Features and Benefits

 • PerfectVoice(tm) voice coding

 • PerfectPitch(tm) voice playback speed control

 • PerfectLevel(tm) voice playback volume control

 • Automatic gain control (AGC)

 • Transaction record

 • Silence compressed record

 • Silence detection

 • Flexible echo cancellation resource

 • PerfectDigit(tm) DTMF signaling

 • PerfectDigit(tm) MF signaling

 • PerfectCall(tm) call progress analysis

 • Positive Voice Detection(tm) (PVD)

 • Positive Answering Machine Detection(tm) (PAMD)

 • Global Tone Detection(tm) (GTD)

 • Global Tone Generation(tm) (GTG)

 • Global Dial Pulse Detection(tm) (GDPD) (optional)

 • Analog Display Services Interface (ADSI)

 • Two way FSK for ADSI

 • Analog Caller ID support

 SpringWare(tm) is the embedded software that provides the signal processing features available on Dialogic voice and intelligent network interface boards. SpringWare software algorithms offer the latest in high-performance, state-of-the-art voice and call processing features.

 Because SpringWare is downloaded from disk on power up, upgrading is easy. Adjustments or enhancements can be loaded from a diskette without replacing existing hardware.

 SpringWare works on all DSP-based Dialogic boards independent of port density. Because of a common source code base, new features and improvements are carried over from one generation of boards to the next and applications developers are assured of uniform performance regardless of board density and system size.

 SpringWare algorithms give you ultimate flexibility for keeping application features up to date while protecting your hardware investment. Default board parameters can be modified from a developer-determined configuration file to adjust performance to meet special application demands.

 SpringWare Compatibility Chart

 Voice Products
 
 
 

Network Interface Products
 
 
 

Voice + Fax Products
 
 
 

† Embedded Base products for existing CT systems. Not recommended for new systems design.

 PerfectVoice Enhanced Voice Coding*

 • Broad variety of voice coding algorithms allows developers to choose the voice quality, data rate, and compatibility that are most appropriate for all applications

 • Dynamically assign different data encoding rates to separate channels while the application is running

 • Open, standard algorithms allow support for a variety of third-party voice recording and editing tools and interoperability with other platforms

 • OKI ADPCM at 24 and 32 Kb/s, the most popular voice coder in the computer telephony industry; offers high quality voice with reasonable hard-disk storage requirements

 • G.711 PCM, µ-law and A-law, at 48 and 64 Kb/s, a worldwide network standard, offers toll quality voice and easy conversion to other coding schemes

 • G.726 ADPCM at 32 Kb/s, the ITU standard, offers toll quality voice with moderate disk storage requirements and interoperates with many existing proprietary voice processing systems (Available on selected products; contact your Dialogic Sales representative for details.)

 • Windows .WAV file data format for 11 kHz 8-bit linear, complying with industry-standard sound card formats

 PerfectPitch Speed Control

 • Speed up or slow down messages while maintaining the recorded voice at its normal pitch. Avoid pitch-shifting "Mickey Mouse" playback effects of linear control algorithms.

 • Map speed adjustment commands to DTMF keypad input (no host interaction required) or control directly from the application

 • Make speed adjustments before or during playback, on a channel-by-channel basis

 PerfectLevel Volume Control

 • Dynamically adjust playback volume over a 40 dB range

 • Map volume adjustment commands to DTMF keypad input (no host interaction required) or control directly from the application

 • Make volume adjustments before or during playback, on a channel-by-channel basis

 Automatic Gain Control (AGC)

 • Automatically adjusts signal level of incoming calls for recording at normal levels

 • Compensates for adverse line conditions, distance, and other factors

 • New and improved algorithm offers better voice quality while maintaining compatibility with existing voice files

 Transaction Record

 • Enables the recording of a two-party conversation. Mixes any two time slots (SCbus only) together for recording to one file. Useful for archiving a verbal transaction or conversation.

 Silence Compressed Record (SCR)*

 • Enables the recording of the caller's message with silent pauses eliminated. This results in smaller size voice files with no loss of intelligibility

 Silence Detection

 • Detects and differentiates speech and tones from quiet line conditions. Useful for controlling a variety of voice and call processing situations.

 Flexible Echo Cancellation Resource*

 • Performs echo cancellation function on an audio stream, improves voice quality, improves operation of the Dialogic conference bridge product, allows cut-through on external speech products, enables Internet telephony applications

 • Enables cut through for automatic speech recognition (ASR) resource boards and host-based ASR systems

 • Improves the Dialogic Conference Bridge quality performance, especially when used with analog loop start interface boards

 • Enables real-time voice over the Internet (VoIP) applications

 PerfectDigit DTMF Signaling

 • DSP-based DTMF (touchtone) detection algorithm optimized for lowest talk-off and play-off susceptibility in the industry. The system will not easily be fooled by mistaking human speech for DTMF tones.

 • Minimum tone duration and interdigit delay times accurately handle speed dialing typical of "power users"

• Utilizes echo cancellation which results in superior cut through for accurate DTMF tone interpretation during voice file playback within a broad range of network/switch environments

 • DTMF outbound dialing generated by DSP for accuracy and flexibility (dialing levels are adjustable to meet a variety of global PTT requirements)

 PerfectDigit MF Signaling

 • R1 MF signaling for use with Automatic Number Identification (ANI) and Dialed Number Identification System (DNIS)

 • R2 MF Compelled Protocol necessary for use in European and other E-1 countries

 Global Tone Detection (GTD)

 • Puts the power of DSP signal analysis into the hands of the applications developer

 • Powerful tool for solving special application situations, such as integrating with a PBX, or recognizing unique tones

 • Used as the basis of many preprogrammed algorithms from Dialogic, such as PerfectDigit and PerfectCall

 Global Tone Generation (GTG)

 • Generates user defined tones for special applications (unique PBX tones); uses DSP for high accuracy

 Global Dial Pulse Detection (GDPD(tm))*

 • DSP-based algorithm for detecting pulses dialed from a rotary dial telephone across the PSTN

 • Useful in countries and regions with little or no DTMF service

 • Designed to accurately recognize DPD pulses in even the most difficult telephony environments where pulse rates vary widely, as in many Latin American and Asian countries

 • Does not require a training digit

 PerfectCall Call Progress Analysis

 • Accurate outbound call monitoring detects when calls are answered and distinguishes

    - line ringing but not being answered

    - line busy

    - problem completing call (such as operator intercept)

    - call answered by a human or answering machine

    - call answered by a fax machine or modem

 • DSP-based algorithm is intelligently tolerant of the wide variation in call progress signaling tones found in Central Offices and PBXs around the globe - offers accurate performance right out of the box

 • Characterization tools available to address unique installation and application situations

 • Offers a choice of outbound tone or pulse dialing with a programmable "wait for dial tone before dialing" feature

 Positive Voice Detection

 • DSP algorithm accurately discriminates live human speech from network tones and noise

 • Designed to meet the needs of outbound telemarketing applications

 • Enables fast and accurate pass through of live calls to agents

 • Verify answered call in as little as 1/10th of a second (only when Positive Answering Machine Detection (PAMD) is not enabled)

 Positive Answering Machine Detection (PAMD)

 • Unique, patented DSP algorithm accurately discriminates human speech from recorded human voice

 • Designed to meet the needs of outbound telemarketing applications

 • Enables fast and accurate pass through of live calls to agents

 Analog Display Services Interface (ADSI)*

 • Physical layer support of the Bellcore standard for driving screen-enhanced telephones over the public network

 • Provides user-friendly visual support for traditional voice applications

 Analog Caller ID Support*

 • Captures calling party identification information, which is transmitted over loop start lines using in-band FSK

 • Versions available to meet both Bellcore CLASS requirements and British Telecom CLIP requirements

 Two Way FSK for ADSI*

 • Half-duplex data communications ability using Bell 202 (1200 Baud FSK). Meets Bellcore SR-346Z. Can be used to enhance ADSI screen phone applications with improved speed and security, or as general purpose data modem for short data burst applications

* See individual voice board datasheets for availability