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

User Manual: Media Board DM/V600BTEPEQ

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Dialogic® DM3 Media Boards Datasheet

DM3 Media Boards

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

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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, speech 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

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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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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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DM/V600BTEPEQ

PCI Bus

E1/ISDN

Trunks to

PSTN or PBX

CT Bus

Third-Party Application Development Toolkit

ASR Engine by Third Party Software

Hardware

Dialogic® DM/V600BTEPEQ

Media Board

Dialogic® Global Call API and

Dialogic® R4 API

Dialogic® DM/V3600BP

Media Board

DM/V1200BTEPEQ

PCI Bus

T1/ISDN

Trunks to

PSTN

T1/ISDN

Trunks to

PSTN

CT Bus

Software

Hardware

Third-Party Application Development Toolkit

ASR Engine by Third Party

Dialogic® DM/N960-4T1 Digital

Telephony Interface Board

Dialogic® DM/V1200BTEPEQ

Media Board

Dialogic® Global Call API

and Dialogic® R4 API

DM/N960-4T1

Figure 1. Terminate Conﬁguration: Uniﬁed Messaging Application

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

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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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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.

• Conﬁgured 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).

• Conﬁgured 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:

• Non-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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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 silence-

compressed 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 listen-

only 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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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stufng

 •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

Technical Speciﬁcations

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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 +12 VDC +3.3 VDC

DM/V300BTEPEQ 0.6 0.58

DM/V600BTEPEQ 1.25 0.82

DMV1200BTEPEQ 1.25 0.82

Operating Requirements

Operating temperature +32°F (0°C) to +122°F (+50°C)

Storagetemperature –4°F(–20°C)to+158°F(+70°C)

Humidity 8% to 80% noncondensing

Approvals and Compliance

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

Technical Speciﬁcations (continued)

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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

Technical Speciﬁcations (continued)

Dialogic® DM3 Media Boards Datasheet

DM3 Media Boards

13

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,congurablebyparameter**

 •(E1)–39dBmto0dBmpertone,congurable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.

Technical Speciﬁcations (continued)

Dialogic® DM3 Media Boards Datasheet

DM3 Media Boards

14

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 Supports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

Technical Speciﬁcations (continued)

Dialogic® DM3 Media Boards Datasheet

DM3 Media Boards

15

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 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 Defaultsettingdesignedtodetect83outof87uniqueringbacktonesusedin96countriesasspecied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-denable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,congurablebyparameter**

 •(E1)–7.0dBmpertone,nominal,country-specic**

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

Technical Speciﬁcations (continued)

Dialogic® DM3 Media Boards Datasheet

DM3 Media Boards

16

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-y

ECM Supported

ASCIItoTIFF Onboard,on-the-y

Pageheaders Generatedonboard,on-the-y

Width A4

Polling Normal and turnaround

Resolution •Standard(100dpi×200dpi)

 •Fine(200dpi×200dpi)

 •Superne(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 Many country-speciﬁc MFC-R2 variants. For more details, refer to the latest Global Call Protocols Package

release notes

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)•

Technical Speciﬁcations (continued)

www.dialogic.com

Dialogic Corporation

9800 Cavendish Blvd., 5th oor

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 specications, 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 Specications forms part of the specications of the

Dialogic DM/V300BTEPEQ, DM/V600BTEPEQ, and DM/V1200BTEPEQ combined media board products and any benets specied are not guaranteed.

Positive Answering Machine Detection/Positive Voice Detection: These performance results were measured using specic computer systems and/or components within

specic lab environments and under specic system congurations. Any difference in system hardware, software design, or conguration 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.

**Congurable to meet country-specic PTT requirements. Actual specication may vary from country to country for approved products.

Copyright © 2009 Dialogic Corporation All rights reserved. 07/09 10339-03

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