Description

Overview

With versatile and robust architecture, The Synway SBC30 Session Border Controller (SBC) offers a complete connectivity solution for SMB enterprises and service provider and enables scalable, reliable and secured connectivity between diverse VoIP networks. Scaling up to 30 concurrent sessions, the SBC30 connects IP-PBXs to any SIP trunking and cloud-based services, and offers superior performance in connecting any SIP to SIP environment. The SBC30 could be customized to multiple voice channels in a 1U platform to enable versatile connectivity between VoIP networks, such as connecting IP PBX systems to any IP-based applications.

• 5~30 Pure IP SBC Sessions with Various Licensing
• High Interoperability with Various SIP Trunks & Platforms
• Enhanced Security and High Resiliency(1+1 Redundancy)

Basic Features & Functions For SBC 

• Dos/DDos protection
• QOS/ TOS/DSCP setting
• Signal encryption(TLS/IPSec)
• Media encryption（SRTP）
• NAT transverse
• SIP/H.323/H.248 interworking
• Support IPV4 , IPV6 and VPN
• Load balancing
• Transmission speed limit
• RTP encoding/decoding
• Anti-phreaking
• Redundancy and Backup

Unique Selling Points 

• High interoperability : Adopted by over 500 SPs and enterprises, and proven interoperability with SIP trunks, SIP platforms and IP cloud services
• Enhanced security :  Security-oriented, robust perimeter defense against cyber, DoS and DDoS attacks, as well as eavesdropping, fraud and service theft
• Superior voice quality :  Integrate decades of SW/HW technologies to obtain advanced capabilities for optimizing and monitoring voice service quality
• High resiliency : Telco-grade reliability, with High Availability (HA) using 1+1 active/standby redundancy, local branch survivability and PSTN fallback
• Flexible scalability :  The SBC30 architecture can scale up from 5 to 30 sessions, and the various licensing options assure economical scalability

Specifications

Capacities

• Max Signaling  : 30(from 5 to 30)
• Max. Transcoding Sessions 30(from 5 to 30)
• Max. RTP/SRTP Sessions 30(from 5 to 30)
• Max. Registered Users 250(upgradeable to 500

Network Interfaces

• 2(10/100/1000 BASE-TX(RJ-45)) & Customizable

Security

• Access Control: DoS/DDoS line rate protection, bandwidth throttling, dynamic blacklisting (Intrusion Detection System)
• Encryption/Authentication: TLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication
• Privacy: Topology hiding, user privacy
• Traffic Separation: Self-adjustable automatic load balance
• Intrusion Detection System: Detection and prevention of VoIP attacks, theft of service and unauthorized access
• VoIP firewall: Optional

Interoperability

• SIP B2BUA: Full SIP transparency, mature and broadly deployed SIP stack, stateful proxy mode
• SIP Interworking: 3xx redirect, REFER, PRACK, early media, call hold
• Registration and Authentication: User registration restriction control, registration and authentication on behalf of users, SIP authentication server for SBC users
• Transport Mediation: Mediation between SIP over UDP/TCP/TLS, IPv4/IPv6, RTP/SRTP
• Header Manipulation: Add/modify/delete SIP headers and message body using simple WireShark-like language with powerful capabilities such as variables and utility functions
• Number Manipulations: Ingress and egress digit manipulation
• Transcoding and Vocoders: Coder normalization including transcoding, coder enforcement and re-prioritization, extensive vocoder support: G.711, G.723.1, G.729, GSM-FR, AMR-NB, SILK-NB/WB, Opus-NB/WB
• Signal Conversion: DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, V.34, packet-time conversion
• NAT: Hosted NAT, RTP self-adaption
• WebRTC controller: Optional or customizable

Voice Quality and SLA

• Call Admission Control: Limit number and rate of concurrent sessions and registers per peer for inbound and outbound directions
• Packet Marking: 802.1p/Q VLAN tagging, DiffServ
• Standalone Survivability: Maintains local calls in the event of WAN failure. Outbound calls can use PSTN fallback (including E911).
• Impairment Mitigation: Dynamic Programmable Jitter Buffer, Silence Suppression/Comfort Noise Generation
• Voice Monitoring and Enhancement: acoustic echo cancellation, fixed and dynamic voice gain control, dynamic programmable jitter buffer, silence suppression, RTP redundancy, broken connection detection
• Direct Media: Hair-pinning (no media anchoring) of local calls to avoid unnecessary media delays and bandwidth consumption
• High Availability: SBC high availability with 1+1 redundancy, active calls preserved
• Test Agent: Ability to remotely verify SIP message flow between SIP UAs
• Echo cancellation: G.168 128 ms tail length
• Advanced Media Processing: T.38 real-time fax, T.38 – G.711 interworking

SIP Routing

• Routing Criteria: Incoming SIP trunk, DID ranges, host names, any SIP headers, codecs, QoE, bandwidth
• Route To: Configured SIP peers, registered users, IP address, request URI
• Advanced Routing Features: Alternative routes, load balancing, least-cost routing, call forking, E911 emergency call detection and prioritization
• SIPREC: SynAPI recording interface

Management

• OAM&P: Browser-based GUI, SNMP, INI Configuration file

Physical/Environmental

• Dimensions: 190*30*120mm
• Weight: About 0.7Kg
• Mounting: Desktop
• Power: 100-240V AC
• Environmental: Operating temperature: 0℃—40℃;Storage temperature: -20℃—85℃ Humidity: 8%— 90% non-condensing;Storage humidity: 8%— 90% non-condensing