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Introduction To DMR Study Guide-Tait Radio Academy PDF

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339 views106 pages

Introduction To DMR Study Guide-Tait Radio Academy PDF

Uploaded by

nikola
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
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You are on page 1/ 106

TaitN

Net DMR
D
DMR R Intrroduction
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Trainiing Ma
anual

DMR-INT: V1.00.04
V
[This page is intentionally blank]
Page 1

CONTENTS

CHAPTER 1
STANDARD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
1.1 Learning Outcomes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5
1.2 The Development of DMR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .6
1.3 What Is DMR? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .6
1.4 DMR ETSI Standards . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .7
1.5 Benefits of DMR (Tier III) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .7
1.5.1 Open Standard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .8
1.5.2 Increased Capacity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .8
1.5.3 Backwards spectrum compatibility . . . . . . . . . . . . . . . . . . . . . . .9
1.5.4 Digital Audio Quality . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .9
1.5.5 Digital Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .12
1.5.6 Longer battery life and greater power efficiency . . . . . . . . . . . .12
1.5.7 Advanced control features . . . . . . . . . . . . . . . . . . . . . . . . . . . . .12
1.6 Benefits of Trunked Radio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .13

CHAPTER 2
INTRODUCTION TO TRUNKING . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
2.1 Learning Outcomes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .15
2.2 What is Trunking? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .16
2.3 Trunking Efficiency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .16
2.3.1 Trunking Developed of to Improve Quality of Service . . . . . . . .16
2.3.2 Trunking Developed to Improve Channel Utilization . . . . . . . . .17
2.3.3 Origin of Trunking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .18
2.4 Trunked Radio Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .19
2.4.1 Trunked Radio Example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .20
2.5 Advantages of a Trunked Radio System . . . . . . . . . . . . . . . . . . . . . . . . .23
2.6 Optimizing Efficiency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .23
2.6.1 Organization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .23
2.6.2 Queuing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .24
2.6.3 Call Timing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .24

CHAPTER 3
NETWORK ARCHITECTURE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
3.1 Learning Outcomes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .25
3.2 DMR Network Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .26
3.3 DMR Network Elements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .27
3.4 Linking Infrastructure (IP Backbone) . . . . . . . . . . . . . . . . . . . . . . . . . . . .28
3.5 DMR Site Equipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .29
3.5.1 Site Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .29
3.5.2 TB9300 Base Station . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .30
3.5.3 TB9300 Stand Alone Node . . . . . . . . . . . . . . . . . . . . . . . . . . . .31
3.5.4 Embedded Node Priority . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .32
3.6 DMR Node Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .33
3.6.1 Block Diagram of a DMR basic network architecture . . . . . . . .33

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3.6.2 Capacity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .34


3.6.3 DMR Node Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .34
3.6.4 Node Equipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .34
3.7 DMR Network Gateways . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .36
3.7.1 TN8271 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .36
3.7.2 Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .36
3.7.3 T1542 Line Dispatch Terminal . . . . . . . . . . . . . . . . . . . . . . . . .37
3.7.4 Conventional Gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .38
3.7.5 MPT Gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .39
3.7.6 Telephone Gateways . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .40
3.8 DMR Network Management Elements . . . . . . . . . . . . . . . . . . . . . . . . . . .42
3.8.1 Node Web UI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .42
3.8.2 Tasks that can be carried out using the Node Web UI: . . . . . . .42
3.8.3 Tait EnableFleet . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .44
3.8.4 Tait EnableMonitor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .45
3.8.5 Tait EnableReport . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .46
3.9 DMR Mobile and Portable Subscriber Units . . . . . . . . . . . . . . . . . . . . . . .47
3.10 DMR Voice Recorder . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .48
3.11 Eventide NexLog Voice Recorder . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .48
3.11.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .48
3.11.2 MediaWorks PLUS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .49

CHAPTER 4
CHANNEL OPERATION AND CONFIGURATION . . . . . . . . . . . . . . . . . . . . . . . . . 51
4.1 Learning Outcomes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .51
4.2 Logical Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .52
4.2.1 Logical Channel Categories . . . . . . . . . . . . . . . . . . . . . . . . . . .52
4.3 DMR Channel Operation and Configuration . . . . . . . . . . . . . . . . . . . . . .53
4.3.1 Control Channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .53
4.3.2 Control Channel Facilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . .53
4.3.3 Control Channel Configurations . . . . . . . . . . . . . . . . . . . . . . . .53
4.3.4 Traffic Channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .54
4.3.5 Traffic Channel Configurations . . . . . . . . . . . . . . . . . . . . . . . . .54
4.4 Channel Numbering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .55
4.4.1 Channel Numbering Calculations . . . . . . . . . . . . . . . . . . . . . . .56

CHAPTER 5
CALL TYPES. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57
5.1 Learning Outcomes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .57
5.2 DMR Call Type Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .58
5.3 Voice Calls - Talkgroup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .58
5.3.1 Talkgroup - Conference Call . . . . . . . . . . . . . . . . . . . . . . . . . . .59
5.3.2 Talkgroup - Broadcast Call . . . . . . . . . . . . . . . . . . . . . . . . . . . .60
5.4 Voice Calls - Individual . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .61
5.5 Data Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .62
5.5.1 Control Channel Data - Status Message . . . . . . . . . . . . . . . . . .62
5.5.2 Control Channel Data - SDM . . . . . . . . . . . . . . . . . . . . . . . . . . .64
5.5.3 Traffic Channel Data - Packet Data Call . . . . . . . . . . . . . . . . . .65
5.5.4 Control and/or Traffic Channel Data . . . . . . . . . . . . . . . . . . . . .66
5.6 Gateway Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .68
5.7 Self Test Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .69

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Page 3

CHAPTER 6
CALL FEATURES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
6.1 Learning Outcomes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .71
6.2 DMR Call Handling Strategies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .72
6.2.1 Transmission Trunked . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .72
6.2.2 Message Trunked . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .72
6.2.3 Individual Call Setup Handling Strategy . . . . . . . . . . . . . . . . . .73
6.2.4 Group Call Setup Handling Strategy . . . . . . . . . . . . . . . . . . . . .73
6.2.5 Queuing Strategy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .74
6.3 DMR Call Timers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .75
6.3.1 Call Time Limits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .75
6.3.2 Other Timers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .75
6.4 DMR Emergency and Priority Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . .77
6.4.1 Emergency Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .77
6.4.2 Priority Group Override . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .78
6.4.3 Priority Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .80
6.4.4 Minimum Access Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .81
6.4.5 Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .82
6.4.6 Operational Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . .82
6.5 DMR Network Optimization Strategies . . . . . . . . . . . . . . . . . . . . . . . . . . .83
6.5.1 User Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .83
6.5.2 Individual Service Areas . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .83
6.5.3 Group Calls - Fixed Service Area . . . . . . . . . . . . . . . . . . . . . . .83
6.5.4 Groups Call - Controlled Service Area . . . . . . . . . . . . . . . . . . .88
6.5.5 Mobility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .90
6.5.6 Asset Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .90
6.5.7 Stun and Revive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .91
6.5.8 Authentication key . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .91
6.6 Subscriber Unit Call Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .92
6.7 Detailed Queuing Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .93
6.7.1 Queuing Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .93
6.7.2 Qued Call Scenarios . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .93
6.7.3 Channel Management for Queuing . . . . . . . . . . . . . . . . . . . . . .95

CHAPTER 7
NUMBERING . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97
7.1 Learning Outcomes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .97
7.2 Numbering Compatibility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .98
7.2.1 Numbering schemes available . . . . . . . . . . . . . . . . . . . . . . . . .98
7.3 DMR Numbering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .98
7.4 MPT 1327 Addressing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .99
7.5 Dialling Calls in MPT1327 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .101
7.5.1 MPT1327 Dialling String Summary . . . . . . . . . . . . . . . . . . . . .101

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1.1 Learning Outcomes Page 5

Chapter 1
Standard

1.1 Learning Outcomes


Upon completion of this chapter, you will be able to do the following:
• Explain the need for the development of DMR
• Define what DMR is
• Identify DMR ETSI standards
• Describe the benefits of DMR Tier III
• Explain the benefits of a trunked radio system

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Page 6 Chapter 1: Standard

1.2 The Development of DMR


Background
It was recognized by vendors and users of radio systems that there is the
need to supersede the existing analog trunking standards with modern
techniques to provide:
1. Improved voice quality
2. Improved functionality (i.e. Location information)
3. Improved security (i.e. Authentication)
4. Improved channel efficiency (2 slot TDMA)

1.3 What Is DMR?


Digital Mobile Radio (DMR) is an international digital radio standard
developed by the European Telecommunications Standards Institute
(ETSI), and first ratified in 2005. The standard now provides a full set of
standards covering voice, data services and conformance tests.
DMR aims to provide a economical, low-complexity digital standard to
replace analog radio. The ETSI DMR Standard, TS102 361, defines three
different tiers.
• Tier I (unlicensed): DMR equipment having an integral antenna and
working in Direct Mode (unit-to-unit) under a general authorization
with no individual rights operation.
• Tier II (licensed conventional): DMR systems operating under
individual licences working in Direct Mode (unit-to-unit) or using a
Base Station (BS) for repeating.
• Tier III (licensed trunked): DMR trunking systems under individual
licences operating with a controller function that automatically
regulates the communications.
Tait has adopted the ETSI DMR Tier III standard for its digital trunked
PMR system. The document ETSI TS 102 361-4 V1.5.1 (2013-02)
contains technical requirements for Digital Mobile Radio (DMR) trunking
systems operating in the existing licensed land mobile service frequency
bands.

Issue 1.00.04
© Tait Limited 2015
1.4 DMR ETSI Standards Page 7

1.4 DMR ETSI Standards


The full list of ETSI standards that define DMR primarily consist of four
documents:

ETSI TS 102 361-1:


"Electromagnetic compatibility and Radio spectrum Matters (ERM);
Digital Mobile Radio (DMR) Systems; Part 1: DMR Air Interface (AI)
protocol".

ETSI TS 102 361-2:


"Electromagnetic compatibility and Radio spectrum Matters (ERM);
Digital Mobile Radio (DMR) Systems; Part 2: DMR voice and generic
services and facilities“.

ETSI TS 102 361-3:


"Electromagnetic compatibility and Radio spectrum Matters (ERM);
Digital Mobile Radio (DMR) Systems; Part 3: DMR data protocol".

ETSI TS 102 361-4:


"Electromagnetic compatibility and Radio spectrum Matters (ERM);
Digital Mobile Radio (DMR) Systems; Part 4: DMR trunking protocol".

1.5 Benefits of DMR (Tier III)


DMR has the following benefits:
• Open Standard (non-proprietary)
• Increased capacity (TDMA)
• Backwards spectrum compatibility with legacy analog systems
• Digital Audio Quality
• Digital services
• Longer battery life and greater power efficiency
• Advanced control features
• Advantages of a DMR trunked radio system
• Trunking efficiency

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Page 8 Chapter 1: Standard

1.5.1 Open Standard


DMR is an ‘open’ standard and is not proprietary to a single
manufacturer. This means that competition is possible between
manufacturers not only when a new system is purchased, but over the life
time of the system (e.g. each time new subscriber units are purchased).
1.5.2 Increased Capacity

Time Division Multiple Access (TDMA)


DMR is a TDMA technology which offers the principle benefit of two
simultaneous and independent talk paths in one single 12.5 KHz channel.

e e
Po Tim Po Tim
we we
r r

12. 12.
5 kH 5 kH
z z

Fre Fre
q ue q ue
n cy n cy

Figure 1: Two logical channels per 12.5kHz physical channel

Each voice burst in the DMR two-slot TDMA carrier provides a ‘vocoder
socket’ for 2 × 108 bits vocoder payload to carry 60 ms of compressed
speech.
This multiplexing technique allows for a predictable doubling of capacity
in existing 12.5 kHz licensed channels, and as such allows for ease of
analog to digital migration.

Note: Vocoder uses 20 ms vocoder frames, the burst will carry three 72-bit
vocoder frames (including FEC) plus a 48-bit synchronization word in a voice
burst, that is 264 bits (27.5 ms) used for the burst contents.

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1.5 Benefits of DMR (Tier III) Page 9

1.5.3 Backwards spectrum compatibility

Analog to Digital Migration


As DMR is designed for ease of analog to digital migration, a major
design goal was that the output spectrum must fit in to the existing 12.5
kHz narrowband FM channels used by legacy analog systems.

DMR Modulation
With this design criteria, the choice of modulation scheme and associated
symbol rate were critical. The outcome was that 4FSK, 4-level Frequency
Shift Keying, modulation was used with an associated symbol rate of
4800 symbols/sec (9600bits/sec, i.e. 2 bits per symbol).

4FSK
4FSK is essentially frequency modulation that employs four frequency
deviation levels. The frequency deviations adopted for 4FSK are +/-648
Hz and +/-1944 Hz.
Mapping of the data bits to the symbols is shown in the table below:

Symbol Bits Frequency offset from Fc


+3 0,1 +1944 Hz
+1 0,0 +648 Hz
-1 1,0 -648 Hz
-3 1,1 -1944 Hz

4FSK modulation contains no amplitude content, so simple transmitters


similar to those found in analog FM systems can be used.
Both the downlink (base-station to terminal) and uplink (terminal to base-
station) use this modulation.
1.5.4 Digital Audio Quality
For the end user, one of the key benefits of a change from analog to DMR
digital radio technology is the improvement in audio quality.

Analog Signal
An analog signal will gradually weaken and become harder to use as the
distance from the site is increased. The user will experience increased
amounts of ‘hiss and crackle’ until finally the received audio is
completely lost in noise.

Digital Signal
A digital signal will remain clear to the edge of coverage.
DMR systems use a device called the AMBE+2™ vocoder to convert
voice information into digital data. During the digitization process, the
background noise, typically present in analog systems, is reduced.

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The data is then protected using Forward Error Correction (FEC), and
Cyclic Redundancy Check (CRC) coders before being transmitted over
the air.
These coders enable receiving radios to detect and automatically correct
transmission errors by analyzing bits inserted into messages that enable
the receiving radio to tell if there is an error.
Through the use of coders and other techniques, digital processing is able
to screen out noise and re-construct signals from degraded transmissions.
The result is that full audio quality is maintained by the built-in error
correction right to the edge of the coverage area. Fringe areas that were
difficult to operate in under the analog system will become loud and clear
under a DMR system.

Key: Digital Analog Digital Advantage

Good Audio Quality 30dB

ity
20dB

l
ua
Q
o
di
Au
12dB
Minimal Acceptable Audio Quality
Coverage

Poor Audio Quality 7dB

2.5 V 0.45 V 0.19 V 0.14 V


-99dBm -114dBm -121dBm -124dBm

Strong Signal Weak Signal

Figure 2: DMR Coverage vs. Analog Coverage

Radiated power and range


Digital coding allows significantly improved recovery of the wanted
signal in the presence of noise. This coding gain is often used to provide
better absolute range. However, to apply this in the case of migrating from
a narrowband analog radio network to a DMR network would have severe
impact on the frequency re-use and interference potential in the land
mobile radio bands.
The DMR standard considers similar transmit powers being used as
analog networks, but the coding gain being employed to provide a good
quality service to the edge of the planned coverage but thereafter a fairly
rapid roll-off.
By this means it is believed that the spectrum planning assumptions used
for the analog service will remain valid for the digital upgraded service.
This is a careful balance to achieve.

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Vocoder
The advanced multi-band excitation (AMBE) vocoder from Digital Voice
Systems, Inc. (DVSI) has been selected by the DMR MoU group as the
preferred vocoder for interoperability.
A vocoder (voice encoder/decoder) compresses the transmitted digital
voice signal to enable it to ‘fit’ into a smaller bandwidth channel and at
the receiving end it un-compresses the signal. Different digital standards
use different vocoder technologies. A full-rate vocoder compresses voice
sufficiently for it to fit in a narrow-band (12.5 kHz) channel. A half-rate
vocoder is necessary to compress it enough to fit into a 6.25 kHz channel
or in one 12.5 kHz TDMA time slot such as used by DMR.

Although the ETSI DMR standard does not specify the use of a particular
vocoder, DMR Association members have agreed to use the Advanced
Multi-Band Excitation (AMBE+2) half-rate vocoder to ensure
compatibility between different manufacturers' equipment. This vocoder
is a proprietary software device produced by Digital Voice Systems Inc.
(http://www.dvsinc.com/products/software.htm).
DVSI AMBE+2™ is based on Multi-Band Excitation (MBE), i.e. a
frequency domain approach.
The main characteristics are:
• very low bit rate 2450 bps (voice) + 1150 bps (FEC) = 3600 bps
• very high voice quality at very low bit rate
• robust to strong background noise and to PMR/LMR channel
• moderate complexity, easy to implement on a low-cost DSP
• language independent
• proven technology MBE family was adopted by TIA for P25 and
in many mobile radio satellite standards
• 20 ms voice frame and FEC optimized for PMR/LMR
applications

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1.5.5 Digital Services

Data Applications
The end-to-end digital nature of DMR enables applications such as text
messaging, GPS, and telemetry to be easily added onto radio devices and
systems. As the DMR standard also supports the transmission of IP data
over the air, this enables the easy development of standard applications. In
a world which increasingly relies on data as well as voice communication,
this ability to add a wide range of data applications to your system results
in the greatest possible return on your investment. In fact, one of the key
drivers for users switching to digital is to add business enhancing data
services and applications to radio systems.
1.5.6 Longer battery life and greater power efficiency
One of the biggest challenges with mobile/portable devices has always
been battery life. In the past, there have been limited options for
increasing the talk time on a single battery charge.
Two-slot TDMA, however, offers a good way forward. Since an
individual call uses only one of the two time slots, it requires only half of
the transmitter's capacity. The transmitter is idle half of the time.
For typical Portable radio operating with the standard portable 5/5/90 duty
cycle (5% Transmit, 5% Receive at full audio, 90% Standby) this
effectively means you are only really transmitting for 2.5% of the time
(half of the 5% Tx allowance). Given that the highest current draws on the
battery occurs when the radio is transmitting, we can see the effective
2.5% transmit time reduction results in a significant increase in the shift
life of the battery (i.e.: the time between charges). This translates to a 40%
increase in battery shift life.
1.5.7 Advanced control features
The DMR standard allows for the ability to use the second time slot for
reverse-channel signaling that is, instructions in the form of signaling
being sent to the radio on the second time slot channel while the first
channel is in a call. This capability can be used for priority call control,
remote control of the transmitting radio or emergency call pre-emption
and gives precise control and flexibility to the operator of a radio system.
FDMA systems cannot deliver similar functionality because they are
limited to one path only per spectrum channel.

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1.6 Benefits of Trunked Radio


A DMR trunked radio system has several important advantages over a
conventional radio system:

Features and Benefits of DMR Trunking


DMR Tier III Feature Benefit
Fast call set-up times and efficient use of channels Less waiting
World-wide acceptance Known technology, reference sites
Caters for small to large networks Flexible and upgradeable
Frequency transparent Can choose best available frequency
Latest technology Future proof
Non proprietary protocol Choice of supplier
4FSK constant envelope modulation No need for linear PAs - hardware cost lower
Priority and emergency call options Safety and security
Use of authentication keys Access security
Provides a data medium Can use text messages, GPS etc.
TDMA (2 logical channels per 12.5Khz) Fewer physical channels required
Supports a multiple of call types PSTN, data, talk groups all possible as standard

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Chapter 2
Introduction to Trunking

2.1 Learning Outcomes


Upon completion of this chapter, you will be able to do the following:
• Define the term trunking as it applies to radio communications.
• Explain how trunking improves service and efficiency of a multi
channel radio system.
• Explain the difference between Transmission trunking, Message
trunking and quasi transmission trunking.
• Explain what is meant by the term Control Channel and describe its
purpose.
• Explain what is meant by the term Traffic Channel and describe its
purpose.

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2.2 What is Trunking?


Trunking describes the process of selecting one clear communications
path from many possibilities. Trunking is based on the premise that if 100
users are sharing a certain communications network, only around 10 users
will actually use the network at any one time. Trunking is used in many
forms of telecommunications.
This principle can be applied to radio systems where a small number of
channels can be shared by many users. In trunked radio communications
channel allocation is:
• Dynamic
• Automatic
An added benefit of trunked radio communications is the ability to free up
(pre-empt) resources for a radio user in the event of an emergency.

2.3 Trunking Efficiency


2.3.1 Trunking Developed of to Improve Quality of Service
Historically, organizations with a significant number of mobile staff had
to rely on conventional radio systems restricted to fixed frequency
channels that the radios were limited to transmitting on the channel it had
been manually set to. The result was that some channels were
overcrowded while other channels were unused. These problems were
compounded for customers requiring communications coverage over
extended areas.
The diminishing availability of radio spectrum began to cause concern in
the early 1980’s and it became obvious that more efficient management of
the frequency spectrum, and allocated channels was necessary to improve
quality of service to users.

A B C

Channel in use Channel in use Channel not used


other users wait other users wait

X%@#! Unavailable to
groups assigned to
channels A and B

Waiting for a channel

Figure 3: Channel Congestion on a Conventional Radio System

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2.3.2 Trunking Developed to Improve Channel Utilization


An analogy for comparing conventional radio systems to trunked radio
systems, could be made to walking into a bank to make a withdrawal and
finding you had to wait in a long queue in front of the one teller who
processes withdrawals while another teller who processes only deposits
had no one waiting. A much better system would be to allow the bank
tellers to process any type of transaction, then you could simply go to the
first one that is available.

Conventional Bank Trunked Bank

Figure 4: Without trunking - queues become longer

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2.3.3 Origin of Trunking

Trunked Phone Lines


The name “trunk” comes from the telephone industry. Trunk lines are the
telephone lines that run between telephone exchanges and are different
from the line that runs to your house. If you call from your telephone
exchange to another telephone exchange, the switching equipment at your
exchange assigns your call a trunk line that runs to the other exchange. In
effect, you “borrow” a trunk line for as long as you are connected.
When you hang up, your exchange recovers the trunk line you were using
and makes it available for assignment to another caller. Therefore, it is not
necessary to install 100 trunk lines to serve 100 telephone customers; only
10 lines will be sufficient to provide a high level of service.






  


 


  

Figure 5: Trunked lines between telephone exchanges

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2.4 Trunked Radio Systems


Advances in technology provided a break-through in the form of low cost
single chip microprocessors. This allowed the concept of trunking to be
applied to mobile radio systems. A better name for trunked radio would
be “computer aided radio” as it is the application of microprocessors and
synthesizers that enables Trunked Radio Systems to share a pool of radio
channels between many groups of users.
A trunked radio system has:
• A Control Channel that is used to send messages between the
trunked system and the subscriber units.
• A number of Traffic Channels used for the voice calls.
Each group of users gets the exclusive use of a Traffic Channel for the
duration of their call. No other groups are using the channel at the same
time. A call has different meanings depending of the type of trunking:
• In Transmission Trunking, a call is a single over (press of the
PTT).
• Quasi-Transmission Trunking uses a “Hang Time”. A reply
within the hang time is part of the same call and uses the same
traffic channel.
• In Message Trunking, a call may consist of several overs (a
conversation) and continues until one of the users presses a button
to end the call. It is typically used for individual calls.
To set up a voice call on a trunking system:
• A subscriber presses the PTT, and the subscriber unit transmits a
call request to the system via the control channel.
• The system sends, via the control channel, a Channel Grant
message to the calling subscriber unit and the subscriber unit (or
group of subscriber units) that they called.
• All the subscriber units involved in the call then tune to the
designated traffic channel, and the call takes place.

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2.4.1 Trunked Radio Example


A trunked site has a control channel and a number of traffic channels. The
number of traffic channels required depends on the number of groups
using the system and the number of calls taking place. In this simple
example there are only two traffic channels, where on a real network there
may be more to handle the required call capacity. There may be hundreds
of subscribers on a trunked system but for simplicity this example shows
just 11 subscribers. The subscribers consist of different groups or teams -
in this example there are fire trucks, ambulances, police cars and highway
patrol cars.
Trunking allows all these different user groups to share the same two
traffic channels. In the picture below a call is in progress between the two
fire trucks and the system has assigned Channel A for this call.

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If one of the ambulances wanted to talk to the other ambulances, then


when they make their call, the subscriber unit would use the control
channel to send a request for a traffic channel to the system. The system
would send a message back, which automatically directs all the
ambulances to the available traffic channel in this case Channel B.

The subscribers would use this channel for their call.

When either of these calls finish that traffic channel is available to any
subscriber for another call. The subscribers talking on the radio do not
need to know what channel they have been allocated for the call; that all
happens automatically in the subscriber units.

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In the more detailed example below the system has a control channel and
two traffic channels, which are sometimes referred to as payload
channels. For simplicity the picture shows just 4 radio users.
Group call • Radio 1 has sent a request to the control channel to initiate a group call
example:
• The system would send a message back which automatically switches
the Radios to Channel A
In this case all radio units are part of a Talk-group (conference) call.

Figure 6: Group Calls

Individual call A call is in progress between Radio 1 and Radio 2 and the system has
example: assigned Channel A for this call.
• If Radio 3 called Radio 4 then the radio unit would use the control
channel to send a request for a traffic channel to the system.
• The system would send a message back which automatically switches
the Radios to Channel B and the units would use this channel for their
call.
The users do not need to know what channel they have been allocated for
the call, as the selection happens automatically in the radio units.

Figure 7: Individual Calls

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2.5 Advantages of a Trunked Radio System


A trunked radio system has several important advantages over a
conventional radio system:
1. Easier operation for users
2. Efficiently allocate channels to a call compared to a conventional sys-
tem.
3. Queuing time decreased therefore reduced waiting for a free channel.
4. Fewer channels required
5. Reduced hardware cost.
6. Private calls as channel allocated exclusively for a single call.
7. Features such as call records are available

2.6 Optimizing Efficiency


The key reason for adopting Trunking technology is that it massively
increases the productivity and usefulness of a multi-channel mobile radio
system.
In fact, the inherent efficiency of trunked radio is such that more than 50
users can potentially be handled per channel. Several hundred radios may
be able to share a multi-channel system without chaos during heavy traffic
periods
The way this is made possible is found in the interrelationship of three
factors:
• Organization.
• Queuing.
• Call timing.

2.6.1 Organization
Trunked radio systems are complex and so good organization is vital. A
trunked radio system may support a number of different organizations,
each using just one or even several fleets each with a number of talk
groups.There is typically many individual radio I.D’s, theoretically; over
one million addresses are available that can be assigned to individuals or
Talk-groups. The network manager is responsible for planning and
organizing the fleets and groups within each customer group.

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2.6.2 Queuing
A “queue” in this instance is a line-up of people wanting to use a two-way
radio system. However, it can be applied to almost any situation where
people line-up to receive some service. The trunked radio system places
callers in a queue when no free channels are available. Queue times are
usually short, and in a Tait trunked system, when a free channel is
available the call is automatically setup. Access to channels is controlled
dynamically. Queuing and channel assignment are handled by the system
infrastructure.

2.6.3 Call Timing


Mobile radio voice transmission times can often be less than 10 seconds
long. With this short transmission time, the trunked system controller can
almost always find a frequency that is open for a transmission. The
Trunking system controls call times to ensure equal access of users.

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Chapter 3
Network Architecture

3.1 Learning Outcomes


Upon completion of this chapter, you will be able to do the following:
• Identify and explain the purpose of the key components of a TaitNet
DMR Trunked network.
• Illustrate a typical DMR network.
• List DMR network elements.
• Explain why IP was chosen as the method of moving voice and data.
• Explain how the architecture scaled to suit customer requirements.
• Describe DMR site equipment and the different system configurations.
• Describe DMR node architecture.
• Describe DMR network gateways.
• Explain how dispatch consoles can be connected to the network and
list the typical functions provided.
• Explain how conventional radio systems can be connected to the DMR
network.
• Explain how telephone systems can be connected to the DMR network
• List DMR network management elements.
• Identify DMR mobile and portable subscriber units.
• Explain how voice recorders can be connected to the DMR network.

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3.2 DMR Network Overview


A typical DMR network consists of one or more node controllers, and a
number of sites, each of which consists of several base stations connected
by an IP backbone that can be either a switched local area network
(LAN), or a routed wide area network (WAN) through the use of routers
and bearers.
This network design is scalable from a single site to a large, wide area
network with multiple node controllers controlling hundreds of sites.
Open standard protocols are implemented to provide gateways to non-
DMR base stations/repeaters and digital or analog dispatch console
equipment.
Radio networks of differing manufacturers and technologies can also be
connected together with the Tait DMR network, creating a simple
migration path or a large scale communication systems

IP interfaces to other
equipment
SITE 1

SITE 2

WAN High Availability Nodes


TP9300 Portable

SITE 3
Network Gateway

TM9300 Mobile

TM9300 Mobile

Figure 8: DMR Network Overview

The DMR network is further enhanced with the addition of solutions from
the Tait Enable Management Tools.

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3.3 DMR Network Elements


The key elements of the network, described in more detail in following
sections, are:
1. Linking infrastructure (IP backbone) - interconnects the various
elements of the DMR network.
2. DMR site equipment (base station/repeater) - provides the RF path to
and from the mobile and portable radios for the voice or data
communications.
3. DMR node(s) - control the call setup, generate and store call records
and raises alarms.
4. Network gateway(s) - provide an audio interface to equipment and
systems outside the DMR system.
5. Telephone gateway(s) - support direct communications between radios
and external telephones through the PSTN/PABX.
6. Network management - including Tait EnableFleet, Tait
EnableMonitor and Tait EnableReport.
7. DMR mobile and portable subscriber units- used to communicate
between radio users and other network connected devices.
8. A full Tait DMR solution will, as indicated in the diagram below,
integrate a wide variety of third-party elements like voice recorders,
dispatchers and applications, (e.g. Automatic Vehicle Location - AVL)

Linking Equipment Linking Equipment IP Switch

LINKING
Microwave
TP9300 Portable E1/T1
Fibre-optic High Availability Nodes

TB9300 Base Station


TM9300 Mobile

TM9300 Mobile

EnableReport EnableMonitor

VOICE
PSTN/PABX GATEWAYS
RECORDER APPLICATIONS DISPATCH NETWORK MANAGEMENT

Figure 9: Tait DMR Solution

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3.4 Linking Infrastructure (IP Backbone)


The DMR trunked network connects the radio sites together using Internet
Protocol (IP).

The advantages of using an IP:


Flexibility
• Fault-tolerant packet-switched connection less architecture
• Scalable - easy to meet future requirements
• Secure
Reduction in costs
• Reduce bandwidth consumption by using high-performance
compression algorithms
• Lower line costs, fully redundant ring architecture can be used
versus a hub and spoke plan
• Lower maintenance costs as only one infrastructure for data and
voice needs to be maintained
Multiplexing voice and data
• Lower maintenance costs as only one infrastructure for data and
voice needs to be maintained
• Network convergence
• Quality of Service (QoS)
• Web enabled applications and interfaces for simple open network
management

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3.5 DMR Site Equipment


3.5.1 Site Architecture
A DMR trunking site consists of the following components:
1. TB9300 base station
• One Primary logical channel is known as the Control Channel.
• Other secondary logical channel(s) are known as the Traffic
Channel(s).
2. Ethernet switch (1 port per physical channel)
3. Transmit and receive antenna equipment
The antenna equipment includes receive multi-coupler equipment and
transmit combiner equipment as well as the antennas themselves.
4. Other equipment that may also be installed at the radio site depending
on system requirements - include:
• Router
• Power supply/uninterruptible power supply (UPS)
• Power distribution equipment
The equipment is mounted in standard 19-inch (483 mm) racks/cabinets.

TB9300
Ethernet connection to node
via IP bearer network
Tx Rx

RF combining TB9300
equipment

RF
cabling TB9300

Figure 10: Diagram of a typical DMR Site

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3.5.2 TB9300 Base Station


The base stations at a site provide the RF interface to the radios using the
network. At each site, there is one control channels and a number of
traffic channels.

Figure 11: TB9300 Base Station

Channel equipment consists of the TB9300 base station, with each 4RU
19” shelf containing one or two RF channels depending on system
requirements. Each base station provides two “logical” channels each
time sharing the full 12.5kHz bandwidth.

Tx output

Transmit Receive Rx input


output input

Power Amplifier Reciter


Power
Power Power Reciter (Receiver
Management Amplifier
Management + Ethernet Interface
Unit
Unit Exciter) cable between
TB9300 & Switch
Router
TB9300
TB9300 Ethernet cable
Reciter - Switch
Ethernet Switch

To DMR
Node
Router Ethernet
Switch

Figure 12: TB9300 DMR Channel Architecture

A single trunking site can consist of up to 20 channels that can operate


independently from a node. For instance, if the base stations at a site
detect that the node has failed, or the link to the node has failed, the site
can be configured so that the site can continue operating as a single site
controlled by a single channel reciter acting as a standalone node.

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3.5.3 TB9300 Stand Alone Node


With this feature a TB9300 base station at an individual site operates as an
embedded node controller. This allows one of the following options to be
implemented.

Fallback Node - Single Site Trunking


This embedded node is expected to be used only as a fallback node. If
configured as ‘Control Channel’, it will try to take control of the site if it
discovers that the site has lost connectivity to the network. When this
node is master, it will send a ‘limited connectivity’ message to
subscribers.

SITE 1
embedded

node
CH 1 CH 2

Link to
node down
CH 3 CH 4 SITE 2

CH 5 CH 6
SITE 3

Node
WAN

Figure 13: Fallback Node - Single Site Trunking

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Single Site Trunking Node


This site does not normally connect to a bigger multi-site network. The
embedded stand alone node is configured to operate the site on a stand-
alone basis. When this embedded node is master, it does not indicate
‘limited connectivity’ to subscriber units over the air interface.

embedded

node
CH 1 CH 2

CH 3 CH 4

CH 5 CH 6

Figure 14: Single Site Trunking Node

3.5.4 Embedded Node Priority


This must be unique for each base station on the site. The lower the
number the higher the priority. The value range is 1 to 255, but it is
recommended that only the higher numbers be assigned to Stand Alone
Nodes to give network nodes priority.
In the event of more than one ‘Control Channel’ Stand Alone Node on the
site, that with the highest priority will become the master node for the site,
with its base station carrying the control channel.

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3.6 DMR Node Architecture


The DMR control node and its associated switching nodes control call
setup, generate and store call records (for fault diagnosis and, where
appropriate, for subscriber billing) and raise alarms in response to fault
conditions. A node consists of a rack-mounted Oracle Netra X4270 server
(or later) running the Tait node controller software.
The node provides a large number of the interfaces to third-party
equipment and other networks:
• An interface to a SIP user-agent server that establishes connections
with third party SIP FXO Gateway devices (such as voice-enabled
Cisco routers and Cisco SPA voice gateways) and allows radio users to
connect to PSTN, analog PABX or digital PABX (IP telephony
networks).
• A digital SIP-based dispatch interface connected to a digital console
that can provide end-to-end encrypted voice and data communication
• An interface to Network Gateways
• An interface to conventional radio networks
• An interface to MPT gateways

3.6.1 Block Diagram of a DMR basic network architecture

SITE 1

SITE 2
TB9300

Tx Rx

RF combining TB9300
equipment

RF SITE 3
cabling TB9300

Node
WAN

Figure 15: Diagram illustrating basic network architecture

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3.6.2 Capacity
A Tait DMR network is scalable from a single site with one base station to
a large, wide area network with multiple nodes, 1000 base stations and
300 network gateways. One node is needed for every 100 talk paths in the
network. One or more additional nodes are desirable for redundancy. Each
reciter in a base station provides two voice channels and each gateway a
single voice channel. A network can have up to 20 nodes and a maximum
of 100 sites A table summarizing the DMR network dimensions can be
found at the end of this section.
3.6.3 DMR Node Operation
In an DMR network, a node has two functions:
• a control node
• a switching node
There is only one control node in a network. In a multi-node network, the
lowest numbered node is the control node. It validates all call requests,
sets up and clears down all calls, and sends call control messages and
routing instructions to the base stations. All nodes in the network can also
function as switching nodes. The switching nodes are used to transfer
audio data between the base stations (as per the control node instructions).
In a single-node network, the node fulfills both functions. Having more
than one node in an DMR network is advantageous, as:
• higher numbers of nodes spreads the load of voice traffic, keeping
queue times down
• if the control node fails, the next lowest numbered switching node will
take over as the control node
3.6.4 Node Equipment

TN9300 Node Controller


Shown below is the Sun Oracle X4270, the platform for the Tait TN9300
Node Controller.
The node controller performs IP packet switching, call control and
network management functions. It handles inter-channel calls and
connections to dispatchers and third party IP interfaces.

Figure 16: Sun Oracle X4270

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The node carries out the following functions:


• Setting up calls. This includes allocating logical channels to calls.
• Switching voice packets between interfaces. The node receives these
voice packets from base stations and SIP (Session Initiation Protocol)
interfaces and switches them to all the logical channels involved in the
call. Where multiple streams are received at the same time, the node
selects a single stream for forwarding to the required destinations.
• Receiving radio registrations, storing them in its registration database
and using them when setting up calls.
• Maintaining a validation database and using this to decide whether to
permit a call request.
• Generating and storing call records. These are used for fault diagnosis
and may be used for subscriber billing.
• Raising alarms in response to fault conditions.
• Providing a SIP user-agent server that controls SIP gateways. This
control is needed for calls involving telephone users and dispatch
consoles respectively.
Smaller networks only need one node (or two for redundancy), but larger
networks will have several, with one being primary. The control node
maintains the validation and registration databases and keeps the other
nodes up-to-date with any changes. Once the control node has set up a
call, it delegates the switching of the call's voice packets to one of the
other nodes.
If the control node fails, the switching node with the lowest node number
automatically takes its place.

Ethernet Switch
The Ethernet Switch is used to connect the node to the IP backbone. This
allows nodes to communicate with each other.

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3.7 DMR Network Gateways


3.7.1 TN8271
The TN8271 Network gateway is a general-purpose device that provides a
single voice channel interface to the network. It converts the AMBE voice
over IP (VoIP) used by the network into analog 4-wire audio.
3.7.2 Applications
The Network Gateway allows simple implementation of the following
solutions that are covered in more detail later in this section:
• Tait Line Dispatch Console (LDT)
• Conventional line interface
• Analog dispatch console
• MPT Gateway
• Analog voice recorder
The TN8271 Network Gateway is also a key element for implementing
telephone interconnect on the DMR network.
It has a web-based application for maintenance, configuration, diagnostics
and firmware upgrade as well as SNMP support for NMS monitoring.

Figure 17: TN8271 Network Gateway

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3.7.3 T1542 Line Dispatch Terminal


The Tait T1542 Line Dispatch Terminal (LDT) is a PC based dispatch
consoles that provides a simple yet powerful interface into the Tait
TN9300 DMR network.
With the LDT software and associated hardware, central dispatch
operations can effectively communicate with a large number of radio
users. The features and functions of the LDT include:
• Simple configurable user interface
• Identify multiple simultaneous emergency and non-emergency
calls from multiple users
• Displays caller/talker ID
• Display historical and missed calls
• simple talkgroup calling
• simple talkgroup monitor/un-monitor
• call hold
• call conferencing
The TN8271 Network gateway provides the 4 wire audio to the LDT
desktop audio interface as illustrated below.

Line Dispatch Terminal

Audio
Interface

4-wire Audio
Network
Gateway

High Availability Nodes


WAN
SITE EQUIPMENT PRIVATE IP NETWORK

Figure 18: TN8271 Network Gateway used for LDT interface

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3.7.4 Conventional Gateway


Conventional analog FM base stations or legacy analog dispatch consoles
with 4-wire E&M interfaces, can connect to the Tait DMR trunked
network through the TN8271 network gateway. This is useful for
interoperability with other organizations still using analog systems, and
also enables a staged transition to a DMR network.

Analog Dispatch Analog Base Station

4-wire E&M 4-wire E&M

Network Gateway

Network Gateway

High Availability Nodes


WAN
DMR SITE EQUIPMENT PRIVATE IP NETWORK

Figure 19: Conventional Gateway Application of the Network Gateway

The network gateway passes voice to and from the analog equipment over
its 4-wire audio interface, and control signals over the E & M connections
and provides a controlled and configurable communication link between
the sub-systems.
When a DMR radio initiates certain group calls, the node can be
configured to either include or exclude the network gateway(s) connected
to the analog equipment. Activity on the analog equipment can be
configured to set up a specific group call to the DMR network.

Note: The mapping between the analog equipment and the DMR
trunked talkgroup is fixed in the network gateway configuration.

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3.7.5 MPT Gateway


The MPT Gateway provides the means to interconnect the DMR network
to an external MPT 1327 network, typically a Nokia Actionet™ network,
allowing a smooth migration. It operates at the MPT 1327 air interface
level. Conceptually the gateway appears to be a population of radios to
the external MPT 1327 network. The gateway simulates MPT 1327
signaling, so that the external network thinks it is talking directly to a
radio. To the external network it looks like all the radios in the DMR
network are registered on the external system’s site that the gateway is
connected to. The following components are required for the gateway:
• T1430 Gateway Site Management (GSM)
• T1431 Gateway Channel Controller (GCC)
• Digi TS2 Port Server
The gateway requires a dedicated site on the external network. Each
channel on the external network site requires a Gateway Channel
Controller on the DMR site.
For Actionet systems a TA2529-01 Actionet Gateway Interface (AGI) is
used to provide electrical level conversion between the Gateway Channel
Controller and the Actionet Line Interface Card. This hardware interface
will correctly interface the RX/TX signals, as well as provide Tone
Generation and inactivity detection.
For other system types, the Gateway Channel Controller can be connected
to the external channel using normal RF equipment or by directly
connecting the 4-wire audio.
The following diagram shows the interconnections:

Serial port Network Network


server Gateway Gateway
TB9300
Intersite
RS232 Intersite Intersite
GSM Audio Audio
TB9300
Site Bus
GCC GCC
TB9300
Gateway Site AGI AGI
DMR Site

LIC LIC

ActioNet Virtual Virtual


TN9300 DMR Node Bridging Site Control Ch. Traffic Ch.

WAN

ActioNet
Central Controller
GSM - Gateway Site Module Mobile Exchange - MX
GCC - Gateway Channel Controller
AGI - Actionet Gateway Interface
LIC - Line Interface Card
ActioNet Site ActioNet Site ActioNet Site

Figure 20: MPT Gateway Architecture Overview

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3.7.6 Telephone Gateways


The Tait DMR network utilizing the TN8271 Network Gateway, can
support direct communications between radios and external telephones
through the PSTN. It also supports communications between radios and
private telephone extensions through a switchboard (PBX), without
dispatcher assistance (although dispatch equipment can also automatically
or manually patch telephone calls through to radios and vice versa.)

Telephony Interconnect / Gateway


Telephony gateways (for example, the Cisco 2911 as pictured below)
provide the interface that enables radio users and telephone users to
communicate with each other. Session Initiation Protocol (SIP) is used for
setting up and clearing down calls. These gateways also convert voice
between the digital format (G.711 ulaw) used within the Tait network
linking infrastructure and the analog format sent to and received from the
PABX or PSTN.

Interfacing with SIP phones


The Tait DMR network can interface directly to a SIP enabled PABX via
an IP connection. The control path goes directly to the node but the voice
path goes through a network gateway for transcoding. A network gateway
is required for each PABX voice path for translating between the DMR
AMBE+2 and PABX G.711 voice streams.

PSTN/PBX

SIP

Network
Gateways
G.711

IP Switch

AMBE+2

High Availability Nodes

Private IP network

Figure 21: SIP Telephone Interface

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Interfacing with non-SIP phones


If an interface to a SIP enabled PABX is not available, then the telephone
interconnect requires one or more third-party telephony gateways (for
example, the Cisco 2911 integrated service router) to interface the PBX or
telephone exchange to the DMR network.
The telephony gateway converts the IP voice traffic from the node to the
signaling format required by the PBX/PSTN and vice versa. SIP is used
for setting up and clearing down calls.

PSTN/PBX

2-wire, 4-wire or E1/T1

Telephone
Telephone Network
Gateway
Gateway Gateways

G.711
G.711
IP Switch

AMBE+2

High Availability Nodes

Private IP network

Figure 22: FXO/FXS Telephone Interface

The telephone user dials the number of a SIP line and then overdials the
number of an individual radio or group. The node receives the over-dialed
string and uses the rules in its in-phone table to find a match, replaces the
dialed digits with a number supplied by the in-phone table. It then sets up
the DMR call to the resulting number of the radio or group.
In the reverse direction, the radio user dials a string of numbers. When the
node receives the string, it uses the rules in its out-phone table to find a
match and replace the string with a telephone number. It then sets up the
call to that telephone number, routing it to a suitable FXO group. Out-
phone tables can be used to limit user access to the PSTN, for example.

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3.8 DMR Network Management Elements


Tait provides a number of configuration, monitoring and reporting tools to
provide effective network management on the DMR radio network.
3.8.1 Node Web UI
The node provides web-based terminal management and monitoring of
network elements, including the base stations, SIP and analog interfaces.

IP Switch

High Availability Nodes

TB9300 Base Station

Web
Browser

NETWORK MANAGEMENT

Figure 23: Node WEB UI Status Screen

3.8.2 Tasks that can be carried out using the Node Web UI:
Connect to the DMR node from a PC using an ordinary web browser to
carry out tasks such as the following:
• Check the alarm status of the network
• View/edit fleet information
• View/edit network parameters
• Monitor network operation
• Download call records and other files
• Make backups of the fleet and node configuration database
• Upload new node firmware

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The TB9300 Base Station equipment and the TN8271 Network Gateway
also provides web-based terminal management and monitoring to allow:
• maintenance
• configuration
• diagnostics
• calibration
• firmware upgrades
There is therefore no need for an installed application on the network
administrator's computer. Because the application is in the node, base
stations and Gateways themselves, there are no problems arising from
configuration software version incompatibilities.
Network administrators can access the TN9300, any TB9300, and any
TN8271 Network Gateway from anywhere on the network, using a secure
session with an ordinary web browser. More than one network
administrator can monitor network elements at the same time.

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3.8.3 Tait EnableFleet


EnableFleet is a configuration management system that can be either
hosted on the customer premises or in the cloud. A defined set of base
configurations is stored within a central database, for easy deployment at
installation time. The configurations managed are programming settings,
software feature licenses and firmware.
Centralized fleet management means improved efficiency, reliability and
cost saving.
The key components of EnableFleet are:
• EnableFleet Manager - for managing and maintaining the fleet
data.
• EnableFleet Client - for software deployment of radio
configurations and capturing installation data.
• EnableFleet Core - the SQL database containing the network
configurations.

data database data

MANAGE + EnableFleet Core DEPLOY


REPORTS - web service
- file generation
- REST interface API
- fleet configuration database Optional
EnableProtect
ASK dongle

EnableFleet Manager EnableFleet Client


- dashboard - local data store
- reports
- device and group configuration

Figure 24: EnableFleet Architecture

EnableFleet has two fundamental principles:


1. Configuration of terminals should be as simple as possible.
• Specialist radio technician skills are not required at
install/deploy/field upgrade time.
• Modifications to configuration are limited to only those permitted
by fleet managers.
2. Data integrity is essential:
• Knowing what version of hardware, firmware and configuration
they have
• Central point of management

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3.8.4 Tait EnableMonitor


EnableMonitor is a flexible network management platform that lets you
monitor, measure and report every aspect of your radio network health.
Using the Tait standards-based design approach, this robust software
application uses SNMP over IP technology, widely proven and trusted to
manage the most complex IT networks. EnableMonitor supports SNMP
v2c and the newer v3, which adds security features such as packet
encryption and message authentication.
Three core functions of EnableMonitor are:
• polling and logging
• alarms
• diagnosis

DMR network

CONTROL CENTER

database

EnableMonitor

REMOTE ACCESS

Figure 25: EnableMonitor Architecture

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3.8.5 Tait EnableReport


EnableReport moves beyond monitoring, to equip you with the expertise
and analysis tools to support your business needs and to optimize your
investment in a new digital network.
With EnableReport you can generate, develop and circulate vital network
performance information to key personnel to inform critical business
decision-making. The report generator reports on your business KPIs,
corporate policy and industry regulation. At the same time, valuable
trending information helps identify traffic hot-spots or latent issues that
may be preventing your network from performing optimally.
Tait can customize reports to suit your individual business needs.

network

database

EnableReport

manager technical analyst finance scheduled reports

reports measuring KPIs

Figure 26: EnableReport Architecture

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3.9 DMR Mobile and Portable Subscriber Units


The TP9300 TM9300 series is a range of high-performance
microprocessor-controlled DMR portable and mobile radios for voice and
data communication.
Tait DMR radios are designed with group communications in mind.
Simple user interfaces and flexible group configurations allow users to
maintain reliable communications within their team and also to other
teams or individuals when required. Group membership can be controlled
both by the radio operator and by a network administrator.

Hand-Held
Control Head
(HHCH)

TM9355 TP9355 TP9360

TM9300 Tait DMR mobile radio TP9300 Tait DMR portable radios

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3.10 DMR Voice Recorder


The TN9300's Voice Recorder Protocol (VRP) enables connection of
VRP compliant Voice Recorders to the TN9300 for recording Unit to Unit
Calls and Group Calls including SIP, AIS and DIP.

3.11 Eventide NexLog Voice Recorder


3.11.1 Overview
The eventide voice recorder can be used to record the voice stream from
the DMR node controller through the VRP interface. An additional DVSI
decoder is required to, on request, decode the AMBE encoded voice
sourced from the the DMR system for playback to the MediaWorks PLUS
browser-based incident replay and instant recall software using G7.11
voice coding.

AMBE Voice
VRP Data
Voice
Node
Recorder
G7.11 Voice AMBE Decoder

G7.11 Voice

Web browser-based recording management

Figure 27: Voice Recorder High Level Architecture

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3.11.2 MediaWorks PLUS


Eventide MediaWorks PLUS browser-based Incident Replay and Instant
Recall software for NexLog recorders allows you to easily search and
replay, and to re-create and save complex incidents that involve multiple
calls. MediaWorks PLUS software permits quick call browsing & replay,
and includes an advanced two-dimensional graphical time line view,
simultaneous multi-channel replay, spoken date & time announcements,
advanced incident recording management features, Instant Recall mode,
and much more.

Figure 28: Eventide MediaWorks PLUS

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Chapter 3 References

The following table gives the dimensions of a DMR network.

Element Limits

Sites / Network 200

Control Channels / Network 200

Physical Channels / Network 1000 (2000 logical channels)

Physical Channels / Site 20 (40 logical channels)

Nodes / Network 20

Active Audio Connections / Node 100


Each active connection consists of:
• reciter logical channel
• network gateway
• a call on an AIS trunk
• a call on an SIP trunk

Audio latency (from PTT including 400 ms


any required call setup to audio out
of b party)

Network Gateways / Network 300 (planned but untested)

DIP Connections 300 (planned but untested)

Concurrent SIP Calls 300 (planned but untested)

Maximum node to base station link 900 ms round trip time (planned but
latency untested)

Table 29:TaitNet DMR network dimensions

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Chapter 4
Channel Operation and Configuration

4.1 Learning Outcomes


Upon completion of this chapter, you will be able to do the following:
• Describe the concept of logical channels
• Describe channel operation and configuration
• Demonstrate channel numbering calculations

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4.2 Logical Channels


Communications between sites and radios are performed on radio
channels. Each channel consists of a frequency pair, one for transmission
and one for reception. DMR being a 2 time slot TDMA standard,
introduces the concept of logical channels.
A logical channel is defined as a communication pathway between two or
more parties, and represent the interface between the protocol and the
radio subsystem. A single RF frequency in a DMR radio system can
maintain two independent logical channels.
4.2.1 Logical Channel Categories
The logical channels may be separated into two categories:
• control channel
• the traffic channel (or payload channel)
A radio channel used for Signaling is known as a control channel. A radio
channel employed for user communications, i.e. speech or packet data
calls, is known as a traffic channel or payload channel.
Each site typically operates a single control channel and multiple traffic
channels. For multi-site group calls or intersite individual calls, a traffic
channel is used at each site involved in the call.
Channels can be configured with one or all of the following options:

Allow Control..
The channel is available for use as a control channel.

Allow traffic..
The channel is available for use as a traffic channel for calls. These can be
calls between two radios that are operating on different sites or a radio
communicating to a Dispatcher or PABX/PSTN line. The channel may
also be used for calls between radios registered on the same site

Inhibit if jammed..
The channel is monitored and will not be used as a traffic channel if
interference is detected. If interference is detected the channel is said to be
jammed. When the interference disappears then the channel is once again
available for carrying calls.
When allocating a traffic channel to the call, only idle channels which are
appropriately configured will be considered.

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4.3 DMR Channel Operation and Configuration


4.3.1 Control Channel
A logical channel, one time slot of a physical RF channel, is assigned as a
Trunk Station Control Channel (TSCC). Any channel at a site can be used
as a control channel but the site will normally select the lowest numbered,
uninhibited, channel. All other channels operate as traffic channels.
If the control channel fails, then it will be removed from service and the
next idle, uninhibited channel will take over as control channel. If there
are no free traffic channels, then the control channel will not be assigned
until one becomes free.
If a failed control channel recovers and comes back in to service it may
take over control from a higher numbered channel. The higher channel
will become a traffic channel again, after it has sent a command over the
air moving all radios operating on that site to the new the new control
channel.
4.3.2 Control Channel Facilities
Subscriber units require a control channel at a site to regulate channel
access. The control channel provides the following facilities:
1. management and control of channel access by subscriber units using a
random backoff mechanism;
2. processing service requests to and from subscriber units and optionally
to and from line connected entities;
3. allocating payload resources to calls;
4. broadcast of system information to subscriber units;
5. subscriber units location management by registration;
6. provision of services such as short data polling and transfer.
4.3.3 Control Channel Configurations

Dedicated Control Channel:


A Trunk Station Control Channel (TSCC) is transmitted continuously.
This channel occupies one DMR TDMA channel. Subscriber unit access
is strictly controlled and access is by invitation only. One TSCC can
support a large number of payload channels. There are a number of Tier
III services (such as short data messaging) that only utilize the TSCC.
This mode of operation yields the highest performance and throughput,
and is straightforward for network planning

Note: TaitNet systems do not support Time Shared Control Channels.

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Control Channel Reassignment


This feature can be used on networks where dedicated frequencies are
unavailable. Each channel operates as a control channel for a set period of
time. The control channel then moves to the next idle, uninhibited
channel. The original control channel, will send a command to all radios
indicating the frequency of the new control channel, before stopping
transmission. When the last available channel has been used, control will
return to the lowest numbered, uninhibited, channel.
4.3.4 Traffic Channel
Any logical channel on time slot 1 or 2, can be allocated as a traffic
channel on a physical RF channel. (excluding the timeslot allocated to the
control channel).
This channel is transmitted continuously from a base station site without
gaps as long as the base station is activated. If there is no information to
transmit, the base station transmits idle messages to fill out the bursts.
4.3.5 Traffic Channel Configurations

Traffic Channel Rotation


When enabled, the Traffic Channel Rotation parameter will cause each
consecutive call to be set up on the next highest channel following the
previously used channel. When the highest channel number is reached,
the call will set up on the lowest available channel, thus forming a
rotational channel allocation pattern. This is to help avoid the situation
where calls are being regularly set up on a low numbered channel that has
a problem.

Channel Partitioning
This feature allows the traffic channels at a site to be split into 20
partitions. Each partition details a set of channels that can be used by a
specified set of radios or call types.

Channel Pooling
This feature allows different sites to use common frequencies; it is useful
if the number of available frequencies is limited. For example a network
with 3 sites, each with 10 channels requires 30 frequency pairs, one pair
for each channel.
If only 20 frequency pairs are available, the network could be set up so
that each site has 5 unique frequency pairs. The remaining channels (5 at
each site) can share the remaining frequency pairs.
A downside of this strategy is that calls on a channel using a shared
frequency may cause interference at the other sites. To prevent this, these
channels can be pooled, i.e. shared between the sites. The node handles
the allocation of pooled channels and ensures only one site uses a shared
frequency at any one time, effectively giving a software ‘cross busy’ of
the channels that would otherwise interfere with each other.

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This feature does however increases the call set up time.

Note: If the sites are geographically far enough apart, interference may
not be a problem. In this case the frequencies can be re-used without the
need to pool the channels.

4.4 Channel Numbering


A particular DMR Tier 3 trunked network typically uses a specified
portion of the RF spectrum, known as a trunked channel block.
A trunked channel block is set of equally spaced channels on which a
trunking system operates. On a DMR trunked network the RF carrier
separation is 12.5 kHz. The control channel, and all traffic channels, are
selected from frequencies in this range.
Subscriber units are directed to payload channels (physical RF frequency
and logical number) by the control channel. The exact frequency the
subscriber unit re-tunes to is not explicitly sent by the control channel, but
a logical channel number and associated time slot is sent.

For example:
A subscriber unit may have service from a control channel with a logical
channel number of 21 that is being broadcast on TDMA timeslot 1. This
can be seen on the subscriber unit as 21A on the display.

During a call this unit may be sent to channel 148 timeslot 2. This will be
seen on the subscriber unit as 148B on the display.

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4.4.1 Channel Numbering Calculations


The subscriber unit now typically calculates the exact RF frequency from
the configured trunked channel block. To do this the subscriber units need
to know a base frequency or ‘Band Edge’.
The base frequency typically represents logical channel 1, i.e the lowest
frequency in the band, and all subsequent channels are calculated from
this point up to the highest frequency in the channel block.
Tait subscriber units use the following formula to calculate the frequency
to retune to.

Equation 1.

where:
FCH = centre frequency
FBASE - base frequency (lower band edge)
n = 1, 2, 3,… - channel number;
ChBW = channel bandwidth (channel spacing always 12.5kHz for DMR)

Example:
Licensing has allocated:-
a base frequency of 151.000 MHz (site transmit)
an upper band edge 163.4875 MHz
a Tx-Rx channel separation of +4 MHz
channel spacing 12.5kHz
Subscriber unit channel block programming is as follows:
Start Rx Start Tx Channel spacing
Start Channel Stop Channel
Frequency (MHz) Frequency (MHz) (kHz)
1 1000 155.0000 151.0000 12.5

If the subscriber unit is directed to channel 16 (FCH16TX & FCH16RX),


the exact transmit and receive frequencies are calculated using Equation 1
as:-
Channel 16 Transmit (FCH16TX):-
FCH16TX = 151.000 + [(16-1) * 0.0125]
FCH16TX = 151.1875MHz

Channel 16 Receive (FCH16RX):-


FCH16RX = FCH16TX + 4.0000
FCH16RX = 151.1875 + 4.0000
FCH16RX = 155.1875MHz

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Chapter 5
Call Types

5.1 Learning Outcomes


Upon completion of this chapter, you will be able to do the following:
• List the different DMR call types
• Describe a talkgroup conference call
• Describe a broadcast group call
• Describe an individual (private) call
• Describe a status message call
• Describe an SDM call
• Describe an packet data call
• Describe a gateway call

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5.2 DMR Call Type Overview


Trunked radio systems enable radio users to make a wide variety of
different call types, including:

Voice Calls:
• talkgroup
• individual (private)
• telephone and gateway
A trunked mobile radio network provides an unrivaled and very efficient
means of communication through the use of group calls.
Private calls extend the features of the radio network to provide a
complete solution for all of an organizations communications needs.

Data Calls:
• Status
• Short data messages (text)
• Packet data (confirmed/unconfirmed)

The technology that enables the various call types is provided by the
user’s radio in some cases and by the DMR network in others. The
network manager can control which call types any particular radio is
allowed to make.

5.3 Voice Calls - Talkgroup


A group or talkgroup is an assigned group identity on a trunked radio
system. When called, the system dynamically assigns frequencies to the
group. There can be single or multiple sites involved in call.
Group calls are either
• conference
• broadcast
and may be programmed on the subscriber unit as either permanent
groups, selectable/subscribed groups and/or scan groups.

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5.3.1 Talkgroup - Conference Call

Overview
Most of the calls made by the average user are group conference calls, and
all group members can contribute to the call.

Use Case
Application - Permanent Groups:
• Groups allocated for important network wide announcements.
• High priority calls that must be heard within a team.
Application - Selectable/subscribed Groups:
• Groups allocated for individual team communications.
• Join a number of different teams talkgroups to manage workload or
incidents efficiently, using a pre-set list, selector position or keypad for
easy access.
Application - Scan Groups:
• Background listening of the traffic of 2 or more talkgroups for general
awareness of work activity or incidents.
(Note some conversations or part of, may not be heard)

What to Expect:
A ‘go ahead’ tone will be heard when pressing the PTT indicating a
channel has been assigned and is available. All the terminals on the
talkgroup are called. Any member of the call is allowed to talk (one at a
time).
On subscriber units in the call the name/number of the currently talking
subscriber is shown (Talker ID) along with the call timer.

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5.3.2 Talkgroup - Broadcast Call

Overview
A broadcast group call prevents the receiving parties from transmitting.
Any subscriber unit is capable of making a broadcast call.

Use Case
Application:
• Announcement call from dispatch to a talkgroup. It is necessary that
the receiving parties do not interrupt so must not transmit.
• A periodic weather announcement, transmitting is not applicable.

This call is made using the menu or can be quickly made by dialing
*11*(group identity).

What to Expect:
Terminals in the called talkgroup can hear the broadcast call but cannot
reply to it.

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5.4 Voice Calls - Individual


Overview
An individual call is made from one subscriber unit to another. Other units
in the fleet do not hear it (private call).
A unit to unit call from subscriber A to subscriber B made by either
dialing the caller party ID or selecting a predefined ID through the radio
menu.

Use Case
Application:
• Private communication between two individual units
• Communication between dispatcher and individual units
• Managers or supervisors use instead of cell-phone
What to Expect:
Both units are assigned to a local or inter-site traffic channel. The call is
“private” as no other radio is included in the call. Units may be configured
to ring like a cell phone until answered.
Call timers may be lower than group calls, as private calls can consume
more system resources.
Once the call finishes, the terminal radios must return to listen the control
channel.

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5.5 Data Calls


A trunked mobile radio network provides one the most convenient and
efficient means of communication available.
A data message can convey the same amount of information as a voice
message, but data messages are transmitted by the system in a small
fraction of the time. A data message provides a clear statement of
requirements and is less likely to be misinterpreted than a spoken
message.
Data calls can be made using the:-
• control channel for status messages and short data messages.
• traffic channel for packet data and GPS position reporting.
5.5.1 Control Channel Data - Status Message
Status messages are sent only as a numeric value on the control channel
and are matched to a pre-programed look-up table label on the receiving
subscriber unit.
Status messages consist of a single request with acknowledgement for
individual calls. Groups Status messages are supported.
• 1 - 30 (MPT networks)
• 1 - 125 (DMR or dual mode networks)
Optionally status: 0 dispatch callback request
126 Status poll
127 dispatch callback cancel

Note: Group Status message are not Acknowledged

Note: Optional status: Implementation of these status will depend on the


manufacture and the DMR revision that was used in the product
development. This may result in different suppliers being incompatible
with other vendors. DMR revision 1.4 has no preassigned Status, while
revision 1.6 assigns 127 as the Status poll.

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Use Case
Application:
Sending clear and concise information regarding work tasks.
For example:-
• free
• next job
• complete

Making the Status Call:


Initiate a Status call on radio by selecting Menu, scroll to Send -> Status
and then select. Scroll through the available status messages. Scroll
through the send options (dispatcher, workgroup, preset or dialled).
What to Expect:
View the received status message by using the left selection key and select
view by scrolling through the options view, call, delete, delete all (if more
than one message).

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5.5.2 Control Channel Data - SDM


Short Data Messages (SDMs) are essentially free format text message.
Short data message is sent on the control channel to an individual or group
of subscriber units.
SDMs require more Signaling than a status message, however they are an
efficient way to send small payloads
• 7-bit ASCII 52 Characters
• 8-bit ASCII 46 Characters

Use Case
Application:
Sending clear and concise information regarding work tasks,
for example:-
• addresses,
• work orders,
• job numbers.

Making SDM Calls:


Initiate a SDM call on a radio by selecting Menu, scroll to Send -> Text
Message then select. Scroll to pre-set messages list and select a message
Select options and scroll through the send options (dispatcher, workgroup,
pre-set and dialled).
What to Expect:
View the received message by using the left selection key and select view
by scrolling through the options view, call, delete, delete all (if more than
one message).

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5.5.3 Traffic Channel Data - Packet Data Call

Unconfirmed Packet Data


Unconfirmed packet data is sent on a traffic channel, which requires
messaging to establish a data call in the same way that voice calls are set
up. The extra call setup and clear down signaling usually means that
packet data is more suited to large payloads, or “chatty” application
protocols.
Once a data call is established on a traffic channel, it is up to the
applications to manage when data packets are sent and to avoid collisions.
Up to 1500 octets can be sent in a single packet. Each packet is
transmitted in blocks over the air.
For unconfirmed data acknowledgement sent by end devices only indicate
it was transmitted and does not guarantee that it was received without
error. It is up to the application to manage retries etc.

Confirmed Packet Data


Confirmed packet data is similar to unconfirmed packet data, with the
addition of over-the-air acknowledgements for each block. The receiving
end (subscriber unit or node controller) can request that individual blocks
are resent if it is detected that they were received with too many errors
that could not be corrected.

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5.5.4 Control and/or Traffic Channel Data


Other messages can be configured to be sent over the control channel, or
as embedded messages during speech calls on traffic channels.

GPS position reporting


• National Marine Electronics Association (NMEA)
• Tait Data Format 1

Overview
The subscriber unit can be connected to a GPS antenna/receiver, which
will send GPS location data directly to the radio. GPS data can then be
packaged and sent via an RF link to a dispatcher Automatic vehicle
location (AVL) server, or shown on the subscriber unit display.

Application:
• Tracking individuals or teams location on allocated jobs.
• Asset tracking.
• Location evidence.
Making the GPS Call:
The subscriber unit will automatically send GPS data periodically or
when polled by the network. No action required of the user.
What to Expect:
View the GPS coordinates by using the menu key and select GPS info by
scrolling through the options.
May be allocated to a function key for easy access.

Trk: the GPS receiver is displaying up-to-date satellite information.


no cnx: the radio has lost serial communications with the GPS receiver.
no fix: the GPS receiver is having trouble connecting to satellites and the
radio is displaying stored information that may not be current.

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Additional information
Tait subscriber units support GPS (Global Positioning System) and AVL
(Automatic Vehicle Location) features. Portable radios have an in-built
GPS antenna. Mobile radios require a standalone GPS receiver to be
attached to a serial port. Features include displaying the radio’s location
reported by GPS, and GPS reporting (polling and unsolicited GPS
reporting)
For DMR networks, GPS data is sent using UDT (Unified Data Transport)
as described in the DMR standards. The following GPS formats are
supported in Tait subscriber units.

NMEA
Unsolicited GPS reports are sent using a UDT format of 0101(NMEA
location coded as per the IEC 61162-1/NMEA 0183 standard).

Tait Data Format 1


Unsolicited GPS reports are sent using a UDT format of 1001 (custom
coded -manufacturer specific). This format will also additional data such
as the last sent status value, and the state of programmable lines set to
User Status Input. This proprietary format is used for specific solutions,
and should only be selected if the GPS system and DMR network
supports it.

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5.6 Gateway Calls


Telephone Calls (PSTN/PABX)
Telephone connections to the DMR network require an appropriate
gateway device that enables radio users to make calls to telephone users.
The network can be configured to control the level of radio subscriber
access to the telephone system, and can also allow telephone subscribers
to make calls to radio users. Inbound telephone calls to the DMR network
can be individual or group.
Tait trunked networks can be an integral part of an organization’s
telecommunications network. Connections can be made directly to
telephone numbers via the PSTN or switched through the organizations
PBX. Radio users and telephone users with a PABX extension number
can then communicate directly.
Conventional Radio Network Calls
Connections to legacy conventional networks is possible through an
appropriate gateway device. This allows cross patching of differing
technologies as well as differing/same frequency bands.

Application Interface Standard (AIS)


The DMR AIS is a SIP based interconnection for DMR Tier II and Tier III
radio network voice and data communications. It is suitable for dispatch
consoles, voice recording, data applications such as text and LIP location,
and interconnection between DMR and other radio networks.
The dispatch interface capabilities include:
An all-digital IP interface (AIS), based on the widely-used SIP (Session
Initiation Protocol) but adds dispatch-specific enhancements or
'extensions' for setting up, modifying and ending dispatch calls. The voice
is transmitted using a Real-Time Transport Protocol (RTP) using either
the AMBE+2 vocoded voice or G.711 coded voice.
• AMBE+2, the speech encoding scheme used for DMR. When this
voice stream is used, no network gateways are required.
• G.711 voice stream, an ITU-T standard for audio pulse code
modulation. This will require the use of network gateways to
convert between AMBE+ and G.711 formats.

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5.7 Self Test Call


The Self Test Call feature:
1. Enables the radio terminal to make a voice call to itself,
2. The transmission that the radio terminal made is played back to itself
on completion of the over.
The self test call will clear down automatically once the audio has been
played back.
This allows the radio terminal making the self call to confirm its transmit
and received audio path is functional and network connectivity is
operational.

Operational Considerations
Any number of self test calls are allowed as long as there are free traffic
channels available at the site. The maximum call time, inactivity and call
answer timeouts will be the standard times as set in the unit profile for the
unit. Audio is limited to a maximum value set in the TT (Talk Timer)
parameter under the radio network settings. At present this has a
maximum limit of 60 Seconds.
Packet data calls will not be accepted as a self test call.
Self test calls can be made with emergency or priority level as well as
normal, and will also be queued in the normal way if there is no channel
resource for the call. The radio may send encrypted audio. All self calls
will be recorded by voice recorders if configured on the system.
In the unit profile there is a tick box to enable "self call tests", allowing
the radio to perform this.
Any number of self calls will be allowed as soon as there are free traffic
channels available at the site.
Self call test will only work for radio calls, they will not work for
disptacher console calls.

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Chapter 6
Call Features

6.1 Learning Outcomes


Upon completion of this chapter, you will be able to do the following:
• Compare DMR Call Handling Strategies
• Define DMR Call Timers
• Describe DMR Emergency and Priority Calls
• Describe DMR Network Optimization Strategies
• Describe Subscriber Unit Call Features

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6.2 DMR Call Handling Strategies


Call processing on the TaitNet System is fully automatic. Generally,
whenever a user wishes to place a call on the system, he, or she, simply
selects from a list, or enters the ID number of the unit or group to call, and
presses the push to talk button (PTT). Each unit or talkgroup may be
called in the same manner, no matter where it is located on the system.
The following features apply to individual and/or group calls as indicated,
on TaitNet DMR networks.
6.2.1 Transmission Trunked
Used only for group calls. When a call is made, the call proceeds to a
traffic channel. The calling radio proceeds to talk. On releasing the PTT
the radio sends a call clear down message and returns to the control
channel. The traffic channel is free to other users. If the system is busy
then a reply to the previous transmission may be queued until a channel is
free. This operation is similar to Logic Trunked Radio (LTR) system that
E.F. Johnson deployed.
6.2.2 Message Trunked
Used for group calls and individual calls. When a call is made, the call
proceeds to a traffic channel and a call time limit is started. The caller and
called parties proceed to talk for that period of time. The call is ended by
either:
• the network call time period expiring.
• the users pressing an end call (clear) button.
• the idle period time expires.
This operation is similar to MPT networks.
The trade off between transmission and message trunking is call setup
speed versus channel reuse efficiency

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6.2.3 Individual Call Setup Handling Strategy

Full Off Air Call Setup (FOACSU)


Trunking systems can set up calls in two ways. In some systems, it is
possible for the system to seek acknowledgement from the called party
before setting up a call. This can improve channel efficiency, and provides
a fast response to PSTN inward calling. The Radio Unit must also be
configured to support the operation.
A unit to unit call with FOACSU behaves like a cellular phone, the called
radio will ring and the user must PTT to answer.
Allowed: (Feature enabled) A subscriber to subscriber call with FOACSU
behaves like a cellular phone, the called radio will ring and the user must
PTT to answer. Acknowledgement will be sought from the called party
before a traffic channel will be allocated for the call. However, if the
system is not capable of seeking acknowledgement, setting FOACSU to
Allowed will have no effect.
Disallowed: (Feature disabled) A traffic channel will be allocated for the
call before the called radio has answered, therefore the call goes to
channel straight away and called party can hear the caller. This is
sometimes called off air call setup (OACSU)

6.2.4 Group Call Setup Handling Strategy

Late Entry to Group Calls (Repeat Go-To-Channel Messaging)


Late entry enables radios to join a group call after it has started. The
control channel sends GTC messages at intervals, inviting radios in the
group to join the call. Radios that are busy in another call, out of range, or
turned off will miss the group call. Late entry makes it possible for them
to join in later.
The late entry feature is enabled (or disabled) individually for each group
on the system. This means that within a system, some groups can be late
entry groups, while other groups are not.

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6.2.5 Queuing Strategy

Overview
This feature allows a call to an individual subscriber unit or group to
queue if there is no channel available. i.e If there are insufficient traffic
channels or network resources available to process a call, the network
puts the call in a queue. When a traffic channel or the called resource
becomes free, the network automatically connects the call.

Use Case
Application:
• Busy systems that have limited frequency allocation.
• Utility company that has adequate frequency allocations for everyday
business, however must retain efficient operation during a storm event
or similar.
• Systems that have very ‘spiky’ busy periods, e.g. airport ground crew,
where 80% - 90% of the time the network has very low usage that does
not require a large number of frequency resource. During aircraft
turnaround, large amounts of traffic needed over a relatively short
period of time.
What to Expect:
A radio user can place a call at any time and the network responds with a
queue message if no traffic channels are available.
Call queue message on screen while the radio waits for a traffic channel.
When one is available, the network allocates this to the queued request
and the call goes ahead.

For more detailed information regarding queuing on the DMR trunked


radio system, please refer to the Detailed Queuing Reference at the end of
this chapter.

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6.3 DMR Call Timers


The network times various stages of a call, and will clear down the call if
the radio is not using the channel efficiently or defined limits have been
reached.
6.3.1 Call Time Limits
The Call Time Limits area configures timers that limit the length of
different types of call. When a timer expires, the node clears down the
call.

Note: If a timer is set to 0 seconds, there is a limit to the length of the call
to 900 seconds [15 minutes] (however, an inactivity time-out can end the
call if no voice or data is sent).

Voice
Defines the call time limit for radio-to-radio normal and priority speech
calls.

Emergency voice
Defines the call time limit for radio-to-radio emergency speech calls.

Phone
Defines the call time limit for radio-to-telephone and telephone-to-radio
normal and priority calls.

Emergency phone
Defines the call time limit for emergency calls to or from a telephone.
6.3.2 Other Timers

Call Inactivity Time-outs


Call inactivity time-outs define how long there can be silence in a call
before the node clears the call down. Once a call is in progress, the node
restarts a timer, each time the carrier signal on the traffic channel ceases.
If a radio doesn’t transmit and the timer expires and the node clears the
call down. Different call inactivity timers are available for the different
call types as described above.
Benefit:- This allows for this 'inactive' channel to be quickly returned to
the available pool of resources so that it can be allocated to a new call.
Example:- A truck driver just puts the microphone down at the end of a
conversation instead of pressing the disconnect key. The channel will be
available for reuse sooner than the call limit timeout may provide.

Note: It is possible to disable inactivity time-outs for emergency calls, by


entering a value of 0.

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Call Answer Time-outs


Call answer time-outs define how long the node waits for a call to be
answered, before clearing the call down. Different call answer timers are
available for the different call types
• Voice
• Phone
• Emergency voice
• Emergency phone

GTC Repeat Interval


This parameter defines the frequency with which repeated GTC (Go to
Channel) messages are sent on the control channel during group calls. It
also defines the frequency with which repeat transmit inhibit messages are
sent on the traffic channel during broadcast calls.
The GTC Repeat Interval parameter is not used with groups that do not
have late entry enabled.The smaller the value chosen for this parameter,
the more difficult it is for radios to leave a group call and call anyone else.

Ignore Received Channel Grant Time


The subscriber unit configuration has timer that works in parallel with the
control channel GTC repeat interval called “Ignore Received Channel
Grant Time”.
This feature is used when receiving an unwanted group call. If this field is
set to 0 and the radio user clears down the call (for example, to make
another call), the radio will re-enter the group call as soon as the next
GTC command is received. That may happen before the radio user has
had a chance to dial the new call.
If a time is entered and that time is still counting down, repeated request
for GTC to re-enter the group call will be ignored.

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6.4 DMR Emergency and Priority Calls


In normal operation, calls in the queue are handled on a first in first out
basis. For urgent communications, priority and emergency calls are
supported. Priority and emergency calls are placed ahead of routine calls
in the queue.
6.4.1 Emergency Calls

Overview
An emergency call is typically mapped to the orange function key on the
top of the portable terminal and the orange key on the front of the mobile.
The call can be to a group or individual. A large number of variables can
be set for emergency mode using the programming application.
Emergency calls can be:
• Individual, conference, or broadcast calls.
• Data calls.
• Made to telephone subscribers as well as to dispatch users.

Emergency Pre-emption
A call made at the emergency level will pre-empt calls in progress to
make a traffic channel available if necessary.

Use Case
Application:
• Individual user with medical/physical emergency.
• Lone worker.
• Dispatch announcement.
• Natural/man-made incident or disaster.
What to Expect:
• Emergency calls shall take precedence over all other calls.
• Emergency call may be pre-emptive causing another call to be cleared
down if the resource requested for the emergency call is not available.

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6.4.2 Priority Group Override


If this feature is enabled, any group call which has been configured as a
priority override group on the subscriber unit, will have priority over
other group calls currently in progress (except for emergency calls). This
is labeled as “Subscribed with override” in the radio programming
software.
The network does this by transmitting a GTC (go to channel) command
on all active traffic channels, directing any units involved in other calls to
move to the channel where the priority override group call is taking place.
Example:-
1. Subscriber A has Talkgroup 6000 set as override, and is currently in a
non-priority group call with Talkgroup 6050 on traffic channel 20.
2. Subscriber B sets up a call to Talkgroup 6000, and is sent to traffic
channel 30.
3. A go to channel command is sent on traffic channel 20 for talkgroup
6000.
4. Subscriber A will obey this ‘override’ command and leave the current
group call (talkgroup 6050 on channel 20) and join Talkgroup 6000 on
channel 30.
A single override group can be nominated and may even be changed by
the subscriber unit.
The override group is always subscribed to and will receive activity if
scanning is on (unless the radio is transmitting when the override is sent).
Certain other calls such as PSTN, individual calls and emergency calls
will not be interrupted by activity on the override group.

Note: ‘Priority Group’ override is different from priority group calls (group
calls dialed with a *8n* prefix).

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Use Case
Application:
Useful when a subscribers primary talk group must always be received,
even if currently listening to a secondary talkgroup. e.g:-
• Due to workload, individuals are temporarily assigned to different
work roles, but must always be available to respond to calls from their
primary work role as their main priority.
What to Expect:
Subscriber units on non-override calls, as well as units not currently in
calls, will join the override group call.
The override group is indicated with an ^ icon, in the subscriber unit
workgroup setup screen.

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6.4.3 Priority Calls

Overview
Four priority levels exist (Priority 3 is the highest priority level). A
priority call is moved to the front of a queue of subscriber unit calls that
are waiting to be processed. Higher priority calls are moved ahead of calls
that are a lower priority, the four levels are:
• No priority
• Priority 1
• Priority 2
• Priority 3

Use Case
Application:
Provide priority to special users for the rapid access of traffic channels.
This call is made using the menu or can be quickly made by dialing
*8n*(group identity)
What to Expect:
A call with priority access is processed before a call with normal priority
access.

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6.4.4 Minimum Access Level


Minimum access level allows a network administrator to set the default
minimum call priority level for a calling or called party (this can be a unit
or a group). This applies for calls which require a payload channel. (Voice
and packet data calls but not SDM or Status calls)
There are five levels of priorities:
0 = Normal (low)
1 = Medium
2 = High
3 = Highest
4 = Emergency
By default, the calling or called parties (individual or group) would have
the minimum priority level set to normal.
If a subscriber requests an access level which is lower than the minimum
access level for that calling party or called party, the access level would be
adjusted to the highest among the three.
1. Subscriber 1 has minimum access level of medium (priority=1).
Group 6000 has minimum access level of emergency (priority=4).
2. Subscriber 1 dials a normal priority call string to call Group 6000
(priority=0).
i.e. User dials > 6000 Send.
3. Node escalates call priority to match Group 6000 Minimum Access
Priority level (priority=4).
4. The final call priority for this call would be an emergency call.

Unit 0001 Group 6000 Group 6000 Group 6000


Medium Priority Dialled as Low Priority Minimum priority = Emergency Emergency
TP9300

6000
Send

Group Min. Access


6000 4 (Emergency)
6001 2 (High)

Figure 30: Example - Minimum Access Level for Emergency Talkgroup

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6.4.5 Configuration
In the node web UI, the unit and group profile now includes two new
parameters under a new heading of 'Default Minimum Priority':
• voice calls
• ip / packet data
There is a drop down menu for each parameter allowing a priority level
of: normal (low), medium, high, highest, or emergency to be set. This
corresponds to how priority levels are now defined in the DMR trunking
specification.
By default all unit and group profiles will set these parameters as normal,
i.e. the lowest value.
6.4.6 Operational Considerations

High Priority Call Request


If a subscriber requests an access level which is higher than the minimum
access level for that calling party or called party, the access level would
NOT be adjusted. The node does not adjust down the call priority level.

Calling Party Access Validation


When a call access level is adjusted, the node would not validate the
access level setting of the calling party.
This means, when a radio is restricted to make only normal call, if its
profile or the called party's profile has minimum access level set to
emergency, then the call would be made as an emergency call. Call record
display only the final call priority level.
This allows for a unit that is not allowed to make emergency priority calls,
for example, but is still able to call a group which has a minimum call
priority that is emergency.

Call Records
Call records indicate the final call priority, not the initial call priority.
There is no indication in the call record that the priority has changed. The
node log records if the call priority has been adjusted.

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6.5 DMR Network Optimization Strategies


6.5.1 User Profiles
The nodes contain validation records for each subscriber unit. If a number
has not been validated, the node bars it from using the network. A
validation record consists of the subscriber unit number, and the call types
it is allowed to make.
• Individual Radio - Intersite / Inter Fleet / Inter Prefix
• Group- Intersite / Inter Fleet / Inter Prefix / Broadcast
• Telephone Calls - PSTN / PABX incoming/outgoing
• Data - IP data / Status / SDM / Group status / Group short data /
self diversion requests
6.5.2 Individual Service Areas

Allowed/Not Allowed
The Network Administrator is able to control a subscriber units service
area. The service area is part of a subscriber units validation record. It is
that part of the network’s coverage area a unit is permitted to operate
from.
6.5.3 Group Calls - Fixed Service Area
The Network Administrator is able to control the sites that are to be
involved in a group call. The following table lists the options that are
available to allow the Network Administrator to create the required or
most efficient Group Call policies.

Table 1: Group Call Service Area types


Option Description

Not Allowed Group calls cannot occur at the site. A radio cannot make a group call and group
members cannot receive a group call.

Allowed When a group call is made, the site uses a channel for the call. If the site is busy, the
call goes ahead and the site joins the call when a channel comes free.

Essential The group call must include the site. If the site is busy for more than the maximum
queuing time, the call fails.

Local Only The group call only uses this site. The calling party must be at the site. Only radios
registered at the site can participate. The node is not involved in call setup. If the
group call originates from another site, this site is not included.

Originate Only The group call can only be originated from the site. Radios can call the group from the
site, but only group members at the originating site and at ‘Allowed’ and ‘Essential’
sites can participate.

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The key shown below is used for the following series of illustrations.

Key

Site part of call service area

Subscriber originating call

Subscriber(s) receiving call

Call unsuccesful

Site busy

Allowed
When a group call is made, the site uses a channel for the call. If the far
site is busy, the call is queued and is then setup when a channel becomes
free at the far site or the maximum queuing time expires, The far site is
included once a channel is free.

Site 1 Site 2

Allowed Allowed

Dedicated Data Link


Site 3 TCP / IP

Dedicated Data Link


TCP / IP

Control / Switching
Node

Allowed

Figure 31: Illustration of “Allowed” fixed service area group call type

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Essential
The group call must include the site. If the site is busy for more than the
maximum queuing time, the call fails.

Site 1 Site 2

Allowed Allowed

Dedicated Data Link


Site 3 Busy TCP / IP

Dedicated Data Link


TCP / IP

Control / Switching
Node

Essential

Site 1 Site 2

Allowed Allowed

Dedicated Data Link


Site 3 Busy TCP / IP

Dedicated Data Link


TCP / IP

Control / Switching
Node

Essential

Figure 32: Illustration of “Essential” fixed service area group call type

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Local Only
The group call only uses this site. The calling party must be at the site.
Only radios registered at the site can participate. If the group call
originates from another site, this site is not included.

Site 1 Site 2

Local Only Local Only

Dedicated Data Link


Site 3 TCP / IP

Dedicated Data Link


TCP / IP

Control / Switching
Node

Local Only

Site 1 Site 2

Local Only Local Only

Dedicated Data Link


Site 3 TCP / IP

Dedicated Data Link


TCP / IP

Control / Switching
Node

Local Only

Figure 33: Illustration of “Local Only” fixed service area group call type

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Originate Only
The Group calls can be made from the originate only site. If a group call
made at another site, for example an allowed site, the originate only site
will not be involved.

Site 1 Site 2

Allowed Originate only

Dedicated Data Link


Site 3 TCP / IP

Dedicated Data Link


TCP / IP

Control / Switching
Node

Allowed

Site 1 Site 2

Allowed Originate only

Dedicated Data Link


Site 3 TCP / IP

Dedicated Data Link


TCP / IP

Control / Switching
Node

Allowed

Figure 34: Illustration of “Originate Only” fixed service area group call
type

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6.5.4 Groups Call - Controlled Service Area


Once a group calls fixed service area has been defined, the network
administrator can further improve the efficiency of the group call by
enabling a group call feature that only uses sites that currently have
talkgroup members located, and who are actually ‘signed-on’ to receive
the talkgroup traffic.

Group Affiliation
Group affiliation allows a subscriber unit to update their group affiliation
over the air. It is implemented by having a unit ‘sign on’ to a group to
indicate that it should now be included as a member of the group’s tracked
units.
When a call request is made to the talkgroup, the call will be set up at only
those sites where at least one signed-on subscriber unit member of the
group’s tracked units had its last registration recorded.
This ensures that traffic channels are only established on sites where
radios are going to participate in that call and so improves network
channel efficiency.

Tait radio controlled group affiliation (RCGA)


This feature is supported by Tait subscriber units. This allows the unit to
join or leave groups by sending a short data message. It is also possible to
manually regroup subscriber units from the node.

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Group Affiliation Illustration


Group calls will only setup on sites that are within the group service area
and on which at least one radio has affiliated (subscribed) with the
talkgroup.

Site 1 Site 2

Allowed Allowed

Dedicated Data Link


Site 3 TCP / IP

Dedicated Data Link


TCP / IP

Control / Switching
Node

Allowed

Site 1 Site 2

Allowed Allowed

Dedicated Data Link


Site 3 TCP / IP
Dedicated Data Link
TCP / IP

Control / Switching
Node

Allowed

Figure 35: Illustration of “RCGA” controlled service area group call.

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6.5.5 Mobility

Diversions
Self Divert:
If allowed a Radio can Divert all of their incoming calls to a radio,
dispatcher or a telephone number.
Divert on Busy:
If this feature is enabled a call made to a radio that is currently involved in
a call will be diverted to a defined number, instead of being queued or
failing with the reason “engaged”. The diversion does not have to be to
another radio, it can be to any valid unit in the system, for example, a line
dispatcher or a phone.
Divert on Not Home:
If this feature is enabled, a call made to a radio that is switched off or out
of contact will be diverted to a defined number. The diversion does not
have to be to another radio, it can be to any valid unit in the system, for
example, a line dispatcher or a phone.

6.5.6 Asset Management

Access Level
There are four access levels a radio may have on a TaitNet network. The
following table lists the options that are available to allow the Network
Administrator to control the radio unit access to the network.

Option Description

Barred Radio is NOT able to make any calls.

Normal Radio is able to make all normal level calls.

Priority Radio is able to make priority calls (priority requires the *8* prefix to be dialled
first). Priority calls jump to front of call queue.

Emergency Radio is able to make emergency calls (emergency requires the *9* prefix to be
dialled first). Emergency calls jump to front of queue and pre-empt a traffic
channel if required.

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6.5.7 Stun and Revive


The TaitNet Network Administrator can stun a Tait radio. This is useful if
the radio has been stolen. A radio that is stunned cannot make or receive
calls. The keyboard is locked. Its display indicates that it is stunned. The
radio cannot be used, even in conventional (non-trunked) mode, but it will
still register so that the NMT can tell you roughly where it is. You can
recover a stunned radio, returning it to normal functioning provided that it
is turned on and registered on the network.
A stunned radio will still receive (and answer) a GPS poll.
6.5.8 Authentication key
Each subscriber unit has a factory-programmed authentication key that is
checked by the node to ensure that a radio registering on the network is
genuine.
The serial number of a subscriber unit is entered in the Authentication key
field of the node subscriber database, and an RC4 Cipher encryption
algorithm is used to generate a 56-bit authentication key.
The node uses this 56-bit key to process a secure validation check of the
subscriber units hardware on the network.
An authentication check is carried out as follows. The node polls a radio
via the control channel of the site where the unit is currently registered,
asking it to provide its authentication key. When a response is received,
the node checks it against the key that is entered in the.
Authentication checks can be configured to be performed on registration
and/or on call requests. The network can also be configured to reject
requests that fail authentication.

Note: The authentication process is based on RC4 [Rivest Cipher 4]


encryption (a popular encryption algorithm commonly used to protect
Internet and wireless traffic). Radios have a 56-bit authentication key.

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6.6 Subscriber Unit Call Features


Various call features are provided by the user’s radio unit and are typically
part of the radios unique configuration.

Pre-set Calls
Preset calls are simply numbers saved along with a name or alpha numeric
tag that allows quick and easy access to the most common numbers
dialled. The preset calls programmed for your radio may be to other
radios, to PABX extensions or to PSTN numbers. To make a preset call
from your radio, you may be able to either:
• use a programmed function key,
• use the selector switch,
• use the main menu, or
• dial the preset call using the keypad
Preset calls can also be used to activate or deactivate a function,
temporarily subscribe your radio to a group, change to conventional
channels and change to a different trunking network. Up to 100 pre-set
numbers can be entered.

Caller ID
Radio displays caller number (and name if recorded in preset list).

Talker ID
Radio displays the ID of the current talker in a group call

Call Back feature


After a call has ended, there is the ability for the radio to allow quick and
easy call back.
• Terminated calls: caller details are stored - pressing PTT re-establishes
the call.
• Missed calls: caller details are displayed with an audio warning - call
back by pressing PTT.

Call Divert
• If allowed, Call Divert. (Call Forwarding) can be performed from the
radio unit.This can either be a self or third party divert. Additionally
the radio can enable a Do Not Disturb or Queue incoming calls feature.

Conventional channels
A number of preset conventional channels (currently 1500) are available
on the subscriber units. It is possible to change to conventional channels
using the channel selector. Radios with an alphanumeric keypad may be
able to dial the channel number. In conventional mode, you communicate
directly with other radios or via a repeater rather than through the trunking
network.

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6.7 Detailed Queuing Reference


6.7.1 Queuing Overview
A new call request at a site will be queued if one of the following occurs:
1. There are no free channels at the site to put the call on. Will be queued
for busy party.
2. The called party is already in a call at that site (this can include parties
that are already queued). Will be queued for busy channel.
3. There are already calls queued at the site. Will be queued for busy
party.
One exception to this rule is amalgamation which will be explained later.
The queue is ordered by:
1. Emergency calls first
2. Priority calls next
3. Normal level calls last
If calls are at the same level they are based on oldest first.
The type of call does not matter (i.e. data vs voice)
The node will not allow two calls to the same party to be queued.
Whenever a channel becomes free at a site it will assign the free channel
to the highest order call in the queue. In this case it will always assign an
emergency call first if there is one in the queue.
The node will only allow calls to be queued for a configurable amount of
time. If a call times-out in the queue it will send "system busy" or
"engaged" depending on how it was queued.
There is also a finite limit on how many calls are queued at each site.

6.7.2 Qued Call Scenarios

Engaged
Individual normal and priority calls that are made to a busy party will not
be queued for busy party. They will be rejected straight away for
"engaged".

Emergency Pre-emption for busy Individual party


An Individual Emergency call made to a busy party will be queued. The
node will attempt to pre-empt the call that the busy party is in.

Amalgamation
For a normal or priority call to a group that is already on channel, the node
will amalgamate the group in to the existing call. This means it will send
a channel grant to the new call straight away, and not queue the call at all.

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Emergency Pre-emption for busy group


If an emergency call is made to a group that is already on channel, the
node will not amalgamate the group call. In this scenario it is important
for all the users to know that the group call has now become an
emergency call. This means the node has to re-establish the group call. It
will:
1. Pre-empt the existing group call. All radios will be cleared back to the
control channel
2. Once all channels are cleared, it will send a new channel grants for the
group indicating it is an emergency call.

Emergency Pre-emption for busy channel


If a site is busy, the node will attempt to clear down calls to free up a
channel. It a call does not clear down (e.g. because someone is
transmitting in the call) then the node will then attempt to clear a 2nd
channel. The node can potentially clear down all channels at a site.

Note: The Node will never pre-empt an existing emergency call.

Emergency pre-emption of the queue


If an emergency call is made to a party or group that is in the queue then
the existing call will be cleared from the queue (unless it was an
emergency call). The node does not allow busy party queueing so you
should only ever have one call in the queue for a particular unit.

Site queues
Each site has an independent queue. This means a call could potentially
be queued at the calling site first, and then the called sites. It will not
attempt to get channel resource at the called sites until the calling site has
resource.

Cross over queued calls


This is an interesting problem that can happen with queuing. If two
intersite calls start at the same time, on different sites, it is possible for
them to both reserve a channel at their initiating sites, but then not have a
channel available at the called site, because of the other call reserving a
channel. One of the calls would only setup once the other call gives up
and clears down.
However if one of these calls is an emergency call, then it would setup
because pre-emption would clear the other call.

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6.7.3 Channel Management for Queuing

Channel Allocation
The called party in a call is not notified of the call until there are channels
available at all sites in the call. These means that a call could be queued
for busy channel, and then fail because the called party is out of service.

Group call Setup times


Whenever a group sets up it will attempt to find channels at every site in
the service area. However, if any of the called sites are queued because
there are no channels available, the node will wait for the sites to become
ready before it sets up the call.
The group can be configured so that the node will only wait a certain
amount of time for this to happen - this is called the group setup time. At
the end of the group setup time it will setup the call on all sites that are
ready. As other channel become ready, they will be brought in to the call
as well.
The group call setup timeout only comes in to effect once we start to find
channels. An emergency group call may be queued first for busy party.
This means the original group must totally clear down first.

Essential sites
The node can be configured so that some sites are essential to the group
call. In this case, at the end of the group setup time, it will fail the call if
any of the essential sites are not ready for the call.

Late Entry
When a radio is in another call, it may miss the channel grant to the group.
In this case once the radio has cleared from its own call it will be pulled in
to the group due to late entry. The node will send late entries for groups
every 10 seconds by default.

Stuck mute timeout


When a call is clearing down the node will wait for the channel to become
totally idle. This includes radios that might be transmitting at the site.
There is currently no way to tell a radio to stop transmitting.
If a cleared channel takes too long to clear (>30s) then the node will mark
the channel as stuck mute, and will consider the call completed anyway.
Depending on configuration it may also mark the channel as jammed and
not use it for further calls.

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Homegroup override
If the homegroup override feature is enabled in the node for a group, it
means that when the group is set up it will attempt to pull in any radios at
the site that may already be in different calls.
It does by transmitting messages on all active channels at the site. These
messages are either:
• Sending a channel grant CSBK on the payload channel if no one is
talking in the call
• Overriding the embedded signalling to send a Tait proprietary message
on channels where there is someone talking.
Radios will only swap to the new group if this group has been configured
for it.

Note: the XPA uses the term 'priority group' instead of homegroup
override.

Emergency group override


This is similar to "homegroup override" but for emergency calls instead.

Clearing a call while someone is talking


If someone is talking in a call, and the node wishes to clear down the call,
it can override the audio to send CLEAR messages that will clear out all
the receiving radios.
However, there is currently no way to clear the talking party,as Tait
terminals are half duplex.

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Chapter 7
Numbering

7.1 Learning Outcomes


Upon completion of this chapter, you will be able to do the following:
• Describe Tait numbering compatibility
• Define DMR Numbering
• Describe DMPT 1327 Addressing
• Demonstrate dialling calls in MPT1327

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7.2 Numbering Compatibility


Existing or new Tait customers that are using a legacy analog MPT
network must have a simple migration path to a DMR network. For this
reason Tait have adopted a numbering format where DMR network
addresses that are used over-the-air by the subscriber units and network
infrastructure, are converted to a programmed numbering scheme (MPT
1327, MPT 1343 or Nokia ANN) on the subscriber unit, and network
management tools.
7.2.1 Numbering schemes available
Examples of interface numbering schemes
• MPT1343 - Mandatory on UK Band III, and used on most other
networks
• ANN - Nokia Actionet infrastructure: Actionet Numbering
• Tait Utilities (Customized MPT 1327) - for utilities customers who
require large numbers of talkgroups

7.3 DMR Numbering


Many DMR data bursts sent on air require a source and/or destination
address to identify subscriber units or talkgroups on the system.
The DMR standards define both individual and group addresses as a
hexadecimal number between 00000116 and FFFCDF16, with additional
numbers reserved for special use.
Normally the DMR address does not have any special meaning and does
not include the fleet identifier. However, as discussed previously on a Tait
network, the DMR address must be compatible with legacy numbering
schemes, the following format is derived when the DMR number is
converted to 24-bit binary:

7-bit prefix
(between decimal 0
and 127)

0000 ppppppp iiiiiiiiiiiii

4-bit lead 13-bit ident


Always 0000 (between decimal 1
and 8100)

Figure 36: DMR numbering

This means that there are over 16 million numbers available for use in the
DMR environment. For convenience Tait have adopted a numbering
system which allows the use of 128 prefixes each with around 8100 idents
(identity numbers). This is identical to the MPT1327 plan.
Every radio unit and talkgroup are assigned their own unique number on
the DMR system.

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7.4 MPT 1327 Addressing


The Tait DMR system uses MPT1327 numbers that map to corresponding
DMR numbers to address all radios and groups in the system, and allows
easy transition for MPT customers.
In MPT 1327 the 20-bit address is divided into two fields and have the
format:- ppp iiii where:-

Item Description Permitted range

ppp prefix (often referred to as PFIX) 000 - 127

iiii identity (often referred to as IDENT) 0001 - 8100

Depending on the numbering scheme configuration, the address can refer


to a group or an individual address.
This gives a total of 128 x 8100 = 1,036,800 unit or group idents in a
system.
Every subscriber unit has a single Individual Address of the form
Prefix / Ident.
A subscriber unit can also typically be configured to be part of one or a
number of groups by being programmed to hold Group Addresses of the
same fromat Prefix / Ident.

Prefix 127 Prefix Independent

idents Independent Ident numbers 8101 - 8191


Ident 0001 are reserved for special purposes
Prefix 002 Ident 8101 - PSTNGI PSTN
Prefix 001
Users Ident 8102 - PABXI PABX
Prefix 000Ident 0001 Ident 8185 - REGI Registration
Users Ident 8100
Ident 0001 Ident 8188 - SDMI Short Data Msg
idents
Ident 0001 Ident 8190 - TSCI Trunk Sys CTRL
Ident 8191 - ALLI All Ident Call
Prefix 8100
Prefix 8100
Ident 8100

Figure 37: 1327 Numbering Structure

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Units and Groups


To manage the subscriber units in a system, a range of groups can be set
up within each prefix. The subscriber units can be assigned to groups as
required.
Currently on Tait DMR networks the MPT 1327 Ident space is split into
two areas;
• Individual idents in the range of 0001 - 5999
• Group idents in the range of 6000 - 8100
• Idents above 8100 are prefix independent and reserved for signalling
purposes

Prefix 127 Prefix Independent

Users Independent Ident numbers 8101 - 8191


Ident 0001 are reserved for special ourposes
Ident 5999
Prefix 002 Ident 8101 - PSTNGI PSTN
Prefix 001 Group
Users Ident 6000 Ident 8102 - PABXI PABX
Prefix 000Ident 0001
Users Ident 8100 Ident 8185 - REGI Registration
IdentIdent
0001 5999
Users Ident 8188 - SDMI Short Data Msg
IdentIdent
0001 5999
Group
Ident 6000 Ident 8190 - TSCI Trunk Sys CTRL
Ident 5999
Group
IdentIdent
6000 8100 Ident 8191 - ALLI All Ident Call
Group
IdentIdent
6000 8100
Ident 8100

Figure 38: MPT132 Individual and Group Ident ranges

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7.5 Dialling Calls in MPT1327


If the subscriber has alphanumeric keys, it can make dialled calls from the
key pad using the dialling strings outlined below. The following tables
lists in detail the available strings, however a well configured subscriber
unit will rarely need the user to know or remember many of these strings.
When configuring a subscriber unit it will be necessary on occasion enter
full dialling strings to achieve desired functionality.

Note: The numbers dialled and dialling features available depend on the
way the radio is programmed and the way the network operates.

7.5.1 MPT1327 Dialling String Summary


The following tables summarize the way calls are dialled to other
subscribers or groups of subscribers and gives an example of each type of
call. Also explained is how you can access special MPT trunking
functions using the * and # keys. In the following examples, the final #
may be replaced by a short press of the PTT key.

Call/Clear Call

Dialling code Function


# (or PTT) End Dialled string or accept an incoming FOACSU call.
Lift Mic. (off hook) on mobile to answer. (programming option)
“Clear” or *# End Call.
“Answer”, # or Accept incoming FOACSU call.
PTT
“Decline” or *# Decline in coming FOACSU call.

Units and Presets

Dialling code Function Example


nnnn# Unit to Unit call n = 0001 – 5999 0021#
2155#
gggg# Group call g = 6000 – 8099 6303#
999# or 112# Emergency operator (does not require Emergency level access)

Emergency and Priority prefixes

Dialling code Function Example


*9*xxx...# Emergency call to Radio, Group, PSTN etc. *9*0021#
*9*2155#
*8*xxx...# Priority call to Radio, Group, PSTN etc. *8*6303#

Group Broadcast

Dialling code Function Example


*11* gggg# Broadcast call to group gggg *11* 6303#

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Status, SDM and Data

Dialling code Function Example


*0s(ss)*nnnn# Status call s(ss) = Status 1 to 126 to radio nnnn *015*2345#
*0s(ss)# Status to prime dispatcher (programmed) *015#

Phone and PABX

Dialling code Function Example


Xnnnn# PABX nnnn: 1000 – 8999 38766#
(extended addressing 4 to 8 digits) X= 3,4,5 or 6
0nnnnnnn# PSTN nnnnnnn: 7 to 31 Digits 01234567#

Dispatcher

Dialling code Function Example


*0# Request base dispatcher to call you back
#0# Cancel request
*0*nnnn# Request another dispatcher to call you back *0*1801#
#0*nnnn# Cancel request #0*1801#

Call Divert (Call forwarding)

Dialling code Function Example


*41*xxxx# Divert own call to radio (Call forwarding) *41*2345#
#41# xxxx = any radio, group PSTN or PABX
Cancel divert
*44*xxxx*yyyy# Divert 3rd party calls xxxx to yyyy *44*2345*2100#
#44*xxxx# Cancel divert for unit xxxx #44*2345#
#45# Cancel incoming call diversions

Queuing

Dialling code Function


*48# Queue incoming calls
#48# Cancel queue
*49# Do not disturb
#49# Cancel do not disturb

Network

Dialling code Function

*700# Display your current MPT number and network


101# to110# Select conventional channel 1 to 10

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Technician

Dialling code Function Example


*50*tcccc# †Select Time slot t(1 or 2) Channel cccc: *50*10001#
(Site Select diagnostic function enabled in pro-
gramming)
#50# Resume normal channel hunt
1pppiiii# Technician Calling (MPT1327) ppp = Prefix iiii = 10011023#
Ident
MPT1343: 201 2495 53 = MPT1327: 0011023

†Certain Dialling strings need to be enabled, requires the radio and the
system to allow it.

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