Nimblevox Engine
The Nimblevox Engine enables you to develop cutting edge communications applications. The Nimblevox Engine is cloud ready and can be hosted in our cloud or your cloud. Deploy your custom IVR application with the Nimblevox Engine and accelerate your time to market with IVR solutions that make business critical processes happen!
The Nimblevox Engine handles media resources such as voice, DTMF, and video. With inherent SIP and VoIP capabilities, the Nimblevox Engine provides enhanced media handling for large conferencing, audio streaming, and most importantly provides a development environment that allows you to make changes quickly and reliably.
Ideal for any automated inbound and outbound services, the Nimblevox Engine allows you to manage and develop applications easily via a web based administration console, while the VoiceXML/CCXML interpreters that come standard with the Nimblevox Engine allow for ease of portability.
Performance: One of the fastest IVR Systems on the market today, the Nimblevox Engine can handle up to 200 CPS, and up to 1400 channel capacity (32 bit), giving our customers the cost-savings and added benefits of requiring fewer ports to power their voice solution.
VXML/CCXML: Our IVR Systems are VoiceXML 2.1 compliant, helping accelerate market adoption for speech-enables applications worldwide, while assuring interoperability across platforms, tools and applications.
Drag and Drop: Our Build service creation environment allows for out of the box drag & drop functionality. Menus, Call Control, and Conferencing is just a few clicks away.
SIP Call Control Features
• Outbound call placement
• Call leg bridging
• Call rejection
• Call routing
• Call transfer
• REINVITE audio re-routing
• REFER support
• Registrar server support
• Proxy server support
• Authentication support
• Configurable timers
• Address restriction
• Early media support
• ISUP/SIP interworking (per SIP-I/SIP-T) for teminations/endpoints
• Configurable RTP port range
SIP Compatability/Interoperability
• Axvoice
• Bandwidth.com
• BroadVoice
• BroadVox
• SER/OpenSER
• Asterisk
• Metaswitch Interoperable with the following SIP Devices:
• SJ Phone
• Cisco/Linksys/Sipura
• Polycom
• Grandstream
• X-Lite Interoperatable with the following Gateways:
• TelcoBridges Media Gateway
• Dialogics IMG1010
• Quintum Tenor VoIP Gateways
• Sucessfully tested against SIP Forum SIPConnect V1.0 specification (PBX)
IVR Functionality
• Answers Inbound Calls
• Plays prerecorded audio prompts/announcements
• Detects DTMF tones (via RFC 2833) in RTP stream
• Records incoming audio
• Plays back recorded audio
• Places outbound calls
• Generate DTMF(RFC 2833) in RTP stream
• Bridges audio between call legs
• Transfers calls (blind, bridge, consultation)
• Conference call management (setup, add/delete participants, announce tones, listen only or active participants, teardown)
Audio/Media Processing
• Packet loss concealment
• Fixed gain control (per channel)
• Configurable jitter buffer
• Codec transcoding
• Voice activity detection
• Live speaker or answering machine discrimination
• Fax tones detection
• T.38 Fax send/receive
• H.263 Video Streaming to/from 3gpp files along with bridged calls
Codecs Supported
• G.711 uLaw (ITU-T unless stated otherwise)
• G.711 aLaw
• G.722 (“HD audio”)
• G.726 ADPCM (32 kbps only)
• G.729a (requires license)
• GSM (RPE-LTP Full Rate – 13kbps)
• AMR-NB (3gpp codec, requires license)
• iLBC (RFC 3952 Internet Low Bit Rate)
• H.263 Video
Audio/Video Mime Types
• audio/x-wav
• audio/x-adpcm
• audio/x-alaw-basic
• audio/x-basic
• audio/x-gsm
• audio/mpeg
• video/3gpp
System Features
• Runs on x86 architecture
• Runs on RHEL5 CentOS5
• Delivered as a tgz file of RPMs, installs using RPM Manager
• Install utility has checkpoint/rollback cpability
• “Installation verification” voice application included in initial install
• Premise or hosted solution
• Web browser based management console
• Documented HTTP management interface
• Online documentation on SPOT SIP Engine server
• Heavily commented ExamplePack (online)
• Worker/standby and N + 1 redundancy server configurations
• Keeps active calls alive during failover
• SNMP Support (Option)
• Open platform, user can add own applications/processes
• SNMP support extensible to user’s add-ins
W3C/ECMA/ITU-T Standards Supported
• VoiceXML 2.0/2.1 Compliant (W3C unless stated otherwise)
• CCXML 1.0 Compliant (Implementation Report participant)
• SRGS 1.0 Speech Grammars
• SSML 1.0 Speech Markup
• SISR 1.0 Speech Semantic Interpretation
• Extensible Markup Language (XML) 1.1
• XML Document Object Model (DOM)
• XML Namespaces
• Standard ECMA-262 ECMAScript Language Specification (ECMA)
• Q.1912.5 Interworking between SIP and BICC or ISUP (ITU-T Recommendation)
IETF Standards
• RFC 1889 – Transport Protocol for Real-Time Applications (Obsoleted by RFC 3550)
• RFC 1890 – RTP Profile for Audio and Video Conferences with Minimal Control (Obsoleted by RFC 3551) • RFC 2246 – The TLS Protocol Version 1.0 (Obsoleted by RFC 4346)
• RFC 2326 – Real Time Streaming Protocol (RTSP)
• RFC 2327 – SDP: Session Description Protocol (Obsoleted by RFC 4566)
• RFC 2616 – Hypertext Transfer Protocol — HTTP/1.1
• RFC 2833 – RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals (Obsoleted by RFC 4733, RFC 4734)
• RFC 2976 – The SIP INFO Method (Obsoleted by RFC 6086)
• RFC 3164 – The BSD Syslog Protocol (Obsoleted by RFC5424)
• RFC 3195 – Reliable Delivery for syslog
• RFC 3204 – MIME media types for ISUP and QSIG Objects
• RFC 3261 – SIP: Session Initiation Protocol
• RFC 3162 – Reliability of Provisional Responses in SIP
• RFC 3263 – SIP: Locating SIP Servers
• RFC 3264 – An Offer/Answer Model with SDP
• RFC 3265 – SIP: Specific Event Notification
• RFC 3266 – SDP: Session Description Protocol (Obsoleted by RFC 4566, Updates RFC 2327)
• RFC 3310 – HTTP Digest Authentication Using Authentication and Key Agreement (AKA)
• RFC 3311 – SIP UPDATE Method
• RFC 3323 – A Privacy Mechanism for SIP
• RFC 3324 – Short Term Requirements for Network Asserted Identity
• RFC 3325 – Private Extensions to SIP for Asserted Identity within Trusted Networks
• RFC 3372 – SIP-T: Context and Architectures • RFC 3515 – SIP REFER Method
• RFC 3550/STD 0064 – RTP: A Transport Protocol for Real-Time Applications (Obsoletes RFC 1889)
• RFC 2551/STD 0065 – RTP Profile for Audio and Video Conferences with Minimal Control (Obsoletes RFC 1890)
• RFC 3581 – An Extension to SIP for Symmetric Response Routing
• RFC 3725/BCP 0085 – Best Current Practices for Third Party Call Control SIP
• RFC 3842 – Message Summary and Message Waiting Indication Event Package for SIP
• RFC 3891 – SIP Replaces Header
• RFC 3952 – RTP Payload Format for iLBC
• RFC 3960 – Early Media and Ringing Tone Generation in SIP
• RFC 4028 – Session Timers in SIP
• RFC 4346 – TLS Protocol Version 1.1 (Obsoletes RFC 2246, Obsoleted by RFC 5246)
• RFC 4733 – RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals (ObsoletesRFC 2833)
• RFC 4734 -Definition of Events for Modem, Fax, and Text Telephony Signals (Obsoletes RFC 2833, Updates RFC 4733)
• RFC 4475 – SIP Torture Test Messages
• RFC 4566 – SDP: Session Description Protocol (Obsoletes RFC 3266, RFC 2327)
• RFC 5424 – The Syslog Protocol (Obsoletes RFC 3164)
• draft-ietf-speechsc-mrcpv2 – Media Resource Control Protocol Version 2
