Dialogic D/41EPCI
4-Port PCI Voice Processing with H.100 Interoperability
Features and Benefits
- Four independent
voice processing ports in a single PCI slot for low- to
medium-density enterprise computer telephony applications
- With approvals
in North and South America and Japan, the D/41EPCI cost-effectively
expands the application's ability to serve several global
markets
- SCbusconnectivity
enables applications to access additional resources to expand
its functionality to include fax, text-to-speech (TTS),
and automatic speech recognition (ASR)
- H.100 connector
allows developers to take advantage of the industry-standard
CT Bus* and increases the board's capacity to interoperate
with other CT Bus compatible boards
- Plug and Play*
ready. Simplifies hardware installation by eliminating DIP
switches and jumper settings and enabling software controlled
configuration
- Configure multiple
boards in a single chassis, PCI bus or mixed PCI/ISA bus,
for easy and cost-effective system expansion up to 64 analog
ports
- Downloadable signal
and call processing firmware, provides field proven performance
based on over 4 million installed ports with access to future
feature enhancements
- DTMF (touch tone)
reception provides reliable detection during voice playback
- allows callers to "type-ahead" through menus
- A-law or µ-law
voice coding at dynamically selectable data rates, 24 Kb/s
to 64 Kb/s, selectable on a channel-by-channel basis for
optimal tradeoff between disk storage and voice quality
- International Caller
ID capability via on-hook audio path. Supports Bellcore
CLASS*, UK CLI, and other international protocols
- Patented outbound
call progress analysis monitors outgoing call status quickly
and accurately
- C-language application
program interface (API) for Windows NT*
- Supports PBX Expert32,
a free software utility that simplifies switch integration
- Optional onboard
Global Dial Pulse Detection (DPD) feature enables callers
with non-touch tone phones to access applications without
additional "pulse-to-tone converter" equipment
- Supported by CT
Media™ Telephony Server software, a standards-based development
platform for building scalable applications that can run
with other vendors' products on converged communications
servers
Applications
- Voice messaging
- Interactive voice
response
- Debit card and
international call back
- Audiotex
- Telemarketing/call
center
- Dictation
- Auto dialers
- Notification systems
- Online data entry/query
The D/41EPCIis a four-port analog voice processing board ideal
for developers building enterprise voice messaging and IVR applications
for the global market. The D/41EPCI provides four telephone
line interface circuits for direct connection to analog loop
start lines. A dual-processor architecture, comprising a DSP
(digital signal processor) and a general-purpose microprocessor,
handles all telephony signaling and performs DTMF (touch tone)
and audio/voice signal processing tasks. The open architecture
enables developers to build CT solutions using products from
multiple vendors. And since you can install multiple D/41EPCI
boards in a single PC chassis, you can build systems scaling
up to 64 ports.
Downloaded Spring
Ware firmware algorithms, executed by the onboard DSP, provide
variable voice coding at 24 and 32 Kb/s ADPCM, and 48 and
64 Kb/s µ-law or A-law PCM, as well as µ-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. Spring Ware firmware also provides reliable
DTMF detection, DTMF cut-through, and talk off/play off suppression
over a wide variety of telephone line conditions.
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 touch tone telephone service. Global DPD can be optimized
on a country-by-country basis to provide superior dial pulse
detection wherever it is used.
The D/41EPCI board
performs the following functions in real time on four independent
channels:
- Connects directly
to analog loop start telephone lines
- Offers application
controlled call answering
- Detects touch tones
- Plays voice messages
to a caller and digitizes, compresses, and records voice
signals
- Places outbound
calls and automatically monitors their progress
Configurations
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 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 (Intel386™, Intel486™, or
Pentium® processor-based PC platforms). 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 selection of Intel products to build scalable,
reliable, and economical CT solutions.
Mixed PCI/ISA
Configuration Example
The D/41EPCI board
can operate within a mixed chassis containing PCI and ISA
products. The design of the D/41EPCI incorporates an H.100
connector to simplify connection to 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 supported
by the System Software and Software Development Kit for Windows
NT* (Native). This package contains a set of tools for developing
complex multichannel applications.
For added flexibility,
the D/41EPCI is also supported in CT Media™ Telephony Server
software. This resource management software makes application
development easier, and enables applications from different
suppliers written to standard APIs like ECTF S.100* to work
together on a converged communications system. CT Media makes
this possible by managing technology resources (boards and
host-based technologies) within the server, and by providing
basic switching functions to multiple client applications.
CT Media software
also runs on Windows 2000*. A minimum of 35 MB should be used
in any field-deployable system - with 64 MB or more recommended.
A Pentium® processor-class system is recommended, as well
as a fast disk I/O system (i.e., SCSI II).
Functional Description
The D/41EPCI uses
a 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 offloads 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 Spring Ware
firmware loaded in DSP SRAM, the DSP performs the following
signal analysis and operations on this incoming data:
- Automatic gain
control to compensate for variations in the level of the
incoming audio signal
- Applies an ADPCM
(Adaptive Differential Pulse Code Modulation) or PCM (Pulse
Code Modulation) 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 tone
- Silence detection
to determine whether 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 outbound dialing
and call progress monitoring:
- Transmits an off-hook
signal to the telephone network
- Dials out (makes
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 fax machine, or a modem
The D/41EPCI also supports optional Global Dial Pulse Detection
(DPD) Software that recognizes dial pulse digits even in the
most difficult telephony environments.
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,
Spring Ware 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 Board Locator Technology (BLT)
installation.
Technical Specifications**
| Number
of ports |
4 |
| Maximum
boards/system |
16 |
| Analog
network interface |
Onboard
loop start interface circuits |
| Resource
sharing bus |
SCbus
OR CT Bus |
| Control
microprocessor |
Intel®
80C186 @ 16 MHz |
| Digital
signal processor |
Motorola
DSP56002* @ 49 MHz, with 32 K word private, 0 wait state
SRAM |
| HOST
INTERFACE: |
| Bus
compatibility |
PCI.
Complies with PCISIG Bus Specification, Rev. 2.1 |
| Bus
speed |
33
MHz max |
| Bus
mode |
32
bit to 16 bit conversion in target mode |
| Shared
memory |
64
KB page |
| I/O
ports |
None |
| TELEPHONE
INTERFACE‡: |
| Trunk
type |
Loop
start |
| Loop
current range |
20
to 120 mA |
| Impedance |
600
Ohms nominal |
| Ring
detection |
15
Vrms min, 13 to 68 Hz (configurable by parameter) |
| Echo
return loss |
Configurable
by software parameter |
| Cross
talk coupling |
Less
than -70 dB at 1 KHz channel to channel |
| Receive
signal/noise ratio |
70
dB referenced to -15 dBm |
| Frequency
response |
200
Hz to 3400 Hz ±3 dB (transmit and receive) |
| Connector |
Four
RJ-11 type |
| POWER
REQUIREMENTS: |
| +5
VDC |
1.22
A, max. |
| +12
VDC |
140
mA max. |
| –12
VDC |
110
mA max. |
| Operating
temperature |
0°
C to +50° C |
| Storage
temperature |
–20°
C to +70° C |
| Humidity |
8%
to 80% non-condensing |
| Form
factor |
PCI
long card, 12.3 in. long. (without edge retainer) or 13.3
in. long (with edge retainer), 0.79 in. wide (total envelope),
3.87 in. high (excluding edge connector) |
| SAFETY
AND EMI CERTIFICATIONS: |
| United
States |
FCC
Part 15 class A; FCC Part 68 EBZUSA-75385-VM-T |
| |
UL:
E96804 UL1950 |
| Canada |
DOC:
885-5542A |
| |
For
specific country approval designation, see the Global
Approvals list or contact a Sales Engineer |
| Warranty |
3
years standard |
Spring Ware Firmware
Technical Specifications**
| AUDIO
SIGNAL: |
| Receive
range |
–50
to –13 dBm (nominal), for average speech signals
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.,
Supplement #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.,
Supplement #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. |
* 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 Intel Sales Engineer.
1 Average speech mandates +16 dB peaks above average and preserves
-13 dB valleys below average.
Hardware System
Requirements
- 80386, 80486, or
Pentium® processor-based PCI bus or mixed PCI/ISA bus PC
or compatible computer
- Operating system
hardware requirements vary according to the number of channels
being used
- System must comply
with PCISIG Bus Specification Rev. 2.1 or later.
Additional Components
- Optional multi-drop
CT Bus cable
- Optional CT Bus/SCbus
adapter
|
