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IVR VoIP > Prophecy SIP Media Server Software

The Voxeo Prophecy SIP Media Server software delivers a resume even more impressive than that of its SIP call control sibling. Built on Voxeo's carrier-grade Host Media Processing (HMP) engine, the Prophecy SIP Media Server has played, recorded and conferenced its way through over three trillion VoIP packets since 1999.

The Prophecy SIP Media Server is driven by Voxeo's certified-compliant VoiceXML IVR and speech media application engine. VoiceXML delivers features to play prompts and speak synthesized text, record audio, and receive input via touch-tone entry or comfortable speech recognition. Voxeo extensions add everything developers need to record calls and to connect callers in high-quality multi-party audio conferences.

The Prophecy SIP Media Server also bundles highly intelligent English-language speech recognition and speech synthesis engines at no additional charge, and supports a wide variety of additional speech engines via support for the IETF Media Resource Control Protocol (MRCP) standard.

Voxeo's Prophecy SIP Platform is a flexible, reliable, VoiceXML-based foundation for any VoIP media, speech, or IVR application with a proven track record of successful real-world deployments by Voxeo and our customers.

You can try Prophecy at no cost, right now, via our free Prophecy download or our free hosted Evolution developer portal.


How does the Prophecy SIP Media Server compare to other solutions?

Voxeo's Prophecy SIP Platform includes many valuable features other platforms lack, including full W3C standards support, comprehensive CCXML call control, call conferencing and recording, and more. The following chart summarizes Voxeo's feature advantages over most other vendors:


 Voxeo
Prophecy
Platform
Other
VoiceXML
Platforms
Other
IVR
Platforms
VoiceXML 2.0
SRGS Grammars
SSML Speech Synthesis
VoiceXML 2.1 Extensions
Supports 100% of W3C IVR Standards
CallXML 2.0
Call Recording
Call Conferencing
Native VoIP Support
Multi-lingual ASR / TTS
Full CTI Support
Intelligent Call Progress Analysis
SYSLOG Logging
Proven Scalability
Premise solution and hosted service
24 x 7 Technical Support
VoiceXML is the W3C standard for voice-driven IVR, including ASR, TTS, prompt playback, and recording.
SRGS is the W3C standard for voice recognition grammar definition. Most VoiceXML Vendors Support SRGS.
SSML is the W3C standard for text-to-speech synthesis definition and guidance. Voxeo's Prophecy platform features full support for the SSML standard, giving you extensive control over speech synthesis.
VoiceXML 2.1 adds support for new, valuable features including improved flow control and built-in data interfaces.
W3C IVR standards include VoiceXML, CCXML, SRGS, and SSML. Many vendors support only VoiceXML and SRGS.
CallXML is a Voxeo-designed language that is much easier to develop applications in than VoiceXML and CCXML.
Many tasks, including voice user interface tuning and third-party verification of telephone sales transactions require the ability to record both sides of a call. Most VoiceXML vendors do not offer this capability.
Most VoiceXML platforms can only bridge or connect two parties. The Voxeo Prophecy platform can connect 2 to 200 parties in a single session.
Most VoiceXML platforms have no - or very limited - Voice over IP support. The Voxeo Prophecy Platform is a full-featured SIP solution, and includes its own SIP Softswitch/proxy server, media server, call control server, and more.
Most VoiceXML platforms support only English, or only one language at a time. The Prophecy platform features dynamic support for English, Spanish, French, German, and Italian speech recognition and synthesis.
Most VoiceXML platforms have little if any CTI (Computer Telephony Integration) support. CTI is required for most call center IVR deployments. Voxeo offers CTI solutions for Cisco ICM, Genesys T-Server, and Nortel and Avaya call center platforms. Additional CTI interfaces can be built quickly via Voxeo's CCXML based CTI-plug-in interface.
The Prophecy platform includes intelligent call analysis to determine if outbound calls are answered by people, answering machines, voicemail systems, or network error messages. This capability is vital when using IVR for outbound notification, collections, and campaign calls.
Prophecy supports the standards SYSLOG logging protocol so you can easily aggregate and analyze vital call logging details.
The Prophecy platform has proven scalability from as few as 1 to as many as 5,000 ports. Most vendors have not deployed VoiceXML IVR systems larger than 200 ports.
Voxeo uniquely offers the Prophecy platform as either a premise solution you can buy outright, or hosted service you can use by the minute.
Simply put, Voxeo is the industry leader in proactive and reactive support services. We offer 24 x 7 support with fully trained, VoiceXML certified technicians and IVR developers. No matter when a problem occurs, our top-notch staff is available to help you.

The present and future direction of the IVR industry is defined by a suite of IVR/Voice standards from the W3C, the same organization that created the highly successful standards that power the web. Voxeo's Prophecy SIP Platform is the first and only platform to support the entire suite of W3C Voice/IVR standards: VoiceXML for IVR control, CCXML for call control, SRGS for speech recognition control, and SSML for text to speech control. The platform supports over 15 additional IT standards, including XML/XPath and SOAP for back-office integration, SYSLOG for logging, ENUM/DNS for call routing, and SIP for both PSTN/PBX and VoIP telephony integration.

You can get a Prophecy proposal today...

Our Prophecy SIP Platform can be custom configured and priced for your enterprise needs by a member of our Extreme Support team. For more information or pricing, just use one of the links below to contact them now.


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Voxeo SIP Platforms & Services
Specification Overview

Call Control Features
  • SIP inbound and outbound call support
  • SIP call redirection (SIP 302 status)
  • SIP call rejection
  • SIP call routing and transfers
  • SIP call leg bridging
  • SIP REINVITE audio re-routing
  • SIP REFER support
  • SIP registrar server support
  • SIP proxy server support
  • SIP authentication support
  • Configurable SIP port range
  • NAT IP translation support
  • Multi-ethernet card bridging support
  • Runs on standard x86 platforms


Media and IVR Features
  • Audio prompt/announcement playback
  • Audio bridging
  • Audio mixing/conferencing
  • Audio noise removal
  • Audio fixed gain control
  • Audio dynamic gain control
  • Audio voice activity detection (VAD)
  • Audio call recording
  • Audio codec transcoding
  • Audio speech recognition (ASR)
  • Audio speech synthesis (TTS)
  • Audio playout jitter buffer
  • DTMF tone detection
  • DTMF tone generation
  • Configurable RTP port range
  • NAT IP translation support
  • Multi-ethernet card bridging support
  • Supports VoiceXML 2.x IVR
  • Supports CallXML 3.0 IVR
  • Runs on standard x86 platforms


Turnkey Server Hardware Features
  • Highly reliable x86 platform
  • 120/240v AC and -48v DC power
  • Full-depth rack mount server
  • Up to 80 servers in a 4 post rack
  • Quad Core 2.33 GHz, 4GB RAM
  • 1U w/ 1 PCI slot, 3U w/ 5 PCI slots


Turnkey Server Built-in PSTN/SIP Gateway
  • Optional add-on to Turnkey Server
  • Supports analog POTS lines
  • Supports T1 and E1 trunks/lines
  • NI1/2, QSIQ, and Euro ISDN
  • DMS100/2500 and 4/5ESS ISDN
  • CAS robbed bit signaling
  • Echo cancel with 32 to 128 MS tail
  • Standard size PCI 3.3v card
  • Up to 96 ports in 1U server
  • Up to 384 ports in 3U server
  • Over 4,000 ports per 4 post rack


Compatibility Features
  • Works with Level(3) SIP
  • Works with BroadVoice SIP
  • Works with Delta3 SIP
  • Works with SER/OpenSER SIP
  • Works with Asterisk SIP
  • Works with Cisco SIP
  • Works with Lucent TNT SIP
  • Works with Sipura SIP
  • Works with Sonus SIP
  • Works with most other SIP services
  • Works with most other SIP devices


Other Features

  • Premise or hosted solution
  • No single points of failure
  • Scales from one to thousands of ports
  • First platform with XML call control
  • First platform with XML conferencing
  • First shipping CCXML implementation
  • First SIP/VoIP IVR platform


IETF Standards Support
  • RFC 3261 SIP
  • RFC 3310 SIP Authentication
  • RFC 1889 RTP Media
  • RFC 1890 RTP Audio
  • RFC 2327 SDP
  • RFC 3264 SDP negotiation
  • RFC 2833 DTMF and events
  • RFC 3263 SRV DNS records
  • RFC 3761 ENUM URI DNS records
  • RFC 3764 ENUM SIP DNS records
  • RFC 3164 UDP Syslog logging
  • RFC 3195 TCP Syslog logging
  • RFC 2865 RADIUS metering
  • RFC 2616 HTTP protocol
  • RFC 2617 HTTP authentication
  • RFC 2964 HTTP state management
  • RFC 2965 HTTP state management
  • RFC 3927 Dynamic IP config
  • RFC 2136 Dynamic DNS updates
  • DNSEXT DNS Service Discovery Draft
  • MMUSIC RTSP Draft
  • SPEECHSC MRCP Draft


W3C Standards Support

  • VoiceXML 2.0 speech/IVR media
  • VoiceXML 2.1 extensions
  • CCXML 1.0 call control
  • SRGS 1.0 speech grammars
  • SSML 1.0 speech markup
  • SISR speech semantic interpretation
  • Extensible Markup Language (XML) 1.1
  • Namespaces in XML 1.1
  • XML Document Object Model
  • XML Path Language (XPath) 1.0
  • XML Event Syntax
  • SOAP Web Services
  • WSDL Web Service Description


ANSI Standards Support
  • T1/DS1 Electrical Interface
  • T1/DS1 Robbed Bit Signaling
  • T1/DS1 Primary Rate ISDN
  • T3/DS3 Electrical Interface
  • Bearer Services for ISDN PRI
  • SS7 SCCP/MTP/IUP/TCAP
  • Telecom Voltage Levels
  • Electrical protection for CO's


ITU Standards Support
  • G.711 uLaw Audio Codec
  • G.712 aLaw Audio Codec
  • G.726 ADPCM Audio Codec
  • G.729 CS-ACELP Audio Codec
  • G.729 (b) Silence Detection
  • GSM RPE-LTP Audio Codec

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