Features and Benefits
Applications
The four-line D/4PCI board is useful for small- and medium-sized enterprise computer telephony (CT) applications that require high-performance, cost aggressive voice processing but don't need the large-scale system sophistication of SCbus or CT Bus* based products. The D/4PCI board uses the same application programming interface (API) as its predecessors, making it easy to scale existing applications to take advantage of its power and features. The D/4PCI board has improved voice quality and automatic gain control (AGC) so that even the weakest telephone signals can be recorded and replayed with clarity.
The D/4PCI board uses digital signal processor (DSP) voice processing technology, making it ideal for server-based CT systems - particularly under the Windows operating systems. Windows support includes TAPI and WAVE APIs which facilitate call control, recording, and playback of voice messages under the Microsoft Windows Open Services Architecture (WOSA)* and lets developers quickly develop robust unified messaging applications. The D/4PCI voice processing board gives Windows NT application developers a powerful platform for creating sophisticated interactive voice response (IVR) applications for the small- and medium-sized enterprise market. Caller ID support lets applications such as IVR receive calling party information via a telephone trunk line. Caller ID is supported for North America (CLASS* protocol), the United Kingdom (CLI protocol), and in Japan (CLIP protocol).
The Global Dial Pulse Detection (DPD) algorithm is available for the D/4PCI board, enabling applications to be deployed in countries with limited touchtone telephone service. Global DPD is optimized for a number of countries and provides superior dial-pulse detection.
With all of these advanced features in a half-size PCI board footprint, the D/4PCI board is optimal for client or small server system development. The board offers DSP power and memory capacity that provide a base level of performance for requirements as well as the "head room" for future application expansion via software-based technologies.
Configurations
Use the D/4PCI board to build sophisticated messaging and IVR systems with optional technologies such as automatic speech recognition (ASR), TTS, Global DPD, and PBX Expert software. The D/4PCI board shares a common hardware and firmware architecture with other Intel® Dialogic® voice boards for maximum flexibility and scalability. More ports and new features can be added to a solution while protecting your original investment in hardware and application code. Applications can be ported to higher line density platforms with only minimum modifications.
The D/4PCI board installs in Intel® processor-based computers (Intel486® or Pentium® processor based PC platforms) and provides for building integrated, non-CT Bus voice solutions, scalable from 4 to 64 ports.
Software Support
The D/4PCI board is supported by the System Software and software development kit (SDK) for Windows NT. The SDK contains all the documentation, demonstration code, and tools necessary for developing complex multichannel applications.
Functional Description
The D/4PCI voice processing board builds on the patented dual-processor architecture that combines the signal processing capabilities of a DSP with the decision-making and data movement functionality of a general-purpose control microprocessor by using faster processors and considerably more memory. This dual-processor approach offloads many low-level decision-making tasks from the host computer, thus enabling 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 the four loop start interfaces receive 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 lets applications go off-hook any time during ring cadence without damaging the board.
Part of the telephone interface for the D/4PCI board includes an on-hook audio path that detects Caller ID information. Depending on the level of service offered by the local telephone provider, Caller ID can include the date, time, caller's telephone number, and in some enhanced Caller ID environments, the name of the person calling. The on-hook audio path can also detect touchtones while the line is on-hook. This capability lets the board operate behind PBXs that require on-hook touchtone detection for their signaling.
Inbound telephony signaling (ring detection and loop current detection) are 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 COder/DECoder (CODEC) circuit. The CODEC filters, samples, and digitizes the inbound analog audio signal and passes this digitized audio signal to a Motorola DSP.
Based on Spring Ware firmware loaded in DSP RAM, the DSP performs the following signal analysis and operations on this incoming data:
For outbound data, the DSP performs the following operations:
The dual-processor combination also performs the following outbound dialing and call progress monitoring
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 popular 11 kHz, 8-bit linear multimedia WAVE format is also supported on the D/4PCI voice board.
Outbound processing is the reverse of inbound processing. The DSP processed speech is transmitted by the control microprocessor to the host PC for disk storage. When replaying a stored file, the microprocessor receives the voice information from the host PC and passes it to the DSP, which converts the file into digitized voice. The DSP sends the digitized voice to the CODEC to be converted into analog voice and then to the line interface for transmission to the telephone network.
The on-board
microprocessor controls all operations of the D/4PCI board via a local bus
and interprets and executes commands from the host PC. This microprocessor
handles real-time events, manages data flow to the host PC to provide faster
system response time, reduces PC host processing demands, processes DTMF and
telephony signaling before passing them to the application, and frees the
DSP to perform signal processing. Communications between this microprocessor
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. All D/4PCI boards installed in the PC share
the same interrupt line. When the system is initialized, Spring Ware firmware
is downloaded from the host PC to the on-board 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.
Technical Specifications**
Number of ports 4
Maximum boards/system 16
Analog network interface On-board loop start interface circuits
Microprocessor Intel® 80C188
Digital signal processor Motorola DSP56002*
HOST INTERFACE:
Bus compatibility PCI (complies with PCISIG Bus Specification, Rev. 2.1)
PCI bus speed 33 MHz
Shared memory 8 KB page, PnP selectable on 16 KB boundaries
Base addresses Selected by PCI BIOS
Interrupt level One IRQ (IntA) shared by all boards
TELEPHONE INTERFACE:
Trunk Type Loop Start (or Ground Start for answer only)
Impedance 600 Ohm for D/4PCI. Matching complex impedance specified in CTR-21
for D/4PCI-Euro.
Ring detection 25 Vrms min., 15.3 Hz to 68 Hz, 150 Vrms max.
Loop current range 20 mA to 120 mA, DC (polarity insensitive), D/4PCI-Euro
current limits at 60 mA per CTR-21 specifications
Crosstalk coupling -80 dB at 3 kHz channel to channel
Frequency response 300 Hz to 3400 Hz ±3 dB (transmit and receive)
Connector Four RJ-11
ENVIRONMENTAL REQUIREMENTS:
+5 VDC 650 mA
+12 VDC 55 mA
-12 VDC 53 mA
Operating temperature 0°C to +50°C
Storage temperature -20°C to +70°C
Humidity 8% to 80% noncondensing
Form factor PC AT (PCI); 6.9 in. long, 0.75 in. wide, 3.85 in. high (excluding
edge connector)
REGULATORY CERTIFICATIONS:
United States FCC part 68 ID#: EBZUSA-65588-VM-E
REN: 1.0B
UL: E143032
Canada IC CS-03, CSA C22.2 No. 950
Load number: 5
ULC: E143032
Warranty Lifetime
Spring
Ware Firmware Technical Specifications**
AUDIO SIGNAL:
Receive range -50 dBm to -13 dBm (nominal), for average speech signals‡
configurable by parameter†
Automatic gain control Application can enable/disable above -30 dBm results
in full scale recording, configurable by parameter†.
Silence detection -40 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 data
rates. Adjustment range: ±50%, adjustable through application or programmable
DTMF control.
WAVE AUDIO:
Supports 11 kHz linear PCM, 8-bit mono mode (available only when running Windows)
DTMF TONE DETECTION:
DTMF digits 0 to 9, *, #, A, B, C, D per Bellcore LSSGR* Sec 6
Dynamic range Programmable, default set at -36 dBm to +0 dBm per tone
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
GLOBAL TONE DETECTION:
Tone type Programmable for single or dual
Max. number of tones Application dependent
Frequency range Programmable within 300 Hz to 3500 Hz
Max. 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 +0 dBm per tone
GLOBAL TONE GENERATION:
Tone type Generate single or dual tones
Frequency range Programmable within 200 Hz 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-000506
Signaling mechanism Complies with Bellcore LSSGR Sec 6, TR-NWT-000506
Dynamic range for detection -25 dBm to -1 dBm per tone
Acceptable twist 6 dB
Acceptable freq. 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 uses both frequency and cadence detection. Application can select
frequency only for faster detection in specific environments.
Ringback detection Default setting designed to detect 83 out of 87 unique
ringback tones used in 96 countries as specified by CCITT Rec E., Suppl #2.
Uses 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 tritone.
Other SIT sequences can be programmed.
Dial tone detection before dialing Application enable/disable. Supports up
to three different user-definable dial tones. Programmable dialtone drop-out
debouncing.
TONE DIALING:
DTMF digits 0 to 9, *, #, A, B, C, D; 16 digits per Bellcore LSSGR Sec 6,
TR-NWT-000506
MF digits 0 to 9, KP, ST, ST1, ST2, ST3
Frequency variation ±0.5% of nominal frequency
Rate 10 digits/s max., configurable by parameter†
Level -5 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 CALLER IDENTIFICATION:
Applicable standards Bellcore TR-TSY-000030
Bellcore TR-TSY-000031
TAS T5 PSTN1 ACLIP: 1994 (Singapore)
British Telecom SIN 242 (Issue 01)
British Telecom SIN 227 (Issue 01)
Japan NTT CLIP
Modem standard Bell 202 or V.23, serial 1200 b/s (simplex FSK signaling)
Receive sensitivity -48 dBm to -1 dBm
Noise tolerance Minimum 18 dB SNR over 0 dBm to -48 dBm dynamic range for
error-free performance
Data formats Single Data Message (SDM) and Multiple Data Message (MDM) formats
via API calls and commands
Line impedance 600 Ohm for D/4PCI. Matching complex impedance specified in
CTR-21 for D/4PCI-Euro.
Message formats ASCII or binary SDM, MDM message content
ANALOG DISPLAY SERVICES INTERFACE (ADSI):
FSK generation per Bellcore TR-NWT-000030. CAS tone generation and DTMF detection
per Bellcore TR-NWT-001273.
*All company
names, products, and services mentioned are the trademarks or registered trademarks
of their respective owners.
** All specifications
are subject to change without notice.
† Analog
levels: 0 dBm0 corresponds to a level of +3 dBm at tip-ring analog point.
Values vary depending on country requirements; contact your Technical Sales
Representative.
‡ Average
speech mandates +16 dB peaks above average and preserves -13 dB valleys below
average.
Hardware
System Requirements
80486, or Pentium® processor-based computer. Operating system hardware
requirements vary according to the number of channels being used.
