ProductsDialogic

 
D/160SC-LS™
16-PORT VOICE PROCESSNG & ANALOG INTERFACE BOARD 


FEATURES & BENEFITS

     
  • 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
  • 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
  • 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
  • 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
  • PerfectDigit™ DTMF (touchtone) provides reliable detection during voice playback - allows callers to "type-ahead" through menus
  • Optional Global Dial Pulse Detection (DPD) feature enables callers without touchtone phones to access applications. No additional "pulse to tone converter" hardware is needed
  • 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
  • Intel 486 GX microprocessor off-loads 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®, Solaris™ and Windows NT™ shorten your development cycle so you can get your applications to market faster
  • High impedance, on-hook record capability enables high-density call logging and transaction record applications
  • 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 fits your needs
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
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

  • connects to 16 analog loop-start telephone channels
  • automatically answers calls
  • detects touchtone
  • plays voice messages to a caller
  • digitizes, compresses, and records voice signals
  • 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:
 
 

  • interference suppression
  • digital-to-analog conversion
  • ring and on-hook/off-hook signaling control
  • tone detection and generation, and
  • 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:
 
 

  • applies 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, or G.726 32Kb/s coding (near future) 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 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

     
  • 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
  • Multidrop SCbus cable
Optional: Station Adapter, 37 pin to 50 pin cable

 


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E-mail : dialogic@foremost-systems.com