THINK LIKE A
MODERN SYSTEM
NO MORE HEADACHES
MODERN SYSTEM
NO MORE HEADACHES
IMPLANT A BRAINBOXTM TRANSFORM
AND THINK FOR THE FIRST TIME
You need a complete upgrade, but you don’t need another soon-to-be legacy system. BrainboxTM Transform is the modern, customizable and scalable VoIP solution that updates with you, so you can exit the vicious circle once and for all.
PURPOSE
Provides new and existing enterprises complete turnkey IP PBX communications hardware and software facilities. Provides cost-competitive solution for resellers. Enables resellers monthly recurring revenue streams and lowers end-customer monthly operational costs.
DESCRIPTION
Provides full functionality of BrainBox™ IP PBX feature set for enterprise use. Enables Expand™ features and functions including LCR, Bucket Routing, Black2Back™ Failover, Gate Routing ACD, and includes engineering, integration and access to (1) SIP trunk, (1) DID line, unlimited incoming calls, domestic and international calls (additional CONUS and Intl. per-minute rates applied).
YOUR SYSTEM GETS SMARTER WITH A BRAIN:
TECH SPECS
Capacities
Max. Signaling/Media Sessions: 150
Max. Transcoding Sessions: 96
Max. SRTP/RTP Sessions: 120
Max. Registered Users: 600
Telephony Interfaces
Modularity and Capacity: 6 slots for hosting voice processing and PSTN termination modules (up to 192 channels)
Digital Module: Up to 6 E1 or 8 T1/J1 spans provided on trunk modules. Each module supports 1, 2, or 4 E1/T1/J1 spans, with an option of PSTN Fallback
Digital PSTN Protocols: Supporting various ISDN PRI protocols such as EuroISDN, North American NI-2, LucentTM 4/5ESS, NortelTM 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.
BRI Module: Up to 20 BRI ports provided on BRI modules. Each module supports 4 BRI ports, with PSTN Fallback. Providing S/T interfaces; NT or TE termination; 2W per port (power supplied)
Analog Module: Up to 24 FXS/FXO interfaces, provided on 4 ports FXO / FXS modules, ground / loop start
Media Processing Module: Up to 4 Media Processing modules (MPM), providing additional DSP resources
Network Interfaces
Ethernets: Up to 6 GE 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, 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, IPv4 / IPv6, RTP / SRTP (SDES)
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, G.727, iLBC, QCELP, GSM EFR
Signal Conversion: DTMF/RFC 2833/SIP, T.38 fax, V.34, packet-time conversion
NAT: Local and far-end NAT traversal for support of remote workers
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 (including E911)
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)
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
Network Interfaces
OAM&P: Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, EMS
Physical / Environmental
Dimensions: 1U x 444 x 355 mm (HxWxD)
Mounting: Desktop or 19″ mount
Operating Temperature: Operational: 0 to 40 C (32 to 104F); Storage: -20 to 70C (-4 to 158F). Relative Humidity: 10 to 85% non-condensing
Weight: Approx. 9.7lb (4.4kg)
Power: Single power supply 100-240V, 50-60 Hz, 1.5A max. optional redundant power supply
Regulatory Compliance
Telecommunications: Safety and EMC Standards
Safety and EMC: UL60950-1; FCC 47 CFR part 15 Class B CE Mark (EN55022 Class B, EN60950-1, EN55024, EN300 386, EN61000-3-2/3-3)
Environmental Storage: ETS 300019-2-1 Storage T1.2, ETS 300019-2-2 Transportation T2.3 ETS 300019-2-3 Operating T3.2