D/160SC-LS™
16-PORT VOICE PROCESSNG & ANALOG INTERFACE BOARD
FEATURES & BENEFITS
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Highest density analog interface voice processing platform in the industry
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
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16 independent loop start telephone interfaces, combined with 16 channels
of voice processing in one ISA slot, provide effective solutions for building
high-density applications
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Create more cost-effective switching solutions via access to the SCSA™
SCbus™ with its 1024 time slot capability; SCxbus™ interbox communications
provides the capability to build higher density systems and large, multi-node
systems
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Downloadable signal and call processing firmware by Dialogic, called SpringWare™,
provides easy feature enhancement and field proven performance based on
over two million installed ports
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PerfectDigit™ DTMF (touchtone) provides reliable detection during voice
playback - allows callers to "type-ahead" through menus
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Optional Global Dial Pulse Detection (DPD) feature enables callers without
touchtone phones to access applications. No additional "pulse to tone converter"
hardware is needed
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Two independent Motorola DSP56002 Digital Signal Processors, clocked at
65 MHz; each with private, high speed SRAM, permit execution of high performance
SpringWare signal processing algorithms
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Intel 486 GX microprocessor off-loads call processing tasks from host PC,
giving more power to the application
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Board Locator Technology™ eliminates confusing DIP switch or jumper settings
and simplifies installation
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C language application program interfaces (APIs) for MS-DOS®, UNIX®,
Solaris™ and Windows NT™ shorten your development cycle so you can get
your applications to market faster
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High impedance, on-hook record capability enables high-density call logging
and transaction record applications
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Caller ID capability for "screen pop" applications (supports Bellcore CLASS
Protocols)
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Configure multiple boards in a single PC (ISA bus) for easy and cost effective
system expansion on the best computing platform that fits your needs
APPLICATIONS
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Voice messaging
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Interactive voice response
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Voice/audio response systems
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Audiotex
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Operator services
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Telemarketing/call center
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Call logging
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Dictation
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Auto dialers
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Notification systems
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On-line data entry/query
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 comprising
DSPs (Digital Signal Processors) and a general purpose microprocessor,
handles all telephony signaling and performs all DTMF (touchtone) and audio/voice
signal processing tasks. The D/160SC-LS board, as a member of the DIALOG/HD™
High Density Voice and Switching products, is based upon Signal Computing
System Architecture™ (SCSA). SCSA provides an open architecture that enables
developers to use products from multiple vendors to build a unified computer
telephony solution. SCSA provides features such as distributed switching,
logical addressing, and location independent resource management.
Downloaded firmware algorithms, SpringWare, executed by the on-board
DSPs, provide variable voice coding at 24 and 32 Kb/s ADPCM, and 48 and
64 Kb/s µ-law or A-law PCM, with G.726 32 Kb/s coding planned as
a near future enhancement.. 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 which is available for the D/160SC-LS, lets you use the product
in countries that have limited touchtone telephone service. Offered as
a software option, Global DPD can also 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 computer telephony system. With industry-standard
ISA bus expansion boards and a variety of channel densities to choose from,
you can integrate Dialogic voice products easily into exactly the type
of system you require at a price and performance level unmatched in the
computer telephony industry.
In real time on all channels, the D/160SC-LS voice board
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connects to 16 analog loop-start telephone channels
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automatically answers calls
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detects touchtone
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plays voice messages to a caller
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digitizes, compresses, and records voice signals
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places outbound calls and automatically reports the result
CONFIGURATIONS
Use the D/160SC-LS board to develop sophisticated, multifunction
computer telephony systems incorporating capabilities such as voice processing,
speech recognition, and text-to-speech. The D/160SC-LS board shares a common
hardware and firmware architecture with other Dialogic SCbus-based boards
for maximum flexibility and scalability. Features can be added or 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/160SC-LS board installs in IBM® PC AT® (ISA bus)
and compatible computers (80386™, 80486™ and Pentium™-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 is supported by Dialogic System Software and SDK
for Windows NT, UNIX, Solaris, and MS-DOS. These packages contain a set
of tools for developing complex multichannel applications.
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:
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interference suppression
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digital-to-analog conversion
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ring and on-hook/off-hook signaling control
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tone detection and generation, and
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digitization and play back 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 EMI (electromagnetic interference) 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 and on-hook/off-hook signals to analog/digital inputs of a signal
converter called a coder/decoder (CODEC) that samples and digitizes these
signals. These digitized signals are sent to an SC2000 chip where they
are routed via the SCbus either to an on-board DSP or to an external resource
on any of the 1024 SCbus time slots. This enables the application to reroute
calls to any added resource, such as speech recognition, facsimile, or
text-to-speech.
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 (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 you use the D/160SC-LS board behind PBXs that require on-hook touchtone
detection for their signaling.
When the on-board call processing resources are used, the network
signals are extracted and passed to the on-board 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 SpringWare firmware
loaded in code/data RAM. Each DSP performs the following signal analysis
and operations:
On the incoming data:
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applies automatic gain control to compensate for variations in the level
of the incoming audio signal
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applies an ADPCM (Adaptive Differential Pulse Code Modulation) or PCM (Pulse
Code Modulation) algorithm, or G.726 32Kb/s coding (near future) to compress
the digitized voice and save disk storage space
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detects the presence of tones - DTMF, MF, or an application defined single
or dual tone
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detects silence to determine whether the line is quiet and the caller is
not responding For outbound data:
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expands stored, compressed audio data for playback
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adjusts the volume and rate of speed of playback upon application or user
request
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generates tones - DTMF, MF, or any application-defined general-purpose
tone The dual-processor combination also performs outbound dialing and
call progress monitoring:
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transmits an off-hook signal to the telephone network
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dials out (makes an outbound call)
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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 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 sends the voice response
to the caller via the telephone line interface. When the system is initialized,
SpringWare firmware to control all board operations is downloaded from
the host PC to the board. This downloadable firmware gives the board all
of its intelligence and enables easy feature enhancement and upgrades.
The onboard control processor controls 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.
| D/160SC-LS 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 |
| Control microprocessor |
Intel 80486 GX @ 28.5 MHz, 0 wait state |
| Digital signal processors |
Two Motorola DSP56002 @ 65 MHz, each with 3 word private,
0 wait state SRAM |
| HOST INTERFACE: |
| Bus compatibility |
EEE 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 3 boundaries. All D/SC boards share the
same base address. Shared memory is page mapped in/out dynamically as needed. |
| Interrupt level |
IRQ2/9, 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 SpringWare) |
| Freq. response |
300 to 3400 Hz ± 3 dB |
| Connector |
DB-37 |
| POWER REQUIREMENTS: |
| +5 VDC |
3.0 A max. |
| +12 VDC |
250 mA max. |
| -12 VDC |
125 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. 0.793 in. wide (total "envelope"),
4.5 in. high (excluding edge connector) |
| SAFETY & EMI CERTIFICATIONS: |
| United States |
UL: 1459 |
| Canada |
CSA: 225 (by UL) |
| Estimated MTBF |
200,000 Hours per Bellcore Method I |
| Warranty |
3 years standard |
| D/160SC-LS SPRINGWARE
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 300-3500 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 2B. |
| 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 6
TR-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.
**Configurable to meet country specific PTT requirements.
....Actual specification may vary from
country to country for
....approved products.
HARDWARE SYSTEM REQUIREMENTS
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80386, 80486 or Pentium IBM PC AT (ISA) bus or compatible computer. Operating
system hardware requirements vary according to the number of channels being
used.
ADDITIONAL COMPONENTS
Optional: Station Adapter, 37 pin to 50 pin cable
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