Dialogic D/160SC-LS
16-Port Voice Processing and Analog Interface Board
Features and Benefits
- High-density
analog interface voice processing platform enables
system integrators and developers to lower costs by
incorporating more ports per chassis, by using less
expensive desktop-style machines, and by easing configuration/installation
effort
- Sixteen independent
loop start telephone interfaces, combined with 16
channels of voice processing in one ISA slot, provide
effective solutions for building high-density applications
- Create more
cost-effective switching solutions via access to the
SCbus with its 1,024 time slot capability; SCxbus
interbox communications provides the capability to
build higher density systems and large, multinode
systems
- Downloadable
signal and call processing firmware provides easy
feature enhancement and field-proven performance based
on over four million installed ports
- DTMF (touch
tone) provides reliable detection during voice playback
- allows callers to "type-ahead" through menus
- Optional
Global Dial Pulse Detection (GDPD) feature enables
callers without touch tone phones to access applications.
No additional "pulse to tone converter" hardware is
needed.
- Two independent
Motorola 56002* digital signal processors (DSPs),
clocked at 65 MHz; each with private, high-speed SRAM,
permit execution of Spring Ware signal processing
algorithms
- Intel® 486
GX microprocessor offloads call processing tasks from
host PC, giving more power to the application
- Board Locator
Technology eliminates confusing DIP switch or jumper
settings and simplifies installation
- C language
application program interfaces (APIs) for MS-DOS*,
UNIX*, and Windows NT* shorten your development cycle
so you can get your applications to market faster
- Caller ID
capability for "screen pop" applications (supports
Bellcore CLASS* Protocols)
- Configure
multiple boards in a single PC (ISA bus) for easy
and cost-effective system expansion on the best computing
platform that best fits your needs
- Supported
by CT Media™ 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
- Voice/audio response systems
- Audiotex
- Operator services
- Telemarketing/call center
- Call logging
- Dictation
- Auto dialers
- Notification systems
- On-line data entry/query
Voice products offer a rich set of advanced features, including
digital signal processor (DSP) technology and signal processing
algorithms, for building the core of any computer telephony
(CT) system. With expansion boards and a variety of channel
densities to choose from, you can integrate voice products
easily into exactly the type of system you require at a price
and performance level unmatched in the CT industry.
The D/160SC-LS board provides 16 channels of call processing
and loop start interfaces in a single PC slot. A unique dual-processor
architecture composed of DSPs and a general-purpose microprocessor
handles all telephony signaling and performs all DTMF (touch
tone) and audio/voice signal processing tasks. Downloaded
firmware algorithms provide variable voice coding at 24 and
32 Kb/s Adaptive Differential Pulse Code Modulation (ADPCM),
and 48 and 64 Kb/s Pulse Code Modulation (PCM) µ-law or A-law.
Sampling rates and coding methods are selectable on a channel-by-channel
basis. Applications can dynamically switch sampling rate and
coding method to optimize data storage or voice quality as
the need arises. Firmware also provides reliable DTMF detection,
DTMF cut-through, and talk off/play off suppression over a
wide variety of telephone line conditions.
Offered as a software option, Global Dial Pulse Detection
(GDPD) converts rotary pulses to DTMF in countries that have
limited touch tone telephone service. Global DPD is also optimized
in several countries to provide superior dial pulse detection
and conversion.
The D/160SC-LS voice board:
- Connects to 16 analog loop-start telephone channels
- Answers calls
- Detects touch tone
- Plays voice messages to a caller
- Digitizes, compresses, and records voice signals
- Places outbound calls and automatically reports the result
And, it does these things in real time on all channels.
Configurations
Use the D/160SC-LS board to develop sophisticated, multifunction
CT systems incorporating capabilities such as voice processing,
speech recognition, and text-to-speech (TTS). The D/160SC-LS
board shares a common hardware and firmware architecture with
other Intel® Dialogic® SCbus based boards for maximum flexibility
and scalability. You can add features or grow the system while
protecting your investment in hardware and application code.
Applications can be easily ported to lower or higher line-density
platforms, with only minimal modifications.
The D/160SC-LS board installs in IBM PC AT* (ISA bus) and
compatible computers (80386, 80486, and Pentium® processor-based
PC platforms). The D/160SC-LS board occupies a single expansion
slot and up to 16 boards can be configured in a system with
each board sharing the same interrupt level. 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.
Software Support
The D/160SC-LS board is supported by System Software and
Software Development Kits (SDKs) for Windows NT, UNIX, and
MS-DOS. These packages contain a set of tools for developing
complex multichannel applications.
For added flexibility, the D/160SC-LS 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 for converged communications.
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 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-class processor is recommended,
as well as a fast (i.e., SCSI II) disk I/O system.
Functional Description
The D/160SC-LS board connects 16 analog (loop start) telephone
lines to 16 on-board call processing resources or to other
resources via the SCbus. This board provides:
- Interference suppression
- Ring and on-hook/off-hook signaling control
- Tone detection and generation
- Digitization and playback of voice files
The signals from the 16 loop start telephone lines connected
to the D/160SC-LS board first pass through a telephone line
interface that provides transient protection and electromagnetic
interference (EMI) suppression (see block diagram). These
telephone line interfaces use 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.
The telephone line interface applies the inbound signal including
the ring or other in-band signaling to analog/digital inputs
of a signal converter called a COder/DECoder (CODEC) for sampling
and digitization. These digitized signals are sent to an SC2000
chip where they are routed via the SCbus either to an onboard
DSP or to an external resource on any of the 1,024 SCbus time
slots. This enables the application to reroute calls to any
added resource, such as speech recognition, facsimile, or
TTS.
Part of the D/160SC-LS board's telephone interface 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 the name of the person calling (in some enhanced
Caller ID environments). The on-hook audio path can also detect
touch tones while the line is on-hook. This capability lets
you use the D/160SC-LS board behind PBXs that require on-hook
touchtone detection for their signaling.
When onboard call processing resources are used, the network
signals are extracted and passed to the onboard control processor,
which can change channel status and communicate channel events
to the application via a shared RAM and the host PC ISA bus.
The DSP processes the digitized voice data based on Spring
Ware firmware loaded in code/data RAM. Each DSP performs the
following signal analysis and operations.
On the incoming data:
- Applies automatic gain control to compensate for variations
in the level of the incoming audio signal
- Applies an ADPCM or 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 tone
- Detects silence to determine whether the line is quiet
and the caller is not responding
For outbound data:
- 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 facsimile,
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 DSP-processed speech is transmitted by the control processor
to the host PC for disk storage. The D/160SC-LS board can
record incoming signals with the telephony interface in either
the high-impedance on-hook state or the normal off-hook state.
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
the digitized voice responses to the CODEC, which is controlled
by a pair of SC2000 chips. The CODEC converts the digitized
voice into analog voice and transmits the voice response to
the caller via the telephone line interface. (Although this
product is capable of recording incoming signals in an on-hook
state, applications such as call logging should use the D/160SC-LS-HiZ,
which is specifically designed for analog high-impedance recording.)
When the system is initialized, firmware is downloaded from
the host PC to the board. It controls all board operations.
Spring Ware gives the board all of its intelligence and enables
easy feature enhancements and upgrades.
The onboard control processor manages all operations of the
D/160SC-LS board via a local bus and interprets and executes
commands from the host PC. This processor 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.
Communication between the 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 ISA bus. All operations
are interrupt-driven to meet the demands of real-time systems.
The Board Locator Technology circuit operates in conjunction
with a rotary switch that eliminates the need to set confusing
jumpers or DIP switches.
Technical Specifications*
| Number of
ports |
16 |
| Max. boards/system |
6 (MS-DOS);
16 (UNIX, Windows NT). Number may be limited by application
and system performance. |
| Analog network
interface |
On-board loop
start interface |
| Resource sharing
bus |
SCbus or PEB |
| Control microprocessor |
Intel 80486
GX @ 32.768 MHz, 0 wait state |
| Digital signal
processors |
Two Motorola
DSP56002 @ 49 - 66 MHz, each with 32 K word private, 0
wait state SRAM |
| HOST
INTERFACE: |
| Bus compatibility |
IEEE P996 ISA
compatible (IBM PC AT) |
| Bus speed |
8 MHz typical |
| Bus mode |
Automatically
configures to 8- or 16-bit transfer mode |
| Shared memory |
32 Kbytes page |
| Base addresses |
8000h to E800h,
on 32K boundaries. All D/SC boards share the same base
address. Shared memory is page mapped in/out dynamically
as needed. |
| Interrupt
level |
IRQ 2/9, 3,
4, 5, 6, 7, 10, 11, 12, 14, 15, software selectable. One
IRQ line must be shared by all D/SC boards. |
| I/O ports |
None |
| TELEPHONE
INTERFACE**: |
| Trunk type |
Loop start;
also works with ground start for inbound applications |
| Impedance |
600 Ohms nominal |
| Loop current
range |
20 to 120 mA |
| Ring detection |
40 to 130 Vrms,
15.3 to 68.0 Hz |
| Echo return
loss |
20 dB min. |
| SNR |
-40 dB |
| Cross talk
coupling |
-70 dB |
| Speech digitization |
64 Kb/s, µ-law
PCM (companding to ADPCM performed in Spring Ware) |
| Freq. response |
300 to 3400
Hz ± 3 dB |
| Connector |
DB-37 |
| POWER
REQUIREMENTS: |
| +5 VDC |
1.5 A max. |
| -12 VDC |
250 mA max. |
| Operating
temperature |
0°C to +50°C |
| Storage temperature |
-20°C to +70°C |
| Humidity |
8% to 80% noncondensing |
| Form factor |
PC AT, 13.3
in. long., 4.5 in. high (excluding edge connector) |
| SAFETY
& EMI CERTIFICATIONS: |
| United States |
UL: 1459, with
optional adapter |
| Canada |
CSA: 225 (by
UL) |
| Estimated
MTBF |
163,000 Hours
per Bellcore Method I |
| Warranty |
3 years standard |
Spring Ware Technical Specifications
| AUDIO
SIGNAL: |
| Receive range |
-40 to +2.5
dBm0 nominal, configurable by parameter** |
| Automatic
Gain Control |
Application
can enable/disable. Above -18 dBm0 results in full scale
recording, configurable by parameter** |
| Silence detection |
-38 dBm nominal,
software adjustable** |
| Transmit level
(weighted average) |
-9 dBm0 nominal,
configurable by parameter** |
| Transmit volume
control |
40 dB adjustment
range, with application definable increments and legal
limit cap |
| 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 OKI®
ADPCM @ 6 kHz sampling
32 Kb/s OKI® 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 |
| DTMF
TONE DETECTION: |
| DTMF digits |
0 to 9, *,
#, A, B, C, D per Bellcore LSSGR Sec 6 |
| Dynamic range |
-36 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. |
| Acceptable
twist and frequency variation |
Meets Bellcore
LSSGR Sec 6 and EIA 464 requirements |
| Noise tolerance |
Meets Bellcore
LSSGR Sec 6 and EIA 464 requirements for Gaussian, impulse
and power line noise tolerance. |
| Cut-through |
Local echo
cancellation permits 100% detection with a >4.5 dB return
loss line. |
| 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 |
| Max. number
of tones |
Application
dependent |
| Frequency
range |
Programmable
within 200-4000 Hz |
| Max. frequency
deviation |
Programmable
in 5 Hz increments. |
| Frequency
resolution |
± 5 Hz. Separation
of dual frequency tones is limited to 62.5 Hz at a signal-to-noise
ratio of 20 dB. |
| Timing |
Programmable
cadence qualifier, in 10 ms increments |
| Dynamic range |
Programmable,
default set at -6 dBm0 to +3 dBm0 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 dBm0 to
-3 dBm0 per tone nominal, programmable |
| MF SIGNALING: |
R1 |
| 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 dBm0 to
+3 dBm0 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 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 |
>99% based
on tests on a database of real world calls in North America.
Performance in other markets may vary. |
| Positive Voice
Detection speed |
Detects voice
in as little as 1/10th of a second. |
| Positive Answering
Machine Detection™ accuracy |
>85% based
on tests on a database of real world calls in North America.
Performance in other markets may vary. |
| Fax/modem
detection |
Pre-programmed
|
| Intercept
detection |
Detects entire
sequence of the North American tri-tone.Other intercept
tone 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 per Bellcore LSSGR Sec 6TR-NWT-000506 |
| Frequency
variation |
Less than ±
1 Hz |
| Rate |
10 digits/s,
configurable by parameter** |
| Level |
-4.0 dBm0 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) |
| Modem standard |
Bell 202 or
V.23, serial 1200 bits/sec (simplex FSK signaling) |
| Receive sensitivity |
-48 dBm (-50
dBv) to -1 dBm |
| Noise tolerance |
Minimum 18
dB SNR over 0 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 |
AC coupled
600 Ohm (@ 1.8 kHz) termination during Caller ID on-hook
detection interval |
| 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 specifications are subject
to change without notice.
*All company names, products, and services mentioned are
the trademarks or registered trademarks of their respective
owners.
**Configurable to meet country specific PTT requirements.
Actual specification may vary from country to country for
approved products.
Hardware System Requirements
- 80386, 80486 or Pentium® processor, IBM PC AT (ISA) bus
or compatible computer. Operating system hardware requirements
vary according to the number of channels being used.
Additional Components
- Multidrop SCbus cable
- Required: Station Adapter, 37-pin to 50-pin cable
- Optional: UL 1459 compliance adapter
|
