Dialogic® DM3 Media Boards. Board DM/V600BTEPEQ 10339 DMVxxx BTEPEQ Ds

User Manual: Media Board DM/V600BTEPEQ

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Datasheet

Dialogic DM3 Media Boards
®

DM3 Media Boards

Products Discussed in This Datasheet
• Dialogic DM/V300BTEPEQ Media Board
®

• Dialogic DM/V600BTEPEQ Media Board
®

• Dialogic DM/V1200BTEPEQ Media Board
®

With One, Two, or Four ISDN PRI Trunks
The Dialogic DM/V300BTEPEQ, DM/V600BTEPEQ, and
DM/V1200BTEPEQ Media Boards are among the industry’s
most powerful media platforms for developers seeking to
rapidly build and globally deploy some of the highest density
media server solutions for the enterprise and public networks.
They provide a truly universal port solution with a robust media feature set — including voice processing,
Large Logospeech recognition, fax,
and conferencing capabilities — combined with an extensive suite of network protocols in a single PCI Express slot.
®

Features

Benefits

Software selectable trunks to configure the board to be either
T1 or E1

Reduces the total cost of ownership by increasing flexibility,
decreasing inventory, and simplifying the purchasing
process and test effort

Universal media loads offer mixed media resources including
voice, fax, and conferencing

By combining three boards into one, reduces development,
inventory, and solution costs by eliminating the need for
dedicated media boards

Provides the ability to mix select protocols on each span

Maximizes slot efficiency and reduces total cost of ownership
in environments where there are multiple protocols (for
example, call centers)

Built on the industry-standard telephony bus — ECTF H.100
CT Bus

Lets applications expand (up to 1200 ports per system)
through access to other communications boards, such as IP
telephony, ATM, HDSI, and SS7

Supports TrueSpeech voice coder (a default coder with
Windows supported by Windows Media Player)

Lets developers play Internet content and develop unified
messaging systems without creating and supporting custom
clients

Ability to select between 16 ms, 32 ms, and 64 ms echo
cancellation tail on select media loads

Longer tail lengths are useful for environments and
applications when optimum audio quality and clarity is a
necessity

PCI Express interface

Supports new full-length PCI Express form factor compatible
with x1 slots

®

®

Medium Logo

To find out more about Dialogic call 800.755.4444 or visit us at: www.dialogic.com

Datasheet

Dialogic DM3 Media Boards
®

DM3 Media Boards

Applications
• Messaging and enhanced services
• Wireless and fixed-line Short Message Service (SMS)
• Color ring back tone
• Voice portal
• Contact center and e-Business
• PC-PBX
• Audio conferencing server
• Web conferencing
• Switching and call completion
• Prepaid/debit card
• Gateway switch
The boards, which are H.100-compliant, are suitable for service providers and large enterprise applications.
As part of the next generation of media span products from Dialogic, these boards include a host of robust features including:
• Software selectable T1/E1 Digital Network Interfaces (DNIs)
• A-law and µ-law conversion with Dialogic System Release 6.0 PCI for Windows and Dialogic System Release 6.1 for Linux
®

®

®

• A universal media load offering simultaneous
— Fax
— Conferencing
— Voice processing
• Improved media densities
• The ability to mix select protocols
• Hardware selectable non-loopback mode
• Different conferencing media loads from which to choose
Support for Continuous Speech Processing (CSP) technology — Dialogic’s Digital Signal Processor (DSP)-based solution optimized
for speech recognition — enables a friendlier user interface and seamless integration of speech recognition software from the
leading speech technology vendors. CSP reduces system latency, increases recognition accuracy, and improves overall system
response time for high-density speech solutions. Also available on select media loads is Enhanced Echo Cancellation (EEC), which
offers the ability to select longer echo cancellation tail lengths of 32 ms and 64 ms (beyond the normal 16 ms) to further improve
and refine audio quality.

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Dialogic DM3 Media Boards
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DM3 Media Boards

Other features include silence-compressed streaming to the host, which improves performance by removing the silence when
data is sent to the host CPU. In addition, streaming to the CT Bus lets echo cancelled data be streamed onto the TDM bus.
The onboard conferencing solution offers an advanced feature set, presenting both a satisfying conferencing experience for
the end user and one that can be used to deploy network-grade conferencing systems with features, audio quality, and density
comparable to typical proprietary solutions, but at significantly reduced costs. Its optimized algorithm prevents noise build-up
and echo in the conference. It also equalizes participant voice volumes, offering optional Dual Tone Multi-Frequency (DTMF and
touchtone) clamping to limit audible enter and exit tones. The DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ are
available with two different types of conferencing — rich and standard — to optimize a conferencing solution based on system
configuration and requirements. (See the Conferencing section for details.)
Another conferencing feature is bridging, also known as cascade conferencing. Bridging lets you bridge together conferences
from different DSPs and boards, consuming just two conferencing resources per bridge. This maximizes the flexibility of your
conferencing solution by letting you create high-density conferences where any party can speak, and be heard, by the other
conference participants.
Powerful DSPs provide a rich set of media processing features, including various rates of voice compression, recording and
playback, conferencing with echo cancellation and active talker algorithm, telephony tone signaling, reliable DTMF detection
using local echo cancellation, and automated outbound call progress analysis with positive voice detection and positive answering
machine detection.
The DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ are based on DM3 architecture, which provides an environment
that accelerates application development while also providing a path for growth. With support for the Dialogic R4 API, it is easy
for these boards to interoperate with other CT Bus boards from Dialogic. Applications can be ported easily to lower or higher
density platforms, or new features can be added with only minimum modifications — thus protecting investment in hardware and
application code.
®

Configurations
Use the DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ to develop sophisticated, mixed-media communication
applications that include voice processing, speech recognition, fax, and conferencing. These boards occupy a single computer
backplane slot and multiple boards can be installed in a single chassis. The maximum number of supported lines and features
depends on the application type, call module, and host computer CPU. For media-intensive applications, 600 ports in a chassis
are reasonable. For other applications, such as call completion, where media processing is less intensive, systems of 1200+ ports
per chassis are possible.
These boards can operate in either terminate or hairpin configurations. In a terminate configuration (see Figure 1), the boards
handle the processing of digital audio and telephony signaling. Additional system resources can access calls via the CT Bus. This
configuration is suitable for voice messaging, unified messaging, voice portal, and Interactive Voice Response (IVR) applications.
In a hairpin configuration (see Figure 2), the boards are connected via the CT Bus and can continuously pass all T1/E1 time slots
through to each other. This configuration can switch call traffic between separate T1 or E1 lines, or can be placed inline between a
T1/E1 public network trunk and a digital switch. Calls on individual channels can either terminate at a call processing resource on
a board, or “flow through” transparently from one board to the other. Even in “flow through” mode, these boards can still monitor
the lines, listening for DTMF or voice commands. This configuration is well suited for call center, PC-PBX, voice portal, prepaid
calling card, international callback, and telecom resale applications.

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Dialogic DM3 Media Boards
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DM3 Media Boards

Third-Party Application Development Toolkit
Dialogic® Global Call API and
Dialogic® R4 API

ASR Engine by Third Party

Hardware

CT Bus

E1/ISDN
Trunks to
PSTN or PBX

Software

Dialogic® DM/V600BTEPEQ
Media Board

Dialogic® DM/V3600BP
Media Board

DM/V600BTEPEQ

PCI Bus

Figure 1. Terminate Configuration: Unified Messaging Application

Third-Party Application Development Toolkit
Dialogic® Global Call API
and Dialogic® R4 API

ASR Engine by Third Party

Hardware

CT Bus

T1/ISDN
Trunks to
PSTN

Software

Dialogic® DM/N960-4T1 Digital
Telephony Interface Board

Dialogic® DM/V1200BTEPEQ
Media Board
DM/V1200BTEPEQ

DM/N960-4T1

PCI Bus

Figure 2. Hairpin Configuration: Call Completion or Prepaid/Debit Card Applications

4

T1/ISDN
Trunks to
PSTN

Datasheet

Dialogic DM3 Media Boards
®

DM3 Media Boards

Software Support
The DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ are supported by system software and Software Development
Kits (SDKs) for Windows and for Linux. They also support the Simple Network Management Protocol (SNMP) agent software for
remote CT board management. See the application note at http://www.dialogic.com/systemreleases for details about the operating
systems currently supported. The SDKs contain all the documentation, demonstration code, and tools necessary for developing
complex multichannel applications.

Dialogic Global Call Software
®

The DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ support Dialogic Global Call Software, a unified call control
programming interface and protocol engine that makes it easier to provide worldwide application portability and can shorten
development time by using the same API for almost any network protocol.
Global Call Software provides a common signaling interface for network-enabled applications, regardless of the signaling protocol
needed to connect to the local telephone network. Global Call is the recommended API for unified call control for Springware and
DM3 architectures. The signaling interface provided by Global Call facilitates the exchange of call control messages between the
telephone network and virtually any network-enabled application. Global Call lets developers create an application that can work
with signaling systems worldwide, regardless of the network to which they are connected.
Global Call is well suited for high-density, network-enabled solutions for voice and data where the supported hardware and signaling
technology can vary widely. Rather than requiring the application to handle the low-level details, Global Call Software offers a
consistent, high-level interface to the user, handling each country’s unique protocol requirements in a way that is transparent to
the application.

Functional Description
The DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ are based on the DM3 mediastream architecture. The
architecture consists of a set of core specifications and firmware modules that are implemented on boards with various processors,
including:
• Intel XScale processor for centralized control
• DSP(s) for mediastream processing
• TDM bus interface (H.100)
• One, two, or four software selectable digital telephony network interfaces
• PCI Express interface
These boards support up to 120 channels of voice processing via a bank of DSPs and up to four T1/E1 digital trunk interface (DTI)
circuits. The DTI circuits contain signaling services (ISDN, CAS, and CCS), plus any alarm handling and line maintenance services
required by the installed networks. Each DTI includes software-switchable clock circuits that can be set to one of three modes:
• Loop — Transmit clocking is slaved to the external network
• Independent — Transmit clocking is derived from an onboard oscillator
• Expansion or system — Transmit clocking is slaved to the TDM; receive clocking is slaved to the trunk interface

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The control processor is the Intel 80200 based on Intel XScale microarchitecture. It is responsible for the initialization, configuration,
and control of the various elements that make up these boards. It controls the TDM bus interface, as well as the signaling protocols
for the DTIs installed on the platform.
The boards support various DSP configurations for voice processing and call progress analysis capabilities. These features are
provided by baseboard and daughterboard configurations using Motorola 56321 DSPs. The DSPs process the digitized voice data
using downloaded resource firmware. Each DSP can perform the following signal analysis and operations.

Incoming Data
• Automatic Gain Control (AGC), which compensates for variations in the level of the incoming audio signal
• Adaptive Differential Pulse Code Modulation (ADPCM), Pulse Code Modulation (PCM), LinearWAV, Global System for Mobile
Communications (GSM), G.726, and TrueSpeech algorithms that compress digitized voice and save disk storage space
• Tone detection of DTMF, MF, or application-defined single or dual tones
• Silence detection to determine if the line is quiet and the caller is not responding

Outbound Data
• Expands stored, compressed audio data for playback
• Adjusts the volume and pitch of playback upon application or user request
• Generates tones — DTMF, MF, or any application-defined general-purpose tone
• Performs outbound dialing
• Monitors call progress functions, including:
— Line busy
— Operator intercept
— Ring
— No answer
— Answered; the DSP detects whether the answering party is a person, answering machine, fax machine, or modem
While recording speech, the DSP can use different digitizing rates from 8.5 kbps to 176 kbps, selectable 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.
DSP-processed speech is transmitted by the control processor to the host for disk storage. When playing back a stored file, the
processor retrieves voice information from the host CPU and passes it to the DSP, which converts the file into digitized voice. The
DSP sends the digitized voice responses to the caller via the network interface or TDM bus. In addition, the cache prompt feature
provides 4 to 8 MB of onboard cache for the storing and playback of voice files directly on the board, eliminating the need to send
voice files to and from the host/server.
Shared RAM on these boards enables communication between the host system and the XScale control processor. A bank of
global memory is also provided to facilitate communication between the control processor and the various DSPs. In addition to
providing a data pathway between processors, the global memory can also serve as a repository for data that is to be shared
among processors, or which may not be storable within local memory associated with the processor.

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Dialogic DM3 Media Boards
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DM3 Media Boards

Downloadable Firmware
The hardware for the DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ consists of a baseboard with a control
processor and one, two, or four DS-1 DNIs. An array of DSPs resides on the baseboard and a low-profile daughterboard on the
DM/V600TEPEQ and DM/V1200TEPEQ. The DM/V300TEPEQ has no daughterboard. Telephony signaling protocols and media
processing features are downloaded as firmware to the board on power up and reside on the various onboard processors. This
downloadable firmware approach enables easy feature upgrade and expansion. Individual firmware components, such as a
network interface protocol or a voice recording function, are referred to as resources.

Network Interface
The DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ have software-selectable trunks that can enable the board to
be configured as T1 or E1 to increase flexibility, simplify the purchasing process and test effort, and help reduce inventory and
the total cost of ownership. With Dialogic System Release 6.0 PCI for Windows and Dialogic System Release 6.1 for Linux, these
boards support A-law/µ-law conversion and the ability to mix individual spans as T1 or E1. They also support ISDN PRI access
for both T1 and E1.
®

®

®

• C
 onfigured as T1 — Support for all T1 robbed-bit signaling protocols and fully compatible with all interface devices that use,
or can be set to use, 1.544 MHz clocking and µ-law PCM. In addition, supports the clear channel feature, thus providing up
to 96 bearer channels when used in this mode. The T1 protocol implementations comply with the North American standard
ISDN PRI and the INS-1500 standard used in Japan. In North America and Japan, the ISDN Primary Rate includes 23
voice/data channels (B channels) and one signaling channel (D channel).
• C
 onfigured as E1 — Support for all CEPT Channel Associated Signaling (CAS) protocols and fully compatible with interface
devices that use, or can be set to use, 2.048 MHz clocking and A-law PCM. In addition, supports the clear channel feature,
thus providing up to 124 bearer channels when used in this mode. The E1 protocol implementations comply with the E1
ISDN PRI protocols. The E1 ISDN Primary Rate includes 30 voice/data channels (B channels) and two additional channels:
one signaling channel (D channel) and one framing channel to handle synchronization.
Key ISDN PRI features include:
• N
 on-Facility Associated Signaling (NFAS) — Lets a single D-channel control up to 10 PRI trunks, providing significant
savings in ISDN service subscription costs available on NI-2, 4ESS, 5ESS, DMS100, and DMS250
• D-channel backup (on NI-2 only) — Lets another D-channel takeover should the main D-channel fail
• Facility, notify, and optional Information Elements (IEs) — Lets applications work with network-specific supplementary
services
• Direct Dialing In (DDI) — Lets an application route incoming calls by automatically identifying the number the caller dialed.
Also known as Dialed Number Identification Service (DNIS).
• Call-by-call service selection — Lets an application select the most efficient bearer channel service, such as a toll-free line
or a WATS line, on a call-by-call basis
• User-to-user information — Lets an application send proprietary messages to remote systems during call establishment
• LAP-D Layer 2 access — Lets developers build a customized Layer 3 protocol
• Protocol timer setting — Lets protocol timers be set dynamically through a configuration file
• Mask-able Layer 2 Control — Lets the application toggle between bringing Layer 2 up and down as desired

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DM3 Media Boards

Dialogic maintains an extensive number of product approvals in international markets. See the list of globally approved products
at http://www.dialogic.com/declarations.

Voice Processing
Voice processing features, downloaded to the onboard DSPs at power up, enable these combined media boards to play and record
voice messages to and from callers through the digital network interface. Messages can be stored using G.711 µ-law or A-law
PCM, at a rate of 64 kbps, as is used by the Public Switched Telephone Network (PSTN). To reduce storage requirements and
help developers implement unified messaging applications that meet Voice Profile for Internet Messaging (VPIM) standards, voice
coding algorithms can compress recordings as low as 8.5 kbps using low-bit rate coders such as GSM, G.726, and TrueSpeech.
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 needed.
AGC is provided to automatically adjust the signal level of incoming calls for recording at normal levels, compensating for adverse
line conditions, distance, and other factors. Playback volume can also be dynamically adjusted over a 40 dB range using DTMF
input or directly from the application.
DTMF detection is provided to control record and play functions using DTMF input. Local echo cancellation techniques are used
to improve DTMF cut-through and talk-off/play-off suppression over a wide variety of telephone line conditions.
The voice player and recorder resources are linked with the DTMF detection resources using Run-Time Control (RTC) messages.
This lets play or record functions be initiated or terminated quickly using DTMF input from the caller. The RTC function off-loads
the host application from involvement in every interaction, thereby enabling voice processing applications to scale to hundreds
of ports per system.
Continuous Speech Processing (CSP) enables software-based Automatic Speech Recognition (ASR) and the ability to speak
over speech prompts. It processes the incoming voice signal using DSP-based Echo Canceller (EC) and Voice Activity Detector
(VAD) integrated on the board. The incoming voice signal is then streamed to the host system and the ASR engines only when
voice energy is detected. Features such as the pre-speech buffer and the onboard VAD let the system attain higher accuracy and
efficiency. Also now available on select media loads is Enhanced Echo Cancellation (EEC), which offers the ability to select longer
echo cancellation tail lengths of 32 ms and 64 ms beyond the standard 16 ms to further improve and refine audio quality.
The transaction record feature lets voice activity on two channels be summed and stored in a single file, or in a combination
of files, devices, and/or memory. When it is necessary to archive a verbal transaction or record a live conversation, the silencecompressed record feature, when enabled, eliminates silence from recorded data, thus saving disk storage space. Speed and
volume control are also provided to let the application or user adjust the speed and volume during playback. Silence compressed
streaming to the host improves performance by removing the silence when data is sent to the host CPU. Streaming to the CT Bus
lets echo cancelled data be streamed into the TDM bus.

Conferencing
The DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ are available with two different types of conferencing (rich
and standard) to maximize conferencing solutions based on system configuration and requirements (see Table 1).
Conferencing Type

Echo Cancellation

Tone Detection and Generation

Tone Clamping

Rich Conferencing

4

4

4

Standard Conferencing

—

4

4

Table 1. Conferencing Media Loads

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The conferencing solution is implemented using onboard DSPs. The conferencing resource sums incoming voice signals on
the board. Higher quality conferencing is attained using sophisticated summing algorithms and Echo Cancellation (EC). The
advanced algorithm dynamically distinguishes between noise and speech and prevents noise buildup. The incoming voice signal
is then streamed out to the CT Bus where it can be transmitted to any network interface — either PSTN or IP telephony.
This enables very large size conferences (1000+) such as analyst calls or broadcast calls where most of the callers are in listenonly mode. In addition, bridging (also known as cascade conferencing) enables conferences to be joined together via a bridge (up
to 64 participants in each conference) from different DSPs and boards, consuming just two conferencing resources per bridge.
This maximizes the flexibility of a conferencing solution by allowing the creation of high-density conferences where any party has
the ability to speak and be heard by other participants.
Both types of conferencing media loads support the active talker feature that identifies which conferees are actively talking at any
given time and suppresses the background noise from all the silent parties. It also lets applications dynamically choose between
the summing mode — active talker vs. pure summation — based on the conference size. Applications can set this parameter
at configuration or change it dynamically at runtime. For a small number of parties, pure summation might be preferred so all
conferees are heard; and as conference size increases, the active talker feature might be enabled so conferees can hear the most
active participants.

Tone Signaling
In addition to the DTMF signaling commonly used for voice processing, the DM/V300BTEPEQ, DM/V600BTEPEQ, and
DM/V1200BTEPEQ also contain a robust set of features used for network tone signaling and control. The Global Tone Detection
(GTD) and Global Tone Generation (GTG) features provide the ability to detect and generate user-defined tones for solving special
application situations, such as integration with Private Branch Exchange (PBX) or dealing with unique tones.
Perfect Call call progress analysis accurately monitors outbound calls, detects when calls are answered, and distinguishes:
• Line ringing with no answer
• Line busy
• Problem completing call (such as operator intercept)
• Call answered by a human or answering machine
• Call answered by a fax machine or modem
Perfect Call is intelligently tolerant of the wide variation in call progress signaling tones found in central offices and PBXs around
the globe and offers accurate performance right out of the box. DSP-based algorithms are used to accurately discriminate human
speech from recorded human voice and from network noise.

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Dialogic DM3 Media Boards
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DM3 Media Boards

Technical Specifications
Digital interfaces

One, two, or four T1/E1

Max. boards/system

Application, call traffic, and CPU dependent

Control processor

Intel 80200

Control processor memory

48 MB

Baseboard global memory

16 MB 32-bit wide DRAM accessible to all signal processors and control processor

Cache prompts

4 MB to 8 MB

Host Interface
Host interface memory

512 KB

Bus compatibility

PCI Express Card Electromechanical Specification Revision 1.0a

Bus mode

Target and DMA master mode operation

Platform
Form factor

Full-length PCI Express (x1 compatible)

Digital Signal Processor (DSP)

Motorola 56321
DM/V300BTEPEQ: 4 DSPs @ 220 MHz each
DM/V600BTEPEQ: 10 DSPs @ 220 MHz each
DM/V1200BTEPEQ: 10 DSPs @ 220 MHz each

DSP memory

512 k word SRAM local to each DSP

Computer telephony bus

ECTF H.100-compliant CT Bus with:
• Onboard switching access to 4096 bi-directional 64 kbps DS0 time slots
• 68-pin ribbon cable connector

Telephone Interface

DSX-1 T1

Clock rate

1.544 Mbps ±32 ppm

Level

3.0 V (nominal)

Pulse width

323.85 ns (nominal)

Line impedance

100 Ohm ±10%

Other electrical characteristics

Complies with AT&T TR62411 and ANSI T1.403-1989

Framing

• SF (D3/D4)
• ESF for ISDN

Line coding

• AMI
• AMI with B7 stuffing
• B8ZS

Clock and data recovery

Complies with AT&T TR62411 and Telcordia TA-TSY-000170

Jitter tolerance

Complies with AT&T TR62411 and ANSI T1.403-1989

Connectors

One, two, or four RJ-48C on front bracket
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DM3 Media Boards

Technical Specifications (continued)
Telephony bus connector

H.100 (PCI)

Loopback

Supports switch-selectable local analog loopback and software-selectable local digital loopback

Zero code suppression

• Bell ZCS (Jam bit 7)
• GTE ZCS (Jam bit 8)
• Digital Data Service ZCS
• No zero code suppression

Telephone Interface

CEPT E1

Network clock rate

2.048 Mbps ±50 ppm

Internal clock rate

2.048 Mbps ±32 ppm

Level

• 2.37 V (nominal) for 75 Ohm lines
• 3.0 V (nominal) for 120 Ohm lines

Pulse width

244 ns (nominal)

Line impedance

• 75 Ohm, unbalanced
• 120 Ohm, balanced

Other electrical characteristics

Complies with ITU-T Rec. G.703

Framing

ITU-T G.704-1988 with CRC4

Line coding

HDB3

Clock and data recovery

Complies with ITU-T Rec. G.823-1988

Jitter tolerance

Complies with ITU-T Rec. G.823, G.737, G.739, G.742-1988

Connectors

One, two, or four RJ-48C on front bracket

Telephony bus connector

H.100 (PCI)

Loopback

Supports switch-selectable local analog loopback and software-selectable local digital loopback

Power Requirements
Configuration
DM/V300BTEPEQ
DM/V600BTEPEQ
DMV1200BTEPEQ

+12 VDC
0.6
1.25
1.25

+3.3 VDC
0.58
0.82
0.82

Operating Requirements
Operating temperature
Storage temperature
Humidity

+32°F (0°C) to +122°F (+50°C)
–4°F (–20°C) to +158°F (+70°C)
8% to 80% noncondensing

Approvals and Compliance
Hazardous substances

RoHS compliance information at http://www.dialogic.com/rohs

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DM3 Media Boards

Technical Specifications (continued)
Safety and EMC
United States

• UL 60950-1 File E96804
• FCC Part 15 Class A

Canada

• ULc CSA 60950-1 File E96804
• ICES-003 Class A

Europe

• EN60950
• EN55022
• EN55024

International

• IEC60950-1
• CISPR 22
• CISPR 24

Telecom Approvals
United States

US:EBZUSA-31207-XD-T

Canada

IC:885A 7969 A

Europe

DoC TBR4

	For country-specific approval information, see the global product approvals database at
http://www.dialogic.com/declarations
Reliability/Warranty
Estimated MTBF

DM/V300BTEPEQ — 270,780 hours per Telcordia Method I
DM/V600BTEPEQ — 239,839 hours per Telcordia Method I
DM/V1200BTEPEQ — 235,000 hours per Telcordia Method I

Warranty

Warranty information at http://www.dialogic.com/warranties

Audio Signal
Usable receive range

–40 dBm0 to 0 dBm0 nominal, configurable by parameter**

Automatic gain control

Application can enable/disable output level, configurable by parameter**

Silence detection

–40 dBm nominal, software adjustable**

Transmit level (weighted average)

–12.5 dBm nominal, configurable by parameter**

Transmit volume control

40 dB adjustment range, with application-definable increments and legal limit cap

Frequency Response
24 kbps

300 Hz to 2600 Hz ±3 dB

32 kbps

300 Hz to 3400 Hz ±3 dB

64 kbps

300 Hz to 3400 Hz ±3 dB

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DM3 Media Boards

Technical Specifications (continued)
Audio Digitizing
8.5 kbps

TrueSpeech

13 kbps

GSM (TIPHON, MSGSM)

16 kbps, 24 kbps, 32 kbps, and 40 kbps

G.726

24 kbps

OKI ADPCM @ 6 kHz sampling

32 kbps

OKI ADPCM @ 8 kHz sampling

32 kbps

IMA ADPCM @ 8 kHz sampling

48 kbps

G.711 PCM (µ-law for T1 and A-law for E1) @ 6 kHz sampling rate

64 kbps

G.711 PCM (µ-law for T1 and A-law for E1) @ 8 kHz sampling rate

64 kbps

Linear 8 kHz 8-bit WAV

128 kbps

Linear 8 kHz 16-bit WAV

88 kbps

Linear 11 kHz 8-bit WAV

176 kbps

Linear 11 kHz 16-bit WAV

A-law/µ-law conversion

Standard (with Dialogic System Release 6.1 for Linux)

Digitization selection

Selectable by application on function call-by-call basis

Playback speed control

• Pitch controlled
• Available on the following 8 kHz coders: OKI ADPCM, G.711 PCM, Linear
• Adjustment range: ±50%
• Adjustable through application or programmable DTMF control

®

DTMF Tone Detection
DTMF digits

0 to 9, *, #, A, B, C, D per Telcordia LSSGR Sec. 6

Dynamic range

• (T1) –36 dBm to +3 dBm per tone, configurable by parameter**
• (E1) –39 dBm to 0 dBm per tone, configurable by parameter**

Minimum tone duration

32 ms, can be increased with software configuration

Interdigit timing

• Detects like digits with a >45 ms interdigit delay
• Detects different digits with a 0 ms interdigit delay

Acceptable twist and frequency variation

• (T1) Meets Telcordia LSSGR Sec 6 and EIA 464 requirements
• (E1) Meets ITU-T Q.23 recommendations**

Noise tolerance	Meets Telcordia LSSGR Sec 6 and EIA 464 requirements for Gaussian, impulse, and power line noise tolerance
Cut-through

• (T1) Local echo cancellation permits 100% detection with a >4.5 dB return loss line
• (E1) Digital trunks use separate transmit and receive paths to network. Performance dependent on far-end
handset’s match to local analog loop.

Talk-off

• Detects less than 10 digits while monitoring Telcordia 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.

13

Datasheet

Dialogic DM3 Media Boards
®

DM3 Media Boards

Technical Specifications (continued)
Global Tone Detection
Tone type

Programmable for single or dual

Max. number of tones

Application-dependent

Frequency range

Programmable within 300 Hz 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

• (T1) Default set at –36 dBm to +3 dBm per tone, programmable
• (E1) Default set at –39 dBm to +0 dBm per tone, programmable

Global Tone Generation
Tone type

Generate single or dual tones

Frequency range

Programmable within 200 Hz to 4000 Hz

Frequency resolution

1 Hz

Duration

10 ms increments

Amplitude

• (T1) –43 dBm0 to –3 dBm0 per tone nominal, programmable
• (E1) –40 dBm0 to +0 dBm0 per tone nominal, programmable

MF Signaling (T1)

R1

MF digits

0 to 9, KP, ST, ST1, ST2, ST3 per Telcordia LSSGR Sec 6, TR-NWT-000506 and ITU-T Q.321

Transmit level

Complies with Telcordia LSSGR Sec 6, TR-NWT-000506

Signaling mechanism

Complies with Telcordia LSSGR Sec 6, TR-NWT-000506

Dynamic range for detection

–25 dBm to +3 dBm per tone

Acceptable twist

6 dB

Acceptable freq. variation

Less than ±1 Hz

MF Signaling (E1)

R2

MF digits

All 15 forward and backward signal tones per ITU-T Q.441

Transmit level

–8 dBm0 per tone, nominal, per ITU-T Q.454; programmable

Signaling mechanism

S upports the R2 compelled signaling cycle and non-compelled pulse requirements per ITU-T Q.457 and
Q.442

Dynamic range for detection

–35 dBm to –5 dBm per tone

Acceptable twist

7 dB

Acceptable freq. variation

Less than ±1 Hz

14

Datasheet

Dialogic DM3 Media Boards
®

DM3 Media Boards

Technical Specifications (continued)
Call Progress Analysis
Busy tone detection

D efault setting designed to detect 74 out of 76 unique busy/congestion tones used in 97 countries as
specified by ITU-T Rec. E., Suppl. #2. Default uses both frequency and cadence detection. Application can
select frequency only for faster detection in specific environments.

Ring back detection

D efault setting designed to detect 83 out of 87 unique ring back tones used in 96 countries as specified by
ITU-T Rec. E., Suppl. #2. Uses both frequency and cadence detection.

Positive voice detection

Standard

Positive voice detection speed

Detects voice in as little as 1/10th of a second

Positive answering machine detection

Standard

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 (when not part of regulatory approval)

Tone Dialing
DTMF digits

0 to 9, *, #, A, B, C, D per Telcordia LSSGR Sec 6, TR-NWT-000506, ITU-T Q.23

Frequency variation

Less than ±1 Hz

Rate

10 dps, configurable by parameter**

Level

• (T1) –4.0 dBm per tone, nominal, configurable by parameter**
• (E1) –7.0 dBm per tone, nominal, country-specific**

Conferencing
Max. parties per conference

Up to 64 (without bridging) on select media loads

Bridging/cascade conferencing	Can bridge together conferences from different DSPs and boards, consuming just two conferencing
resources per bridge
Echo cancellation

16 ms

Tone clamping

Enable/disable at board level

Summing modes	Automatically configures to active talker or pure summation based on number of parties in a conference.
Application can specify minimum number of parties before active talker mode is used.
Automatic gain control	Normalizes the parties’ power levels to a unified target. Key features include speech/noise discrimination,
tolerance to impulsive noise, faster convergence, and increased steady-state stability.
Tone detection/generation	Generates tariff tones and warning tones. Detects DTMF from each party and can clamp those tones so that
other members of the conference do not hear them.
Active talker notification	Can notify the application as to which party is talking so the application can process that information and
act accordingly
Number of active talkers

Dynamically selectable

15

Datasheet

Dialogic DM3 Media Boards
®

DM3 Media Boards

Technical Specifications (continued)
Modes

• Duplex
• Monitor
• Coach
• Pupil

Facsimile
Fax compatibility

T.30, T.4, T.6, V.17, V29, V27ter, V.21

Speed

14.4 kbps with automatic fallback send and receive concurrently on all channels

TIFF

• Single page
• Multipage

Compression

• MH (ITU T.4, 1D)
• MR (ITU T.4 2D)
• MMR (ITU T.6)
• Onboard, on-the-fly

ECM

Supported

ASCII to TIFF

Onboard, on-the-fly

Page headers

Generated onboard, on-the-fly

Width

A4

Polling

Normal and turnaround

Resolution

• Standard (100 dpi × 200 dpi)
• Fine (200 dpi × 200 dpi)
• Superfine (200 dpi × 400 dpi)

JPEG/JBIG

Color fax and gray scale fax pass-through feature

Protocols
T1 CAS

E&M (wink start, immediate start), loop start, ground start; feature group A, B, and D

T1 ISDN

NI-2, 4ESS, 5ESS, DMS100, DMS250, INS1500, Q.Sig

E1 CAS
release notes

Many country-specific MFC-R2 variants. For more details, refer to the latest Global Call Protocols Package

E1 ISDN

NET5, DPNSS, DASS2, QSIG

Additional Components
• Multidrop CT Bus cables (CBLCTB3DROPQ, CBLCTB4DROPQ, CBLCTB8DROPQ, CBLCTB12DROPQ, CBLCTB16DROPQ)
• 120 Ohm to 75 Ohm converter (supplied by a third party)

16

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www.dialogic.com

Dialogic Corporation
9800 Cavendish Blvd., 5th floor
Montreal, Quebec
CANADA H4M 2V9
INFORMATION IN THIS DOCUMENT IS PROVIDED IN CONNECTION WITH DIALOGIC PRODUCTS. NO LICENSE, EXPRESS OR IMPLIED, BY ESTOPPEL OR OTHERWISE, TO ANY
INTELLECTUAL PROPERTY RIGHTS IS GRANTED BY THIS DOCUMENT. EXCEPT AS PROVIDED IN A SIGNED AGREEMENT BETWEEN YOU AND DIALOGIC, DIALOGIC ASSUMES NO
LIABILITY WHATSOEVER, AND DIALOGIC DISCLAIMS ANY EXPRESS OR IMPLIED WARRANTY, RELATING TO SALE AND/OR USE OF DIALOGIC PRODUCTS INCLUDING LIABILITY OR
WARRANTIES RELATING TO FITNESS FOR A PARTICULAR PURPOSE, MERCHANTABILITY, OR INFRINGEMENT OF ANY INTELLECTUAL PROPERTY RIGHT OF A THIRD PARTY.
®

Dialogic products are not intended for use in medical, life saving, life sustaining, critical control or safety systems, or in nuclear facility applications.
Dialogic may make changes to specifications, product descriptions, and plans at any time, without notice.
Dialogic is a registered trademark of Dialogic Corporation. Windows is a registered trademark of the Microsoft Corporation in the United States and/or other countries.
Other names of actual companies and products mentioned herein are the trademarks of their respective owners. Dialogic encourages all users of its products to procure
all necessary intellectual property licenses required to implement their concepts or applications, which licenses may vary from country to country.

None of the information provided in this datasheet other than what is listed under the section entitled Technical Specifications forms part of the specifications of the
Dialogic DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ combined media board products and any benefits specified are not guaranteed.
Positive Answering Machine Detection/Positive Voice Detection: These performance results were measured using specific computer systems and/or components within
specific lab environments and under specific system configurations. Any difference in system hardware, software design, or configuration may affect actual performance.
The results are furnished for informational use only and should not be construed as a commitment by Dialogic. Dialogic assumes no responsibility or liability for any errors
or inaccuracies.
Outbound Dialing/Telemarketing: Outbound dialing systems may be subject to certain laws or regulations. Dialogic makes no representation that Dialogic products will
satisfy the requirements of any such laws or regulations (including, without limitation, any regulations dealing with telemarketing).
Any use case(s) shown and/or described herein represent one or more examples of the various ways, scenarios or environments in which Dialogic products can be used.
Such use case(s) are non-limiting and do not represent recommendations of Dialogic as to whether or how to use Dialogic products.
**Configurable to meet country-specific PTT requirements. Actual specification may vary from country to country for approved products.
Copyright © 2009 Dialogic Corporation

All rights reserved.

07/09

10339-03



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Subject                         : Dialogic, DM3 Media Boards, DM3, DM/V300BTEPEQ, DM/V600BTEPEQ, DM/V1200BTEPEQ, media server, enterprise, public networks, voice processing, speech recognition, fax, and conferencing capabilities, PCI Express, PCIe, ISDN PRI, PRI, ISDN, Primary Rate Interface, Messaging and enhanced services, Wireless, Short Message Service, SMS, color ring back tone, voice portal, contact center and e-Business, PC-PBX, audio conferencing server, web conferencing, switching, gateway switch, System Release 6.0 PCI for Windows, System Release 6.1 for Linux, Global Call, R4, Continuous Speech Processing, CSP, Enhanced Echo Cancellation, EEC, Dual Tone Multi-Frequency, DTMF, TrueSpeech, Windows Media Player, SDK, software development kits, Simple Network Management Protocol, SNMP, Direct Dialing In, DDI, Non-Facility Associated Signaling, NFAS, D-channel backup, Information Elements, IE, Call-by-call service selection, user-to-user information, LAP-D Layer 2 access, protocol timer setting, mask-able Layer 2 Control
Creator                         : Dialogic Corporation
Description                     : Datasheet.DM3 Media Boards
Title                           : Dialogic® DM3 Media Boards.
Startup Profile                 : Print
Producer                        : Adobe PDF Library 8.0
Trapped                         : False
Page Count                      : 17
Author                          : Dialogic Corporation
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