 |
D/120JCT-LS
12-Port PCI Analog Voice Processing Board
Ideal for Multimedia Communications Applications |
 
Features and Benefits
-
Full-size PCI board form factor is compatible with industry standards and
trends
-
With 12 analog telephony ports, the D/120JCT-LS™ board is the first high-density
analog voice processing solution in PCI form factor
-
CTR21 approval allows for easier globalization of solutions and reduces
inventory costs by supporting a single board for multiple countries
-
Support for advanced voice coders used in unified messaging solutions
-
H.100 support ensures a single universal bus standard allowing simplified
expansion to the new industry-standard CT Bus™*
-
Support for Windows NT® and UNIX® operating systems ensure high
reliability in multitasking environments*
-
Caller ID and Global Dial Pulse Detection™ (GDPD™) support for advanced
global CT solutions
-
Configure multiple boards in a single PCI chassis for easy and cost-effective
system expansion
-
Dialogic SpringWare™ downloadable signal and call processing firmware provides
easy feature enhancement and field-proven performance
-
PerfectDigit™ DTMF (touchtone) provides reliable detection during voice
playback — lets callers "type-ahead" through menus
-
Silence-compressed recording eliminates silence and preserves hard disk
space
-
Board Locator Technology™ eliminates confusing DIP switch or jumper settings
and simplifies installation
-
Support for BoardWatch™, the industry-standard Simple Network Management
Protocol (SNMP) used for remote CT board diagnostics/management
-
Dialogic DSP-based fax with a fax resource manager self-manages available
resources to support multiple fax instances*
-
Earth Recall capability provides support for switches in the UK in addition
to leveraging existing infrastructure
-
Full-duplex echo cancellation and barge-in capabilities for advanced messaging
solutions*
-
Dialogic CT Media™ support facilitates multi-application development*
* Features will be available in future releases
Applications
-
Interactive media response
-
Web-enabled call center
-
Unified messaging
-
Speech-enabled auto-attendant
The D/120JCT-LS™ board is part of the next generation of Dialogic analog
PCI boards, offering the enhanced capabilities that an evolving computer
telephony (CT) market demands. The product is ideal for advanced CT-based
communications applications that require multimedia resources. This high-performance,
scalable CT product, based on SpringWare™ technology, offers a rich set
of advanced features in addition to supporting state-of-the-art digital
signal processing (DSP) technology and signal processing algorithms, ensuring
a competitive edge for your solutions.
Advanced features include new voice coders such as G.726 and GSM,
the de facto standard when complying with Voice Profile for Internet Messaging
(VPIM) standards. Advanced features such as software-based fax and speech
recognition also let you add robust enhancements without additional hardware.
With the support of the industry-standard PCI bus architecture and the
international standard CTR21 in a single board, you can integrate the D/120JCT-LS
board at a price and performance level unmatched in the CT industry.
In addition, high impedance, on-hook record capability enables
high-density call logging and transaction record applications supported
in Dialogic CT Media™ for Windows NT. CT Media provides a standards-based
application development software platform and run-time environment for
building open telecommunication servers.
With all these features and advanced technologies, the 12-port
D/120JCT-LS PCI analog voice processing board is well positioned as the
principle building block for developing multimedia communications applications
such as Web-enabled call centers, unified messaging, and speech-enabled
interactive media response (IMR) systems.
Configurations
D/120JCT-LS Configuration Diagram
The D/120JCT-LS board is based upon the Signal Computing System Architecture™
(SCSA™). SCSA provides an open architecture that lets developers use products
from multiple vendors to build unified CT solutions. SCSA provides features
such as distributed switching, logical addressing, and location-independent
resource management. This PCI model incorporates a CT Bus™ connector that
also supports SCbus™ operation.
The D/120JCT-LS board provides 12 channels of call processing
and loop start interfaces in a single PCI slot. The unique dual-processor
architecture comprised of DSPs and a general-purpose microprocessor handles
all telephony signaling and performs all DTMF (touchtone) and audio/voice
signal processing tasks.
Downloaded firmware algorithms such as SpringWare provide variable
voice coding at 24 and 32 Kb/s ADPCM, and 48 and 64 Kb/s µ-law or
A-law PCM. Sampling rates and coding methods are selectable on a channel-by-channel
basis. Applications can dynamically switch the 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.
Offered as a software option, Dialogic Global Dial Pulse Detection™
(GDPD™) converts rotary pulses to DTMF in countries that have limited touchtone
telephone service. Global DPD is also optimized in several countries to
provide superior dial pulse detection and conversion.
Functional Description
The D/120JCT-LS board connects 12 analog loop start telephone
lines to 12 onboard call processing resources or to other resources via
the CT Bus.
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 12 loop start telephone lines connected to the D/120JCT-LS
board first pass through a telephone line interface that provides transient
protection and electromagnetic interference (EMI) suppression (see daughterboard
block diagram). These telephone line interfaces use reliable, solid-state
hook switches (no mechanical contacts) and FCC-Part 68 Class A ring voltage
and Class B ring frequency 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 a QSLAC chip where they are
routed via the CT Bus either to an onboard Motorola 56303 Onyx DSP or to
an external resource on any of the 1024 CT Bus time slots. This enables
the application to reroute calls to any added resource, such as speech
recognition, facsimile, or text-to-speech (TTS).
Part of the D/120JCT-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 touchtones while the line is on-hook. This capability
lets you use the D/120JCT-LS board behind PBXs that require on-hook touchtone
detection for their signaling. When the 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 PCI bus.
Baseboard Block Diagram
Daughterboard Block Diagram
The Motorola 56303 Onyx DSP processes the digitized voice data based
on SpringWare firmware loaded in code/data RAM. Each Motorola 56303 Onyx
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 Adaptive Differential Pulse Code Modulation (ADPCM), Pulse Code
Modulation (PCM) algorithm, G.726 32 Kb/s coding, or GSM 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 Motorola 56303 Onyx DSP can use 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/120JCT-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 Motorola
56303 Onyx DSP, which converts the file into digitized voice. The DSP sends
the digitized voice responses to the CODEC, which is controlled by three
QSLAC 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 is downloaded
from the host PC to the board. It controls all board opera-tions. SpringWare
gives the board all of its intelligence and enables easy feature enhancements
and upgrades.
The onboard control processor controls all operations of the D/120JCT-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 Motorola 56303 Onyx DSPs to perform signal processing.
Communication between the processor and the host PC is via 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. The Board Locator Technology™ circuit operates in conjunction
with a rotary switch that eliminates the need to set confusing jumpers
or DIP switches.
|
SpringWare Technical Specifications*
| AUDIO SIGNAL: |
| Receive range |
–40 to +2.5 dBm0 nominal, configurable by parameter** |
| Automatic gain control |
Applications can enable/disable. Above -18 dBm0
results in full-scale recording, configurable by parameter.** |
| Silence detection |
–40 dBm nominal, software adjustable** |
| Transmit level(weighted average) |
–9.5 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 |
–38 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 |
(T-1) 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 tone |
| Max. number of tones |
Application dependent |
| Frequency range |
Programmable within 300 to 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 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 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., 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 |
Preprogrammed |
| 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; 16 digits per Bellcore
LSSGR Sec 6, TR-NWT-000506 |
| Frequency variation |
Less than ±1 Hz |
| Rate |
10 digits/s, configurable by parameter** |
| Level |
–7.5 dBmO 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
-
Pentium® based (PCI) bus or compatible computer. Operating system hardware
requirements vary according to the number of channels being used.
Additional Components
-
Multidrop CT Bus cables
-
Six-strand RJ-type cable (RJ-11 connectors to standard 50-pin Amphenol
connector)
— Support for both US and Euro cards
— One cable per board required
-
"Two-into-one" conversion cable
— Six cables per board required
— US and Euro cables sold separately
|
