Adopted by over 500 SPs and enterprises, and proven interoperability with SIP trunks, SIP platforms and IP cloud services
Fit
to complex networks, a sophisticated combo SBC and gateway architecture
for gradual migration, low CAPEX and reduced space and power footprints
Security-oriented,
robust perimeter defense against cyber, DoS and DDoS attacks, as well
as eavesdropping, fraud and service theft
Integrate decades of SW/HW technologies to obtain advanced capabilities for optimizing and monitoring voice service quality
Telco-grade
reliability, with High Availability (HA) using 1+1 active/standby
redundancy, local branch survivability and PSTN fallback
Basic Features and 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
Capacities
Max Signaling 250(from 120 to 250)
Max. RTP/SRTP Sessions 250(from 120 to 250)
Max. Transcoding Sessions 250(from 120 to 250)
Max. Registered Users 2000(upgradeable to 4000)
Telephony Interfaces
Analog Optional
Digital Up to 8E1/T1 Interfaces
Clock Source 50 ppm High Precision
Digital
PSTN Protocols: ISDN: ISDN User Side, ISDN Network Side, SS1: SS1
Signaling; SIP signaling: SIP V1.0/2.0, RFC3261; SS7 MTP1~3,SS7 TCAP,
SS7 ISUP, SIGTRAN, SS7 1+1 active/standby redundancy
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: 44*440*267mm
Weight: About 3.1Kg
Mounting: 19” rack mount
Power: 100-240V AC redundant dual feed
Environmental: Operating temperature: 0℃—40℃;Storage temperature: -20℃—85℃
Humidity: 8%— 90% non-condensing;Storage humidity: 8%— 90% non-condensing
