrency: Support for 1500 concurrent media encoding and decoding processes
Call CPS: Maximum processing of 150 calls per second
User Registration Count: Maximum support for 10000 users to register
Registration CPS: Maximum processing of 200 registration messages per second
SIP Trunks: Maximum number of SIP trunks that can be added is 128
Trunk SBC Hot Standby (1+1)
Concurrent Call Licenses: 500 (non-transcoding)
Scalable to 3000 concurrent calls (non-transcoding)
WebRTC SBC Hot Standby (1+1)
WebRTC Registered Users: 2000, scalable to 10000
Concurrent Call Licenses (non-transcoding): 200, scalable to 3000
Transcoding Licenses: 200 (transcoding G.711/G.729/OPUS)
Scalable to 800 (transcoding G.711/G.729/OPUS)
Support for SDK (4-end: Windows/Mac/iOS/Android) integration

Security:
DOS/DDOS Attack Defense
Access Control Policies
IP and SIP Attack Prevention based on Policies
Malformed Message Detection and Processing
UDP-Flood Attack Defense
TCP-Flood Attack Defense
SRTP/DTLS-SRTP Encrypted Sessions
TLS Security Protection
Main Caller Number Blacklist/Whitelist
ACL Control
VoIP Firewall

Physical Specifications:
Power Supply: 100-240VAC, 50-60 Hz
Power Consumption: 70W
Operating Temperature: 0℃ ~ 45℃, Storage Temperature: -20℃ ~ 80℃
Humidity: 10%-90% without condensation
Dimensions (W/D/H): 43732044mm (1U)
Weight: 6kg

Featured Features

  1. 2,000 concurrent sessionsand **1,500-channel voice media transcoding processingper standalone unit; 
  2. Dual-machine hot standby (HA) for both **Trunk SBCand **WebRTC SBC**; 
  3. WebRTC, voice, and video calls; 
  4. Application docking with SDK (supporting 4 terminals: Windows/mac/iOS/Android) and network voice coding optimization; 
  5. Standard SIP protocol and flexible routing rules, with perfect compatibility for IMS systems; 
  6. Topology hiding and DoS/DDoS security attacks prevention to protect the core network; 
  7. Intelligent bandwidth limitation and dynamic blacklist; 
  8. Cross-network and NAT/PAT traversal, adapting to various networking environments; 
  9. Encrypted sessions via SIP over TLS and SRTP/DTLS-SRTP for security and reliability.