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Features and Benefits
D/41EPCI supports SpeechWorks Host which enables you to create applications that allow hands-free speed dialing from cellular car phones, hands-free voice mail, and automatic dialing of spoken numbers or names. Complicated numeric menu systems can be reduced to a small set of user-friendly spoken commands. Downloaded SpringWare™ firmware algorithms, executed by the onboard DSP, provide variable voice coding at 24 and 32 Kb/s ADPCM, and 48 and 64 Kb/s m-law or A-law PCM, as well as m-law to A-law conversion. Sampling rates and coding methods are selectable on a channel-by-channel basis. Applications may dynamically switch sampling rate and coding method to optimize data storage or voice quality as the need arises. SpringWare firmware also provides reliable DTMF detection, DTMF cut-through, and talk off/play off suppression over a wide variety of telephone line conditions. Dialogic Global DPD™ dial pulse detection algorithm, available as a software option for the D/41EPCI, lets you use the product in countries that have limited touchtone telephone service. Global DPD can be optimized on a country-by-country basis to provide superior dial pulse detection wherever it is used. Dialogic voice products offer a rich set of advanced features, including state-of-the-art DSP technology and signal processing algorithms, for building the core of any CT system. With industry-standard PCI bus expansion boards you can integrate Dialogic voice products easily into exactly the type of system you require at a price and performance level unmatched in the CT industry. The D/41EPCI board
Use the D/41EPCI board to build sophisticated CT systems to which
capabilities such as speech recognition, facsimile, and text-to-speech
can be added. The D/41EPCI shares a common hardware and firmware architecture
with other Dialogic SCbus™ based boards for maximum flexibility and scalability.
Features can be added and systems can grow while protecting investment
in hardware and application code. With only minimum modifications, applications
can be easily ported to lower or higher line-density platforms.
The D/41EPCI installs in any PCI-based personal computer or server (PCI bus or mixed PCI/ISA) and compatible computers (Intel 80386, 80486, or Pentium™-based PC platforms). The D/41EPCI provides everything required for building integrated voice solutions scalable from 4 ports to 64 ports. The maximum number of lines that can be supported is dependent on the application, the amount of disk I/O required, and the host computer CPU and power supply. Applications developed to run on the Proline/2V™, DIALOG/4™, D/41D™,
D/41H™, or D/41ESC™ family will run on a similar D/41EPCI configuration.
Developers can choose from a wide selection of Dialogic products to build
scalable, reliable, and economical CT solutions.
The D/41EPCI board can operate within a mixed chassis containing Dialogic PCI and ISA products. The forward-looking design of the D/41EPCI incorporates the new H.100 connector to simplify connection to next generation CT Bus™ products. The D/41EPCI can also connect to existing SCbus products through the use of an optional CT Bus/SCbus adapter. The adapter provides both SCbus and H.100 physical connectors required to link the D/41EPCI to current SCbus products. Software Support The D/41EPCI is currently supported by the Dialogic System Software and Software Development Kit for Windows NT® (Native). This package contains a set of tools for developing complex multichannel applications. Functional Description The D/41EPCI uses a unique dual-processor architecture that combines the signal processing capabilities of a DSP with the decision-making and data movement functionality of a general purpose 80186 control microprocessor. This dual processor approach off-loads many low-level decision-making tasks from the host computer and thus enables easier development of more powerful applications. This architecture handles real time events, manages data flow to the host PC for faster system response time, reduces host PC processing demands, processes DTMF and telephony signaling, and frees the DSP to perform signal processing on the incoming call. Each of four analog loop-start telephone line interfaces on the
D/41EPCI receives analog voice and telephony signaling information from
the telephone network (see block diagram). Each telephone line interface
uses reliable, solid state hook switches (no mechanical contacts) and FCC-part
68 class B ring detection circuitry. This FCC-approved ring detector is
less susceptible to spurious rings created by random voltage fluctuations
on the network. Each interface also incorporates circuitry that protects
against high-voltage spikes and adverse network conditions and allows applications
to go off-hook any time during ring cadence without damaging the board.
Inbound telephony signaling (ring detection, loop-current detection, and Caller ID information) is conditioned by the line interface and routed via a control bus to the control processor. The control processor responds to these signals, informs the application of telephony signaling status, and instructs the line interface to transmit outbound signaling (on-hook/off-hook) to the telephone network. The audio voice signal from the network is bandpass filtered and conditioned by the line interface and then applied to a CODEC (COder/DECoder) circuit. The CODEC filters, samples, and digitizes the inbound analog audio signal and passes this signal to a Motorola DSP. Based on SpringWare firmware loaded in DSP SRAM, the DSP performs the following signal analysis and operations on this incoming data:
When recording speech, the DSP can use different digitizing rates from 24 to 64 Kb/s as selected by the application for the best speech quality and most efficient storage. The digitizing rate is selected on a channel-by-channel basis and can be changed each time a record or play function is initiated. The DSP processed speech is transmitted via the control processor to the host PC for disk storage. When replaying a stored file, the processor retrieves the voice information from the host PC and passes it to the DSP, which converts the file into digitized voice. The DSP sends digitized voice and appropriate signaling responses to the CODEC to be converted into analog format for transmission to the telephone network. Signaling data (on-/off-hook, ringing, Caller ID, etc.) is passed to the onboard control processor and transmitted to the application via a dual-port shared RAM and the host PCI bus. When using the D/41EPCI board and the SCbus, digital voice and signaling information from a network board or other resource enter the board via the H.100 connector and SCbus interface. A SC2000 chip manages these signals and acts as the traffic coordinator and matrix switch to buffer the high-speed digital data from the bus until the data for each channel can be transmitted to the DSP. The SC2000 chip transmits several lower speed data streams over the SCbus high speed channel. The bus configuration is set when the firmware is downloaded at system initialization. This chip incorporates matrix switching capabilities. Under control of the onboard control processor, the SC2000 chip can connect any call being processed to any of the four analog lines or to any of the 1024 SCbus time slots. This enables the application to switch calls to or from other resources, such as facsimile or speech recognition, as they are needed, or to reroute calls. The onboard control processors control all operations of the D/41EPCI board via a local bus and interpret and execute commands from the host PC. These processors handle real-time events, manage data flow to the host PC to provide faster system response time, reduce PC host processing demands, process DTMF and telephony signaling before passing them to the application, and free the DSP to perform signal processing. Communications between a processor and the host PC is via the Shared RAM that acts as an input/output buffer and thus increases the efficiency of disk file transfers. This RAM interfaces to the host PC via the PCI bus. All operations are interrupt-driven to meet the demands of real-time systems. When the system is initialized, SpringWare firmware is downloaded from the host PC to the onboard code/data RAM and DSP RAM to control all board operations. This downloadable firmware gives the board all of its intelligence and enables easy feature enhancement and upgrades. With the rotary switch on the D/41EPCI set to 0, the D/41EPCI board is Plug and Play enabled. Configuration is handled exclusively by software. Alternatively, you can set the rotary switch to another value to manually control board location for ease of cabling or backwards compatibility with Dialogic Board Locator Technology™ (BLT) installation. Technical Specifications*
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SpringWare Technical Specifications*
| AUDIO SIGNAL: | |
|---|---|
| Receive range | –50 to –13 dBm (nominal), for average speech signals1 configurable by parameter‡ |
| Automatic gain control | Application can enable/disable. Above –18 dBm results in full scale recording, configurable by parameter‡ |
| Silence detection | –38 dBm nominal, software adjustable‡ |
| Transmit level | |
| (weighted average) | –9 dBm nominal, configurable by parameter‡ |
| Transmit volume control | 40 dB adjustment range, with application definable increments |
| Frequency response | |
| 24 Kb/s | 300 Hz to 2600 Hz ±3 dB |
| 32 Kb/s | 300 Hz to 3400 Hz ±3 dB |
| 48 Kb/s | 300 Hz to 2600 Hz ±3 dB |
| 64 Kb/s | 300 Hz to 3400 Hz ±3 dB |
| AUDIO DIGITIZING: | |
| 24 Kb/s | ADPCM @ 6 kHz sampling |
| 32 Kb/s | ADPCM @ 8 kHz sampling |
| 48 Kb/s | µ-law PCM @ 6 kHz sampling |
| 64 Kb/s | µ-law PCM @ 8 kHz sampling |
| Digitization selection | Selectable by application on function call by call basis |
| Playback speed control | Pitch controlled; available for 24 and 32 Kb/s ADPCM data rates; adjustment range: ±50%; adjustable through application or programmable DTMF control. |
| DTMF TONE DETECTION: | |
| DTMF digits | 0 to 9, *, #, A, B, C, D per Bellcore LSSGR Sec 6 |
| Dynamic range | –45 dBm to +3 dBm per tone, configurable by parameter‡ |
| Minimum tone duration | 40 ms, Can be increased with software configuration |
| Interdigit timing | Detects like digits with a 40 ms interdigit delay. |
| Detects different digits with a 0 ms interdigit delay. | |
| Twist and frequency variation | Meets Bellcore LSSGR Sec 6 and EIA 464 requirements |
| Acceptable twist | 10 dB |
| Signal/noise ratio | 10 dB (referenced to lowest amplitude tone) |
| Noise tolerance | Meets Bellcore LSSGR Sec 6 and EIA 464 requirements for Gaussian, impulse, and power line noise tolerance |
| Cut through | Detects down to –36 dBm per tone into 600 Ohm load impedance |
| Talk off | Detects less than 20 digits while monitoring Bellcore TR-TSY-000763 standard speech tapes (LSSGR requirements specify detecting no more than 470 total digits). Detects 0 digits while monitoring MITEL speech tape #CM 7291. |
| GLOBAL TONE DETECTION™: | |
| Tone type | Programmable for single or dual |
| Maximum number of tones | Application dependent |
| Frequency range | Programmable within 300 to 3500 Hz |
| Maximum frequency deviation | Programmable in 5 Hz increments. |
| Frequency resolution | Less than 5 Hz.—Note: Certain limitations exist for dual tones closer than 60 Hz apart. |
| Timing | Programmable cadence qualifier, in 10 ms increments |
| Dynamic range | Programmable, default set at –36 dBm to +3 dBm per tone |
| GLOBAL TONE GENERATION™: | |
| Tone type | Generate single or dual tones |
| Frequency range | Programmable within 200 to 4000 Hz |
| Frequency resolution | 1 Hz |
| Duration | 10 msec increments |
| Amplitude | –43 dBm to –3 dBm per tone, programmable |
| MF SIGNALING: | |
| MF digits | 0 to 9, KP, ST, ST1, ST2, ST3 per Bellcore LSSGR Sec 6, TR-NWT-000506 and CCITT Q.321 |
| Transmit level | Complies with Bellcore LSSGR Sec 6, TR-NWT-506 |
| Signaling mechanism | Complies with Bellcore LSSGR Sec 6, TR-NWT-506 |
| Dynamic range for detection | –25 dBm to +3 dBm per tone |
| Acceptable twist | 6 dB |
| Acceptable frequency variation | Less than ±1 Hz |
| CALL PROGRESS ANALYSIS: | |
| Busy tone detection | Default setting designed to detect 74 out of 76 unique busy/congestion tones used in 97 countries as specified by CCITT Rec E., Suppl #2. Default utilizes both frequency and cadence detection. Application can select frequency only for faster detection in specific environments. |
| Ring back detection | Default setting designed to detect 83 out of 87 unique ring back tones used in 96 countries as specified by CCITT Rec E., Suppl #2. Utilizes both frequency and cadence detection. |
| Positive Voice | |
| Detection™ Accuracy | >98% based on tests on a database of real world calls |
| Positive voice detection speed | Detects voice in as little as 1/10th of a second. |
| Positive answering | |
| machine detection™ accuracy | 80 to 90% based on application and environment |
| Fax/modem detection | Preprogrammed |
| Intercept detection | Detects entire sequence of the North American tri-tone. |
| Other SIT sequences can be programmed. | |
| Dial tone detection | |
| before dialing | Application enable/disable; supports up to three different user definable dial tones; programmable dial tone drop out debouncing. |
| TONE DIALING: | |
| DTMF digits | 0 to 9, *, #, A, B, C, D; 16 digits per Bellcore LSSGR Sec 6, TR-NWT-506 |
| MF digits | 0 to 9, KP, ST, ST1, ST2, ST3 |
| Frequency variation | Less than ±1 Hz |
| Rate | 10 digits/s max., configurable by parameter‡ |
| Level | –4.0 dBm per tone, nominal, configurable by parameter‡ |
| PULSE DIALING: | |
| 10 digits | 0 to 9 |
| Pulsing rate | 10 pulses/s, nominal, configurable by parameter‡ |
| Break ratio | 60% nominal, configurable by parameter‡ |
| ANALOG DISPLAY SERVICES INTERFACE (ADSI): | |
| FSK generation per Bellcore TR-NWT-000030. | |
| CAS tone generation and DTMF detection per Bellcore TR-NWT-001273. | |
Hardware System Requirements
Office :F2 Everest, 7th Flr,
Tardeo Rd,Mumbai 400 034. Tel. : +91-22-2352 0968, 5660 3222, 2352
2050 Fax.: +91-22-2351 6881
E-mail : dialogic@foremost-systems.com