Dialogic D/4PCI
4-Port Voice Processing for Small and Medium Enterprise
Applications
Features and Benefits
- Build flexible,
cost-effective messaging and voice response platforms
for small- and medium-sized enterprise applications
- Supports
Windows NT* including TAPI/WAVE*
- CTR-21 approvals
mean expanded markets
- Caller ID
lets applications perform intelligent call handling
- Delivers
advanced call processing features and enables competitive
differentiation by supporting software-based features
such as
- Global Dial Pulse Detection
- PBX Expert tone characterization utility
- Provides
reliable DTMF detection during voice playback, letting
callers "type-ahead" through voice menus for quicker
completion of call transactions
- Ensures reliability
via call progress analysis which monitors outgoing
call status quickly and accurately
- Flexible
voice coding at dynamically selectable data rates,
24 to 64 Kb/s, selectable on a channel-by-channel
basis for optimal tradeoff in disk storage and voice
quality
- Superior
voice quality through enhanced telephone circuitry
and automatic gain control
- Half-size
PCI form factor enables developers to build cost-effective
systems by using the most up-to-date industry-standard
chassis. The ability to mix form factors offers a
cost-effective transition to the PCI form factor.
- Compatible
with legacy telephone switches in the United Kingdom
and Northern Europe that use Earth Recall signaling
Applications
- Networked voice messaging
- Automated attendant
- Interactive voice response
- Enhanced messaging
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:
- uses AGC to compensate for variations in the level
of the incoming audio signal. The D/4PCI board also
includes special circuitry to detect and amplify extremely
weak line signals due to harsh telephone line conditions
or back-to-back local loops often found in 800 (toll-free)
service environments
- applies an adaptive differential pulse code modulation
(ADPCM) or pulse code modulation (PCM) algorithm to
compress the digitized voice and save disk storage
space
- detects the presence of tones - DTMF, MF, or an
application-defined single- or dual-frequency tone
- uses silence detection to determine when the line
is quiet and the caller is not responding
For outbound data, the DSP performs the following operations:
- expands stored, compressed audio data for playback
- adjusts the volume and rate of speed of playback
upon application or user request
- generates tones — DTMF, MF, or any application-defined
general-purpose tone
The dual-processor combination also performs the following
outbound dialing and call progress monitoring
- transmits an off-hook signal to the telephone network
- dials out (places an outbound call)
- monitors and reports results: line busy or congested;
operator intercept; ring, no answer; or if the call
is answered, whether answered by a person, an answering
machine, a facsimile machine, or a modem
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.
|