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.


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.


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).



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


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


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