IP-PBX, GSM/WCDMA 3G/VoLTE 4G Gateway, Analog/PRI Gateway, SBCs, Call Recording, HMP -synway.net-Products

SBC30

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

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.

Highlights
Specifications
Resources

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

Basic Features and Functions For SBC

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

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

Network Interfaces
Ethernet:                   2(10/100 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

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