IT’S STILL YOUR SYSTEM, WE JUST PUT A BRAIN IN IT
IMPLANT BRAINBOX™ EXPAND TO ADD
LINES SIMPLY, EFFICIENTLY AND AFFORDABLY
You need more phone lines, but you don’t need a six-figure capital investment upgrade to get there. BrainBox™ Expand is our exclusive technology that augments your current system with VoIP, so you can add as many lines as you need, simply, efficiently and affordably.
PURPOSE
Expands the shelf life of existing legacy PBX capabilities with IP-enabling technologies. Enables customers to use existing PBX facilities and begin migrating to new IP technologies simultaneously without “throwing away” the old PBX. Ideal for properties undergoing renovation that want to minimize CAPEX while minimizing OPEX in the process. Enables resellers monthly recurring revenue streams and lowers end-customer capital expenditures and monthly operational costs.
DESCRIPTION
Provides a hybrid communications system that extends the shelf life of legacy PBX systems while utilizing new IP-based communications simultaneously. Enables BrainBox™ IP PBX feature set including LCR and Bucket Routing for additional cost savings intelligence along with Black2Back™ Failover, Gate Routing ACD at 1 enterprise location. This product includes engineering, integration and access to (1) SIP trunk, (1) utility DID line, unlimited incoming calls, domestic and international calls (additional CONUS and Intl. per-minute rates applied).
YOUR SYSTEM GETS SMARTER WITH A BRAIN:
* These features are enabled using BrainBox with proper coordination and access to legacy switching platforms.
TECH SPECS
Capacities
Max. Signaling/Media Sessions: 250
Max. Transcoding Sessions: 571
Max. SRTP/RTP Sessions: 180
Max. Registered Users: 800
Telephony Interfaces
Analog: 4/8/12 FXS ports; 4/8/12 FXO ports
Digital: Up to two E1/T1 interfaces with an option for PSTN Fallback
Clock Source: 5 ppm High Precision
Digital PSTN Protocols: Supporting various ISDN PRI protocols such as EuroISDN, North American NI-2, Lucent™ 4/5ESS, Nortel™ DMS-100 and others. It also supports different variants of CAS protocols, including MFC R2, E&M immediate start, E&M delay dial / start and others.
Network Interfaces
Ethernets: 4 GE or 4 GE + 8 FE interfaces configured in 1+1 redundancy or as individual ports
Security
Access Control: DoS/DDoS line rate protection, bandwidth throttling, dynamic blacklisting
VoIP Firewall: RTP pinhole management, rogue RTP detection and prevention, SIP message policy, advanced RTP latching
Encryption/Authentication: TLS, DTLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication, RADIUS Digest
Privacy: Topology hiding, user privacy
Traffic Separation: VLAN/physical interface separation for multiple media, control and OAMP interfaces
Intrusion Detection System: Detection and prevention of VoIP attacks, theft of service and unauthorized access
Security
Access Control: DoS/DDoS line rate protection, bandwidth throttling, dynamic blacklisting
VoIP Firewall: RTP pinhole management, rogue RTP detection and prevention, SIP message policy, advanced RTP latching
Encryption/Authentication: TLS, DTLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication, RADIUS Digest
Privacy: Topology hiding, user privacy
Traffic Separation: VLAN/physical interface separation for multiple media, control and OAMP interfaces
Intrusion Detection System: Detection and prevention of VoIP attacks, theft of service and unauthorized access
Interoperability
SIP B2BUA: Full SIP transparency, mature and broadly deployed SIP stack, stateful proxy mode
SIP interworking: 3xx redirect, REFER, PRACK, session timer, early media, call hold, delayed offer
Registration and Authentication: User registration restriction control, registration and authentication on behalf of users, SIP authentication server for SBC users
Transport Mediation: SIP over UDP/TCP/TLS/WebSocket, IPv4 / IPv6, RTP / SRTP (SDES/DTLS)
Message Manipulation: Ability to add/modify/delete SIP headers and message body using advanced regular expressions (regex)
URI and Number Manipulations: URI user and host name 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.726, G.729, GSM-FR, AMR-NB/WB, SILK-NB/WB, Opus-NB/WB
Signal Conversion: DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, V.34, packet-time conversion, V.150.1
WebRTC Controller: Interworking between WebRTC devices and SIP networks Supports WebSocket, Opus, VP8 video coder, lite ICE, DTLS, RTP multiplexing, secure RTCP with feedback
NAT: Local and far-end NAT traversal for support of remote workers
Visit our partners,shoes – leaders in fashionable footwear!
Voice Quality and SLA
Call Admission Control: Based on bandwidth, session establishment rate, number of connections/registrations
Packet Marking: 802.1p/Q VLAN tagging, DiffServ, TOS
Standalone Survivability: Maintains local calls in the event of WAN failure. Outbound calls can use PSTN fallback for external connectivity
Impairment Mitigation: Packet Loss Concealment, Dynamic Programmable Jitter Buffer, Silence Suppression/Comfort, Noise Generation, RTP redundancy, broken connection detection
Voice Enhancement: Transrating, RTCP-XR, Acoustic echo cancellation, replacing voice profile due to impairment detection, Fixed & dynamic voice gain control
Direct Media(No Media Anchoring): Hair-pinning of local calls to avoid unnecessary media delays and bandwidth consumption
Voice Quality Monitoring: RTCP-XR, AudioCodes Session Experience Manager (SEM)
High Availability (Redundancy): SBC high availability with two-box redundancy, active calls preserved
Quality of Experience: Access control and media quality enhancements based on QoE and bandwidth utilization
Test Agent: Ability to remotely verify connectivity, voice quality and SIP message flow between SIP UAs
SIP Routing
Routing Methods: Request URL, IP address, FQDN, ENUM, advanced LDAP, third-party routing control through REST API
Advanced Routing: Criteria QoE, bandwidth, SIP message (SIP request, coder type, etc.), Layer-3 parameters
Routing Features: Least-cost routing, call forking, load balancing, E911 gateway support, emergency call detection and prioritization
SIPRec: IETF standard SIP recording interface
Management
OAM&P: Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, EMS
Physical / Environmental
Dimensions: 1U x 320mm x 345mm (HxWxD)
Mounting: Desktop or 19” rack mount
Operating Temperature: 5°-40° C
Weight: Approx. 5.95lb (2.7kg) loaded with OSN
Power: 100-240V 4A 50-60 Hz
Regulatory Compliance
Telecommunications: TIA/EIA-IS-968 (FXO, T1) interface, ETSI ES203 021 (FXO interface), TBR-4 (ISDN over E1 interface), TBR13/13 (E1 lines), TBR-3 (BRI interface)
Safety and EMC: IEC60950-1, UL60950-1, FCC Part 15 Class A, EN55022 Class A, EN55024, EN300 386
Environmental Storage: ETS300019-2-1 class T1.2
Transportation: ETS300019-2-2 class T2.3
Operating: ETS300019-2-3