brainboxbrown

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

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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

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