Bridge-IT
IP Codec User Manual
Software Version: 2.14.88
Manual Version: v.2.2_20151125
November, 2015
2
Bridge-IT Manual
Table of Contents
Part I How to Use the Documentation
5
Part II Warnings and Safety Information
6
Part III Glossary of Terms
7
Part IV Introduction to the Codec
9
Part V Front Panel Controls
11
Part VI Rear Panel Connections
13
Part VII Navigating Codec Menus
15
Part VIII Adjusting Input/Meter Levels
20
Part IX Configuring AES3 Audio
24
Part X Headphone/Output Monitoring
26
Part XI Language Selection
27
Part XII About Program Dialing
28
Part XIII Getting Connected Quickly
30
1 10 ...................................................................................................................................
Quick Steps to Connect Bridge-IT
30
2 Monitoring
...................................................................................................................................
IP Connections
33
3 Load
...................................................................................................................................
and Dial Custom Programs
35
4 Disconnecting
...................................................................................................................................
a Connection
35
5 Redialing
...................................................................................................................................
a Connection
36
6 Configuring
...................................................................................................................................
Auto Reconnect
36
7 Speed
...................................................................................................................................
Dialing Connections
37
8 Dial/Disconnect
...................................................................................................................................
Multiple Connections
37
9 Creating
...................................................................................................................................
a Multicast Server Program
38
10 Creating
...................................................................................................................................
a Multicast Client Program
41
11 Dialing
...................................................................................................................................
SIP Peer-to-Peer
43
12 Dialing
...................................................................................................................................
SIP Addresses
44
13 Deleting
...................................................................................................................................
Programs
45
14 Selecting
...................................................................................................................................
Algorithm Profiles
45
15 SDHC
...................................................................................................................................
Card Backup
48
16 SDHC
...................................................................................................................................
Card File Playback
48
17 Lock
...................................................................................................................................
or Unlock a Program in the Codec
49
18 Locking
...................................................................................................................................
the Front Panel
50
© Tieline Pty. Ltd. 2015
Contents
Part XIV Connecting to the ToolBox
Web-GUI
3
51
1 Opening
...................................................................................................................................
the Java or HTML5 Web-GUI & Login
52
2 Changing
...................................................................................................................................
the Default Password
55
3 Installing
...................................................................................................................................
USB Drivers
56
4 Launching
...................................................................................................................................
the GUI over USB
57
Part XV Java Toolbox Web-GUI
Introduction
57
Part XVI Java Toolbox Web-GUI Codec
Configuration
65
1 Configuring
...................................................................................................................................
IP Settings
65
2 Configuring
...................................................................................................................................
Input/Output Settings
69
3 Configuring
...................................................................................................................................
Mono or Stereo Peer-to-Peer Programs
73
4 Configuring
...................................................................................................................................
Multi-Unicast Dialing Programs
79
5 Configuring
...................................................................................................................................
Multicast Server Programs
84
6 Configuring
...................................................................................................................................
Multicast Client Programs
88
7 Configuring
...................................................................................................................................
SIP Settings
91
8 Configuring
...................................................................................................................................
SIP Programs
93
9 Dial
...................................................................................................................................
and Disconnect a Program
97
10 Dial
...................................................................................................................................
and Disconnect Multi-unicast Connections
98
11 Lock
...................................................................................................................................
or Unlock Programs
100
12 View/Edit/Delete
...................................................................................................................................
Programs
100
13 Reset
...................................................................................................................................
Factory Default Settings
103
14 Backup
...................................................................................................................................
and Restore Functions
104
15 Web-GUI
...................................................................................................................................
Software License Installation
106
16 Download
...................................................................................................................................
Logs
107
17 RS232
...................................................................................................................................
Data Configuration
108
18 Creating
...................................................................................................................................
Rules
109
19 Upgrading
...................................................................................................................................
Codec Firmware
113
Part XVII HTML5 Toolbox Web-GUI
Introduction
115
Part XVIII HTML5 Toolbox Web-GUI
Configuration
126
1 Using
...................................................................................................................................
the HTML5 Toolbox Quick Connect Web-GUI
126
2 Configuring
...................................................................................................................................
IP Settings
131
3 Configuring
...................................................................................................................................
Input/Output Settings
135
4 Configure
...................................................................................................................................
SIP Settings
137
5 Load,
...................................................................................................................................
Unload and Dial a Program
140
6 Lock
...................................................................................................................................
or Unlock Programs
143
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...................................................................................................................................
144
7 Reset
Factory Default Settings
8 Backup
...................................................................................................................................
and Restore Functions
145
9 HTML5
...................................................................................................................................
Software License Installation
146
10 Download
...................................................................................................................................
Logs
147
11 RS232
...................................................................................................................................
Data Configuration
148
12 Creating
...................................................................................................................................
Rules
150
13 Upgrading
...................................................................................................................................
Codec Firmware
154
Part XIX Front Panel Configuration Tasks
155
1 Configuring
...................................................................................................................................
IP via the Front Panel
155
2 Selecting
...................................................................................................................................
an Algorithm
158
3 Configuring
...................................................................................................................................
the Jitter Buffer
164
4 Configuring
...................................................................................................................................
Forward Error Correction
167
5 Configuring
...................................................................................................................................
Encode/Decode Direction
169
6 Enabling
...................................................................................................................................
Relays & RS232 Data
169
7 Configuring
...................................................................................................................................
TCP/UDP Ports
172
8 Configuring
...................................................................................................................................
QoS for Broadcasts
173
9 Configuring
...................................................................................................................................
Data Packet Time-to-Live
174
10 Reset
...................................................................................................................................
and Restore Factory Default Settings
175
11 Installing
...................................................................................................................................
Software Licences
176
Part XX Reference
178
1 Installing
...................................................................................................................................
the Codec at the Studio
178
2 Understanding
...................................................................................................................................
IP Networks
184
3 Tips
...................................................................................................................................
for Creating Reliable IP Connections
186
4 Testing
...................................................................................................................................
IP Network Connections
187
5 Software
...................................................................................................................................
Licences
189
6 Compliances
...................................................................................................................................
and Certifications
197
7 Trademarks
...................................................................................................................................
and Credit Notices
198
Part XXI Specifications
199
Index
200
© Tieline Pty. Ltd. 2015
Bridge-IT Manual
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5
How to Use the Documentation
Overview of this User Manual
Use this manual to learn how to:
· Connect the codec to an IP network and configure peer-to-peer, multicast or multi-unicast
connections.
· Configure the codec over a LAN or USB cable.
· Adjust audio and other settings within the codec.
· Configure automatic SDHC card backup.
Please read Getting Connected Quickly for an overview of how to configure the codec using
'programs' to store connection settings.
Manual Conventions
Warnings: Instructions that, if ignored, could result in death or serious personal injury
caused by incorrect operation of the equipment. These must be observed for safe operation.
Cautions: Instructions warning against potential hazards, or to detail practices that must be
observed for safe operation and to prevent damage to equipment or personnel.
Important Note: Information you should know to connect and operate your codec
successfully.
Typographic Conventions
· Codec software elements are in Arial bold, e.g. Contacts
· Codec hardware elements are in bold Capitals, e.g. KEYPAD
Help Button
Press the
(information/help) button on the codec when navigating codec menus to display a
help dialog on the LCD screen suggesting actions which can be performed from the current menu
item.
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Warnings and Safety Information
THUNDERSTORM AND LIGHTNING WARNING:
DO NOT USE Tieline codecs during thunderstorms and lightning. You may suffer an injury
using a Tieline codec, or any device connected to a LAN connection during a thunderstorm.
This can lead to personal injury and in extreme cases may be fatal. Protective devices can
be fitted to lines, however, due to the extremely high voltages and energy levels involved in
lightning strikes, these devices may not offer protection to users, the Tieline codec and
equipment connected to the codec.
Secondary strikes can occur. These secondary strikes are induced by lightning strikes and
also produce dangerously high currents and energy levels. You only need to be near an
object struck by lightning to lead to personal injury or damage to equipment. e.g. if located
near a lighting tower at a sports facility, water features and drains on golf courses you may
be affected by these secondary strikes.
Damage to personnel and Tieline codecs may occur during thunderstorm, even if the codec
is turned off but is connected to the system or the power.
ANY DAMAGE TO A TIELINE PRODUCT CAUSED BY LIGHTNING or an ELECTRICAL
STORM WILL VOID THE WARRANTY. Use of this product is subject to Tieline's
SOFTWARE LICENSE
and WARRANTY conditions, which should be viewed at
www.tieline.com/support before using this product.
Disclaimer
Whilst every effort has been made to ensure the accuracy of this manual we are not responsible for
any errors or omissions within it. The product specifications and descriptions within this manual will
be subject to improvements and modifications over time without notice, as changes to software and
hardware are implemented.
© Tieline Pty. Ltd. 2015
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7
Glossary of Terms
AES/EBU
Digital audio standard used to carry digital audio signals between devices.
AES3
Official term for the audio standard referred to often as AES/EBU.
DNS
The Domain Name System (DNS) is used to assign domain names to IP
addresses over the World-Wide Web.
The Differentiated Services Code Point is a field in an IP packet header for
prioritising data when traversing IP networks
Method of switching to an alternative audio stream if the primary connection is
lost.
Acronym for Graphic User Interface
DSCP
Fail over
GUI
ISP
IP
LAN
Latency
Multicast
Internet Service Providers (ISPs) are companies that offer customers access to
the internet
Internet Protocol; used for sending data across packet-switched networks.
Local Area Network; a group of computers and associated devices sharing a
common communications link
Delay associated with IP networks and caused by algorithmic, transport and
buffering delays.
Efficient one to many streaming of IP audio using multicast IP addressing.
Multi-unicast
A multi-unicast program (also known as multiple unicast) can transmit a single
audio stream with common connection settings to a number of different
destinations.
Network Address A system for forwarding data packets to different private IP network addresses
Translation
that reside behind a single public IP address.
(NAT)
Packet
A formatted unit of data carried over packet-switched networks.
Port
Address Related to NAT; a feature of a network device that allows IP packets to be
Translation
routed to specific ports of devices communicating between public and private IP
(PAT)
networks.
QoS (Quality of Priority given to different users or data flows across managed IP networks. This
Service)
generally requires a Service Level Agreement (SLA) with a Telco or ISP.
Redundancy
RTP
SDP
SIP
SLA
STL
TCP
TTL
© Tieline Pty. Ltd. 2015
Choosing an alternative audio stream to use if a primary audio connection is
lost.
A standardized packet format for sending audio and video data streams and
ensures consistency in the delivery order of voice data packets.
SDP defines the type of audio coding used within an RTP media stream. It
works with a number of other protocols to establishes a device’s location,
determines its availability, negotiates call features and participants and adjusts
session management features.
SIP works with a myriad of other protocols to establish connections with other
devices. It is used to find call participants and devices and is the method used
by most broadcast codecs to connect to competing brands of codec for
interoperability.
Service Level Agreements (SLAs) a contractual agreement between an ISP and
a customer defining expected performance levels over a network
Studio to transmitter link for program audio feeds.
TCP protocol ensures reliable in-order delivery of data packets between a
sender and a receiver. Its two functions include controlling the transmission
rate of data and ensuring reliable transmission occurs. Generally not wellsuited to streaming live audio because buffering (latency) is employed to
ensure data packets are received in order
Time-to-Live is the setting used in multicast servers to ensure data packets
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Bridge-IT Manual
UDP
Unicast
WAN
have a finite life and don't cause congestion over networks.
The protocol most commonly used for sending internet audio and video
streams. UDP packets include information which allows them to travel
independently of previous or future packets in a data stream. In general, UDP is
a much faster and more efficient method of sending audio over IP.
Broadcasting of a single stream of data between two points.
Wide Area Network; a computer network spanning regions and/or countries to
connect separate LANs
© Tieline Pty. Ltd. 2015
Bridge-IT Manual
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9
Introduction to the Codec
Welcome to Tieline’s Bridge-IT, the ultimate low-cost, high-performance, stereo IP audio codec
solution for broadcast and professional applications. Capable of both peer-to-peer or multi-point
connections, Bridge-IT transports audio streams reliably, simply and effectively over IP data
networks such as wired and wireless LANs, WANs, the internet, satellite IP, Wi-MAX and Wi-Fi.
Bridge-IT is perfect for a large range of broadcast and professional applications that include:
·
·
·
·
·
·
Studio-to-Transmitter Link (STL) applications
Stereo multi-unicast IP audio distribution (stereo to up to 6 endpoints)
Simple remote broadcast links
IP multicasts over compatible IP networks
Low-latency audio over IP bridging solutions
Multiple codec installations (2 codecs fit in 1 x 19” rack unit)
Codec Features
The following table outlines the features available in Bridge-IT.
Bridge-IT Features
Peer-to-peer mono and stereo IP audio
SmartStream IP software for automatic management of IP connection
streaming
High quality low-delay linear audio
G.711 G.722, MPEG Layer 2, plus low-delay Opus, Tieline Music and
MusicPLUS algorithms
Simultaneous analog and digital XLR AES/EBU outputs
Automatic SD/SDHC card connection failover
Web-GUI for remote control and configuration
EBU N/ACIP Tech 3326 compatibility over IP
2 relay inputs and 2 opto-isolated outputs plus RS-232 for local and remote
control of equipment at either end of your codec link
Multi-unicast and send 6 stereo connections, one with bidirectional audio
AAC-LD, AAC-ELD, LC-AAC, HE-AAC v.1 and HE-AAC v.2 algorithms
TieServer for automatic firmware upgrade notification
16 bit and 24 bit aptX® Enhanced algorithm
= included
O = optional
Package Contents
Your codec is delivered with:
· Bridge-IT IP codec
© Tieline Pty. Ltd. 2015
O
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Bridge-IT Manual
· Multi-region plug pack 12 volt 1 Amp power supply
· Phoenix 7-way connector for control port activation
If any of the parts are incorrect, missing, or damaged, contact Tieline or your nearest authorised
dealer.
© Tieline Pty. Ltd. 2015
Bridge-IT Manual
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11
Front Panel Controls
Bridge-IT features a full hardware front panel
interface with menu navigation buttons, an LCD
display with PPM metering and a dialing
keypad.
The codec also features an SDHC card slot,
which can be used for automatic program audio
backup, audio playout and in-store audio
recordings.
Navigation Buttons
Bridge-IT has four arrow shaped navigation
buttons for navigating codec menus and an
OK button for selecting menu items.
Dialing Keypad
The keypad has alpha-numeric buttons and
operation buttons used to:
·
·
·
·
Launch codec functions.
Navigate menus.
Dial and hang up connections.
Configure contact details.
Operation Button Descriptions
Features
1 Return Button
Operation Button Descriptions
Press to navigate back through menus & delete characters
2 Function Button 1
Press to open codec user functions
3 Function Button 2
Press to open codec user functions
4 Connect Button
Press to dial IP connections
5 Home Button
Press to return to home screen
6 Information Button
Press to view a help menu on-screen
7 Settings Button
Press to configure codec settings
8 Hangup Button
Press to end a call
Adjusting LCD Screen Contrast Levels
1. Press and hold the
button and then press the arrow up
button to display the Contrast
adjustment screen.
2. Use the left
and right
arrow buttons to adjust the LCD screen contrast until viewing is
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optimised.
3. Press
when you have finished.
Contrast can also be adjusted by pressing the HOME
and using the down button to navigate to Contrast.
button, selecting Settings, then System,
Enabling and Disabling LCD Screen Auto Dim Mode
By default the codec LCD screen has Auto Dim mode enabled. This dims the intensity of the
display 30 secs after inactivity and is designed to maximise the working life of the screen. Disable
this mode if you want the screen to be illuminated at all times.
1.
2.
3.
4.
Press the HOME
button to return to the Home screen.
Use the navigation buttons on the front panel to select Settings and press the
button.
Navigate to System and press the
button.
Navigate to Auto Dim and press the
button to toggle between Enabled and Disabled.
Important Note: The default Auto Dim time-out is reduced from 30 seconds to 10 seconds
when the Auto Lock function is enabled (to lock the front panel controls). Disabling Auto
Dim mode will override all time-out periods and the LCD will remain fully illuminated at all
times.
© Tieline Pty. Ltd. 2015
Bridge-IT Manual
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13
Rear Panel Connections
1
Features
IN1/AES3 IN
2
IN 2
Rear Panel Connection Descriptions
Balanced Female XLR Mic/Line input 1 or AES3 (AES/EBU)
input for mono and stereo AES3 sources
Balanced XLR Line input 2
3
Out 1/Out2
Balanced Male XLR Analog Line Outputs 1 and 2
4
AES3 OUT
5
CTRL PORT
AES3 (AES/EBU) output via male XLR for mono and stereo
AES3 signals
2 inputs/2 opto-isolated relay outs
6
RS232
RS-232 (DB9) connection for serial device control
7
Headphone Output
Stereo 6.35 mm (¼”) headphone out
8
USB 2.0 port
USB 2.0 for codec configuration
9
LAN
RJ-45 port for Ethernet 10/100 BaseT network connection
10 12v DC input
2.1mm 12V DC power supply connection
XLR Analog and Digital Inputs
Bridge-IT features two XLR microphone inputs.
Input 1 is a balanced mic/line input with the ability to connect high, medium and low gain mics, as
well as an unbalanced source. It has switchable phantom power of 15 volts that is turned off by
default and can also be used as an AES3 (AES/EBU) digital input. This input accepts both mono
and stereo digital AES3 signals. Input 2 is a line input only.
Important Note: Channel 2 input gain can only be adjusted using the Input Audio screen
on the codec (See: Adjusting Input Levels)
XLR Analog and AES3 Outputs
Bridge-IT features two balanced XLR analog audio outputs and a digital XLR AES3 (AES/EBU) audio
output. Both the analog and digital outputs can be used simultaneously and the AES3 output can
send both mono and stereo signals via the single XLR output.
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Bridge-IT Manual
Stereo Headphone Jack Output
Bridge-IT has a 6.35mm (1/4") stereo headphone output jack for monitoring audio inputs 1 and 2 and
return link audio. Channel 1 is mapped directly to the left headphone output and channel 2 is
mapped directly to the right headphone output. When listening to return link audio channel 1 is
mapped directly to the left headphone output and channel 2 is mapped directly to the right
headphone output. (See Headphone Monitoring)
LAN Port
The codec features a RJ-45 port for Ethernet 10/100 BaseT network connections.
Command & Control Interfaces
Bridge-IT features:
1. 2 relay inputs and 2 opto-isolated outputs for machine control via the CTRL PORT.
2. A nine pin RS-232 connection for local and remote control of equipment at either end of the
link
3. A USB 2.0 (slave) connection for codec web-GUI configuration.
DC Power Input
The codec is powered by a 12 volt DC power supply using a standard polarised DC plug.
© Tieline Pty. Ltd. 2015
Bridge-IT Manual
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Navigating Codec Menus
The codec has simple and intuitive menu navigation screens. All main codec menus can be
launched from the Home screen and audio levels remain visible throughout all menus.
Features
1 PPM Meters
Bridge-IT Home Screen
Codec Home Screen Elements
Left (top) and right channel audio levels
2 Screen Name
The name of the current screen being navigated
3 Connect
Select to dial & adjust connection settings
4 Cxns
Displays the number of current connections
5 Programs
View and edit Program dialing configurations
6 Settings
Select to configure codec settings
Press the RETURN
button to navigate backwards through menus, or press the HOME
button to return to the Home screen from any menu. If a full menu cannot be viewed on the codec
screen then arrows on the right hand side of the screen indicate that the current menu has items
below and/or above the items currently visible. Use the navigation arrows to scroll up and down.
Features
1 Up Arrow
Codec Home Screen Elements
Arrow indicating menus can scroll upwards
2 Down Arrow
Arrow indicating menus can scroll downwards
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Codec Menu Overview
© Tieline Pty. Ltd. 2015
Bridge-IT Manual
Connect Menu
© Tieline Pty. Ltd. 2015
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IP Setup Menu Navigation
© Tieline Pty. Ltd. 2015
Bridge-IT Manual
Settings Menu
© Tieline Pty. Ltd. 2015
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Bridge-IT Manual
Adjusting Input/Meter Levels
The codec uses dBu to express nominal operating level, headroom and noise floor levels. The PPM
meters display input audio by default when the codec is not connected and they then switch to
monitor decoded return program audio after making a connection.
Mono and Stereo Audio Capabilities
The codec sends input 1 directly to the left output and input 2 directly to the right output. When
sending mono analog audio select a mono algorithm in the IP Connect Setup menu and connect
audio to input 1 of the codec. Input audio is replicated and sent to both channel 1 and 2 analog XLR
outputs in this mode. The AES3 outputs are directly mapped to both the analog and digital inputs,
therefore if a mono analog profile is selected, only channel one will have audio on it over AES3.
Note: It is not possible to mix channels 1 and 2 into dual mono outputs.
The codec will provide both analog and digital audio out at all times and this is not dependent on
whether your audio source is analog or digital. The only point to note is that when you configure a
mono analog connection the codec will only send audio on one of the AES3 outputs, but it will send
audio on both the left and right channels of the analog outputs.
Adjusting Audio Meter Reference Scale Settings
When connected to stereo sources the top PPM meter is the left channel and the bottom PPM
meter is the right channel. The codec is configured by default to automatically connect to other
Tieline codecs using the correct audio reference meter scales. The audio reference level settings in
the codec are:
Reference Setting
1 Auto (default)
2 Tieline G5
Description
When connecting to a Tieline codec with session data enabled
the codec will automatically adjust the reference level for G5 and
G3 codecs. When connecting to a non-Tieline codec, or a Tieline
codec without session data enabled, the codec will use the
Tieline G5 setting.
The audio reference scale is -16dBu to +22dBu
3 Tieline G3
The audio reference scale is -11dBu and +18dBu
To configure this setting manually:
1.
2.
3.
4.
Press the SETTINGS
button.
Navigate to Audio and press
.
Navigate to Ref Level and press
.
Select the preferred setting and press
.
Audio Metering when Connecting to Tieline G5 Codecs
The Tieline G5 audio reference scale displayed on the codec screen is -16dBu to +22dBu when
you connect to a codec in Tieline's Merlin, Genie or Bridge-IT IP codec families. Set audio levels so
that audio peaks average at the nominal 0vu point. This represents a program level of +4 dBu leaving
the codec. Audio peaks can safely reach +22 dBu without clipping, providing 18dBu of headroom
from the nominal 0vu point. The default PPM audio meter indications are as follows.
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Bridge-IT Manual
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Features
1 -16dBu
Description
PPM meter low point
2 +4dBu
Nominal 0vu reference level at +4dBu
3 +20dBu
+20dBu indication that should not be exceeded to prevent clipping
at +22dBu
PPM indication displays a solid section at the right-hand end
when audio is in danger of clipping
4 PPM meter in clip
Audio Metering when Connecting to Tieline G3 Codecs
New generation Genie, Merlin and Bridge-IT IP codecs have more audio headroom than Tieline G3
audio codecs, therefore metering needs to be adjusted when connecting to a Commander or i-Mix
G3 codec. The G3 metering scale is between -11dBu and +18dBu and audio levels should average
around the nominal 0vu point. Audio peaks should not exceed +16dbu as indicated on the PPM
meter.
Features
1 -11dBu
Description
PPM meter low point
2 +4dBu
Nominal 0vu reference level at +4dBu
3 +16dBu
+16 indication where audio will clip/distort
Channel 1 Mic/Line Level Audio Adjustment
The default input level setting in the codec for channel 1 is line level. To adjust this setting for a miclevel or unbalanced source:
1. Press the SETTINGS
button.
2. Navigate to Audio and press
.
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Bridge-IT Manual
3. Ensure Input Type is set to Analog.
4. Use the arrow-down
button to highlight and select the Ch 1 Gain setting and press the
button.
5. Use the navigation buttons to select the appropriate gain setting and press the
button to
save the setting.
Important Note: 15 volt phantom power is not supplied to input 1 by default. To turn this on:
1. Select Settings > Audio and use the arrow-down
button to highlight the Phantom
setting.
2. Press the
button to toggle between Enabled and Disabled.
Channel 2 is a line input only and gain can only be adjusted using the Input Audio screen
on the codec.
Quick Adjustment of Input Levels
1. Press the
button and the right
arrow button to open the Input Audio level adjustment
screen.
2. Press
on the numeric keypad to toggle channel 1 on and off and press
to toggle channel
2 on and off.
3. Use the up
and down arrow buttons to navigate to the channel you want to adjust. Note: A
channel is highlighted when selected.
4. Use the left
and right
arrow buttons to adjust the input levels up or down.
5. Press the RETURN
button to exit the screen.
Input Audio Features
1 Channel On Symbol
Description
Symbol indicates a channel is turned on
2 Channel Off Symbol
Symbol indicates a channel is turned off
3 Input 1 Level Control
Ch 1 level indication with percentage of gain indicated, i.e. 68.
4 Input 2 Level Control
Ch 1 level indication with percentage of gain indicated, i.e. 68
5 Ch1/2 Gang Indication
Indicates whether ganging is enabled or disabled
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Intelligent Gain Control (IGC)
When the broadcast action really starts to heat up, the codec's inbuilt DSP limiter automatically
takes care of any instantaneous audio peaks that occur in demanding broadcast situations. IGC
(Intelligent Gain Control) is enabled by default and is activated at +20 dBu (G5 audio scale) and
+14dBu (G3 audio scale) to prevent audio clipping. IGC automatically adjusts high audio input levels
downwards until they are acceptable. If IGC auto level recovery (IGC Level) is not enabled, the input
level will remain at the adjusted point until the input gain is manually adjusted again by the user. If
IGC is active in the codec it is indicated in the PPM meter section. To adjust this setting:
1. Press the SETTINGS
button.
2. Navigate to Audio and press
.
3. Navigate to IGC and press
to toggle between Enabled and Disabled.
IGC Auto Level Recovery
IGC Level works with IGC to detect when incoming audio levels have reduced sufficiently.
There are two settings; Auto and Fixed.
If the IGC Level setting is Auto then the codec will return input levels to the gain setting prior
to IGC being activated. The codec takes just 250 milliseconds to detect audio levels have
returned to normal (after IGC Level has been initiated) and will then return the levels to the
previous setting within half a second. This response is linear.
If the setting is Fixed then audio levels will remain lower and not return to the original setting.
To adjust this setting:
1. Press the SETTINGS
button.
2. Navigate to Audio and press
.
3. Navigate to IGC Level and press
to toggle between Auto and Fixed.
Ganging Audio Channels
Ganging allows you to adjust the audio level of both inputs simultaneously.
1. Press the
button and the right
arrow button to open the Input Audio level adjustment
screen.
2. Use the up
and down arrow buttons to navigate to and select Gang 1 + 2 Enabled or
Disabled.
3. Press the
button to select Enabled.
4. Use the up
and down arrow buttons to highlight and select the audio channels.
5. Use the left
and right
arrow buttons to adjust the levels for both inputs up or down
simultaneously.
6. Press the RETURN
button to exit the screen.
When channels 1 and 2 are ganged together:
· Both channels highlight together when selected.
· You can adjust the audio of both channels simultaneously.
· The gain setting for both channels is automatically set to match the gain level of the lowest of
the two channels when ganging is configured.
· If one channel is turned on when ganging is configured then the other one will be turned on
automatically.
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Configuring AES3 Audio
If your input source is AES3 (AES/EBU) format use the IN1/AES3 IN input on the rear panel of the
codec. This is a balanced 110 ohm female XLR input and can operate effectively over distances of up
to 100 meters. The input accepts both mono and stereo AES3 signals as only a single 3 pin XLR
input or output is required for sending and receiving two channels of AES3 data. To configure the
codec to accept AES3 data signals:
1. Press the SETTINGS
button.
2. Navigate to Audio and press
.
3. Select Input Type and press the
button to toggle from Analog to AES/EBU.
The 3 pin male XLR AES3 output on the rear panel is labeled AES3 OUT. It is capable of sending
both mono and stereo AES3 signals.
Important Note: Input levels are set at 100% automatically for AES3 connections. If you
switch back to the analog input setting after selecting AES3, the previous analog settings
will be recovered.
AES/EBU Sample Rate Conversion
The codec implements an Asynchronous Sample Rate Converter (ASRC) to convert the sample rate
of the AES3 input to the sample rate set in the codec. The codec sample rate is determined by the
selected algorithm. For example, if you select the Music algorithm, the sample rate will be set to
32kHz when the codec is connected.
The codec accepts AES3 input sample rates of 32kHz, 44.1kHz or 48kHz. The output sample rate
is determined by the algorithm used when connected.
AES3 Audio Out & SD Card Sample Rate
1. When you are not connected, you can adjust the AES3 output sample rate manually in the
Audio menu via the AES3 SR setting.
2. If file playback occurs before a connection is initiated, the AES3 output sample rate will match
the audio file sample rate.
3. For best performance, the SD card file sample rate should match the AES3 SR setting in the
codec and the algorithm sample rate. If they don't match, the codec will re-sample the SD card
file audio to match the connection sample rate, and this will be used by the AES3 output. For
example, if you are streaming audio using Tieline Music (32kHz sampling), file playback will be
re-sampled to 32kHz if the SD card file is 44.1kHz or 48kHz. In this example the AES3 output
sample rate will be 32kHz.
Adjusting the Codec Output Sample Rate
As there is no external reference clock for the codec it is necessary to set the output sample rate of
the codec when you are not connected. The AES3 SR setting in the Audio menu will configure
audio outputs and audio monitoring, as well as SD card playback at this sample rate. The default
setting is 48kHz sampling:
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1.
2.
3.
4.
Press the SETTINGS
button.
Navigate to Audio and press
.
Navigate to AES3 SR and press
.
Select a preferred sample rate then press
25
.
Tieline normally recommends selecting STL Silence Mode for audio monitoring if using AES3. If
using Auto Select make sure the algorithm sample rate and the AES3 SR sample rate setting are
the same.
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Headphone/Output Monitoring
The 6.35mm (1/4") stereo headphone output on the codec can be used for monitoring audio inputs 1
and 2 and return link audio.
Important Note: When the codec makes a connection it will monitor the decoded return
audio link by default.
Adjusting Headphone Output Levels
When using analog or digital inputs you will see input audio on the PPMs and hear it in the
headphones.
1. Press and hold the
button and then press the right
arrow button to display the H/P
Volume adjustment screen.
2. Use the left
or right
navigation buttons to adjust the volume levels up or down. The screen
displays level adjustments in real-time.
3. Press
when you have finished.
Headphone levels can also be adjusted by navigating to SETTINGS
down button to navigate to HP Level and press
.
> Audio and using the
Output/Headphone Monitoring Settings
There are three Output/HP monitoring configurations available in the codec. The default setting is
Auto Select and to adjust this setting:
1. Press the SETTINGS
button.
2. Navigate to Audio and press
.
3. Navigate to Output/HP and press
to select either Monitor Input or STL Silence.
The table below displays how audio is routed to the codec outputs/headphones based on the
configuration selected and the current codec state (IP streaming or idle). Please note:
· The same audio is always routed to the headphone and XLR outputs.
· Output 1 corresponds to headphone left and output 2 corresponds headphone right.
· The same audio is routed to the analog and AES XLR outputs.
· When inputs are routed to the outputs, input 1 is sent to output 1 and input 2 is sent to output
2.
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Codec Connection State
Mode
Idle
Streaming (Mono)
Streaming (Stereo)
Auto Select
(default)
Inputs
Same decoded audio
on all outputs
Channel 1 decoded
on output 1, and
channel 2 decoded on
output 2
STL Silence
No audio
Same decoded audio
on all outputs
Channel 1 decoded
on output 1, and
channel 2 decoded on
output 2
Monitor Input
Inputs
Inputs
Inputs
Auto Select Mode
Auto Select is the default Output/HP monitoring setting in the codec. Use this setting if you
want to monitor the inputs when not connected and return audio when connected.
Monitor Input Mode
Select Monitor Input to configure the codec to always monitor input audio. This may be useful
if an announcer wants to monitor their own voice and not return audio when connected.
STL Silence Mode
1. In STL Silence mode input audio is not monitored on the PPMs or via the headphone output
before the codec is connected. It is necessary to check input audio levels using Auto Select
or Monitor Input modes prior to connecting.
2. When the codec connects in STL Silence mode it automatically monitors decoded incoming
audio.
3. If the connection is lost for any reason then silence is enabled, ensuring input audio cannot
be misconstrued as return program audio for STL connections.
4. If the connection is subsequently restored the codec will again monitor decoded incoming
audio.
11
Language Selection
English is the default language in the codec. To adjust this setting:
1.
2.
3.
4.
Press the SETTINGS
button.
Navigate to System and press
.
Use the navigation buttons to select Language and press
Select a language and press
.
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About Program Dialing
What Defines a Program?
The codec uses the concept of 'program' dialing to connect using peer-to-peer, multicast or multiunicast connections. A Program configures a Tieline codec to send or receive an Audio Stream.
The attributes of each audio stream and associated connections are embodied within a program
when it is created, including the configuration, dialing and answering parameters. Essentially a
program is like a connection profile with:
· A Program Name.
· IP address dialing details for up to 6 connection end-points or a multicast IP address.
· Specific connection profile details pertaining to algorithm, FEC, jitter buffer and bit-rate
settings etc.
Custom programs allow you to store connection settings for a range of peer-to-peer, multicast and
multi-unicast connections and retrieve or edit them easily at the touch of a button. Simple peer-topeer or multicast profiles can be created using the codec front panel, whereas multi-unicasts must
be created with the Programs panel in the Toolbox web-GUI.
Using Programs to Dial between Two Tieline Codecs
Tieline Bridge-IT and Bridge-IT XTRA codecs operate similarly to Tieline G3 codecs. By default,
Tieline codecs send proprietary session data when connecting to each other in order to establish,
manage and terminate connections. When a connection between two codecs is established:
1. The dialing codec sends information about how the codec receiving the call should be
configured.
2. Once the codec receiving session data from the dialing codec has received information
successfully, it sends an acknowledgement to the dialing codec and streaming can
commence.
If you configure a stereo program on the dialing codec using a particular algorithm and bit rate etc.,
these settings will be configured on the answering codec when it connects. It is also possible to
lock a loaded program in a codec to ensure the currently loaded program cannot be unloaded by a
codec dialing in with different program settings.
Creating Programs
Simple peer-to-peer (point-to-point) programs can be created using the codec front panel. The
Toolbox web-GUI contains a Programs panel with a wizard for configuring program settings and
backup connections. Edit settings easily at the touch of a button and use existing programs as
templates for creating other programs.
Mono and Stereo Peer-to-Peer Programs
New peer-to-peer programs can be created using the codec front panel keypad (see 10 Quick
Steps to Connect). If you know the IP address of the codec you want to dial then all you need to
do is enter this into the codec, choose your preferred connection settings and then press
CONNECT
.
Front panel configured programs are automatically saved as Recent Programs which retain all
the audio stream dialing and configuration information. These Recent Programs are displayed
when you press the CONNECT
button from within any menu except the IP Mode or SIP
Mode screens, or the Connect IP or Connect SIP screens.
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Ensure you configure all the correct connection settings when using the codec front panel,
because these are stored as part of the program's profile when you first connect. They cannot
be adjusted afterwards without using the editing features in the Program panel within the
Toolbox web-GUI.
Important Note: When configuring a connection use the Save function in the Connect IP
and Connect SIP screens to save programs permanently to the codec's Programs menu.
Otherwise they are stored to the Recent Programs list and will be overwritten after several
calls have been made.
Multi-unicast Programs:
Multi-unicast programs can contain dialing information for up to 6 connection end-points. They
are useful for distributing audio to several studios and can be created using the Programs
panel in the Toolbox web-GUI. (See Configuring Multi-Unicast Programs for configuration
details). Once multi-unicast connections have been created they can be dialed via the codec
keypad without using the web-GUI.
Multicast Programs:
Multicasts can be used to broadcast to 'subscribers' who wish to connect to a multicast stream.
Multicast server transmissions are sent using a dedicated IP multicast address that looks
similar to a regular IP address and multicast (client) subscribers request transmissions from
this address. New programs can be created using either the codec front panel or the Programs
panel in the Toolbox web-GUI. (For web-GUI configuration see Configuring a Multicast Server
Program or Configuring a Multicast Client Program).
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Getting Connected Quickly
Preparing to Connect
Before attempting a new connection please connect and adjust the following:
1.
2.
3.
4.
Attach the supplied 12 volt power supply to the codec.
Attach an RJ45 Ethernet cable to the LAN port on the rear panel of the codec.
Attach headphones to the 6.35mm (1/4") headphone jack on the rear panel of the codec.
Check that the correct country is selected in the codec.
i. Press the SETTINGS
button.
ii. Navigate to System and press the
button.
iii. Navigate to Country and press the
button.
iv. Use the navigation buttons to select your country of operation.
5. Make sure you have the IP address of the codec you are dialing, or have used the Toolbox
web-GUI to load the programs you will be using to dial onto the codec. (see Configuring IP
Addresses).
13.1
10 Quick Steps to Connect Bridge-IT
Important Notes:
· The following procedure will create a custom peer-to-peer connection program using the
codec front panel keypad and navigation buttons. It instructs how to connect your codec
over IP for the very first time without using the Toolbox web-GUI and your computer for
configuration.
· See the Java Toolbox Web-GUI Introduction for details on configuring connections
remotely via a computer. Creation of programs is not currently supported in the HTML5
Toolbox Web-GUI.
· See Installing the Codec at the Studio for valuable information about installing your
codec, negotiating firewalls and port forwarding.
· See Tips for Creating Reliable IP Connections for a range of IP information to assist with
setting up IP services for your codecs.
· See Testing IP Network Connections to learn how you can test and verify the reliability of
your IP connection.
1. Press the
button and right
navigation button to open the Input Audio Level adjustment
screen and adjust audio levels.
· Press
on the numeric KEYPAD to toggle channel 1 on and off and press
to toggle
channel 2 on and off.
· Use the up
and down
navigation buttons to select Gang 1 + 2 and press the
button
to toggle ganging on/off.
· Use the up
and down
navigation buttons to select a single channel, or ganged
channels. Note: A channel is highlighted when selected.
· Use the left
and right
navigation buttons to adjust the input levels up or down.
Important Note: 15 volt phantom power is not supplied to input 1 by default. To adjust this
setting select Settings, then Audio and then Phantom. Press the OK button to toggle
between Disabled and Enabled.
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2. Press the HOME
button to return to the Home screen, select Connect and press the
button, then select IP and press the
button.
3. Select your preferred IP Session mode. In this Peer-to-Peer connection example we have
selected Tieline, which uses Tieline session data, then press the
button. Note: Select SIP
or Sessionless if these connections are required.
4. Use the numeric KEYPAD to enter the IP address of the codec you want to dial, using the
or
buttons to enter the periods in the IP address. Use the RETURN
button to delete
numbers already entered. Next, press the down
navigation button to select Setup and press
.
Important Note: The codec remembers recent IP addresses just like a cell-phone. To
view these addresses just press the OK button when you select the Connect IP screen.
The most recent addresses and programs are listed first and you can use the navigation
buttons to scroll up and down. Press the OK button to select the address you have
highlighted.
4. Press the down
navigation button to select Algor'm (algorithm) and press
.
5. Use the navigation buttons to select a preconfigured algorithm profile, or manually enter algorithm
settings, then press
.
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6. If you decide to manually configure the algorithm, use the navigation buttons to select your
preferred sample rate (if displayed) and bit rate. Press
after selecting each option.
7. Press the down
navigation button to select Jitter (jitter buffer) and press
to select a
different automatic jitter buffer setting for your connection. Alternatively you can enter a fixed jitter
buffer value in milliseconds (maximum 5000 ms). The default Auto, Best Compromise setting is
a good starting point for most internet connections.
8. Press the down
navigation button to select FEC and press
to view selection options. Use
the navigation buttons to choose the FEC percentage you want to use and press
.
9. When configuration is complete press the RETURN
Connect IP screen that the IP address was entered into.
button to navigate backwards to the
Important Note: At this point you can navigate to Save on the Connect IP screen and
press
to save the settings as a custom program for subsequent recall and dialing. Use
the numeric KEYPAD to give the program a name and press
to save the program. A
confirmation message is displayed after the program is saved.
10.Press the CONNECT
button to make a connection. The Wait Connecting screen appears
during the connection process.
After successfully connecting the codec will display connection details. Use the down
navigation
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button to view connection Status and press
to view connection statistics for IP packets being
sent over the connection. To negotiate higher bit-rates press
then 3 on the numeric KEYPAD;
for lower bit-rates press
then 9.
13.2
Monitoring IP Connections
The number of active audio streams and connections is displayed on the Home screen via Cxns. In
the following image the program has two connections configured and both are active, expressed as
2/2. If only one was active the bracketed number would display 1/2.
To view more detailed connection information:
1. Press the HOME
button to return to the Home screen.
2. Use the navigation buttons to select Cxns and press the
button.
The Connected IP screen displays all connections. The IP address dialed and the LQ (link quality)
is displayed on the screen and you can use the down
navigation button to view the algorithm
being used, the connection bit rate, total bytes used and the jitter buffer latency over IP network
connections.
Link Quality (LQ) Readings
Send and return LQ numbers can also help to determine if a problem is occurring at either end of a
connection. For example, on an IP connection the Return reading represents the audio being
downloaded from the network locally (i.e. audio data is being sent by the remote codec).
Conversely, the Send link quality reading represents the audio data being sent by the local codec
(i.e. being downloaded by the remote codec). To ensure a stable connection, try to maintain a
reliable reading of 80 or higher for both the Send and Return LQ reading.
Important Note:
· The Return link quality reading is the same as the Local (L) setting displayed on a G3
codec.
· The Send link quality reading is the same as the Remote (R) setting displayed on a G3
codec.
Viewing Connection Statistics
Navigate to Status in the Connected IP screen and press the
button to display the Cxn Stats
(connection statistics) screen. This displays the performance of the codec in sending IP audio
packets across the network. Analysis is historic and assessed over 60 seconds and 10 minutes of
connection time.
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Feature
1 Lost Packets
Description
Packets sent that failed to arrive
2 Empty
(Jitter Indicates how often the jitter buffer ‘reservoir’ empties causing loss of
Buffer)
audio
3 Late Packets
The number of packets that arrive late, i.e. after audio play out
4 FEC Packets
5 1 minute
Indicates the number of forward error correction (FEC) packets that
have been sent if it is enabled in the codec
Statistics listed for the last minute of network activity
6 10 minutes
Statistics for the last 10 minutes of network activity
Important Note: If the jitter buffer, FEC or the connection bit rate is changed, we
recommend assessing a minute of recent connection performance in preference to 10
minutes of historical connection performance. 10 minutes of data will include connection
settings which may no longer be relevant. ‘Packet arrival history’ is cleared when you hang
up a connection.
Following is an analysis of possible causes and solutions for the packet analysis statistics
displayed on the screen.
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35
Packet
Analysis
Displays
Possible Causes
Possible Solutions
Loss
Packets sent
and that failed
to arrive.
·
·
·
·
LAN/WAN congestion
Unreliable ISPs
Unreliable networks
Inferior IP hardware
· Renegotiate connection bit rate
downwards
· If link quality good add or increase
FEC as required
· Assess ISP’s QOS if very bad
performance
Empty
Indicates how
often the jitter
buffer
‘reservoir’
empties
causing loss
of audio.
· High number of
packets being lost or
arriving late
· Signal drop-outs
using cell-phone
networks
· Renegotiation
causes the jitter
buffer reservoir to
empty
· Once could be an anomaly –
assess lost & late packets
· If many lost packets network is
unreliable – renegotiate bit rate
and /or FEC down
· If many late packets increase jitter
buffer
Late
The number of
packets that
arrive late and
after audio
play out.
· Network congestion
· Jitter Buffer depth is
too low
· Auto-jitter buffer will adjust
automatically
· For manual jitter buffer settings
increase jitter buffer depth 50-100
ms & reassess (if only a few
packets arrive late over time, audio
repairs will be automatic and may
not require buffer changes).
FECd
Indicates the
number of
FEC repaired
packets if FEC
active.
· Packets have been
lost or corrupted over
the network
· Assess audio quality & the number
of FEC repairs – if many packets are
being ‘lost’ perhaps reduce FEC &/
or renegotiate bit rate down.
Load and Dial Custom Programs
Custom programs stored on the codec are simple to load and dial from the codec front panel.
1. Press the HOME
button to return to the Home screen.
2. Use the navigation buttons to select Programs and press the
button.
3. Use the up
and down
navigation buttons to select the program you want to connect
with, then press the CONNECT
button to dial.
4. The Wait Connecting screen appears during the connection process and then connection
details are displayed.
13.4
Disconnecting a Connection
1. Press the red DISCONNECT
button on the numeric KEYPAD at any time to hangup a
connection.
2. Use the right
navigation button to select Yes and press the DISCONNECT
button or the
button to confirm the connection hangup.
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13.5
Bridge-IT Manual
Redialing a Connection
Press the CONNECT
button from any codec menu to redial previous connections (except
menus accessed via the Connect > IP screen).
Manually dialed connections are saved as programs - retaining all the dialing and configuration
information programmed into the codec. A program is identified in the Recent Program redial
screen using either a previously entered name, or by an IP address (manually dialed connections).
A multi-unicast connection will display the first IP address dialed and the number of additional
connections dialed.
Screen Display
1 Manual
peer-to-peer
connection
2 Manual
multi-unicast
connection (via web-GUI)
3 Program name (via web-GUI)
Description
Displays the IP address of an IP connection to a single
end-point
Displays the IP address of first codec dialed and the
number of additional codec connections
Displays the name of a program configured by the
Toolbox web-GUI
Redialing Manually from the Connect IP Screen
From the Home screen select Connect > IP > [Select an IP Session mode] and the codec will
assume you want to dial a new ad hoc manual connection. Press the CONNECT
button when
the Connect IP (or Connect SIP) screen is displayed to retrieve previously dialed IP addresses.
Codec settings for any connection dialed from the IP Connect (or Connect SIP) screen will include
the current settings in the Setup menu.
13.6
Configuring Auto Reconnect
Auto Reconnect is disabled by default. When enabled the dialing codec attempts to reconnect if
data is temporarily lost over an IP connection. To adjust the setting:
1. Press the HOME
button to return to the Home screen, select Connect, then select IP
and press the
button.
2. Select the IP Session mode you are using to connect.
3. Select Setup and press
.
4. Navigate to A/Recon and press
to toggle between Enabled and Disabled.
Important Note: When Auto Reconnect is enabled, the dialing codec will continue to
attempt a connection with the remote codec until Disconnect is pressed either on the
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dialing codec's keypad, or in the web-GUI. This setting should only be configured on the
dialing codec.
13.7
Speed Dialing Connections
Assigning Speed Dial Numbers
1.
2.
3.
4.
Press the HOME
button to return to the Home screen.
Use the navigation buttons to select Programs and press the
button.
Navigate to the program you want to assign a speed dial number, then press the
Navigate to Speed Dial and press the
button.
button.
5. Navigate to the speed dial number you want to assign to the selected program and press the
button.
6. A confirmation message will display the number assigned.
Speed Dialing
1. Press the HOME
button to return to the Home screen.
2. Use the numeric KEYPAD to enter the speed dial number.
3. When the Speed Dial screen appears, press the
button or the CONNECT
connect.
13.8
button to
Dial/Disconnect Multiple Connections
Multiple Connections within Programs
Multi-unicast programs allow you to simultaneously transmit a mono or stereo audio stream to up to
6 destination codecs. Multi-unicast programs can only be created using the Toolbox web-GUI. There
are two ways to simultaneously dial multiple IP audio stream connections multi-unicast programs:
1. Load the program into the codec via the front panel and dial.
2. Connect to the codec using the Toolbox web-GUI and use the Master panel to load the
program and connect.
Dialing Multiple Connections via the Front Panel
1. Press the HOME
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2. Use the navigation buttons to select Programs and press the
button.
3. Use the up
and down
navigation buttons to select the multi-unicast program you want
to connect with, then press the CONNECT
button to make a connection.
4. The Wait Connecting screen appears briefly and then the Home screen is displayed.
It is also possible to redial the connection.
Disconnect All Connections
1. Press the red DISCONNECT
button on the numeric KEYPAD at any time to hangup all
connections.
2. Use the right
navigation button to select Yes and press the DISCONNECT
button or the
button to confirm the disconnection.
Disconnecting Individual Multi-unicast Connections
It is only possible to disconnect individual connections via the Master panel in the dialing codec's
Toolbox web-GUI.
13.9
Creating a Multicast Server Program
Two different types of multicast programs need to be created when multicasting:
· A multicast server program is used by the broadcasting codec to send multicast IP packets
to multicast routers on a network.
· A multicast client program is used by codecs to receive multicast IP audio packets.
Prerequisites:
· Bridge-IT firmware v.1.01.00 r4219 or higher.
· ToolBox web-GUI v.1.2.2.3 or higher.
· A multi-unicast license installed in the dialing codec (Note: the multi-unicast license includes
multicast server capability).
Important Notes:
· You cannot edit a program when it is currently loaded in the codec.
· Ensure all connection related settings like the port, algorithm, bit rate (etc) match on
both multicast server and client programs or they will not be able to join multicast
streaming sessions.
· There is no jitter buffer setting in a multicast server program because it is an encode
only program and never receives audio packets.
· You can lock a loaded custom program in a codec to ensure the currently loaded
program cannot be unloaded by a codec dialing in with a different program type.
· Always dial the multicast server codec connection first before connecting multicast
client codecs.
· Multicast client codecs will display return link quality (LQ) only. The Return reading
represents the audio being downloaded from the network locally. Multicast server
codecs do not display LQ readings.
· Forward Error Correction (FEC) is not available for multicast connections.
· It is not possible for a G3 codec to receive multicast IP audio streams.
· To learn more about programs see the section titled "About Program Dialing".
· See Toolbox web-GUI documentation for more detailed information about "Configuring
Multicast Server Programs" or "Configuring Multicast Client Programs"
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1. Press the HOME
button to return to the Home screen, select Connect > IP > Sessionless
and press the
button.
2. Select M'cast Server to configure a server codec program.
3. Use the RETURN
button to delete any numbers already entered, then use the numeric
KEYPAD to enter the multicast IP address you want to dial, using the
or
buttons to enter
the periods in the IP address. Note: The same multicast address and audio port must be used for
both the server and client programs. Next, press the down
navigation button to select Setup
and press
.
4. Press the down
navigation button to select Algor'm (algorithm) and press
.
5. Use the navigation buttons to select an algorithm profile or manually choose algorithm settings,
then press
.
6. Navigate to and select TTL and enter the IP Time-to-Live value, then press
. Please note: The
TTL value you need to use is dependent upon your network infrastructure. Please consult your
network administrator if you are unsure about how to configure this setting.
7. Select Data to enable or disable data in the audio stream. See RS232 Data Configuration for
more information on data connections.
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8. Select Proto (protocol) to select the audio protocol and adjust the Remote Audio Port. Select
UDP/IP +RTP for RFC-compliant IP streaming. Press
to save settings.
9. If required, enable A/Recon (auto reconnect) and use Via to specify which IP streaming interface
is used to dial this connection, e.g. Primary (Ethernet port) or VLAN (if configured). Note: By
default Any will select Primary.
10.Press the RETURN
button when configuration is complete to navigate backwards to the
Connect IP screen that the multicast IP address was entered into.
Important Note: At this point you can navigate to Save on the Connect IP screen and
press
to save the settings as a custom program for recall and dialing. Use the numeric
KEYPAD to give the program a name and press
to save the program. A confirmation
message is displayed after the program is saved.
Connecting a Multicast Program
1. After you have created multicast server and client programs on your codecs you can dial
multicast connections. First select the multicast server program you want to use on the server
codec:
a. Press the HOME
button to return to the Home screen.
b. Use the navigation buttons to select Programs and press the
button.
c. Use the up
and down
navigation buttons to select the multicast server program you
want to connect with, then press the
button to load the program.
d. Press the CONNECT
button to make a connection.
2. Select and load the multicast client program on each of the multicast client codecs and dial the
multicast IP address to begin receiving multicast audio packets.
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Bridge-IT Manual
Navigate to Cxns on the Home screen to view a codec's connection Status, then press
connection statistics for IP packets being sent over the connection.
13.10
41
to view
Creating a Multicast Client Program
Use the procedure which follows to configure a multicast client program and allow the codec to
receive multicast IP audio packets.
Prerequisites:
· Bridge-IT firmware v.1.01.00 r4219 or higher.
· ToolBox web-GUI v.1.2.2.3 or higher.
· A multi-unicast license installed in the dialing codec (Note: the Multi-Unicast license includes
multicast server capability).
Important Notes:
· You cannot edit a program when it is currently loaded in the codec.
· Ensure all connection related settings like the port, algorithm, bit rate (etc) match on
both multicast server and client programs or they will not be able to join multicast
streaming sessions.
· The default UDP audio port is 9000 for a multicast client program configured via the
codec front panel.
· You can lock a loaded custom program in a codec to ensure the currently loaded
program cannot be unloaded by a codec dialing in with a different program type.
· Always dial the multicast server codec connection first before connecting multicast
client codecs.
· Multicast client codecs will display return link quality (LQ) only. The Return reading
represents the audio being downloaded from the network locally.
· Forward Error Correction (FEC) is not available for multicast connections.
· It is not possible to connect to a G3 codec and receive multicast IP audio streams.
· To copy multicast client programs onto multiple codecs see Backup and Restore
Functions.
· To learn more about programs see the section titled "About Program Dialing".
· See Toolbox web-GUI documentation for more detailed information about "Configuring
Multicast Client Programs"
1. Press the HOME
button to return to the Home screen, select Connect > IP > Sessionless
and press the
button.
2. Select M'cast Client to configure a client codec program.
3. Use the RETURN
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button to delete any numbers already entered, then use the numeric
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Bridge-IT Manual
KEYPAD to enter the multicast IP address you want to dial, using the
or
buttons to enter
the periods in the IP address. The same multicast address and audio port must be used for both
the server and client programs. Next, press the down
navigation button to select Setup and
press
.
4. Press the down
navigation button to select Algor'm (algorithm) and press
.
5. Use the navigation buttons to select an algorithm profile or manually choose algorithm settings,
then press
.
6. Click to configure your preferred Jitter Buffer setting. Select Auto Adapt and your preferred
automatic jitter setting, or Fixed Level. For a fixed buffer setting enter the Jitter Buffer Depth,
which has a maximum setting of 5000ms, then press
.
Important Notes: Automatic or fixed jitter buffer settings can be adjusted on individual
client codecs as required. There is no jitter buffer setting on the server codec because it
never receives audio packets.
7. Select Data to enable or disable data in the audio stream. See RS232 Data Configuration for
more information on data connections.
8. Select Proto (protocol) to select the audio protocol and adjust the Local Audio Port. Select
UDP/IP +RTP for RFC compliant IP streaming. Press
to save settings.
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7. If required, enable A/Recon (auto reconnect) and navigate to Via to specify which IP streaming
interface is used to dial this connection, e.g. Primary (Ethernet port) or VLAN (if configured).
Note: By default Any will select Primary.
8. Press the RETURN
button when configuration is complete to navigate backwards to the
Connect IP screen that the multicast IP address was entered into.
Important Note: At this point you can navigate to Save on the Connect IP screen and
press
to save the settings as a custom program for subsequent recall and dialing. Use
the numeric KEYPAD to give the program a name and press
to save the program. A
confirmation message is displayed after the program is saved.
Connecting a Multicast Client Program
1. After you have created multicast server and client programs on your codecs you can dial
multicast connections. First select the multicast server program you want to use on the server
codec and dial to connect.
2. Select and load the multicast client program on each of the multicast client codecs and dial the
multicast IP address to begin receiving multicast audio packets.
a. Press the HOME
button to return to the Home screen.
b. Use the navigation buttons to select Programs and press the
button.
c. Use the up
and down
navigation buttons to select the multicast client program you
want to connect with, then press the
button to load the program.
d. Press the CONNECT
button to make a connection.
Navigate to Cxns on the Home screen to view a codec's connection Status, then press
connection statistics for IP packets being received over the connection.
13.11
to view
Dialing SIP Peer-to-Peer
Important Notes:
· The codec is fully EBU N/ACIP Tech 3326 compliant when connecting using SIP
(Session Initiation Protocol) to other brands of IP codecs.
· Both codecs connecting over SIP need to be configured
· SIP dialing is only supported over peer-to-peer connections, not multi-unicast
connections.
· Tieline G3 codecs do not support AAC and will default to MPEG Layer 2 if a Bridge-IT
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Bridge-IT Manual
codec configured for AAC attempts to connect.
· Some Telcos and ISPs may block SIP traffic over UDP port 5060.
· When connecting to a Tieline G3 codec using SIP you need to manually select the G3
audio reference level in the codec. To do this select SETTINGS
> Audio > Ref
Level > Tieline G3. In addition, select the following on the G3 codec prior to dialing:
 Select either a mono or stereo profile
 Select [Menu] > [Configuration] > [IP1 Setup] > [Session Type] > [SIP]
 Select [Menu] > [Configuration] > [IP1 Setup] > [Algorithm] > [G711/G722 or
MP2]
Dialing Peer-to-Peer SIP IP Connections
SIP can be used to make direct peer-to-peer calls to different brands of IP codecs with public IP
addresses, or between two codecs over a LAN which do not pass through firewalls. Peer-to-peer SIP
calls are usually used to connect to other brands of codecs and perform call and session
management tasks. Peer-to-peer SIP calls between two codecs are detected automatically and
require no special configuration.
To make a peer-to -peer call between codecs we recommend both codecs use public IP addresses:
·
·
·
·
Find out the IP address of the remote codec being dialed.
Program each codec with a compatible algorithm and sample rate etc.
Dial using SIP within the Connect menu.
If the remote codec has a private IP address then it should be configured for port forwarding
and should dial the public IP address at the studio (see Configuring TCP/UDP Protocols for
more details on port forwarding).
1. To dial peer-to-peer press the HOME
button to return to the Home screen, select Connect >
IP > SIP.
2. Use the numeric KEYPAD to enter the IP address of the codec you want to dial, using the
or
buttons to enter the periods in the IP address and use the RETURN
button to delete
numbers already entered.
3. Then press the down
navigation button to select Setup and press
to adjust the algorithm,
jitter buffer and encode/decode direction if required.
4. Press the RETURN
button to navigate backwards to the Connect SIP screen.
5. Press the CONNECT
button to make a connection.
13.12
Dialing SIP Addresses
Dialing a SIP Address via the Codec Front Panel
1. Press the HOME
button to return to the Home screen, select Connect > IP > SIP.
2. Use the KEYPAD to enter any combination of alphabetic and numeric characters in the SIP
address of the codec you want to dial. Use the
or
buttons to enter the periods in the SIP
address and use the RETURN
button to delete any numbers already entered. Alternatively, if
you have dialed the SIP address previously, press the RETURN
button to view the Recent
Call screen and select the SIP address you want.
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3. Press the down
navigation button to select Setup and press
, then adjust the algorithm,
jitter buffer, encode/decode direction, port and auto reconnect settings if required.
4. Press the RETURN
button to navigate backwards to the Connect SIP screen and select
Save to name and save the program.
5. Press the CONNECT
button to make a connection.
Important Notes:
· See Configuring SIP Settings for instructions on entering SIP account details into the
codec using the Toolbox web-GUI. If your codec is registered with same SIP registrar as
the destination codec then you only need to enter the SIP user name to dial
successfully.
· Each codec should be registered to a different SIP server account to avoid connection
conflicts.
· If you don't save the program during configuration, a temporary program is created after
you dial the SIP connection for the first time using the codec KEYPAD. The temporary
program will appear in the recent calls list if you want to redial the program.
13.13
Deleting Programs
1. Press the HOME
button to return to the Home screen.
2. Use the navigation buttons to select Programs and press the
button.
3. Navigate to the program you want to delete and press the
button.
4. Navigate to Delete and press the
button.
5. Confirm program deletion and press the
13.14
button.
Selecting Algorithm Profiles
The codec has a number of preconfigured mono and stereo dialing profiles available. These can be
used to configure the codec quickly with the most popular settings that provide high quality
connections using each available algorithm.
If you are unsure about which algorithm to use, see Selecting an Algorithm for more details on each
algorithm available in the codec.
1. Press the HOME
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button to return to the Home screen.
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Bridge-IT Manual
2. Use the navigation buttons to select Connect and press the
button.
3. Select IP and press the
button.
4. Select Tieline session mode and press the
button. Note: algorithm profiles are only
available for Tieline session connections.
5. Use the down
navigation button to select Setup and press the
button.
6. Press the
button to select Algor'm.
7. Use the right
navigation button to select Profile.
8. Select the profile you want from the Favorite, Mono or Stereo menus.
Features
1 Favorite
2 Mono
Codec Home Screen Elements
Displays a list of favorite profiles that have been selected manually
within the codec by users
Displays preconfigured mono profiles within the codec
3 Stereo
Displays preconfigured stereo profiles within the codec
Adding a Profile into the Favorite Menu
1.
2.
3.
4.
5.
6.
7.
8.
9.
Press the HOME
button to return to the Home screen.
Use the navigation buttons to select Connect and press the
button.
Select IP and press the
button.
Select your preferred IP Session mode and press the
button.
Use the down
navigation button to select Setup and press the
button.
Press the
button to select Algor'm.
Use the right
navigation button to select Profile.
Select the profile you want from the All, Mono or Stereo menus.
Press the hatch button
to add the profile into the Favorite menu.
Profiles that have been added into the Favorite menu are identified by the hatch (pound) symbol
next to their name after they have been selected.
Deleting a Profile from the Favorite Menu
1.
2.
3.
4.
5.
6.
Press the HOME
button to return to the Home screen.
Use the navigation buttons to select Connect and press the
button.
Select IP and press the
button.
Select your preferred IP Session mode and press the
button.
Use the down
navigation button to select Setup and press the
button.
Press the
button to select Algor'm.
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7. Use the right
navigation button to select Profile.
8. Select the profile you want to delete from the Favorite menus.
9. Press the hatch button
to delete the selected profile from the favorite menu.
Available Profiles
The following profiles are pre-configured in all Bridge-IT codecs. Note: AAC and aptX Enhanced
algorithm profiles are not available unless valid licenses have been installed in the codec.
Profiles
Algorithm
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
AAC
AAC
AAC
HE-AAC
HE-AAC
HE-AAC
AAC-LD
AAC-LD
AAC-ELD
AAC-ELDv2
Enhanced apt-X
Enhanced apt-X
Enhanced apt-X
Enhanced apt-X
G.711
G.722
MPEG 1 Layer 2
MPEG 1 Layer 2
MPEG 1 Layer 2
MPEG 1 Layer 2
MPEG 1 Layer 2
MPEG 1 Layer 2
Music
Music
Music
Music
MusicPLUS
MusicPLUS
MusicPLUS
MusicPLUS
MusicPLUS
PCM Mono
PCM Stereo
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Mono/Stereo
Sample Rate (kHz)
Bit-rate (Kbps)
Mono
Stereo
Stereo
Mono
Stereo
Stereo
Mono
Stereo
Mono
Stereo
Mono
Mono
Stereo
Stereo
Mono
Mono
J-Stereo
J-Stereo
Mono
Mono
Stereo
Stereo
Mono
Mono
Stereo
Stereo
Mono
Mono
Stereo
Stereo
Stereo
Mono
Stereo
48
48
48
32
32
32
32
32
32
32
32 (16 bit)
48 (24 bit)
32 (16 bit)
48 (24 bit)
8
16
32
48
24
48
32
48
32
32
32
32
48
48
48
48
48
48 (16bit)
48 (16bit)
64
128
256
16
32
48
48
64
24
48
128
288
256
576
64
64
128
192
64
256
128
256
28.8
48
64
96
48
96
96
128
192
768
1,540
48
13.15
Bridge-IT Manual
SDHC Card Backup
SDHC File Backup
The codec features an SD/SDHC card slot for automatic backup to MP2 or MP3 recordings if an IP
connection is interrupted. Backup connections are configured using the web-GUI and this is outlined
in Configuring Mono or Stereo Peer-to-Peer Programs. USB file backup is automatic and occurs:
1. If encoded audio streaming from a remote codec is lost for a time period predetermined within
the web-GUI (default 30 seconds). Note: Loss of audio is measured by whether data packets
can be received or decoded. If audio is muted at the remote codec and it continues to send
audio packets containing 'silence', the receiving codec will not activate USB file backup.
2. Immediately if a connection to another codec is lost.
Backup will occur according to the silence threshold parameters configured for audio file backup.
After SD/SDHC file backup is activated the audio file plays continuously in loop mode until a backup
connection is dialed and connects, or the primary connection is restored. The codec Home screen
indicates failover to the backup SD/SDHC card has occurred by displaying (F) in the Cxns display.
Playback continues during reconnection attempts and ceases when a connection is restored.
Important Notes:
· File playback will occur automatically if the silence threshold parameters are breached; if
the codec is not connected for any reason file playback will commence. To stop file
playback open the Master panel in the web-GUI, click to select the file playback
connection, then click Disconnect.
· The USB drive can be inserted or removed at any time as long as the codec is not
already playing audio in failover mode. Avoid removing the card while audio is playing or it
will result in poor audio quality. If it is removed accidentally you must reboot the codec to
ensure backup audio will continue to operate reliably.
13.16
SDHC Card File Playback
Playing Audio from the SDHC Card
SDHC card files can be played back using the codec front panel controls.
1.
2.
3.
4.
5.
Press the HOME
button to return to the Home screen.
Select Connect, then select File and press the
button.
Use the navigation buttons to select a file.
Press the
button or the CONNECT
button to play the selected file.
Press the red DISCONNECT
button on the numeric KEYPAD to stop file playback.
Important Notes for SDHC Card File Playback:
· A FAT32 formatted SDHC Card is required (SD cards may be less reliable and are not
recommended).
· Create MP2 or MP3 files using a 32kHz, 44.1kHz or 48kHz sample rate.
· Ensure recordings used are not variable bit rate files.
· SDHC file audio is not sent to codec encoders and cannot be transmitted via an audio
stream to another codec.
· File playback audio is sent directly to the codec outputs and therefore IGC is not
available. When you create your MP2 or MP3 files ensure the audio levels match the
audio reference level of your codec and that peaks average at the correct levels.
· If you create a single file name ensure you add the file extension, e.g. "test.mp3", or the
file will not play back.
· If you create a directory name, all the files within the directory will be played back. We
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recommend you save all audio files as a playlist and link to this if you want them to play
out sequentially. Please note that "M3U" is the playlist file format supported by the
codec.
13.17
Lock or Unlock a Program in the Codec
By default Tieline codecs will attempt to answer a call from another codec if possible. For example, if
a mono program is loaded in the codec and a stereo incoming call is detected, the codec will adjust
and load a compatible answering program.
It is also possible to lock a loaded custom program in a codec to ensure a program with your
preferred settings is not unloaded when a codec dials in. Incoming calls are generally down or up
sampled to accommodate a locked program where possible. Scenarios in which you may wish to
lock a program in the codec include:
1. Locking a dialing program to ensure the codec only dials and never answers an incoming call.
2. Locking an answering program to ensure an incoming codec call is not allowed to:
· Unload the current codec program, e.g. mono or stereo.
· Change the preferred local site settings like the jitter buffer and FEC configuration etc.
Incoming calls to an answering codec with a locked program can still specify different connection
parameters such as algorithm preferences and bit rates via session data.
1.
2.
3.
4.
Press the HOME
button to return to the Home screen.
Select Settings and press
.
Navigate to System and press
.
Navigate to Lock Pgm (lock program) and press
to toggle between Enabled and
Disabled.
5. When program lock is Enabled a warning message confirms program status.
6. When program lock is Disabled a warning message confirms incoming calls may load any
supported factory program.
7. Press the RETURN
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button to exit the warning message.
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Important Note: It is only possible to lock custom programs in a codec. If Lock Program
is enabled and you load a new custom program in the codec, Lock Program remains
enabled and locks the most recently loaded custom program.
13.18
Locking the Front Panel
The codec features a front panel lock feature for tamper-proof operation. This feature is disabled by
default.
There are two levels of panel lock and each requires a user to enter a PIN to access different
features:
1. Admin PIN: Required to change codec connection or configuration settings accessed via
the SETTINGS
button. (Default PIN is: 456789)
2. User PIN: Required to use the codec front panel buttons and dial/hangup a connection
(Default PIN is: 123456)
Enabling the Front Panel Lock Feature
1. Press the SETTINGS
button.
2. Navigate to System and press
.
3. Navigate to Auto Lock and press
to toggle from Disabled to Enabled.
4. Navigate down to the panel Lock Timeout field and press
to enter the desired time-out
period in seconds. Note: The time-out period is the time in seconds before the codec front
panel is relocked after being used.
5. If you want to change the default Admin PIN or User PIN, navigate to each in turn and press
to enter a new PIN.
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51
Connecting to the ToolBox Web-GUI
There are three graphical user interface (GUI) options for configuring Tieline G5 codecs:
1. Java Toolbox Web-GUI: codecs can be fully configured including program creation, dial and
hangup, command and control.
2. HTML5 Toolbox Web-GUI: most codec settings can be configured, dial and hangup existing
programs only, plus codec command and control.
3. HTML5 Toolbox Quick Connect Web-GUI: designed for simple peer-to-peer connections and
non-technical users.
About the Java Toolbox Web-GUI
Codecs can be fully configured using the Java ToolBox Web-GUI and this can be launched using an
IP/LAN connection with the codec. Instructions for using the Java web-GUI are contained in the
application itself from the Help panel and additional information is available at http://
www.tieline.com/support/toolbox. The Tieline Java Toolbox Web-GUI application runs on:
· Internet Explorer 6 or greater on Windows® XP, Windows Vista ® and Windows 7 ®.
· Firefox® 3 or greater on Windows® XP, Windows Vista ® and Windows 7 ®, Solaris™ and
Linux®.
Java Toolbox Web-GUI Prerequisites
1. To use the Java ToolBox Web-GUI you will need to download the latest version of Java™ by
visiting http://www.java.com. The Web-GUI will prompt you to do this if Java is not installed
and you attempt to launch the Java ToolBox Web-GUI.
2. After updating to the latest version of Java you need to refresh your browser.
About the HTML5 Toolbox Web-GUI
The HTML5 Toolbox Web-GUI was developed to improve the user experience with G5 codec
command and control. With some major web-browsers moving away from Java compatibility, Tieline
has delivered an HTML5 configuration option which runs seamlessly on modern browsers.
The HTML5 Toolbox Web-GUI will run on computers and tablets, as well as iOS and Android
smartphones, which expands the range of devices engineers can now use for configuration. In
addition, many users have previously experienced connectivity issues due to regular Java updates
designed to mitigate exposure to security vulnerabilities. By using the HTML5 Toolbox these issues
will be avoided.
Most codec settings can be configured using the HTML5 Toolbox Web-GUI, including:
· Dial, monitor and hangup existing programs only (currently program creation wizard is not
available).
· Extensive command and control of codec settings.
About the HTML5 Toolbox Quick Connect
The HTML5 Toolbox Quick Connect Web-GUI has a reduced feature-set and allows non-technical
users to load existing programs and dial via the Quick Connect panel. Users can dial a simple peerto-peer connection over IP or SIP.
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14.1
Bridge-IT Manual
Opening the Java or HTML5 Web-GUI & Login
1. Attach an Ethernet cable to the ETH1 port on the codec.
2. Press the SETTINGS
button and select Unit to display the IP address programmed into your
codec.
3. Ensure your PC is connected to the same LAN.
4. Open your web browser and type the IP address of your codec into the address bar of your
browser, e.g. http://192.168.0.xxx (the last digits are the private address details unique to your
codec over a private LAN).
5. Refresh the browser and the Web-GUI landing page will display the various command and control
options.
Launching the Java Toolbox Web-GUI
1. Click to launch the Java ToolBox Desktop Application (this is recommended in preference to
launching the Java Toolbox Browser Applet). Note: When you launch for the first time the
application will download and launch the desktop Toolbox application that will allow you to
configure your codec. A desktop short-cut will also be created.
Desktop
Icon
2. When you launch Toolbox an authentication dialog prompts you to enter a password to login. The
first time you log in you can enter the default setting "password" and click the OK button. Tieline
highly recommends you click the hyperlink in the login dialog or visit Changing the Default
Password to change the password. This will provide better network security to maintain reliability
during live broadcasts.
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Important Note: If you update Java software or clear the Java cache on your computer you
will need to repeat the preceding steps. If you have trouble launching the Web-GUI in a
browser, type http://<insert codec IP address>.htm directly in your browser.
Launching the HTML5 Toolbox Web-GUI
1. Click to launch the HTML5 Toolbox Web-GUI.
2. When you launch Toolbox for the first time an authentication dialog prompts you to enter the user
name "admin" and password "password" to login, then click the OK button. Tieline highly
recommends you change the password (see Changing the Default Password). This will provide
better network security to maintain reliability during live broadcasts.
Launching the HTML5 Toolbox Quick Connect
1. Click to launch the HTML5 Toolbox Quick Connect Web-GUI.
2. When you launch Toolbox for the first time an authentication dialog prompts you to enter the user
name "admin" and password "password" to login, then click the OK button. Tieline highly
recommends you change the password (see Changing the Default Password). This will provide
better network security to maintain reliability during live broadcasts.
Using the Web-GUI over the Internet
If your codec is connected over the internet via a public static IP address it is possible to connect
and configure it from any PC which is also connected to the internet.
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LAN Troubleshooting
PC LAN Settings
Check the LAN settings on your PC if it is connected to a LAN and is having trouble opening the
Toolbox Web-GUI in a web-browser.
1.
2.
3.
4.
Open Internet Explorer.
Click Tools > Internet Options > Connections.
Click the LAN settings button.
If the PC is using a proxy server over the LAN you may need to select the Bypass
proxy server for local addresses option box.
5. If you still can't connect, click the Advanced button in the LAN Settings dialog and ask
your IT administrator to assist you with entering the IP address of the codec into the
Exceptions pane of the Proxy Settings dialog.
Port Selection
By default port 80 is used by your PC to communicate with the codec and launch the web-GUI.
If port 80 cannot be used across your network for some reason, type the IP address of your
codec into your browser with a full colon and the port number 8080.
E.g. 192.168.0.176:8080
It is also possible to specify a different port for connecting the Toolbox web-GUI to your codec.
button on the codec to return to the Home screen.
Press the HOME
Use the navigation buttons to select Settings and press the
button.
Use the navigation button to navigate down to WebGUI and press the
button.
Select Alt. Port and press
.
Use the KEYPAD to enter a new port number and press the
button to save the new
setting.
6. Type the IP address of your codec into your browser with a full colon and then the new
port number.
1.
2.
3.
4.
5.
Important Note: Any new port specified must be within the range 2000 to 65535 inclusive.
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14.2
55
Changing the Default Password
The default password for the Toolbox Web-GUI is password. Enter this in the authentication dialog
to use the Web-GUI initially and then Tieline highly recommends changing the default password to
protect your codec from being tampered with during live broadcasts. Note: In the HTML5 Web-GUI
authentication dialog it is necessary to enter admin as the User Name. This field is not visible in
the Java Web-GUI authentication dialog.
Toolbox Java Web-GUI Login Dialog
Toolbox HTML5 Web-GUI Login Dialog on a Merlin Codec
Caution: Codecs connected to the internet can be accessed by anyone with knowledge of
the codec's public IP address. Setting a strong password protects your equipment from
being tampered with during live broadcasts.
Creating a New Password
The authentication login password can be changed at any time using the codec keypad and LCD
screen. Note that passwords are case sensitive:
1.
2.
3.
4.
Press the SETTINGS
button.
Use the navigation button to select WebGUI and press the
Select Password and press
.
Use the KEYPAD to enter a new password and press the
(Note: there is no character limit for passwords).
button.
button to save the new setting
If you forget the password for the Toolbox web-GUI then you can always press the SETTINGS
button on the codec and navigate to WebGUI to view the current password and change it if
required.
Important Note: The Username in the codec menu is permanently set to admin and
cannot be changed; only the Password can be changed.
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14.3
Bridge-IT Manual
Installing USB Drivers
USB drivers need to be installed on your PC in order to connect successfully to the codec using
the USB port. To install drivers:
1. Download the zipped USB driver ".inf" file from the Bridge-IT firmware download pages at
http://www.tieline.com/Support/Latest-Firmware.
2. Unzip the file and save it to your PC.
3. Connect a USB cable between your PC and the Bridge-IT USB port on the rear panel of the
codec.
4. The PC should detect that a new device has been attached and launch the Found New
Hardware Wizard.
5. Select Install from a list or specific location (Advanced) and click Next.
6. Select the folder in which you have saved the downloaded ".inf" file and click Next.
7. When the drivers have been installed click the Finish button.
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Launching the GUI over USB
1. Install USB drivers into your PC.
2. Connect a USB cable between your PC and the USB port on the rear panel of the codec.
3. From the codec Home screen navigate to Settings > Unit > USB0 and press the
button to
display the USB address details configured into your codec.
4. Open your web browser and type the USB address of your codec into the address bar of your
browser, i.e. http://169.254.x.y (the last 2 blocks of digits are the USB address details unique to
your codec).
5. Refresh the browser and the web-GUI should launch automatically.
6. When you open the ToolBox web-GUI an authentication dialog prompts you to enter a password
to login. The first time you log in enter the default setting "password" and click the OK button.
(See Changing the Default Password for instructions on changing the default password to
increase your network security).
15
Java Toolbox Web-GUI Introduction
Important Note: The default password for the web-GUI is password. This has to be
entered to initially use the Toolbox web-GUI. For additional security Tieline highly
recommends changing the authentication login password using the codec keypad and
screen (see Changing the Default Password for more info).
The following sections provide an overview of the different programming panels available within the
codec's Toolbox web-GUI. Navigate with the mouse pointer to a symbol at the top of the web-GUI
screen and click to open the panel selected. When a panel is opened in the web-GUI, the text
below the symbol at the top of the screen is highlighted (see Master in the following image).
Web-GUI Symbols for Opening Panels
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Master Panel to Load Programs and Connect Audio Streams
1
Feature
Input/Output PPMs
Description
4 PPM meters to display audio levels for inputs and outputs
2
Connections
3
Locked program
Provides a summary of audio stream and connection details audio
streams
Symbol indicates the currently loaded program is locked
4
Maximize/Minimize
5
Close button
6
Connect button
7
Load button
8
Programs list
9
Disconnect button
Click to maximize a panel to view it in full-screen mode, or click to
minimize back to the default panel size
Click to close the Master panel
Click Connect to connect all audio streams configured within the
currently selected program in the Programs list; this button also
loads the program currently selected in the Programs list
Click to Load the codec with the program currently selected in the
Programs list
Lists all configured programs which have been added into the
codec. Click to select a program before loading or connecting
Click to disconnect the currently selected audio stream or a
specific connection. Note: this button becomes a Connect or
Unload button when all audio streams are disconnected.
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Programs Panel for Connection Configuration
1
Feature
Programs List
Description
Displays all programs in the codec
2
New Program button
Click to add a new program.
3
Program Name
The name of the currently selected program in the panel.
4
Edit Name
Click to edit the name of the currently selected program.
5
Audio Stream overview
6
Delete Program
7
Maximize/Minimize
8
Close button
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Click the blue arrows
to expand audio stream and
connection information; click the Edit symbol
to adjust
program settings. This panel displays the program wizard
when creating a new program.
Click to delete the currently selected program (Note:
Ensure the program is not loaded or the delete function will
not work).
Click to maximize a panel to view it in full-screen mode, or
click to minimize back to the default panel size
Click to close the Connect panel.
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Inputs Panel for Adjusting Input Levels
1
Feature
Channel ON/OFF Buttons
Description
Click to turn each channel ON or OFF
2
Reference levels menu
3
Lock Button
4
View local
Click the drop-down arrow to select the codec input
reference level (default setting Auto)
Click to lock all Input panel settings (greys out when
locked)
Click to view local codec inputs (default)
5
Settings button
Click to adjust input Name, Type, IGC and Ganging
6
Maximize/Minimize
7
Close button
Click to maximize a panel to view it in full-screen mode, or
click to minimize back to the default panel size
Click to close the panel
8
Input Sliders/Faders
Input gain control sliders/faders
9
Analog/AES3 Indication
Indicates whether the codec input is configured for analog
or digital audio sources
Input PPM meter
10 Input PPM meter
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Statistics Panel for Monitoring Connection Stability
1
2
Feature
Headings
Connection Statistics
3
Maximize/Minimize
4
Close button
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Description
Headings for the various packet arrival statistics available
Right-click to view audio stream bit-rate and jitter buffer
statistics
Click to maximize a panel to view it in full-screen mode, or
click to minimize back to the default panel size
Click to close the panel
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Rules Panel for Creating Relay Activation Rules
5
Rule
Connect/Disconnect a program
by toggling a relay input
Connect when an input is
switched ON; Disconnect when
another input is switched ON
Synchronise a local relay input
with a remote relay output
Toggle a relay based on
connection status
Maximize/Minimize
6
Close button
1
2
3
4
Description
Click to program Connection and Disconnection by
toggling an input
Click to program Connection and Disconnection after
different relay inputs are switched ON
Click to program a local relay input to Synchronise with
the state of a remote relay output
Click to program a relay to toggle based on connection
status
Click to maximize a panel to view it in full-screen mode,
or click to minimize back to the default panel size
Click to close the Rules panel
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Settings Panel
Feature
1 Network tab
Description
Click to edit or view codec network configuration settings
2 Options tab
3 Audio tab
Click to configure RS232 and QoS data settings, lock a loaded
user Program and Adjust Session Port settings
Click to configure the AES Output Clock sample rate
4 SIP tab
Click to edit or view SIP configuration settings
5 Firmware tab
Click to view software versions and perform an upgrade
6 Licensing tab
Click to select a license file and install it into the codec
7 Reset/Restore tab
Click to reset codec default settings and perform backup/restore of
codec programs and settings
Activate to specify DNS addresses and domains to search.
8 DNS Pane
9 Maximize/Minimize
10 Close button
Click to maximize a panel to view it in full-screen mode, or click to
minimize back to the default panel size
Click to close the panel
11 Network Interface
Select a network interface for configuration options
12 Network
Identifier
13 IPv6 details
Interface Control and streaming configuration options for each network
interface, e.g. Ethernet Port or LAN.
IPv6 addressing details and configuration
14 MAC Address
Device MAC address
15 IPv4 details
IPv4 addressing details and configuration
16 Save Settings button
Saves all configuration settings
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Help Panel for Product Support
Feature
1 About
2 Resources
3 Support Logs
4 Event Logs
5 Maximize/Minimize
6 Close button
Description
Details about the Toolbox web-GUI and codec firmware version, as
well as the codec type and codec serial number
Links to open the user manual in a new browser, or view support
information
Click to download diagnostic information that can be sent to Tieline
support
Click to download user-viewable event logs
Click to maximize a panel to view it in full-screen mode, or click to
minimize back to the default panel size
Click to close the Help panel
Language Selection
The Toolbox web-GUI offers language support for several languages.
1. Click on the language drop-down menu arrow in the top right-hand corner of the web-GUI page.
2. Select your language of choice.
3. Click to refresh your web-browser and download the user manual for the language selected.
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Java Toolbox Web-GUI Codec Configuration
The following sections describe how to configure your codec using the Java Toolbox web-GUI.
16.1
Configuring IP Settings
Click the Settings
symbol to open the Settings panel and click the Network button to view
Ethernet and VLAN interface settings in the web-GUI.
Important Note: For assistance with configuration of IPv4 or IPv6 network connections
contact your IT Administrator.
IPv4 versus IPv6
An IP address is a unique address to identify a device on a TCP/IP network. Your codec uses dual
IP protocol stacks to allow your codec to work on both IPv4 and IPv6 networks. Your Tieline codec
supports both DHCP (default) IP addressing and static IP addresses for dialing IPv4 connection
endpoints.
If you want to dial a codec with a public IP address you simply dial the IP address to connect. If you
want to dial a codec with a private IP address you need to perform network address translation
(NAT). NAT allows a single device, such as a broadband router, to act as an agent between the
public internet and a local private LAN. Usually this will be set up at the studio end so you can dial
into the studio from the remote codec.
Support for IPv6 connections allows you to use IPv6 infrastructure to connect to other codecs
globally.
Configuring Ethernet Ports and VLANs
The codec features a physical Ethernet port and up to four additional VLAN interfaces.
VLAN interfaces have features similar to the Ethernet port; however, your network administrator will
need to configure VLAN support throughout your network for them to be supported in your codec.
As an example, VLANs can be used to separate codec Control and Streaming functions if required.
Ethernet and VLAN interfaces can be configured for:
· Control Only: codec control and command only from the Ethernet port.
· Control and Streaming: stream audio and control and command the codec via the Ethernet
port.
· Streaming audio: stream audio only from an Ethernet port (VLANs only).
· Nothing: Disable the Ethernet port from streaming audio and codec command and control
(VLANs only).
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The name entered into the right-hand text box, e.g. Primary or Secondary, is an interface identifier
used when configuring new programs via the Programs panel.
IPv4 Address Configuration
The codec is capable of automatic DHCP address assignment, or manually configured static IPv4
address configuration via the drop-down Configure IPv4 menu. If you want to ignore IPv4 settings
select Off.
DHCP IP addresses are automatically assigned and can change each time you connect to your
Internet Service Provider or to your own local area network (LAN). By default the codec is configured
for DHCP-assigned IP addresses.
Static IP addresses are fixed addresses that are recommended for studio installations, so that IP
address dialing remains the same over time for incoming codec connections.
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Click Save Settings to store all configuration settings.
Note: The Subnet Mask is used by the TCP/IP protocol to determine whether a host is on
the local subnet or on a remote network. The default Gateway is the router linking the
codec's subnet to other networks. See your IT administrator for more details.
IPv6 Address Configuration
An IPv6 address is represented by 8 groups of 16-bit hexadecimal values separated by colons (:).
The drop-down Configure IPv6 menu provides three address configuration options:
1. Auto: An address is automatically assigned to the codec when you connect the codec to an
IPv6 router. This process is similar to how an IPv4 DHCP address is assigned.
2. Manual: Select to enter static IPv6 address details.
3. Off: Select to ignore IPv6 address details.
Important Note: Select Off in the drop-down Configure IPv6 menu if you are not using
IPv6 to connect to another device. This ensures your codec will attempt to connect using
IPv4 at all times.
Types of IPv6 Addresses
There are two types of addresses displayed in the IPv6 section:
1. IPv6 address (normally global): A router-allocated IP address with 'global' visibility,
details of which are displayed in the Address, Prefix and Gateway text boxes.
2. Link Local Address: A local address which can only be used to connect to another
device directly over a LAN. This address is allocated by the codec internally based on
MAC address details.
Auto Address Assignment
By default the codec is programmed for connecting to an IPv6 router which automatically
allocates IPv6 address details, as displayed in the following example.
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Manual IPv6 Address Assignment
To configure IPv6 address details into the codec manually, select Manual and enter details into
the Address, Prefix and Gateway text boxes.
Click Save Settings to store all configuration settings.
Specifying DNS Settings
It is possible to specify Domain Name Server (DNS) settings to allow easy look up of codecs within
the specified DNS Addresses or Domains.
The codec can be configured to tag IP data packets sent across a network by entering a value into
the Differentiated Services Code Point (DSCP) field within the header of data packets transmitted
over the network.
Configuring QoS
1. Open the web-GUI and click the Settings
symbol at the top of the screen to display the
Settings panel.
2. Click the Options button at the top of the Settings panel.
3. Click in the QoS text box and enter the new value.
4. Click the Save Settings button at the bottom of the panel to save the new setting.
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Important Note: Check with your IT administrator before changing this setting. By default
the codec is programmed for Assured Forwarding and more details about DSCP are
available on Wikipedia at http://en.wikipedia.org/wiki/Dscp. For more information on
configuring QoS see Configuring QoS for Broadcasts in this manual.
16.2
Configuring Input/Output Settings
Click the Inputs button
to view input controls available within the Toolbox web-GUI.
Important Note: 15 volt phantom power can only be supplied on analog input 1; this is
disabled by default.
Configuring Input Channel Settings
Renaming Input Channels:
1. Click the Input Settings
symbol on the input channel you want to rename.
2. Select Name and click in the text box to edit or enter a new name.
3. Click Change Name to confirm the name change.
Selecting Analog and Digital Audio Sources:
Codec inputs are configured for analog high-gain mic level audio sources by default.
1. Click the Input Settings
symbol.
2. Select Type and click to select either Analog or AES3.
3. When you select AES3, the display changes to reflect 100% input levels; slider and
input on/off controls are locked on.
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Important Note: Input levels can only be adjusted on analog inputs. See Configuring AES3
Audio for more information about the digital inputs and outputs.
Ganging Channels:
Ganging is useful because it allows you to adjust the audio level of both inputs simultaneously.
1. Click the Input Settings
symbol on either channel.
2. Select Gang and click to either gang or ungang channels.
3. When ganged, the two channel sliders move in sync with each other when dragged
using a mouse-pointer.
4. Click the Link symbol to temporarily disable the ganging function and fine-tune channel
audio levels. Click the Link symbol again to resume ganging.
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Setting Analog Audio Levels
Audio levels on the Input panel should be set to ensure audio peaks average at the first yellow
indications on the PPM meters, which represents +4dBu. These levels should also be checked
against the Input PPM Meters on the Master panel. When the codec is not connected the Master
panel will display the input audio going to the outputs (default setting).
Other Input Controls
Adjust the IGC (Intelligent Gain Control) input
settings to Auto, Fixed or Off as required.
Channel 1 Phantom Power
Input 1 can be configured to supply 15 volt phantom power if required; this is disabled by default.
1. Click the Input Settings
symbol on channel 1.
2. Click to select the Phantom check box.
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Locking Input Settings
1. Click the Lock symbol to lock all Input panel settings.
2. When locked, the Input panel is greyed out and the lock symbol appears in the bottom-left
corner. Note: the lock function does not affect the codec front panel controls.
AES3 Output Sample Rate Configuration
The AES3 output sample rate can be configured using the Toolbox web-GUI.
1. Click the Settings
symbol at the top of the screen to open the Settings panel.
2. Click the Audio tab and use the drop-down menu to select your preferred AES Output Clock
setting, then click Save Settings.
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Configuring Mono or Stereo Peer-to-Peer Programs
The Programs panel incorporates a wizard to configure a new program and all audio stream
settings. Before you configure a new codec program consider if:
· You want your codec to be capable of dialing and answering, dialing only or answering only.
· A backup connection is required.
This section contains instructions for:
1. Configuring Peer-to-Peer Programs: Dialing and Auto reconnect
2. Configuring Answering Connections
For more information about programs and audio streams within programs see the section titled
About Program Dialing. Note: The following connection setup instructions will display how to
configure a dial and answer program, with a backup connection. If you want the codec to either dial
or answer only, select the option and the wizard will automatically display relevant screens to allow
you to configure the codec correctly.
Configuring Peer-to-Peer Programs: Dialing
Important Notes: Before you start program configuration please note:
· You cannot edit a program when it is currently loaded in the codec.
· You can lock a loaded custom program in a codec to ensure the currently loaded
program type cannot be unloaded by a codec dialing in with a different program type.
· Some drop-down menus and settings may be greyed out intentionally depending on
features available.
· It is possible to save a program at several points throughout the program wizard and
use default settings to save configuration time.
· To learn more about programs see the section titled About Program Dialing.
1. Open the web-GUI and click the Programs
Programs panel.
symbol at the top of the screen to display the
2. Click the New Program button to open the wizard and:
· Click in the text box to name the new program.
· Select Mono/Stereo Peer-to-Peer, or if you want to use an existing program as a template,
select this option. Then click Next.
Important Notes: When you decide to use an existing program as a template, the new
program inherits all the settings of the template program and you can adjust these settings
as required by continuing through the program wizard.
3. Enter a name for the Audio Stream and configure the codec to dial, answer or dial and answer.
Then click Next.
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4. This audio stream connection in the wizard will allow the codec to dial. Enter the name of the
connection in the text box, then click Next.
5. Follow the instructions on the right-hand side of the panel to configure the transport settings for
the connection, then click Next.
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Important Notes:
· If you select Sessionless as the Session Protocol select UDP/IP +RTP for RFCcompliant IP streaming.
· See RS232 Data Configuration for detailed information on RS232 data and see "Enabling
Relays and RS232 Data" for more information on relay operations.
6. Configure the IP address, ports, and then specify which streaming interface is used to dial this
connection, e.g. Primary (LAN / ETHERNET port) or VLAN if configured. VLANs can be used to
configure separate control and streaming interfaces if required. Note: By default Any will select
Primary as the streaming interface.
Important Notes: The Remote Audio Port is the codec port at the remote end of the link
to which you are sending audio. The Local Audio Port is the port used by the local codec
to receive audio from the remote codec. When Tieline Codecs is the Session Protocol
selected (using Tieline sesssion data), the Local Audio Port is automatically configured as
UDP audio port 9000 by default. This is the default audio port used by all Tieline IP codecs
for the first audio stream connection. Click to deselect the Automatic check-box to change
this setting.
When you select Sessionless as the Session Protocol the Session Port is not configurable and
you can manually configure the Remote Audio Port and Local Audio Port.
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7. Click Save Program to save the program with the default algorithm, jitter and FEC settings
which are physically entered in the codec. Alternatively, click Next to specify individual
algorithm, jitter buffer and FEC settings for this audio stream (recommended).
8. Click the drop-down arrows on the right-hand side of each text box to adjust the Encoding,
Sample rate and Bit rate options, then click Next.
9. For IP connections click to configure:
· Auto Jitter Adapt and the preferred auto jitter setting using the drop-down arrow for Buffer
priority, or
· Fixed Buffer Level and enter the Jitter Depth (5000ms maximum).
· Local and Remote FEC settings if required.
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10.Click Save Program or click Next to Enable Auto Reconnect.
11.Click Save Program to complete configuration of a dialing only program, or click Next to
configure an answering connection
Configuring Answering Connections
The codec is capable of being configured to use specific settings when answering calls. This useful
to configure the answering codec with appropriate jitter buffer and FEC settings to suit the prevailing
conditions at the codec's location:
1. Enter a name for the answering connection and click Next.
2. Follow the instructions on the right-hand side of the panel to configure the Transport,
Session Protocol and Audio Port settings for the connection, then click Next.
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3. Click to configure jitter buffer and FEC settings:
· Auto Jitter Adapt and the preferred auto jitter setting using the drop-down arrow for
Buffer priority, or
· Fixed Buffer Level and enter the Jitter Depth, which must be between 12ms and
5000ms depending on the algorithm you select.
· Local and Remote FEC settings if required.
4. After configuring these settings there are 3 options:
i. If you want to create another answering connection, select the check-box for Create
another answering connection at the bottom of the panel and continue through the
wizard.
ii. Click Next and select Enable File Playback on silence detection.
iii. Click Save Program to save the program at this point.
When you save the program it will be confirmed by the following message.
5. Click Finish and the newly created program will be displayed in the left pane within the
Programs panel and in the Master panel. Select and connect audio streams in a program
using the Master panel, or dial the program manually using the codec front panel.
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Configuring Multi-Unicast Dialing Programs
16.4
The codec can transmit a mono or stereo multi-unicast audio stream to a maximum of 6 endpoints
in total. The first connection in a multi-unicast program is capable of bidirectional audio. Multiunicast connections can only be created using the ToolBox web-GUI and require a software license
which supports multi-unicasting.
Prerequisites:
·
·
·
·
Bridge-IT firmware v.1.01.00 r4219 or higher.
G3 codec firmware v.1.6.56 or higher (if connecting to a G3 codec).
ToolBox web-GUI v.1.2.2.3 or higher.
A multi-unicast license installed in the dialing codec.
Important Notes: Before you start program configuration please note:
· You cannot edit a program when it is currently loaded in the codec.
· You can lock a loaded custom program in a codec to ensure the currently loaded
program cannot be unloaded by a codec dialing in with a different program type.
· If you are dialing to both G3 and Bridge-IT, Merlin or Genie codecs, by default the
Audio Reference Level will be configured for the compatibility of the codec dialed first.
I.e. if you dial a G3 codec first then the G3 Audio Reference Level will be configured for
all connections.
· Connections are dial only for multi-unicast programs.
· Some drop-down menus and settings may be greyed out intentionally depending on
features available.
· It is possible to save a program at several points throughout the program wizard and
use default settings to save configuration time. The first connection in each multiunicast stream determines default settings (e.g. algorithm, sample rate, bit rate), for
all subsequent connections in that stream, except for dialing settings.
· Select any algorithm for multi-unicast connections except aptX Enhanced and PCM.
· Bidirectional audio is only available on the first connection dialed.
· Forward Error Correction (FEC) is not available on multi-unicast connections.
· Renegotiation of connection bit rates is not possible when connected.
· Ensure you have sufficient connection bandwidth at the local codec to support all the
connections to which you are connecting.
· To learn more about programs see the section titled About Program Dialing.
Creating a Multi-Unicast Program
1. Open the web-GUI and click the Programs
Programs panel.
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2. Click the New Program symbol to open the wizard and then:
· Click in the text box to name the new program.
· Select Multi-unicast, or if you want to use an existing program as a template, select this
option. Then click Next. Note: In this example we are creating a Multi-unicast program to dial
2 end points.
Important Note: When you decide to use an existing program as a template, the new
program inherits all the settings of the template program and you can adjust these settings
as required by continuing through the program wizard.
3. Enter a name for the Audio Stream and then click Next.
4. This audio stream connection in the wizard will allow the codec to dial. Enter the name of the
connection in the text box, then click Next.
5. Follow the instructions on the right-hand side of the panel to configure the transport settings for
the connection, then click Next. Note: only the first connection dialed in a multi-unicast program
can encode and decode audio. All other connections are unidirectional and encode only.
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Important Note: Auxiliary IP data connections are not possible between the codec and G3
Commander and i-Mix codecs over multi-unicast connections. See RS232 Data
Configuration for detailed information on RS232 data and see "Enabling Relays and RS232
Data" for more information on relay operations.
6. Configure the IP address, ports, and then specify which streaming interface is used to dial this
connection, e.g. Primary (LAN / ETHERNET port and default setting) or VLAN if configured.
Note: By default Any will select Primary.
Important Notes: The Remote Audio Port is the codec port at the remote end of the link
to which you are sending audio. The Local Audio Port is the port used by the local codec
to receive audio from the remote codec. When Tieline Codecs is the Session Protocol
selected (using Tieline sesssion data), the Local Audio Port is automatically configured as
UDP audio port 9000 by default. This is the default audio port used by all Tieline IP codecs
for the first audio stream connection. Click to deselect the Automatic check-box to change
this setting.
Click Save Program to save the program with the default algorithm, jitter and FEC settings which
are physically entered in the codec. Alternatively, click Next to specify individual algorithm, jitter
buffer, FEC and auto reconnect settings for this audio stream (recommended).
7. Click the drop-down arrows on the right-hand side of each text box to adjust the Encoding,
Sample rate or Bit rate options.
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Click to configure:
· Auto Jitter Adapt and the preferred auto jitter setting using the drop-down arrow for Buffer
priority, or
· Fixed Buffer Level and enter the Jitter Depth (5000ms maximum).
· Local and Remote FEC settings if required.
8. Select the Enable Auto Reconnect check-box if you want to enable this feature. This is enabled
by default.
9. Select the Create another dialing connection check-box to configure a new connection for an
additional endpoint, then click Next.
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10.Enter the name of the second connection in the text box, then click Next.
11.Continue through the program wizard and configure all multi-unicast connections in a similar
manner.
12.After configuring all connections there are 2 options:
i. Click Next and select Enable File Playback on silence detection.
ii. Click Save Program to save the program.
When you save the program it will be confirmed by the following message.
13.Click Finish and the newly created program will be displayed in the left pane within the
Programs panel and in the Master panel. Select and connect audio streams in a program
using the Master panel, or dial the program manually using the codec front panel.
Monitoring Multi-Unicast Programs
The Master panel in the ToolBox web-GUI is used to monitor connection details and view PPMs,
as displayed in the following image. Click the blue arrow in the Program/Cxn column to expand
and minimize connection details (as displayed below).
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Configuring Multicast Server Programs
How Multicasting Works
Multicast transmissions are sent using a dedicated IP multicast address that looks similar to a
regular IP address and multicast subscribers request transmissions from this address. This unique
address allows multicast routers to identify multicast requests from a group of codecs interested in
a particular transmission and packets are replicated depending on demand. This can create large
demands on network bandwidth if the multicast group is significant in size.
As a result, only small sections of the internet are multicast enabled and many internet service
providers (ISPs) block multicast traffic over wide area networks. This restricts most multicast
broadcasts to private local area networks. Some ISPs provide quality of service (QoS) priority to
multicast streams for an increased service charge. You need to check with your ISP to find out what
multicast services, if any, are available over WANs.
Important Notes:
· When a connection is dialed Tieline codecs normally use session data to configure
settings like the algorithm, connection bit rate and sample rate etc. Multicast connections
are sessionless and do not use Tieline session data. As a result, it is imperative that all
codecs are configured with the same connection settings prior to connecting, or they will
not be able to join multicast streaming sessions.
· Automatic or fixed jitter buffer settings can be adjusted on individual client codecs as
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required. There is no jitter buffer setting on the server codec because it never receives
audio packets.
Prerequisites:
Bridge-IT firmware v.1.01.00 r4219 or higher.
G3 codec firmware v.1.6.56 or higher (if connecting to a G3 codec).
ToolBox web-GUI v.1.2.2.3 or higher.
A multi-unicast license installed in the dialing codec (Note: the Multi-Unicast license includes
multicast server capability).
· Use firmware higher than 2.8.xx in the Bridge-IT, Genie and Merlin families of codecs to transmit
auxiliary data.
·
·
·
·
Multicast Server versus Multicast Client Programs
Two different types of multicast programs need to be created when multicasting:
· A multicast server program is used by the broadcasting codec to send multicast IP packets
to multicast routers on a network.
· A multicast client program is used by codecs to receive multicast IP audio packets.
A multicast server codec sends audio packets only and a multicast client codec receives audio
packets only. Codecs using the client program request multicast packets (sent from the server
codec), which are distributed by multicast routers.
Creating Multicast Server Programs
Important Notes: Before you start program configuration please note:
· Ensure all connection related settings like the port, algorithm, bit rate (etc) match on
both multicast server and client programs or they will not connect successfully.
· You cannot edit a program when it is currently loaded in the codec.
· You can lock a loaded custom program in a codec to ensure the currently loaded
program cannot be unloaded by a codec dialing in with a different program type.
· Some drop-down menus and settings may be greyed out intentionally depending on
features available.
· It is possible to save a program at several points throughout the program wizard and
use default settings to save configuration time.
· To learn more about programs see the section titled "About Program Dialing".
· Always dial the multicast server codec connection first before connecting multicast
client codecs.
· Multicast client codecs will display return link quality (LQ) only. The Return reading
represents the audio being downloaded from the network locally. Multicast server
codecs do not display LQ readings.
· The default UDP audio port setting is 9000 for a multicast and the client and server port
settings must match.
· Forward Error Correction (FEC) is not available for multicast connections.
· It is not possible to connect to a G3 codec and receive multicast IP audio streams.
· To copy multicast client programs onto multiple codecs see Backup and Restore
Configuration Files.
1. Open the web-GUI and click the Programs
Programs panel.
symbol at the top of the screen to display the
2. Click the New Program button to open the wizard and:
· Click in the text box to name the new program.
· Select Multicast Server or if you want to use an existing program as a template, select
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this option. Then click Next.
Important Notes: When you decide to use an existing program as a template, the new
program inherits all the settings of the template program and you can adjust these settings
as required by continuing through the program wizard.
3. Enter a name for the Audio Stream, then click Next.
4. This audio stream connection in the wizard will allow the codec to dial. Enter the name of the
connection in the text box, then click Next.
5. Follow the instructions on the right-hand side of the panel to configure the transport settings for
the connection, then click Next. Note: select UDP/IP +RTP for RFC-compliant streaming. If
auxiliary data is enabled the audio stream will not be RFC-compliant.
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Important Notes:
· The encode and decode direction is configured automatically for Encode Only (server
program) or Decode Only (client program). This setting is configured when you select
either Multicast Server or Multicast Client when you first create the program in the
wizard.
· The TTL value you need to use is dependent upon your network infrastructure. Please
consult your network administrator if you are unsure about how to configure this setting.
· Use firmware higher than 2.8.xx in the Bridge-IT, Genie and Merlin families of codecs to
enable auxiliary data.
6. Configure the multicast IP address and Remote Audio Port (the same multicast address and
port must be used for both the server and client programs), then specify which IP streaming
interface is used to dial this connection, e.g. Primary (LAN / ETHERNET port and default setting)
or VLAN if configured. Note: By default Any will select Primary.
7. Click the drop-down arrows on the right-hand side of each text box to select the Encoding,
Sample rate , Bit rate or Sample size options. Click Next to continue.
8. Click to select Enable Auto Reconnect to enable automatic reconnection, then click Save
Program.
Important Note: There is no jitter buffer setting on the server codec because it never
receives audio packets.
9. Configure multicast server and multicast client programs and load all codecs with the appropriate
program. Dial the multicast server program connection first and then connect multicast client
codec programs to begin receiving multicast audio packets. Select and connect audio streams in
a program using the Master panel, or dial the program manually using the codec front panel.
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Configuring Multicast Client Programs
Important Notes: Before you commence program configuration please note:
· Ensure all connection related settings like the port, algorithm, bit rate (etc) match on
both multicast server and client programs or they will not connect successfully.
· You cannot edit a program when it is currently loaded in the codec.
· You can lock a loaded custom program in a codec to ensure the currently loaded
program cannot be unloaded by a codec dialing in with a different program type.
· Some drop-down menus and settings may be greyed out intentionally depending on
features available.
· It is possible to save a program at several points throughout the program wizard and
use default settings to save configuration time.
· To learn more about programs see the section titled About Program Dialing.
· Always dial the multicast server codec connection first before connecting multicast
client codecs.
· Multicast client codecs will display return link quality (LQ) only. The Return reading
represents the audio being downloaded from the network locally. Multicast server
codecs do not display LQ readings.
· The default UDP audio port setting is 9000 and the client and server port settings must
match.
· Forward Error Correction (FEC) is not available for multicast connections.
· Use firmware higher than 2.8.xx in the Bridge-IT, Genie and Merlin families of codecs to
enable auxiliary data.
· It is not possible to connect to a G3 codec and receive multicast IP audio streams.
· To copy multicast client programs onto multiple codecs see Save and Restore
Configuration Files.
Configuring Multicast Client Programs
1. Open the web-GUI and click the Programs
Programs panel.
symbol at the top of the screen to display the
2. Click the New Program button to open the wizard and:
· Click in the text box to name the new program.
· Select Multicast Client to configure a multicast program, or if you want to use an
existing program as a template, select this option. Then click Next.
Important Notes: When you decide to use an existing program as a template, the new
program inherits all the settings of the template program and you can adjust these settings
as required by continuing through the program wizard.
3. Enter a name for the Audio Stream, then click Next.
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4. This audio stream connection in the wizard will allow the codec to dial. Enter the name of the
connection in the text box, then click Next.
5. Follow the instructions on the right-hand side of the panel to configure the transport settings for
the connection, then click Next. Note: select UDP/IP +RTP for RFC compliant streaming. If
auxiliary data is enabled the audio stream will not be RFC-compliant.
Important Notes:
· The encode and decode direction is configured automatically for Encode Only (server
program) or Decode Only (client program). This setting is configured when you select
either Multicast Server or Multicast Client when you first create the program in the
wizard.
· Use firmware higher than 2.8.xx in the Bridge-IT, Genie and Merlin families of codecs to
enable auxiliary data.
6. Configure the multicast IP address and audio port (the same multicast address and port must be
used for both the server and client programs), then specify which IP streaming interface is used to
dial this connection, e.g. Primary (LAN / ETHERNET port and default setting) or VLAN if
configured. Note: By default Any will select Primary.
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7. Click the drop-down arrows on the right-hand side of each text box to select the Encoding,
Sample rate , Bit rate or Sample size options. Click Next to continue.
8. Click to configure:
· Auto Jitter Adapt and the preferred auto jitter setting using the drop-down arrow for Buffer
Priority, or
· Fixed Buffer Level and enter the Jitter Depth (5000ms maximum).
Important Notes: Automatic or fixed jitter buffer settings can be adjusted on individual
client codecs as required. There is no jitter buffer setting on the server codec because it
never receives audio packets.
9. Click to select Enable Auto Reconnect to enable automatic reconnection, then either:
i. Click Next to select Enable File Playback on silence detection, or
ii. Click Save Program to save the program at this point.
When you save the program it will be confirmed by the following message.
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10.Click Finish and the newly created program will be displayed in the left pane within the
Programs panel and in the Master panel.
11.Configure multicast server and multicast client programs and load all codecs with the appropriate
program. Select and connect audio streams in a program using the Master panel, or dial the
program manually using the codec front panel. Dial the multicast server program connection first
and then connect multicast client codec programs to begin receiving multicast audio packets.
16.7
Configuring SIP Settings
The codec is fully EBU N/ACIP Tech 3326 compliant when connecting using SIP (Session Initiation
Protocol) to other brands of IP codecs.
About SIP
SIP provides superior interoperability between different brands of codecs due to its standardized
protocols for connecting devices and is intended to be used when connecting Tieline codecs to nonTieline devices. Devices primarily use SIP to dial another device’s SIP address and find its location
with a minimum of fuss. This task is usually performed by SIP servers, which communicate between
SIP-compliant devices to set up a call.
When connecting two devices, SDP performs similar tasks to Tieline’s proprietary session data,
which is used to configure all non-SIP IP connections. There are two very distinct parts to a call
when dialing over IP. The initial stage is the call setup stage and this is what SIP is used for. The
second stage is when data transference occurs and this is left to the other protocols used by a
device (i.e. using UDP to send audio data).
All the mandatory EBU N/ACIP 3326 algorithms are supported (G.711, G.722, MPEG-1 Layer 2 and
16 bit PCM), as well as optional algorithms including LC- AAC, HE-AAC and aptX Enhanced. The
default algorithm selected when connecting using SIP is G.711.
Important Notes:
· Each codec should be registered to a different SIP server account to avoid connection
conflicts.
· SIP dialing is only supported over peer-to-peer connections, not multi-unicast
connections.
· Tieline G3 codecs do not support AAC and will default to MPEG Layer 2 if a Bridge-IT
codec configured for AAC attempts to connect.
· Some Telcos and ISPs may block SIP traffic over UDP port 5060.
· SIP account registration can only be configured via the Ethernet port and not a VLAN.
· When connecting to a Tieline G3 codec using SIP you need to manually select the G3
audio reference level in the codec. To do this select SETTINGS
> Audio > Ref
Level > Tieline G3. In addition, select the following on the G3 codec prior to dialing.
 Select either a mono or stereo profile
 Select [Menu] > [Configuration] > [IP1 Setup] > [Session Type] > [SIP]
 Select [Menu] > [Configuration] > [IP1 Setup] > [Algorithm] > [G711/G722 or
MP2]
SIP Server Connections: Getting Started
Registering codecs for SIP connectivity is simple. First, choose the SIP server that you wish to
register your codec with. On a LAN this may be your own server, or it could be one of the many
internet servers available. We recommend that you use your own SIP server and configure it to use
G.711, G.722, MP2 and AAC algorithms. This is because most internet SIP servers are for VoIP
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phones and are only configured for G.711 and GSM algorithms.
When you register an account with a SIP server you will be provided with:
·
·
·
·
·
The SIP server IP address.
A username (often the same as a SIP number).
A password.
Domain details.
Realm details (sometimes).
Configure the Codec for SIP using the Web-GUI
Use the Toolbox web-GUI to program SIP account registration details into your codec. Once these
details have been entered into the codec, each time it is connected to a public IP address it will
contact the SIP server automatically to acknowledge its presence over a wide area network.
1. Connect your codec to a LAN connection with a public IP address, then login to the Toolbox
web-GUI and click the Settings
symbol at the top of the screen to display the Settings
panel.
2. Click the SIP button.
3. Enter the account details into the relevant text boxes.
4. Enter the Registration Timeout (this shouldn't need to be adjusted from the default setting).
5. Click to select Activate Account and click the Save Settings button to create the account
in the codec. Settings Saved is displayed in the bottom-left corner of the Settings panel if
the account details are saved successfully.
5. Navigate to SETTINGS
> SIP > Accounts to verify that the account has been registered
to the SIP server. The registration symbol
appears when it is activated successfully.
Important Notes: Some ISPs may block SIP traffic over UDP port 5060.
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Configuring SIP Programs
SIP programs are like a normal IP program to configure, with two small differences; entering a SIP
address and selecting SIP as the Session Protocol.
Important Notes: Before you start program configuration please note:
· SIP can only operate using the LAN / ETHERNET port on the rear panel of the codec.
· You cannot edit a program when it is currently loaded in the codec.
· Some drop-down menus and settings may be greyed out intentionally depending on
features available.
· To learn more about programs see the section titled About Program Dialing.
1. Open the web-GUI and click the Programs
Programs panel.
symbol at the top of the screen to display the
2. Click the New Program button to open the wizard and:
· Click in the Program Name text box to name the new program.
· Select Mono/Stereo Peer to Peer, or if you want to use an existing program as a template,
select this option. Then click Next.
Important Notes: When you choose to use an existing program as a template, the new
program inherits all the settings of the template program and you can adjust these settings
as required by continuing through the program wizard.
3. Enter a name for the Audio Stream and configure the codec to dial, answer or dial and
answer. Then click Next.
Note: The following example will display how to configure a dial and answer program. If you want the
codec to either dial or answer only, select the option and the wizard will automatically display
screens to allow you to configure the codec correctly.
4. This audio stream connection in the wizard will allow the codec to dial. Enter the name of the
connection in the text box, then click Next.
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5. Follow the instructions on the right-hand side of the panel to configure the transport settings
for the connection: Ensure that you select SIP as the Session Protocol menu. Then click
Next.
6. Configure the destination codec Address and Audio Port, then the network interface used to
dial the connection, e.g. Primary (LAN / ETHERNET port and default setting) or VLAN if
configured. Note: By default Any will select Primary.
At this point you can click Save Program and save the program with the default jitter and FEC
settings in the codec. Alternatively, click Next to specify individual algorithm, jitter buffer and
FEC settings for this connection and configure backup audio for this audio stream
(recommended).
Important Notes:
· If your codec is registered with same SIP registrar as the destination codec then you
only need to enter the SIP user name to dial successfully.
· The default UDP audio port when using SIP is 5004 in Tieline codecs. To contact a codec
that is behind a firewall or NAT-enabled router, it is essential that this and all other
relevant ports are open and forwarded to the other device.
7. Click the drop-down arrows on the right-hand side of each active drop-down menu to adjust
the Encoding, Sample rate or Bit rate parameters. Click Next to continue.
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8. Click to configure:
· Auto Jitter Adapt and the preferred auto jitter setting using the drop-down arrow for Buffer
Priority, or
· Fixed Buffer Level and enter the Jitter Depth (5000ms maximum).
9. Click to select Enable Auto Reconnect to enable automatic reconnection, then click Next.
10.Enter a name for the answering connection and click Next.
11.Follow the instructions on the right-hand side of the panel to configure the Transport,
Session Protocol and Audio Port (preconfigured for SIP) settings for the connection, then
click Next.
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12.Click to configure jitter buffer and FEC settings:
· Auto Jitter Adapt and the preferred auto jitter setting using the drop-down arrow for Buffer
priority, or
· Fixed Buffer Level and enter the Jitter Depth (5000ms maximum).
13.After configuring these settings there are 3 options:
i. If you want to create another answering connection, select the check-box for Create
another answering connection at the bottom of the panel and continue through the
wizard.
ii. Click Next and select Enable File Playback on silence detection.
iii. Click Save Program to save the program at this point.
When you save the program it will be confirmed by the following message.
5. Click Finish and the newly created program will be displayed in the left pane within the
Programs panel and in the Master panel. Select and connect audio streams in a program
using the Master panel, or dial the program manually using the codec front panel.
Caution: If the codec LAN cable is disconnected and the IP address changes when dialing
in SIP mode, you will need to reboot the codec, otherwise the codec will not be able to
reconnect.
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Dial and Disconnect a Program
Connecting a Program
1. Click to select the program you want to load from the Programs list in the Master panel.
2. Click Connect to load the program and connect all audio streams.
Disconnecting a Program
1. Click to highlight the audio stream in the Connections pane of the Master panel.
2. Click Disconnect to end the connection.
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Dial and Disconnect Multi-unicast Connections
Dial Multiple Connections within a Program
1. Click to select the program you want to load from the Programs list.
2. Click Connect to load the program and connect all connections.
Disconnect All Audio Stream Connections
1. Click to select the program in the Connections pane, e.g. Multi-unicast to Codec A & B in the
following example.
2. Click the Disconnect button.
3. Click Yes in the confirmation dialog to disconnect all audio stream connections.
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Disconnect a Single Audio Stream Connection
1. Click to select the audio stream connection you want to disconnect.
2. Click the Disconnect button.
3. Click Yes in the confirmation dialog to disconnect all audio stream connections.
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Lock or Unlock Programs
It is possible to lock a loaded custom program in a codec to ensure the currently loaded program
type, e.g. mono, cannot be unloaded by a codec dialing in with a different program type, e.g. stereo.
For example, if you require the codec at the studio to always connect in mono, simply load and lock
a mono program in the codec.
1. Open the web-GUI and click the Settings
symbol at the top of the screen to display the
Settings panel.
2. Click the Options button at the top of the Settings panel.
3. Click the Lock Loaded User Program check-box to lock or unlock a user program in the codec.
4. Click Save Settings at the bottom of the panel to save the new configuration.
Important Note:
· It is only possible to lock custom programs in a codec.
· If Lock Program is enabled and you load a new custom program in the codec, Lock
Program remains enabled and locks the most recently loaded custom program.
· A red Padlock symbol appears in the Status section of the Master panel to indicate a
program is locked in the codec.
16.12
View/Edit/Delete Programs
Important Notes: You cannot edit or delete a program when it is currently loaded in the
codec. Ensure you have unloaded a program prior to editing the current configuration.
To view configuration settings for an existing program, or edit settings:
1. Open the web-GUI and click the Programs
Programs panel.
2. Click to select a program in the left-hand pane.
symbol at the top of the screen to display the
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3. Click the blue arrow
to expand audio stream information and click the Edit symbol
to
adjust program settings.
The program wizard will open at the relevant point to facilitate editing of connection parameters.
Click Save Program to store settings.
Deleting Programs
There are two ways to delete a program.
1. Ensure the program you want to delete is not currently loaded in the codec.
2. Click to select a program listed on the left hand side of the Programs panel and then rightclick to display menu options.
3. Select and click Delete Program.
4. Click Yes in the confirmation dialog.
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5. Alternatively, click Delete Program next to the program name in the top-right corner of the
Programs panel.
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Reset Factory Default Settings
There are several options which allow you to restore factory default settings within the codec. See
Reset and Restore Factory Defaults for more details on each option.
1. Open the web-GUI and click the Settings
symbol at the top of the screen to display the
Settings panel.
2. Click the Reset/Restore button at the top of the Settings panel.
3. Click one of the reset options available.
4. A confirmation dialog appears for each option, click Yes to proceed or No to cancel the reset
function.
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Backup and Restore Functions
The Toolbox web-GUI can be used to backup and restore codec settings, including:
· Programs containing a variety of connection settings.
· All system settings that have been adjusted to change the factory default codec settings
(current runtime settings).
Files can also be used to copy configurations onto other similar codecs. Programs are essentially
connection profiles that may include:
· Program, audio stream and connection names.
· IP address, port, algorithm, jitter buffer, FEC and bit rate settings (etc.) for audio stream
connections.
Creating Backup Files
1. Open the web-GUI and click the Settings
symbol at the top of the screen to display the
Settings panel.
2. Click the Reset/Restore button at the top of the Settings panel.
3. Click Backup.
4. Use your mouse-pointer to click and select the check boxes to confirm your backup
requirements, then click Backup.
5. Click Save and select a location on your PC to save the configuration file.
Restoring Configuration File Settings
1. Open the web-GUI and click the Settings
Settings panel.
symbol at the top of the screen to display the
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2. Click the Reset/Restore button at the top of the Settings panel.
3. Click Restore.
4. Navigate to the configuration file on your PC that you want to load, then click Open.
5. Use your mouse-pointer to click and select the check boxes for restoring items. For example,
you could select the Include programs check-box and deselect the Include system check-box
if you are only copying programs onto codecs.
6. Click Restore to copy the configuration file settings onto the codec; confirmation of successful
file restoration is provided.
7. The codec will reboot automatically to ensure the restored configuration is loaded successfully.
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Web-GUI Software License Installation
Prior to installation you will need connect your codec to a PC and ensure you are connected to the
internet. You should also check that you have received notification by email from Tieline that your
new license file is ready to download from TieServer.
Perform an Automatic Software License Install using the Java Toolbox WebGUI
1. Open the Java Toolbox web-GUI in a browser on your PC by typing either the IP address of the
codec (LAN connection), or the USB address of the codec (USB connection) into the address
bar.
2. Click the Settings
button at the top of the web-GUI screen to open the Settings panel.
3. Click Licensing in the System panel.
4. Select the Get license file from TieServer button.
5. Click the Upload License File button.
6. After the upgrade is completed click Finish and the codec screen should display a confirmation
message within a short period of time.
7. Reboot the codec via Settings > Reset > Reboot Codec and press the
reboot by removing the power cable from the codec.
button. Note: do not
Download a License File and Install Manually
Prior to installing any new software license you will need to connect your codec to a PC and save
the license file on this computer.
1. Open the Toolbox web-GUI in a browser on your PC by typing either the IP address of the
codec (LAN connection), or the USB address of the codec (USB connection) into the address
bar.
2. Click the Settings
button at the top of the web-GUI screen to open the Settings panel.
3. Click Licensing in the System panel.
4. Click Upload a selected file.
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5. Click the Select File button to open a dialog and navigate to the ".lcf" license file on your
PC, then click the Open button.
6. Click the Upload License File button to upload the license file into the codec.
7. Click the Finish button.
8. Reboot the codec via Settings > Reset > Reboot Codec and press the
button. Note: do
not reboot by removing the power cable from the codec.
16.16
Download Logs
The codec is capable of providing diagnostic information via user logs, which can either be sent to
Tieline support, or downloaded for user diagnostics.
Procedure for Sending Logs to Tieline
1. Open the web-GUI and click the Help
panel.
2. Click Download Logs.
symbol at the top of the screen to display the Help
3. Save the file to your computer and then send it as a .zip file to Tieline support via
support@tieline.com
Download Event Logs
Event logs can be downloaded from the codec and viewed using any text editor, e.g. Microsoft®
Word.
1. Open the web-GUI and click the Help
symbol at the top of the screen to display the
Help panel.
2. Click Download Event Log and select a location to save the log file.
Clearing Logs
This option should only be used if instructed to by Tieline support staff. To clear all event and other
logs in the codec via the front panel see the Reset and Restore Factory Default Settings section of
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this manual, or see Reset Factory Default Settings to clear recent log history using the web-GUI.
16.17
RS232 Data Configuration
The codec supports both in-band and out-of-band data depending on the algorithm you are using.
RPTP data is automatically enabled when using the Tieline Music or MusicPLUS algorithms over
IP. It is also possible to enable synchronized out-of-band data using any algorithm.
Algorithm Selected
Tieline Music and MusicPLUS
All other algorithms
IP
· In-band RPTP data enabled automatically
· Synchronized out-of-band data can be enabled and
disabled
· Synchronized out-of-band data can be enabled and
disabled
The codec can be connected to external devices and send RS232-compatible data via the serial
port on the rear panel of the codec. To enable RS232 data within a connection, select Enable
Auxiliary Data when creating a program in the Programs panel wizard. Alternatively, select using
the codec Setup menu (see Enabling RS232 Data) ).
Setting RS232 Data Rates and Flow Control
1. Open the web-GUI and click the Settings
symbol at the top of the screen to display the
Settings panel.
2. Click the Options button.
3. Click the Baud rate drop-down menu arrow to select the serial port baud rate which matches the
baud rate of the external device connected to the RS232 port on the codec.
4. Click to select the Enable flow control check box and enable flow control, then click Save
settings.
Important Notes:
· When connecting to G3 codecs over IP only in-band data is available via the Music and
MusicPLUS algorithms.
· Use firmware higher than 2.8.xx in the Bridge-IT, Genie and Merlin families of codecs to
enable auxiliary data over multicast connections.
· It is important to enable serial port flow control as it regulates the flow of data through the
serial port. If disabled, data will flow unregulated and some may be lost.
· Ensure you configure the serial port baud rate to match the setting of the external device
to which you are connecting. Ideally the settings on both codecs should match, or you
could have data overflow issues.
· Only the dialing codec needs to be configured to send RS232 data. Session data sent
from the dialing codec will configure all other compatible codecs (non-G3) when you
connect.
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· RS232 data can be sent from the dialing codec to all endpoints of a multi-unicast
connection if your codec is capable of these connections. Note: Bidirectional RS232 data
is only available on the first connection dialed when multi-unicasting.
16.18
Creating Rules
The Rules panel in the Toolbox web-GUI is used to configure actions which are dependent upon
changes to GPIO control port states or connection events. Rules can only be created with the webGUI while the codec is disconnected. Note: Data must be enabled in the Connection menu to
enable contact closure operation and RS232 data. This is disabled by default. Data can be enabled
in the codec as follows:
1.
2.
3.
4.
5.
6.
Press the HOME
button to return to the Home screen.
Use the navigation buttons on the front panel to select Connect and press the
Select IP and press the
button.
Select your preferred IP Session mode (Tieline or Sessionless).
Use the down
navigation button to select Setup and press the
button.
Navigate to Data and press
to toggle between Enabled and Disabled.
button.
For more information please see "Enabling Relays & RS232 Data" in this user manual.
Configuring Rules
Default rules have been configured into the codec to facilitate programming the most common
events required by broadcast engineers. To view rules options:
1. Click the Rules
button at the top of the web-GUI screen to open the Rules panel.
2. Click Add New Rule.
3. Click to select the appropriate rule for your requirements. See the Web-GUI Introduction section
for explanations of what actions each rule can perform.
Any previously configured rules are displayed when the Rules panel is first opened.
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Rule 1: Toggle a Control Port Input to Connect and Disconnect a Program
This rule is used to connect and disconnect a selected program when a control port input is toggled.
1. Click the first rule in the Rules panel.
2. Click the drop-down Input arrow and select the control port input which will trigger program
connection and disconnection.
3. Click the drop-down Program arrow to select the program to be connected.
4. Check the Rule Summary and click Create Rule to save the settings.
Rule 2: Switch Different Control Port Inputs On to Connect and Disconnect
a Program
This rule is used to connect and disconnect a selected program when different codec control port
inputs are turned on.
1. Click the second rule in the Rules panel.
2. Click the drop-down arrows to select the control port input for connecting and the alternative
one for disconnecting.
3. Click the drop-down Program arrow to select an individual program which will be connected
and disconnected by the change in the control port input states.
4. Check the Rule summary and click Create Rule to save the settings.
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Rule 3: Synchronise Local Control Port Input Status with a Remote Relay
Output
Use this rule allow a local codec's control port input to change the state of a remote relay output.
1. Click the third rule in the Rules panel.
2. Click the drop-down arrow to select the local control port input used to control a remote relay
output.
3. Check the Rule summary and click Create Rule to save the settings.
Rule 4: Toggle a Relay Output with each Change in Connection Status
This rule is used to toggle a codec's control port relay output each time a program connects and
disconnects.
1. Click the fourth rule in the Rules panel.
2. Click the drop-down Relay arrow and select the relay output you want to toggle.
3. Click the drop-down Program arrow to select a specific program which will affect the relay
toggle function, or use the default setting whereby any program will toggle the relay output.
4. Check the Rule summary and click Create Rule to save the settings.
Invalid Rules
Any rule in the Rules panel related to a deleted program is invalid. An Invalid error warning will be
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displayed when this type of rule exists.
Deleting Rules
1. Click the Rules
button at the top of the web-GUI screen to open the Rules panel.
2. Click the Delete button next to the rule you want to delete.
3. Click Yes in the confirmation dialog.
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Upgrading Codec Firmware
To download the latest codec firmware visit http://www.tieline.com/Support/Latest-Firmware.
Manual Firmware Upgrades
The following procedure explains how to perform codec firmware upgrades with a downloaded
firmware file saved to your PC.
1. Click the Settings
button at the top of the web-GUI screen.
2. Click Firmware in the Systems panel.
3. Click Update from a selected file and click the Select File button.
4. Select the .bin file you are using to perform the upgrade and click Open.
5. Press the Update Firmware button to commence the upgrade.
Automatic Firmware Upgrades
By default the web-GUI application integrates with TieServer to automatically update users when a
firmware upgrade is available.
1. Connect your codec to your PC using either a LAN or USB connection and open the web-GUI
(See Connecting to the Web GUI)
2. If new software is available the Update symbol appears in the top-left hand side of the
screen.
3. Position your mouse-pointer over the Update symbol and click the update dialog when it
appears to download the new software.
4. Click More Information in the Updating firmware dialog to display details of the upgrade
process.
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+
Important Note: Firmware upgrade files are very large and it is usually much quicker to
download the file to your PC first and then upgrade the codec manually using the Manual
Firmware Upgrade procedure.
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115
HTML5 Toolbox Web-GUI Introduction
The following sections provide an overview of the different configuration panels available within the
codec's HTML5 Toolbox Web-GUI. Navigate with the mouse pointer to the Menu bar at the top of
the Web-GUI screen and click to select and open each panel in turn.
HTML GUI Menu Bar for Opening Panels
When you first open the HTML5 Toolbox Web-GUI the Program Loader panel, Connections
panel and PPMs panel are loaded by default. If you retain cookies in your browser, any panels
opened previously in the Web-GUI are automatically populated when you log in next. The default
panel view is displayed on login if cookies have been cleared.
Adjusting the Theme
To adjust the Theme or 'skin' of the HTML5 Toolbox Web-GUI, navigate to the Menu bar at the top
of the screen and click Theme, then click to select your preferred option. Note: this manual uses
the White theme for most images.
Opening a Panel & Adjusting Screen Position
Click an item in the Menu bar to display available panel options, then click to select and open a
panel. New panels automatically open in the top left of the screen.
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Position the mouse pointer over a panel's Title bar and click and drag to move a panel and
reposition it in a preferred screen position.
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Master Panels: Load Programs & Manage Audio Streams
Program Loader Panel
Description
Lists all configured programs which have been added into the
codec. Click to select a program before loading.
Click to close the panel.
Feature
1 Programs list
2 Close button
3 Check-box symbol
4 Load and
button
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The Check-box symbol identifies the currently loaded program
in the codec.
Unload Click Load to load the currently selected program in the
Programs list; the button changes automatically to Unload after
loading, to allow unloading of a program when required.
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1
2
3
4
Feature
Program
Connect/
Disconnect button
Audio
Stream
Connect/Disconnect
button
Connection Connect/
Disconnect button
Show/Hide Arrow
Connections Panel
Description
Click to connect/disconnect all audio streams in a program.
Click to connect/disconnect all connections in an audio stream.
Click to connect/disconnect an individual connection.
Click to show/hide audio stream and connection details.
5 Close button
Click to close the panel.
Feature
1 Show/Hide Arrow
Statistics Panel
Description
Click to show/hide audio stream statistics.
2 Show/Hide Arrow
Click to show/hide individual connection statistics.
3 Close button
Click to close the panel.
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Feature
1 PPM Meters
PPMs Panel
Description
4 PPM meters.
2 Close button
Click to close the panel.
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Inputs Panel for Input Adjustments
Important Note: Tieline codecs have different input configurations, therefore the image
shown may not reflect the number of inputs displayed in your codec Web-GUI.
1
Feature
Settings button
Description
Click to adjust input Name, Type and IGC.
2
Input PPM meter
Input PPM meter.
3
Close button
Click to close the panel.
4
On/Off button
Click to toggle an input on or off.
5
Input Sliders/Faders
Input gain control sliders/faders.
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Rules Panel for Creating Relay Activation Rules
5
Rule
Connect and disconnect a
program when an input is
toggled
Connect when an input is
switched
ON;
Disconnect
when
another
input
is
switched ON
Synchronise a local relay input
with a remote relay output
Toggle a relay based on a
connection's status
Back / Add New Rule button
6
Close button
1
2
3
4
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Description
Click to configure connection and disconnection by
toggling an input.
Click to configure connection and disconnection after
different relay inputs are switched ON.
Click to configure a local relay input to synchronise
with the state of a remote relay output.
Click to configure a relay to toggle based on
connection status.
Click to add a new rule, or exit the rule creation
function.
Click to close the panel.
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Settings Panels
There are 6 Settings panels which can be
opened in the Web-GUI. Each panel provides
specific codec configuration settings and options.
Click to select and open each panel.
As an example, the Network panel is displayed
with network interface configuration options. A
brief description of the other panel options is also
provided.
Settings panels
Feature
1 Network tab
2 Network Interface
3 IPv4 details
4 DNS details
Description
Click to select and edit, or view network configuration settings for
each Ethernet, VLAN and USB interface.
Control and streaming configuration options for each network
interface.
IPv4 address details and configuration.
5 Edit / Save button
Select the check-box and specify DNS addresses and domains
to search.
Click to edit Network settings, or save configured settings.
6 IPv6 details
IPv6 address details and configuration.
7 MAC Address / Link Click to open the panel and view the device MAC address and
Local
IPv6 local network address created by the codec.
8 Reset
9 Licensing tab
10 Firmware tab
11 SIP tab
Click to open the panel; reset codec default settings and perform
backup/restore of codec programs and settings.
Click to open the panel; select a license file and install it in the
codec.
Click to open the panel; view software versions and perform an
upgrade.
Click to open the panel and edit or view SIP configuration
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12 Options tab
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settings.
Click to open the panel; configure RS232 and QoS data settings,
lock a loaded user Program and adjust Session Port settings and
SNMP. Also configure the AES Output Clock sample rate.
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Help Panels
Feature
1 Close button
2 User manual link
Resources Panel
Description
Click to close the panel.
3 Support website link
Click to open the codec user manual in a new browser, or view
support information (Note: the codec name displayed will vary by
product type)
Click to visit the support page on the Tieline website.
4 Email Support
Click to email Tieline support.
5 Event Logs
Click to download user-viewable event logs
6 Support Logs
Click to download diagnostic information that can be sent to
Tieline support
The About panel provides details of the codec Toolbox and firmware version, as well as the
codec serial number. Note: the codec name displayed will vary by product type.
Feature
1 Close button
About Panel
Description
Click to close the panel.
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Language Selection
The HTML5 Toolbox Web-GUI offers language support for several languages.
1. Click on the Language drop-down menu arrow in the top right-hand corner of the Web-GUI
page.
2. Select the preferred language to display.
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HTML5 Toolbox Web-GUI Configuration
The following sections describe how to configure your codec using the HTML5 Toolbox web-GUI.
18.1
Using the HTML5 Toolbox Quick Connect Web-GUI
The HTML5 Quick Connect Web-GUI is designed for simple peer-to-peer connections and nontechnical users. It has a reduced feature-set and allows users to:
1. Load existing programs in a codec via the Program Loader panel and then dial via the
Quick Connect panel.
2. Use the Quick Connect panel to create and dial a simple peer-to-peer connection using IP/
SIP.
Important Note: Simple peer-to-peer connections are not saved as programs with unique
names. Details of the last ad hoc dial are retained in the Quick Connect panel, even after
a program is loaded and unloaded using the Program Loader panel.
Launching the HTML5 Quick Connect Web-GUI
1.
2.
3.
4.
Type the codec IP address in your web-browser.
Click to launch the HTML5 Toolbox Quick Connect Web-GUI.
Enter the authentication Password for the codec and click OK.
The panels in the Quick Connect Web-GUI will automatically be displayed.
Important Note: To change the password using the codec front panel navigate to Settings
> WebGUI > Password and press the button. Use the keypad to enter a new password
and press the button to save the new setting.
Loading and Unloading an Existing Program
If programs are saved in the codec they are displayed in the Program Loader panel.
1. Click to select a program in the Program Loader panel and click the Load button to load
it in the codec.
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2. The Check-box symbol appears next to the program name to confirm it has been loaded
and the Load button changes to an Unload button.
To unload a program click the Unload button.
Dial a Loaded Program
1. Click the Connect button in the Quick Connect panel to dial a loaded program. Note: After
connecting, the Connect button changes to a Disconnect button.
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Disconnect a Loaded Program
1. Click the Disconnect button in the Quick Connect panel.
2. Click Yes in the confirmation dialog to disconnect the connection.
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Dial Peer-to-Peer over IP with Quick Connect
Important Notes:
· Click the Unload button in the Program Loader panel if a program is currently loaded.
· The transcriptor algorithm is for closed captioning and not normal broadcast
configurations.
1. Click the drop-down Transport menu arrow in the Quick Connect panel and select IP.
2. Click the drop-down Algorithm menu and select an algorithm.
3. Click the select the appropriate Sample Rate and Bit Rate for the connection. Note: If only one
sample rate is available this will be automatically selected.
4. Click in the Destination text box and enter the IP address of the destination codec.
5. Click the Connect button to dial.
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Monitoring PPMs
Set audio levels so that audio peaks average at the nominal 0vu point indicated below on the PPM
meters. This represents a program level of +4 dBu leaving the codec. Audio peaks can safely reach
+22 dBu without clipping, providing 18dBu of headroom from the nominal 0vu point. Note: the audio
metering reference scale is automatically adjusted by default when a Merlin codec connects to a
Commander G3 codec. The G3 metering scale is between -11dBu and +18dBu.
Mono and Stereo PPM Metering
When connected with a mono program the codec will display a mix of inputs 1 and 2 on PPM1.
PPM 3 displays the level of return audio.
When connecting with a stereo program, the codec displays audio on PPM1 and 2 for inputs 1
and 2 and PPM 3 and 4 for the return audio.
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Configuring IP Settings
Open the HTML5 Toolbox Web-GUI and click Settings and then click Network to view and
configure Ethernet and VLAN interface settings in the Web-GUI.
Important Note: For assistance with configuration of IPv4 or IPv6 network connections
contact your IT Administrator.
IPv4 versus IPv6
An IP address is a unique address to identify a device on a TCP/IP network. Your codec uses dual
IP protocol stacks to allow your codec to work on both IPv4 and IPv6 networks. Tieline codecs
support both DHCP (default) IP addressing and static IP addresses for dialing IPv4 connection
endpoints.
If you want to dial a codec with a public IP address you simply dial the IP address to connect. If you
want to dial a codec with a private IP address you need to perform network address translation
(NAT). NAT allows a single device, such as a broadband router, to act as an agent between the
public internet and a local private LAN. Usually this will be set up at the studio end so you can dial
into the studio from the remote codec.
Support for IPv6 connections allows you to use IPv6 infrastructure to connect to other codecs
globally.
Configuring Ethernet Ports and VLANs
The codec features a physical Ethernet port and up to four additional VLAN interfaces.
VLAN interfaces have features similar to physical Ethernet interfaces. However, your network
administrator will need to configure VLAN support throughout your network for them to be supported
in your codec.
As an example, VLANs can be used to separate codec Control and Streaming functions if required.
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Ethernet and VLAN interfaces can be configured for:
· Control Only: codec control and command only from the Ethernet port.
· Control and Streaming: stream audio and control and command the codec via the Ethernet
port.
· Streaming audio: stream audio only from an Ethernet port (VLANs only).
· Nothing: Disable the Ethernet port from streaming audio and codec command and control
(VLANs only).
The Name text box, e.g. Primary or Secondary, is an interface identifier used when configuring
new programs via the Programs panel.
IPv4 Address Configuration
Click the Edit button in the Network panel to configure settings. The codec is capable of automatic
DHCP address assignment, or manually configured static IPv4 address configuration via the dropdown Configure IPv4 menu. If you want to ignore IPv4 settings select Off.
DHCP IP addresses are automatically assigned and can change each time you connect to your
Internet Service Provider, or to your own local area network (LAN). By default the codec is
configured for DHCP-assigned IP addresses.
Static IP addresses are fixed addresses that are recommended for studio installations, so that IP
address dialing remains the same over time for incoming codec connections.
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Click Save to store all configuration settings.
Note: The Subnet Mask is used by the TCP/IP protocol to determine whether a host is on
the local subnet or on a remote network. The default Gateway is the router linking the
codec's subnet to other networks. See your IT administrator for more details.
IPv6 Address Configuration
An IPv6 address is represented by 8 groups of 16-bit hexadecimal values separated by colons (:).
The drop-down Configure IPv6 menu provides three address configuration options:
1. Automatically: An address is automatically assigned to the codec when you connect the
codec to an IPv6 router. This process is similar to how an IPv4 DHCP address is assigned.
2. Manually: Select to enter static IPv6 address details.
3. Off: Select to ignore IPv6 address details.
Important Note: Select Off in the drop-down Configure IPv6 menu if you are not using
IPv6 to connect to another device. This ensures your codec will attempt to connect using
IPv4 at all times.
Types of IPv6 Addresses
There are two types of addresses displayed in the IPv6 section:
1. IPv6 address (normally global): A router-allocated IP address with 'global' visibility,
details of which are displayed in the Address, Prefix size and Gateway text boxes.
2. Link Local: A local address which can only be used to connect to another device directly
over a LAN. This address is allocated by the codec internally based on MAC address
details.
Auto Address Assignment
1. Click the Edit button in the Network panel to configure settings.
2. By default the codec is configured to connect to an IPv6 router which automatically allocates
IPv6 address details, as displayed in the following example.
3. Click Save to store all configuration settings.
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Manual IPv6 Address Assignment
1. Click the Edit button in the Network panel to configure settings.
2. To configure IPv6 address details into the codec manually, select Manually and enter
details into the Address, Prefix and Gateway text boxes.
3. Click Save to store all configuration settings.
Specifying DNS Settings
It is possible to specify Domain Name Server (DNS) settings to allow easy look up of codecs within
the specified DNS Addresses or Domains.
1. Click the Edit button in the Network panel to configure settings.
2. Click Save to store all configuration settings.
The codec can be configured to tag IP data packets sent across a network by entering a value into
the Differentiated Services Code Point (DSCP) field within the header of data packets transmitted
over the network.
Configuring QoS
1.
2.
3.
4.
Open the HTML5 Toolbox web-GUI in a browser on your PC.
Click Settings at the top of the screen and then click Options to display the Options panel.
Click the Edit button at the bottom of the panel.
Click in the QoS DSCP text box and enter the preferred value.
5. Click the Save button at the bottom of the panel to save the new setting.
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Important Note: Check with your IT administrator before changing this setting. By default
the codec is programmed for Assured Forwarding and more details about DSCP are
available on Wikipedia at http://en.wikipedia.org/wiki/Dscp. For more information on
configuring QoS see Configuring QoS for Broadcasts in this manual.
18.3
Configuring Input/Output Settings
Open the HTML5 Toolbox Web-GUI and click Inputs to display the Inputs panel.
Important Note: 15 volt phantom power can only be supplied on analog input 1; this is
disabled by default.
Configuring Input Channel Settings
Renaming Input Channels:
1. Click the Input Settings
symbol on the input you want to rename.
2. Click in the Name text box to enter a new name, or edit an existing name.
3. Click Save to confirm the name change.
Selecting Analog and Digital Audio Sources:
Codec inputs are configured for analog line level audio sources by default.
1. Click the Input Settings
symbol.
2. Select Type and click to select either Analog or AES3.
3. When you select AES3 each input is automatically configured for 100% input levels;
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input level and input on/off controls are removed and unable to be adjusted. See
Configuring AES3 Audio for more information about the digital inputs and outputs.
Adjusting Analog Audio Levels
Gain on Input 1 can be configured for mic or line level sources. Input 2 accepts line level only.
Audio levels on the Input panel should be set to ensure audio peaks average at the first yellow
indications on the PPM meters, which represents +4dBu. These levels should also be checked
against the Input PPM Meters on the Master panel.
Other Input Controls
Adjust the IGC (Intelligent Gain Control)
input settings to Auto, Fixed or Off as
required.
AES3 Output Sample Rate Configuration
The AES3 output sample rate can be configured using the HTML5 Toolbox Web-GUI.
1. Open the HTML5 Toolbox Web-GUI and click Settings, then click Options to open the
Options panel.
2. Click the Edit button to configure settings.
3. Click the AES Output Clock drop-down menu to select your preferred AES Output Clock
setting, then click Save.
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137
Configure SIP Settings
The codec is fully EBU N/ACIP Tech 3326 compliant when connecting using SIP (Session Initiation
Protocol) to other brands of IP codecs.
About SIP
SIP provides superior interoperability between different brands of codecs due to its standardized
protocols for connecting devices and is intended to be used when connecting Tieline codecs to nonTieline devices. Devices primarily use SIP to dial another device’s SIP address and find its location
with a minimum of fuss. This task is usually performed by SIP servers, which communicate between
SIP-compliant devices to set up a call.
When connecting two devices, SDP performs similar tasks to Tieline’s proprietary session data,
which is used to configure all non-SIP IP connections. There are two very distinct parts to a call
when dialing over IP. The initial stage is the call setup stage and this is what SIP is used for. The
second stage is when data transference occurs and this is left to the other protocols used by a
device (i.e. using UDP to send audio data).
All the mandatory EBU N/ACIP 3326 algorithms are supported (G.711, G.722, MPEG-1 Layer 2 and
16 bit PCM), as well as optional algorithms including LC- AAC, HE-AAC and aptX Enhanced. The
default algorithm selected when connecting using SIP is G.711.
Important Notes:
· Each codec should be registered to a different SIP server account to avoid connection
conflicts.
· SIP account registration can only be configured via Ethernet port 1.
· SIP dialing is only supported over point-to-point connections, not multi-unicast
connections.
· Tieline G3 codecs do not support connections using AAC and will default to MPEG Layer
2 if an incoming call is programmed to use this algorithm.
· Failover and SmartStream PLUS redundant streaming are not available with SIP
connections.
· When connecting to a Tieline G3 codec using SIP you need to manually select the G3
audio reference level in the codec. To do this select SETTINGS
> Audio > Ref Level
> Tieline G3. In addition, select the following on the G3 codec prior to dialing.
 Select either a mono or stereo profile
 Select [Menu] > [Configuration] > [IP1 Setup] > [Session Type] > [SIP]
 Select [Menu] > [Configuration] > [IP1 Setup] > [Algorithm] > [G711/G722 or
MP2]
SIP Server Connections: Getting Started
Registering codecs for SIP connectivity is simple. First, choose the SIP server that you wish to
register your codec with. On a LAN this may be your own server, or it could be one of the many
internet servers available. We recommend that you use your own SIP server and configure it to use
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G.711, G.722, MP2 and AAC algorithms. This is because most internet SIP servers are for VoIP
phones and are only configured for G.711 and GSM algorithms.
When you register an account with a SIP server you will be provided with:
·
·
·
·
·
The SIP server IP address.
A username (often the same as a SIP number).
A password.
Domain details.
Realm details (sometimes).
Configure the Codec for SIP using the Web-GUI
Use the HTML5 Toolbox Web-GUI to configure SIP account registration details in your codec. Once
these details have been entered into the codec, each time it is connected to a public IP address it
will contact the SIP server automatically to acknowledge its presence over a wide area network.
1. Connect your codec to a LAN connection with a public IP address, then login to the HTML5
Toolbox Web-GUI.
2. Click Settings at the top of the screen and then click SIP to display the SIP panel.
3.
4.
5.
6.
Click the Edit button to configure settings.
Enter the account details into the relevant text boxes.
Enter the Registration Timeout (this shouldn't need to be adjusted from the default setting).
Click to select the Activate Account check-box and click Save to create the account in the
codec.
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7. Navigate to SETTINGS
> SIP > Accounts to verify that the account has been registered
to the SIP server. The registration symbol
appears when the account has been activated
successfully.
Important Notes: Some ISPs may block SIP traffic over UDP port 5060.
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Load, Unload and Dial a Program
To load and dial a program it is necessary to open the Program Loader panel and Connections
panel. Use the Program Loader panel to load a program and then manage connecting and
disconnecting using the Connections panel.
Important Notes: The HTML5 Toolbox Web-GUI currently does not support the creation of
new programs. Use the Java Toolbox Web-GUI to create a new program.
Loading a New Program
1. Open the HTML5 Toolbox Web-GUI and click Master and then click Program Loader to open
the Program Loader panel.
2. Click to select a program in the Program Loader panel and then click Load to load the
program in the codec.
Note: the currently loaded program has the check-box symbol displayed next to its name.
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Connecting a Program
To connect audio streams and connections within an existing program there are three options:
1. Click the program Connect/Disconnect
symbol and then click Connect; this connects
all active audio streams and connections associated with the program.
2. Click the audio stream Connect/Disconnect
symbol and then click Connect; this
connects all connections associated with this audio stream.
3. Click the connection Connect/Disconnect
connects an individual audio stream connection.
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symbol and then click
Connect; this
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Disconnect a Program
To disconnect audio streams and connections within an existing program there are three options:
1. Click the program Connect/Disconnect
symbol and then click Disconnect to disconnect
a program; this includes all audio streams and connections associated with the program.
2. Click the audio stream Connect/Disconnect
symbol and then click Disconnect to
disconnect an individual audio stream and all associated connections.
3. Click the connection Connect/Disconnect
stream connection.
symbol to disconnect an individual audio
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Unloading a Program
1. Click to select the loaded program in the Program Loader panel and then click Unload to
unload the program in the codec.
18.6
Lock or Unlock Programs
It is possible to lock a loaded custom program in a codec to ensure the currently loaded program
type, e.g. mono, cannot be unloaded by a codec dialing in with a different program type, e.g. stereo.
For example, if you require the codec at the studio to always connect in mono, simply load and lock
a mono program in the codec. Generally programs will be up or down-mixed by the answering codec
to match the loaded program type. In some situations incompatible program types will be rejected.
1. Open the HTML5 Toolbox Web-GUI and click Settings at the top of the screen, then click
Options to display the Options panel.
2. Click the Edit button to configure settings.
3. Click the Lock Loaded User Program check-box to lock or unlock a user program in the
codec.
4. Click Save to store the new configuration.
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Important Note:
· A black Padlock symbol appears next to the program name in the Connections panel
and in the Program Loader panel, to indicate a program is locked in the codec.
· It is only possible to lock custom programs in a codec.
· If Lock Program is enabled and you load a new custom program in the codec, Lock
Program remains enabled and locks the most recently loaded custom program.
18.7
Reset Factory Default Settings
There are several options which allow you to restore factory default settings within the codec. See
Reset and Restore Factory Defaults for more details on each option.
1. Open the HTML5 Toolbox Web-GUI and click Settings at the top of the screen, then click
Reset to display the Reset panel.
2. Click one of the available reset options to adjust codec settings, or reboot the codec. Note:
Hover with the mouse pointer over the Information
symbol to view a tool-tip for each reset
option.
4. A confirmation dialog appears for each option; click Yes to proceed.
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145
Backup and Restore Functions
The HTML5 Toolbox Web-GUI can be used to backup and restore codec settings, including:
· Programs containing a variety of connection settings.
· All system settings that have been adjusted to change the factory default codec settings
(current runtime settings).
Files can also be used to copy configurations onto other similar codecs. Programs are essentially
connection profiles that may include:
· Program, audio stream and connection names.
· IP address, port, algorithm, jitter buffer, FEC and bit rate settings (etc.) for audio stream
connections.
Creating Backup Files
1. Open the HTML5 Toolbox Web-GUI and click Settings at the top of the screen, then click
Reset to display the Reset panel.
2. Click Backup.
3. Click to select the check-boxes to confirm your backup requirements, then click Backup.
4. Select a location on your PC to save the configuration file. Note: You may need to "allow" your
browser to display the pop-up dialog.
Restoring Configuration File Settings
1. Open the HTML5 Toolbox Web-GUI and click Settings at the top of the screen, then click
Reset to display the Reset panel.
2. Click Restore.
3. Click to select the check-boxes and confirm your restore settings. For example, you could select
the Include programs check-box and deselect the Include system check-box if you are only
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copying programs onto codecs.
4. Click Restore and select the .tgz file you want to load onto the codec. A Success dialog
confirms the files have been restored.
Note: The codec will automatically reboot if you restore system settings.
18.9
HTML5 Software License Installation
Prior to installation you will need connect your codec to a PC and ensure you are connected to the
internet. You should also check that you have received notification by email from Tieline that your
new license file is ready to download from TieServer.
Perform an Automatic Software License Install with the HTML5 Toolbox
Web-GUI
1. Open the HTML5 Toolbox web-GUI in a browser on your PC by typing either the IP address of the
codec (LAN connection), or the USB address of the codec (USB connection) into the address
bar.
2. Ensure you have unloaded any currently loaded program in the codec via the Program Loader
panel .
3. Click Settings at the top of the screen and then click Licensing to display the Licensing panel.
4. Click Get license file from TieServer.
5. A Success dialog in the web-GUI Licensing panel confirms when installation is complete and
the codec screen should display a confirmation message.
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6. Reboot the codec via Settings > Reset > Reboot Codec and press the
reboot by removing the power cable from the codec.
147
button. Note: do not
Download a License File and Install Manually
Prior to installing any new software license you will need to connect your codec to a PC and save
the license file on this computer.
1. Open the HTML5 Toolbox web-GUI in a browser on your PC.
2. Click Settings at the top of the screen and then click Licensing to display the Licensing
panel.
3. Click Upload a selected file.
4. Navigate to the ".lcf" license file on your PC, then click the Open button to commence
license installation.
5. A Success dialog in the web-GUI Licensing panel confirms when installation is complete
and the codec screen should display a confirmation message.
6. Reboot the codec via Settings > Reset > Reboot Codec and press the
not reboot by removing the power cable from the codec.
18.10
button. Note: do
Download Logs
The codec is capable of providing diagnostic information via user logs, which can either be sent to
Tieline support, or downloaded for user diagnostics.
Procedure for Sending Logs to Tieline
1. Open the HTML5 Toolbox Web-GUI and click Settings in the Menu Bar, then click Help to
display the Help panel.
2. Click Download Logs.
3. Save the file to your computer and then send it as a .zip file to Tieline support via
support@tieline.com
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Download Event Logs
Event logs can be downloaded from the codec and viewed in your browser.
1. Open the HTML5 Toolbox Web-GUI and click Settings in the Menu Bar, then click Help to
display the Help panel.
2. Click Download Event Log to view the log in a new web-browser window.
Clearing Logs
This option should only be used if instructed to by Tieline support staff. To clear all event and other
logs in the codec via the front panel, see the Reset and Restore Factory Default Settings section of
this manual, or see Reset Factory Default Settings to clear recent log history using the Web-GUI.
18.11
RS232 Data Configuration
The codec supports both in-band and out-of-band data depending on the connection transport and
algorithm you are using. RPTP data is automatically enabled when using the Tieline Music or
MusicPLUS algorithms over any transport. Over IP it is also possible to enable synchronized out-ofband data using any algorithm.
Algorithm Selected
Tieline Music and MusicPLUS
All other algorithms
IP Transport
· In-band RPTP data enabled automatically
· Synchronized out-of-band data can be enabled and
disabled
· Synchronized out-of-band data can be enabled and
disabled
The codec can be connected to external devices and send RS232-compatible data via the serial
port on the rear panel of the codec. To enable RS232 data within a connection, select Enable
Auxiliary Data when creating a program in the Programs panel wizard.
Setting RS232 Data Rates and Flow Control
1. Open the HTML5 Toolbox Web-GUI and click Settings in the Menu Bar, then click Options
to display the Options panel.
2. Click the Edit button to configure settings.
3. Click the Baud rate drop-down menu arrow to select the serial port baud rate which matches
the baud rate of the external device connected to the RS232 port on the codec.
4. Click to select the Enable flow control check box and enable flow control, then click Save
to store the new settings.
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Important Notes:
· When connecting over IP to G3 codecs only in-band data is available via the Music and
MusicPLUS algorithms.
· Use firmware higher than 2.8.xx in the Bridge-IT, Genie and Merlin families of codecs to
enable auxiliary data over multicast connections.
· It is important to enable serial port flow control as it regulates the flow of data through the
serial port. If disabled, data will flow unregulated and some may be lost.
· Ensure you configure the serial port baud rate to match the setting of the external device
to which you are connecting. Ideally the settings on both codecs should match, or you
could have data overflow issues.
· Only the dialing codec needs to be configured to send RS232 data. Session data sent
from the dialing codec will configure all other compatible codecs (non-G3) when you
connect.
· RS232 data can be sent from the dialing codec to all endpoints of a multi-unicast or
multicast connection if your codec is capable of these connections. Note: Bidirectional
RS232 data is only available on the first connection dialed when multi-unicasting.
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18.12
Bridge-IT Manual
Creating Rules
The Rules panel in the Toolbox Web-GUI is used to program events for specific codec actions.
Typically these 'rules' are based on a change in the state of a GPIO control port or the codec being
connected or disconnected. Rules can only be created with the Web-GUI while the codec is
disconnected.
Important Note: Data transmission is disabled by default. Data must be enabled in the
Connection menu to enable contact closure operation and RS232 data.
1.
2.
3.
4.
5.
6.
Press the HOME
button to return to the Home screen.
Use the navigation buttons on the front panel to select Connect and press the
Select IP and press the
button.
Select your preferred IP Mode and press the
button.
Use the down navigation button to select Setup and press the
button.
Navigate to Data and press
to toggle between Enabled and Disabled.
button.
Configuring Rules
Default rules have been preprogrammed into the codec to facilitate programming the most common
events required by broadcast engineers. To view rules options:
1. Open the Toolbox HTML5 Web-GUI and click Rules in the Menu Bar to display the Rules
panel.
2. Click Add New Rule.
3. Click to select the appropriate rule for your requirements. See the Rules panel section in Using
the Toolbox HTML5 Web-GUI for an explanation of the action each rule can perform.
Note: When rules have been configured previously they are displayed when the Rules panel is
opened.
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Rule 1: Toggle a Control Port Input to Connect and Disconnect a Program
This rule is used to connect and disconnect a selected program when a control port input is toggled.
1. Click the first rule in the Rules panel titled Connect and disconnect a program when an
input is toggled.
2. Click the drop-down Input arrow and select the control port input which will trigger program
connection and disconnection.
3. Click the drop-down Program arrow to select the program to be connected.
4. Check the Rule Summary and click Create Rule to save the settings.
Rule 2: Switch Different Control Port Inputs On to Connect and Disconnect
a Program
This rule is used to connect and disconnect a selected program when different codec control port
inputs are turned on.
1. Click the second rule in the Rules panel titled Connect a program when an input is
switched ON and disconnect it when another input is switched ON.
2. Click the drop-down arrows to select the control port input used to connect and the alternative
input for disconnecting.
3. Click the drop-down Program arrow to select an individual program which will be connected
and disconnected by the change in the control port input states.
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4. Check the Rule summary and click Create Rule to save the settings.
Rule 3: Synchronise Local Control Port Input Status with a Remote Relay
Output
Use this rule allow a local codec's control port input to change the state of a remote relay output.
1. Click the third rule in the Rules panel titled Synchronise a local input to a remote relay.
2. Click the drop-down arrow to select the local control port input used to control a remote relay
output.
3. Check the Rule summary and click Create Rule to save the settings.
Rule 4: Toggle a Relay Output with each Change in Connection Status
This rule is used to toggle a codec's control port relay output each time a program connects and
disconnects.
1. Click the fourth rule in the Rules panel titled Toggle a relay based on a connection's
status.
2. Click the drop-down Relay arrow and select the relay output you want to toggle.
3. Click the drop-down Program arrow to select a specific program which will affect the relay
toggle function, or use the default setting whereby any program will toggle the relay output.
4. Check the Rule summary and click Create Rule to save the settings.
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Deleting Rules
1. Open the Toolbox HTML5 Web-GUI and click Rules in the Menu Bar to display the Rules
panel.
2. Click to select the rule you want to delete.
3. Click the Delete button.
4. Click Yes in the confirmation dialog.
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18.13
Bridge-IT Manual
Upgrading Codec Firmware
To download the latest codec firmware visit http://www.tieline.com/Support/Latest-Firmware.
Firmware Upgrades
The following procedure explains how to perform codec firmware upgrades with a downloaded
firmware file saved to your PC.
1. Open the Toolbox HTML5 Web-GUI and click Settings in the Menu Bar, then click
Firmware to display the Firmware panel.
2. Click Browse firmware to search for the firmware for your codec and download it to your
computer.
3. Once the firmware has been saved, click Update from a selected file in the Firmware
panel.
4. Select the .bin file you are using to perform the upgrade and click Open to start the upgrade.
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155
Front Panel Configuration Tasks
The following sections explain how to configure codec settings using the front panel LCD screen
and KEYPAD.
19.1
Configuring IP via the Front Panel
Checking IP Address Details in the Codec
1.
2.
3.
4.
5.
Press the Home
button to return to the home screen.
Use the navigation buttons on the front panel to select Settings and press the
button.
Select Unit and press the
button.
Select ETH1 and press the
button.
IP address details and other relevant information is listed. Use the arrow up
and down
buttons to scroll and view all details listed.
Important Note: See the Configuring IP Connections sections for more details about IP
connections. For assistance with configuration of IPv4 or IPv6 network connections contact
your IT Administrator.
Ethernet and VLAN Configuration Options
The codec features a Ethernet port and up to four additional VLAN interfaces.
VLAN interfaces have features similar to physical Ethernet ports. However, your network
administrator will need to configure VLAN support throughout your network for VLANs to be
supported in your codec.
As an example, when only one physical Ethernet interface is available, VLANs can be used to
separate codec Control and Streaming functions if required.
Following are a range of Ethernet and VLAN settings which can be configured in the LAN menu.
After completing configuration ensure you navigate to Apply Setting and press the
button to
apply the new settings.
Configure an IPv4 DHCP Address
By default the codec is programmed for DHCP-assigned IP addresses. DHCP IP addresses are
automatically assigned and can change each time you connect to your Internet Service Provider or
by a router on your local area network (LAN).
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1.
2.
3.
4.
Press the SETTINGS
button.
Select LAN and press the
button.
Use the down
navigation button to select ETH1 or a VLAN interface.
Select Usage and choose the appropriate control and/or streaming mode for the connection,
then press the
button.
5. Select IPv4 and press the
button.
6. Select DHCP and press the
button.
7. Use the up
navigation button to scroll to the top of the menu and select Apply Setting,
then press the
button to confirm the new settings.
Configure a Static IPv4 Address
Static IP addresses are fixed addresses which are recommended for studio installations. Using a
static IP address ensures remote codecs can connect reliably using the same IP address over time.
1.
2.
3.
4.
5.
6.
7.
8.
Press the SETTINGS
button.
Select LAN and press the
button.
Use the down
navigation button to select ETH1 or a VLAN interface.
Select Usage and choose the appropriate control and/or streaming mode for the connection,
then press the
button.
Select IPv4 and press the
button.
Select Static and press the
button.
Navigate to v4 Static and enter the IP address, then press the
button.
Navigate to v4 Snet and enter the Subnet Mask, then press the
button.
9. Navigate to v4 Gway and enter the Gateway details, then press the
button.
10.Use the up
navigation button to scroll to the top of the menu and select Apply Setting,
then press the
button to confirm the new settings.
11.Check the Unit Details menu to ensure the new static IP address has been entered
correctly.
IPv6 Address Assignment
There are three IPv6 settings available for the Ethernet port and any VLANs which are configured.
1. Auto: An address is automatically assigned to the codec when you connect the codec to an
IPv6 router. This process is similar to how an IPv4 DHCP address is assigned.
2. Manual: Select to manually enter IPv6 address details.
3. Off: Select to ignore IPv6 address details.
Important Note: Select Off if you are not using IPv6 to connect to another device. This
ensures your codec will attempt to connect using IPv4 at all times.
To adjust this setting:
1. Press the SETTINGS
button.
2. Select LAN and press the
button.
3. Use the down
navigation button to select ETH1 or a VLAN interface.
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4. Select IPv6 and press the
button.
5. Select Auto, Manual or Off and press the
157
button.
By default the codec is configured to allow the codec to automatically receive IPv6 address
information from an IPv6 enabled router.
Manual IPv6 Address Assignment
Select Manual mode using the previous procedure and enter information into the v6 Static
(Address), v6 Prefix and IPv6 Gway fields in the codec to manually configure address details.
DNS Server
It is possible to specify Domain Name Server (DNS) settings to allow easy look up of codecs within
the specified DNS Addresses or Domains section within the web-GUI. This feature can be turned
on or off in the LAN codec menu.
1.
2.
3.
4.
5.
Press the SETTINGS
button.
Use the navigation buttons on the front panel to select LAN and press the
Use the down
navigation button to select ETH1 or a VLAN interface.
Use the down
navigation button to scroll to Auto DNS.
Press the
button to toggle between Yes and No.
button.
VLAN ID (VLAN configuration only)
The VLAN ID is encapsulated in IP packets to facilitate routing throughout your network.
1.
2.
3.
4.
5.
Press the SETTINGS
button.
Use the navigation buttons on the front panel to select LAN and press the
button.
Use the down
navigation button to select a VLAN interface.
Select Usage and press the
button.
Select the mode of operation for this VLAN (e.g. Control & Streaming, Streaming only,
Control Only) and press the
button.
6. Use the down
navigation button to scroll to VLAN ID.
7. Press the
button to enter a number between 1-4094 inclusive.
8. Press the
button to confirm this setting.
VLAN Priority (VLAN configuration only)
The VLAN Priority setting represents a prioritization scheme for forwarding data packets throughout
Virtual Local Area Networks.
1.
2.
3.
4.
5.
Press the SETTINGS
button.
Use the navigation buttons on the front panel to select LAN and press the
button.
Use the down
navigation button to select a VLAN interface.
Select Usage and press the
button.
Select the mode of operation for this VLAN (e.g. Control & Streaming, Streaming only,
Control Only) and press the
button.
6. Use the down
navigation button to scroll to Priority.
7. Press the
button to enter a number from 0 to 7 inclusive.
8. Press the
button to confirm this setting.
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VLAN Interface (VLAN configuration only)
This setting applies the VLAN settings to a physical Ethernet port in the codec. Only one physical
Ethernet port is available which cannot be reconfigured.
19.2
Selecting an Algorithm
The codec offers a range of high quality algorithm options as well as 16 Bit 22kHz linear audio at
less than 12 ms encode delay for high quality, uncompressed audio.
All Bridge-IT and Bridge-IT XTRA codecs include Opus, MPEG Layer 2, G.711 and G.722
algorithms, as well as the AAC suite of algorithms and Tieline Music and MusicPLUS as standard.
Music and Music PLUS are optimized for wired and wireless IP connections.
aptX® Enhanced is included in Bridge-IT XTRA and can be purchased separately as a license
upgrade in Bridge-IT codecs.
Bridge-IT Algorithm Encode License Options
AAC-LD, AAC-ELD, LC-AAC, HE-AAC v.1 and HE-AAC v.2
algorithms
16 bit and 24 bit aptX® Enhanced algorithm
Included
* Option available for purchase separately if required.
Bridge-IT
Bridge-IT XTRA
*
Note: Bridge-IT has a range of default connection profiles that make it very simple to easily program
your codec to connect using all available algorithms. See Choosing Dialing Profiles for more details.
Overview of Tieline Algorithms
1. The Tieline Music algorithm is optimized for audio bit rates as low as 19.2kbps with only a 20
millisecond encode delay. It offers 15 kHz mono from 24Kbps to 48Kbps.
2. Tieline MusicPLUS delivers up to 20 kHz mono from 48kbps upwards. It can also deliver up to 20
kHz stereo from 96kbps upwards, offering huge savings on your IP data bills and outstanding
audio quality.
Overview of AAC Algorithms
Important Notes: AAC algorithms are only available in Bridge-IT if the AAC license has
been purchased and uploaded into the Bridge-IT codec. For more information see Installing
Software Licenses.
AAC-LC
LC-AAC is optimised for audio bit-rates of 64Kbps per channel or higher using a sample rate of
48kHz. Tieline recommends using LC-AAC instead of HE-AAC if bandwidth of 64Kbps or higher
per channel is available, to optimise audio quality. If lower than 64Kbps is available, consider
using HE-AAC, Tieline Music or Tieline MusicPLUS.
AAC-HE
Codecs include both HE-AAC v.1 and HE-AAC v.2, which are optimized for low bit rate
connections. Selection of HE-AAC v.1 and v.2 is automatically managed within the codec, so
only AAC-HE is displayed on the screen. When used for mono connections, HE-AAC v.1
performs best at bit rates of 24kbps per channel or higher. HE-AAC v.1 is also used for stereo
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connections when audio connection bandwidth is 48kbps or higher.
HE-AAC v.2 is used for stereo connections when audio connection bandwidth is below 48kbps
and is capable of delivering 15kHz quality stereo audio at audio bit rates as low as 24kbps.
A sample rate of 32kHz is used in the codec's default profiles to achieve ultra-low bit-rate
connections, but this is adjustable to 44.1kHz or 48kHz if required.
AAC-LD
AAC-LD (Low Delay AAC), AAC-ELD (Enhanced Low Delay AAC) and AAC-ELDv 2 are
optimized for low latency real-time communication. AAC-LD is suited to bit rates of 96kbps or
higher for stereo audio.
AAC-ELD
AAC-ELD is optimised for high quality stereo connections from 48 - 96kbps and performs better
at these bit rates when compared with AAC-LD.
AAC-ELD v 2
For stereo connections below 48kbps AAC-ELD v2 will deliver better performance than AACELD down to 24kbps.
Overview of aptX® Enhanced Audio Coding
Important Notes: aptX® Enhanced is only available if the aptX® Enhanced license has
been purchased and uploaded into the codec. For more information see Installing Software
Licenses.
aptX® Enhanced audio coding is used by thousands of radio stations to deliver very low delay audio
for studio to transmitter links, audio distribution and remote broadcasts. It delivers outstanding audio
quality with exceptionally low delay across a range of IP networks. It is ideal for high quality studioto-transmitter links and audio distribution.
32kHz, 44.1kHz or 48kHz sampling rates are available at either 16 bit or 24 bits per sample. aptX®
Enhanced has a minimum connection bit-rate of 128Kbps per channel and offers 10Hz to 24kHz
frequency response. 24 bit, 48kHz aptX® Enhanced at the maximum bit-rate of 576Kbps delivers
>120dB of dynamic range.
Overview of Opus Algorithm
Opus is a highly versatile open source audio coding algorithm. It incorporates technology from the
well-known SILK and CELT codecs to create a low latency speech and audio codec. It is a variable
bit rate algorithm ideal for live broadcast situations because of its capacity to deliver high quality,
real-time Audio over IP (AoIP) at low bit rates. Visit http://www.opus-codec.org for more info.
There are three Opus algorithm configurations available:
Algorithm
Recommended connection for on-air use
Opus Voice
Opus Mono
Opus Stereo
High quality low bit rate remotes (9.6kbps -64kbps)
Very high quality mono remotes, STLs and audio distribution (48kbps -128kbps)
Very high quality stereo remotes, STLs and audio distribution (64kbps -256kbps)
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Configuring an Algorithm in the Codec
1.
2.
3.
4.
5.
6.
7.
Press the HOME
button to return to the Home screen.
Use the navigation buttons on the front panel to select Connect and press the
button.
Select IP and press the
button.
Select your preferred IP Session mode and press the
button.
Use the down
navigation button to select Setup and press the
button.
Navigate to Algor'm and press
.
Select the mono or stereo algorithm that you want to connect with and press
.
How do I choose the right algorithm?
The algorithm you select will not only affect the quality of the broadcast but it will also contribute to
the amount of latency or delay introduced. For example, if MP2 algorithms are used, program
delays will be much longer than when using Tieline Music or MusicPLUS algorithms. This is due to
the additional inherent encoding delays involved when using MP2 algorithms. This can be a major
consideration for live applications that integrate remote-crosses into a broadcast. The algorithm you
choose to connect with will also depend upon:
·
·
·
·
The codecs you are connecting to (Tieline versus non-Tieline)
Whether you are creating multi-unicast connections.
Whether you are connecting using SIP or not.
The uplink bandwidth capability of your broadband connection.
Important Notes: Music and MusicPLUS algorithms cannot be used over SIP connections.
Use MP2 algorithms at 64kbps mono or 128kbps stereo for high quality connections when
using SIP, or use G.711 and G.722 if required. Tieline G3 codecs do not contain AAC
algorithms so choose one of the other available algorithms if you are connecting to a G3
codec.
It can be a good idea to listen to the quality of your program signal using each algorithm and to see
how it sounds when it is sent at different connection bit-rates (as well as different FEC and jitterbuffer millisecond settings). This will assist you to determine what the best algorithm setting is for
the connection you are setting up. Please see the following table for details on the connection
requirements of the different algorithms available.
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Algorithm
Audio
Bandw idth
Algorithm ic
Delay
IP bit rate
per channel
IP overAudio Quality and
head per Features
connectio
n
PCM/Linear
(Uncompressed)
16/24 bit
up to
24kHz
0ms
sample rate x
bits per
sample x no.
channels
80kbps
Tieline Music
Up to
15kHz
20ms
24 kbps
minimum
16kbps
Tieline MusicPLUS
Up to
22kHz
20ms
48 kbps
minimum
(Optimised
for 64kbps
per audio
channel)
16kbps
G.711
3kHz
1ms
64kbps
minimum
80kbps
G.722
7kHz
1ms
64kbps
minimum
80kbps
MPEG Layer 2
Up to
22kHz
24 to
36ms
64kbps
minimum
8.5 13.3kbps
LC-AAC
Up to
15kHz
64ms
64kbps
15kbps
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Recom m ended
applications for on-air
use
· Full bandw idth,
· Extremely high quality
perfect audio
uncompressed audio
quality for voice
for STLs and audio
and music
distribution.
· No error
· Ideal for fiber or high
concealment/
bandw idth links.
correction or
artefacts
· High quality voice · Great for live voice or
and music
music remotes as w ell
· Very low delay
as STLs and audio
at low bit rates
distribution w ith limited
connection bandw idth
(e.g. 3G w ireless)
· Suitable w hen
bidirectional
communication
betw een announcers
is required
· Very high quality · Very high quality, very
voice and music
low delay STLs and
· Very low delay
audio distribution
at low to
· Remote connections
moderate bitable to achieve
rates
48kbps for each audio
channel
· Suitable w hen
bidirectional
communication
betw een announcers
is required
· Low quality 3kHz · Highly compatible w ith
POTS phone
other brands of audio
quality audio
codec
· Very low delay
· Low quality and used
at moderate bit
generally for
rates
compatibility
· Good quality
· Highly compatible w ith
7kHz voice
other brands of audio
· Better quality
codec
than a standard
· Good voice quality
POTS phone call
audio for remotes and
· Very low delay
other voice quality
at moderate bit
applications
rates
· Very high quality · Highly compatible w ith
voice and music
other brands of audio
· Low to moderate
codec
delay at
· Very high quality audio
moderate to high
for remotes, STLs and
bit rates
audio distribution
· High quality voice · Voice or music
and music at
remotes as w ell as
low est bit rate;
STLs and audio
better quality at
distribution w here
higher bit rates
some delay is tolerable
· Moderate delay
· Tieline Music or
at moderate to
MusicPLUS deliver
high bit rates
low er delay
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HE-AAC v.1
Up to
15kHz
128ms
48kbps
7.4kbps
HE-AAC v.2
Up to
15kHz
128ms
Minimum
16kbps
(Mono);
24kbps
(stereo)
7.4kbps
AAC-LD
Up to
20kHz
20ms at 48kbps
48kHz
minimum
30kbps
AAC-ELD
Up to
20kHz
15-30ms 24 kbps
minimum
15-30kbps
AAC-ELDv.2
Up to
20kHz
35ms
Pending
release
aptX Enhanced
10Hz24kHz
2.5ms at 128kbps
80kbps
48kHz
minimum
(16bit; 32kHz)
to 288kbps
(24bit;48kHz)
Opus
4Hz20kHz
20ms
Pending
release
9.6-256kbps
16kbps
· High quality voice · Live voice or music
and music at the
remotes as w ell as
low est bit rate;
STLs and audio
better quality at
distribution w ith limited
higher bit rates
connection bandw idth
· Low to Moderate · Use w hen bidirectional
bit rates
communication
· High delay
betw een announcers
is not required
· High quality voice · Used for DAB+ radio
and music
streaming
· Low bit rates
· Ideal for low bit rate
· High delay
remotes
· Use w hen bidirectional
communication
betw een announcers
is not required
· Very high quality · Very high quality, very
voice and music
low delay STLs and
· Very low delay
audio distribution
at low to
· Remote connections
moderate bit
able to achieve
rates
48kbps for each audio
channel requiring
· Suitable w hen
bidirectional
communication
betw een announcers
is required
· Very high quality · Great for live voice or
voice and music
music remotes
· Very low delay
· Suitable w hen
at low bit rates
bidirectional
communication
betw een announcers
is required
· High quality voice · Great for live voice or
and music
music remotes w here
· Low delay at low
limited connection
bit rates
bandw idth is available
· Suitable w hen
bidirectional
communication
betw een announcers
is required
· Very high quality · Ideal for STLs and
voice and music
audio distribution
· Extremely low
w here high
delay at high bit
connection bandw idth
rates
is available and very
· Highly cascade
low delay is highly
resilient
desirable.
· Resilient w ith multiple
encodes/decodes
w hen required
· Very high quality · "Opus Voice" is ideal
voice and music
for high quality, and
· Very low delay
low delay voice quality
at low bit rates
remotes at extremely
low bit rates.
· "Opus Mono" and
"Opus Stereo" are
perfect for high fidelity
remotes, STLs and
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audio distribution at
higher bit rates
Algorithm Selection Guide
Algorithm
Very Low
Delay
Moder-
Excell-
Preferr-
ate to High
Delay
ent Perform ance at
Low
ed for Live
Rem otes
Preferred for
Highly
STLs and
Com patAudio
ible w ith
Distribu-tion other Codecs
Bit rates
Linear/PCM
Opus
Tieline Music
Tieline MusicPLUS
apt-X Enhanced
LC-AAC
HE-AACv1
HE-AACv2
*
AAC-LD
AAC-ELD
AAC-ELDv2
MPEG Layer 2
G.722
G.711
* Use with caution for remotes due to high delay; not suitable when bidirectional communications is
required.
Sampling Rates
When selecting linear (PCM) uncompressed audio or AAC, MPEG and aptX® Enhanced algorithms,
it is possible to select different either 32kHz, 44.1kHz and 48kHz sample rates as required. Tieline
Music runs at 32kHz sampling and MusicPLUS runs at 48kHz sampling. G.711 and G.722 will
always run at a 32kHz sampling rate (downsampled to 8kHz and 16kHz respectively).
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19.3
Bridge-IT Manual
Configuring the Jitter Buffer
Jitter, (also known as latency or delay), is the amount of time it takes for a packet of data to get
from one point to another. A jitter buffer is a temporary storage buffer used to capture incoming data
packets. It is used in packet-based networks to ensure the continuity of audio streams by
smoothing out packet arrival times during periods of network congestion. Data packets travel
independently and arrival times can vary greatly depending on network congestion and the type of
network used, i.e. LAN versus wireless networks. The concept of jitter buffering is displayed visually
in the following image.
Jitter buffer management is encompassed within Tieline's SmartStream IP technology which can:
·
·
·
·
·
Remove duplicate packets.
Re-order packets if they arrive out-of-order.
Repair the stream in the event of packet loss (error concealment).
Manage delay dynamically based on current network congestion.
Manage forward error correction (FEC).
Tieline codecs can be used to program either a fixed or automatic jitter buffer and the setting you
use depends on the IP network you are connecting over. Over LANs, WANs and wireless networks
the automatic jitter buffer generally works well. It adapts automatically to the prevailing IP network
conditions to provide continuity of audio streaming and minimizes delay.
A fixed jitter buffer is preferable over satellite connections to ensure continuity of signals.
CAUTION: If a Tieline codec connects to a device that is using non-compliant RTP streams
then the last fixed setting entered into the codec will be enabled (default is 500ms). Noncompliant devices include some other brands of codec, web streams and other devices.
Tieline ‘Auto Jitter Buffer’ Settings
Least Delay: This setting attempts to reduce the jitter buffer to the lowest possible point, while still
trying to capture the majority of data packets and keep audio quality at a reasonable level. This
setting is the most aggressive in its adaptation to prevailing conditions, so jitter buffer may vary
more quickly than with the other settings. It is not recommended in situations where jitter variation is
significant and/or peaky. (E.g. 3G/multi-user wireless networks). It is best for stable and reliable
links such as dedicated or lightly-loaded WAN/LANs.
Highest Quality: This setting is the most conservative in terms of adapting down to reduce delay.
The jitter-buffer setting will actually stay high for a longer period after a jitter spike is detected – just
in case there are more spikes to follow. This setting is best used where audio quality is most highly
desired and delay is not so critical. Unless delay is irrelevant, this setting is also not recommended
over peaky jitter networks (such as 3G) and is best used on more stable networks where large jitter
peaks are not as common.
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Best Compromise: This (default) setting is literally the midpoint between the jitter buffer levels that
would have been chosen for the Highest Quality and Least Delay settings. It is designed to provide
the safest level of good audio quality without introducing too much extra delay.
Good Quality and Less Delay: These two settings lie between the mid-point setting of Best
Compromise and two settings Highest Quality and Least Delay. They indicate a slight preference
and may assist in achieving better performance from a connection without incurring extreme delays
in transmission or packet loss.
Which Algorithms can use Automatic Jitter Buffering?
The following table provides an overview of which algorithms are capable of using the automatic jitter
buffer feature over SIP and non-SIP connections.
Algorithm
Non-SIP Connections
SIP Connections
Linear (Uncompressed)
Tieline Music
Tieline MusicPLUS
G.711
G.722
MPEG Layer 2
LC-AAC
HE-AAC v.1
HE-AAC v.2
AAC-LD
AAC-ELD
Opus
aptX Enhanced
Configuring Automatic Jitter Buffering (Default Setting)
1.
2.
3.
4.
5.
6.
7.
8.
Press the HOME
button to return to the Home screen.
Use the navigation buttons on the front panel to select Connect and press the
Select IP and press the
button.
Select your preferred IP Session mode and press the
button.
Use the down navigation button to select Setup and press the
button.
Navigate to Jitter and press
.
Select Auto Adapt and press
.
Select your preferred jitter buffer setting and press
.
button.
How to get the Best Jitter Buffer Results
When configuring automatic jitter buffer settings, establish the IP connection for a while before
‘going live’, to let the codec evaluate the prevailing network conditions. The initial jitter buffer setting
when a codec connects is 500ms and it is kept at this level for the first minute of connection (as
long as observed delay values are lower than this point).
After the initial connection period the jitter buffer is adapted to suit the current network conditions
and is usually reduced. Establish a connection for at least 5 minutes prior to broadcasting, so that
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the codec has been provided with enough jitter history to ensure a reliable connection.
There are five states or stages that jitter buffer may display and these can be observed in the
connection status screen by selecting HOME
> Cxns button while connected and use the
down
and up
navigation buttons to scroll through connection statistics.. The first four stages
are observed in “auto” jitter buffer mode.
1. Stabilization period (a1): A few seconds at the start of a connection where no action is
taken at all while the establishment of a stable connection means analysis of jitter data is not
valid.
2. Stage 2 (a2): A compatibility check to ensure the RTP connection is compliant and RTP
clocks are synchronized enough to perform jitter analysis.
3. Stage 3 (a3): If the compatibility check is successful, this is the analysis hold-off period.
During a minute, the jitter buffer is held at a safe, fixed value of 500ms while enough history is
recorded to start jitter buffer adaptation.
4. Stage 4 “live” (A): This is where the codec determines it is safe enough to start
broadcasting using the auto-jitter buffer level. We recommend running the codec for a few
more minutes to obtain a more comprehensive history of the connection’s characteristics.
5. Fixed (F): This state is displayed if the jitter buffer is fixed.
Important Note: The jitter buffer setting in the codec can only be adjusted when a
connection is off-line. Automatic jitter buffering is disabled for a Raw (linear
uncompressed) audio connection.
Auto Jitter Buffer and Forward Error Correction (FEC)
If forward error correction is configured then additional data packets are sent over a connection to
replace any data packets lost. There is no need to modify jitter buffer settings if you are sending
FEC data, only if you are receiving FEC data.
The jitter buffer depth on the receive codec needs to be increased if forward error correction is
employed. We recommend you add 100ms to the jitter buffer on a codec receiving FEC at a setting
of 20% and 20ms at a setting of 100%.
Tieline’s auto jitter buffer detects the amount of FEC that is being used and automatically
compensates to increase the codec jitter buffer if FEC is being used.
Fixing Jitter Buffer Settings
The default jitter-buffer setting in Tieline codecs is 500 milliseconds. This is a very reliable setting
that will work for just about all connections. However, this is quite a long delay and we recommend
that when you set up an IP connection you test how low you can set the jitter-buffer in your codec.
1.
2.
3.
4.
5.
6.
7.
8.
Press the HOME
button to return to the Home screen.
Use the navigation buttons on the front panel to select Connect and press the
button.
Select IP and press the
button.
Select your preferred IP Session mode and press the
button.
Use the down navigation button to select Setup and press the
button.
Navigate to Jitter and press the
button.
Select Fixed Buffer and press the
button.
Use the numeric KEYPAD to enter the fixed buffer value in milliseconds and press the
button.
If you change the jitter buffer setting in a codec it will only adjust to the new level when link quality is
high (e.g. above 70%). This is done to ensure audio quality is not compromised. When manually
configuring the jitter-buffer delay in a codec it is necessary to think carefully about the type of
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connection you will be using. Following is a table displaying rule of thumb settings for configuring
jitter-buffer delays into your codec.
Connection
Jitter-Buffer Recommendation
Private LAN
60 milliseconds
Local
100 - 200 milliseconds
National
100 - 300 milliseconds
International
100 – 400 milliseconds
Wireless Network
250 - 750 milliseconds
Satellite IP
500 - 999 milliseconds
Important Note: The preceding table assumes Tieline Music is the algorithm in use. Do not
use PCM (uncompressed) audio over highly contended DSL/ADSL connections without
enough bandwidth to support the high connection bit-rates required.
19.4
Configuring Forward Error Correction
Forward Error Correction (FEC) is designed to increase the stability of UDP/IP connections in the
event that data packets are lost. FEC works by sending a secondary stream of audio packets over
a connection so that if your primary audio stream packets are lost or corrupted, then packets from
the secondary stream can be substituted to replace them. The amount of FEC required depends on
the number of data packets lost over the IP connection.
Both the local and remote codec FEC settings can be configured in the codec before dialing. These
settings can also be changed ‘on the run’ while the codecs are connected. FEC should only be
used if link quality displayed on the codec is below S:99 R:99, as it is of no benefit otherwise.
Configuring FEC into the Codec
1. Press the HOME
button to return to the Home screen.
2. Use the navigation buttons on the front panel to select Connect and press the
3.
4.
5.
6.
7.
8.
9.
Select IP and press the
button.
Select Tieline and press the
button.
Use the down
navigation button to select Setup and press the
button.
Navigate to FEC and press
.
Select the local codec FEC setting in the Local FEC screen and press
.
Select the remote codec FEC setting in the Remote FEC screen and press
Check that the settings are correct in the RTP Settings screen.
button.
.
The four FEC settings in Tieline codecs are outlined in the following table with their bit rate ratios.
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FEC Setting
Bit rate Ratios
Connection Use
100%
A simultaneous dual-redundant
stream (1:1 ratio) is sent from the
codec. Twice the connection bit
rate is required to operate the
codec using the 100% setting.
E.g. if your connection is
14,400Kbps, you will require an
additional
14,400
Kbps
of
bandwidth to allow for the FEC
data stream.
Recommended to be used over
wireless
and
international
connections.
50%
Additional data is sent by FEC in
a ratio of 2:1.
Recommended for international &
national connections
33%
Additional data is sent by FEC in
a ratio of 3:1.
Recommended for national and local
connections.
20% (Highest Additional data is sent by FEC in
delay)
a ratio of 5:1.
Recommended for local and LAN
connections.
Off
Recommended for wired LAN
connections & managed T1 & E1
connections for STLs that have
connections that aren’t shared &
have quality of service (QoS).
(Lowest
delay)
FEC is off in the codec and the
connection bandwidth is equal to
the connection bit rate setting in
the codec.
Important Note: FEC can only be configured for use with the Music and MusicPLUS
algorithms.
How does FEC work?
If you enter a FEC setting of 20% and you are losing one packet in every five sent, the lost packet
will be replaced by FEC to maintain the quality of the connection. If you are losing more packets
than this, say one in three, it will be necessary to increase the FEC setting to 33% to compensate.
Note: There is an inverse relationship between FEC settings and the jitter-buffer millisecond setting
that you use for IP connections.
So why not use 100% FEC every time? The answer is because you need twice the bit rate to
achieve full redundancy and depending on the link conditions, this could potentially cause more
drop-outs because of network congestion than it fixes. Here is a simple rule to remember: Your
maximum uplink speed is all the bandwidth you have to play with. As a rule of thumb, try not to
exceed more than 80% of your maximum bandwidth. If your link is shared, be even more
conservative.
You should also consider the remote end too. What is their maximum upload speed? Is the
connection shared at either end? Your bit rates, FEC settings and buffer rates must be preconfigured at both ends before you connect, so it's always better to set your connection speed and
balance your FEC according to the available uplink bandwidth at each end for best performance.
As an example, if you want 15 kHz mono (using the Tieline Music Algorithm) you will need at least
a 24kbps connection for audio. Adding 100% FEC will add another 24kbps making your bit rate
48kbps plus some overhead of around 10kbps is required. If you're on a 64kbps uplink, you should
consider reducing your FEC to minimise the likelihood of exceeding your bandwidth capacity.
Here is another example, if you want 15 kHz stereo, you need at least 56kbps for the audio. 100%
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FEC requires at least 112kbps and 50% FEC requires at least 84kbps. If your uplink speed is
256kbps and you're on a shared connection, then choosing a lower FEC setting of 20%-33% may
give you better results.
19.5
Configuring Encode/Decode Direction
By default the codec is configured to encode and decode data. However, it is possible to encode or
decode audio data only. This is useful for:
· Conserving connection bandwidth when unidirectional data streaming is required.
· Lowering data costs.
· Increasing overall connection reliability.
Configure the transmitting codec to encode only, and configure the receive codec to decode only
when using this feature.
To adjust this setting:
1.
2.
3.
4.
5.
6.
7.
19.6
Press the HOME
button to return to the Home screen.
Use the navigation buttons on the front panel to select Connect and press the
Select IP and press
.
Select your preferred IP Session mode and press the
button.
Use the down
navigation button to select Setup and press
.
Navigate to Dir and press
.
Select Encode Only or Decode Only and press
.
button
Enabling Relays & RS232 Data
The codec supports both in-band and out-of-band data depending on the algorithm you are using.
(See RS232 Data Configuration for more details). RPTP data is automatically enabled when using
the Tieline Music or MusicPLUS algorithms over IP. It is also possible to enable synchronized outof-band data using any algorithm via the RS232 data port on the codec.
Data must be enabled to activate contact closure operation and RS232 data.
1.
2.
3.
4.
5.
6.
Press the HOME
button to return to the Home screen.
Use the navigation buttons on the front panel to select Connect and press the
button.
Select IP and press the
button.
Select Tieline (or Sessionless) as the IP Session mode and press the
button.
Use the down
navigation button to select Setup and press the
button.
Navigate to Data and press
to toggle between Enabled and Disabled (Note: default
setting is Disabled).
Configuring Control Port Contact Closure Operation
The Rules panel on the web-GUI can be used to configure switch inputs and relay outputs. See the
section titled Creating Rules for more information.
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About Relays
The codec has two CMOS solid state relays for the control of equipment, consisting of two relay
closures and two opto-isolated outputs.
Inputs
The input signal is referenced to chassis ground, i.e. the ground reference terminal on the
terminal block is connected the chassis. The input device is a high impedance CMOS device
with a 330 ohm pull-up resistor to +5 volts.
Operation is as simple as joining the input pin to the ground terminal. This can be via a remote
relay contact or the open circuit collector of a transistor or FET. DO NOT feed voltages into the
inputs.
Outputs
CMOS field effect transistors switch a low impedance path between the two pins when
activated. These are opto-isolated and floating above ground. It is important to current-limit the
source as damage will result where the current exceeds 100mA peak-to-peak. No more than 48
volts peak-to-peak should be used as a safety precaution. The resistance of the CMOS element
is approximately 25 ohms in the ON state.
Relay Operation and Pinouts
A closing contact across Input 1 or 2 (pins 5 or 6) to Ground (pin 7) will provide a closing contact on
the remote codec Output 1 (pins 1 and 2) or Output 2 (pins 3 and 4).
For multi-unicast connections to multiple codecs, a contact closure will appear on each of the
compatible (non-G3) remote codecs' corresponding contacts. I.e. Input 1 shorted, Output 1 contacts
on all codecs closed.
Pins
1
2
3
4
5
6
7
Pin Function
Output 1
Output 1
Output 2
Output 2
Input 1
Input 2
Ground
Important Note: For more information about how to configure relay operations with a
PC using the Toolbox web-GUI, please see Creating Rules.
Configuring the Codec to Send RS232 Data
Once Data is enabled, the codec can also be connected to external devices and send RS232-
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compatible data via the serial port on the rear panel of the codec.
1. Press the SETTINGS
button.
2. Navigate to System and press
.
3. Select RS232 Cfg and press
.
4. Use the navigation buttons to select the correct baud rate.
5. Select Enable for flow control and press
to save all settings.
RS232 Pin-outs and Data Connections
Pin INTERFACE Female
DB9 (RS232) DCE
DATA Male DB9
(RS232) DTE
1
No Connection
No connection
2
TX Data
RX Data
3
RX Data
TX Data
4
No connection
No connection
5
Signal Ground
Signal Ground
6
No Connection
No connection
7
CTS
RTS
8
RTS
CTS
9
No connection
No connection
DB9 Male
Connector Pins
DB9 Female
Connector Pins
Important Notes:
· Codecs using Bridge-IT firmware lower than v2.8.xx cannot activate relays on Tieline G3
codecs or send RS232 data to them.
· It is important that you enable serial port flow control within the codec. Flow control
regulates the flow of data through the serial port. If disabled, data will flow unregulated and
some may be lost.
· Ensure you match the serial port baud rate to match the rate of the external device you are
connecting to. Ideally the settings on both codecs should match, or you could have data
overflow issues.
· Only the dialing codec needs to be configured to send RS232 data. Session data sent from
the dialing codec will configure all other compatible codecs (non-G3) you connect with.
· RS232 data can be sent from the dialing codec to all end-points of a multi-unicast
connection. Note: Bidirectional RS232 data is only available on the first connection dialed
when multi-unicasting.
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19.7
Bridge-IT Manual
Configuring TCP/UDP Ports
Tieline Codec Default Port Settings
By default, the codec uses a TCP session port to send session data and a UDP port to send audio.
The session port uses the TCP protocol because it is more likely to get through firewalls – ensuring
critical session data (including dial, connect and hang-up data) will be received reliably.
The default session and audio port settings in Tieline codecs, for both TCP and UDP connections,
are outlined in the Installing the Codec at the Studio section of the manual. This section also
contains useful information for configuring port forwarding and troubleshooting IP connections.
Changing Codec Port Numbers
Reasons for adjusting the port setting on your codec include:
· Having to create a path through gateways and firewalls.
· Another IP device is using the codec’s default port number.
· You are sharing a single IP address with multiple codecs and each codec requires a different
port number to perform port forwarding.
Configuring the Session and Audio Port Numbers used when Dialing a
Program
Codecs require matching port numbers to connect successfully. When you create a program the
session and audio ports can be adjusted from the defaults as required. Note: If there is a need to
change codec port settings please consult your organization’s resident IT professional. To adjust
port settings:
1.
2.
3.
4.
5.
6.
7.
8.
Press the HOME
button to return to the Home screen.
Use the navigation buttons on the front panel to select Connect and press the
button.
Select IP and press the
button.
Select an IP Session mode.
Use the down navigation button to select Setup and press the
button.
Navigate to either Session (session protocol) or Proto (audio protocol) and press
.
Select the session or audio ports you want to adjust and press
.
Use the numeric KEYPAD add a new port number and press
.
Configuring the Tieline Session Ports when Answering
To adjust the local Tieline session data port used by your codec to answer incoming calls:
1.
2.
3.
4.
5.
Press the HOME
button to return to the Home screen.
Use the navigation buttons on the front panel to select Settings and press the
Select Tieline Session and press
.
Navigate to Port or Alt. Port (alternative session port) and press
.
Adjust the setting and press the
button to store the new configuration.
button.
Audio Port Settings for Tieline Session Data and Sessionless IP Calls
The codec supports sessionless IP streaming, whereby the codec does not send Tieline session
data when attempting to connect. When using this mode you need to configure the "send" audio
port (codec port at the remote end of the link to which you are sending audio) and "return" audio port
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(port used by the local codec to receive audio from the remote codec).
It is also possible to configure the remote and local audio ports for a codec using Tieline session
data to establish IP connections. This may be required because some firewalls require symmetric
port configuration.
Sessionless Audio Port Configuration
When you select Sessionless as the Session Protocol:
·
·
·
·
The default value for both the Send and Return (audio) Ports is 9000
The range of values for the audio ports is 2000 to 65535
The audio port values can be set independently
Both audio ports can always be configured, i.e. there is no dependency on encode/
decode direction
"Tieline Codec" Port Configuration
If using the Tieline Codec setting for call establishment (i.e. Tieline session data is enabled),
you can also change the default audio ports if required.
· The default value for the Send (audio) Port is 9000
· The range of values for the Send Port is 2000 to 65535
· The default port value for the Return (audio) Port is Automatic. Note: Automatic
indicates that the codec will allocate the return port value and send this information to
the codec to which you are dialing
· The range of values for the Return Port is 2000 to 65535
Sessionless Multicast Connections
For a sessionless multicast server connection:
· Only the Send Port is available
· The default value for the port is 9000
· The range of values for the port is 2000 to 65535
For a sessionless multicast client connection:
· Only the Return Port is available
· The default value for the port is 9000
· The range of values for the port is 2000 to 65535
19.8
Configuring QoS for Broadcasts
It is possible for IP networks to differentiate between and prioritise data packets being transmitted
through routers across networks. This is useful because in modern data networks many different IP
services like email, voice, web pages, video and streaming music coexist within the same network
infrastructure.
Prioritising IP Data Packets when Broadcasting
Broadcast IP audio data packets can be configured for expedited or assured forwarding (Quality of
Service or QoS) when traversing different networks. Routers can also be configured to ignore these
forwarding priorities so they are not assured across all networks.
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The codec can be configured to prioritise IP data packets sent across a network by entering a value
into the Differentiated Services Code Point (DSCP) field within the header of data packets
transmitted by the codec over the network. Check with your IT administrator before changing this
setting. By default the codec is configured for Assured Forwarding and more details about DSCP are
available on Wikipedia at http://en.wikipedia.org/wiki/Dscp.
Configuring Bridge-IT for QoS
1. Press the SETTINGS
button.
2. Use the navigation buttons to select QoS and press the
button.
3. Press the
button and use the RETURN
button to delete numbers already entered,
then use the numeric KEYPAD to enter the new setting.
4. Press the
button to save the new setting.
See Configuring IP Settings for instructions on configuring this setting using the HTML5 Toolbox
web-GUI.
19.9
Configuring Data Packet Time-to-Live
Time-to-Live (TTL) is a value you can configure to set a finite life for data packets sent by the codec.
This avoids situations where packets can keep circulating through routers causing network
congestion.
The Time-to-Live setting is configurable and sets the maximum number of router hops allowable for
multicast data packets. In most situations the default value of 1 is used, to ensure packets are sent
through a single LAN router and not over multiple router hops and networks.
This setting is only used in Multicast Server mode.
1. Press the HOME
button to return to the Home screen, select Connect, then select IP
and press the
button.
2. Select Sessionless and press
3. Press the right
.
navigation button to select Multicast Server and press
.
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3. Press the down
navigation button to select Setup and press
4. Press the down
navigation button to select TTL and press
175
.
.
5. Press the RETURN
button to delete the current setting and use the numeric KEYPAD to
enter a new value. Press
to confirm the new setting.
19.10
Reset and Restore Factory Default Settings
There are several options in the Reset menu which allow you to restore factory default settings within
the codec.
Function
1 Reset Audio and
'Connect' Settings
2 Restore
Factory
Defaults
3 Delete Programs &
Call History
4 Reboot Codec
Description
Click to restore factory default settings for Audio and Connect
menu settings
Click to restore factory default settings, excluding user defined
programs and call history
Deletes custom programs and recent calls in the codec
5 Clear Logs
Deletes codec event and log history. Note: This should only be
performed if instructed to by Tieline support staff.
Click to restart the codec
Important Note: After restoring factory defaults, always reboot the codec using the
Reboot Codec function, not by removing power from the codec.
1. Press the SETTINGS
button.
2. Navigate to Reset and press the
button.
3. Navigate to the preferred option from those available and press the
4. Select Yes and press the
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button to confirm the menu function that you are performing.
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Reset and Restore Factory Defaults using the Web-GUI
The web-GUI can also be used to reset and restore factory defaults. See Reset Factory Default
Settings for in the web-GUI section of this manual for more details.
19.11
Installing Software Licences
Bridge-IT XTRA codecs include all software and algorithm feature options when purchased. Other
Bridge-IT codecs can be upgraded to include aptX® Enhanced encoding via an additional license
purchase and upgrade.
Contact Tieline at sales@tieline.com, or your favorite dealer, if you need to purchase a software
license upgrade. When a software license has been purchased there are two ways to perform an
upgrade:
1. Automatically download the software license from TieServer and perform the upgrade.
2. Download the new software license file onto your PC and upgrade using the web-GUI.
To install a software license using the Toolbox Java or HTML5 web-GUI options see Web-GUI
Software License Installation.
Checking Installed Licenses
The codec License Manager is used to view which licenses are installed in each codec. To view
the licenses installed in your codec:
1. Press the SETTINGS
button.
2. Use the navigation buttons to select Licenses and press the
button.
3. A list of all possible licenses is displayed and all licenses that have been installed have a tick
next to them.
Update and Install Licenses from the Codec
1. Navigate to Update from TieServer in the License Manager screen and press the
button.
2. The codec will contact TieServer and automatically install all valid licenses.
3. The screen will indicate the update is in progress and then confirm it has been completed
successfully.
4. Press the RETURN
button a few times until you return to the Home screen.
5. Use the navigation buttons to select Settings and press the
button.
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6. Navigate to Reset and press the
button.
7. Navigate to Reboot Codec and press the
button.
8. Select Yes and press the
button to reboot the codec.
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Reference
The following sections contain reference and troubleshooting information.
20.1
Installing the Codec at the Studio
Studio IP Streaming Setup for Tieline Audio Codecs
The following instructions are intended to help you configure your internet connection and Tieline
codecs at the studio to enable incoming calls over the internet from a remote Tieline codec. It is
assumed that you have a basic understanding of your IP network and how to configure IP devices. If
you have limited IT network knowledge, we recommend you engage the services of an IT
professional to install the public IP address and perform the Network Address Translation (NAT) and
port forwarding between the public internet and your private Local Area Network (LAN) at the studio.
Prerequisites
The following procedures are valid for:
· All firmware versions in the Genie and Merlin codec families.
· All Bridge-IT Basic and Pro and Bridge-IT XTRA codecs with firmware release v.2.x or higher.
· All Commander G3 and i-Mix G3 codecs.
Getting Started at the Studio
To perform a typical codec installation at the studio you will need to:
1. Contact your Internet Service Provider and organize a dedicated high speed broadband
connection at the studio for your codec with a public static IP address. Do not share this
connection with other devices.
2. Install your codec at the studio and attach an active RJ-45 LAN cable to the “LAN” or
“Ethernet” port on the rear of the codec. Please note:
· The green LED underneath the “LAN” or "Ethernet" port will illuminate and the orange
LED will flash steadily if you are connected to an active LAN connection.
· The Genie and Merlin families of IP codecs support two simultaneous Ethernet
connections.
3. If you are connecting a single codec to a router without a firewall you can enter the public IP
address, Subnet Mask and Gateway directly into the codec and your work is done. Note:
your Telco should be able to provide this information.
4. Alternatively, if you are connected to a router with a firewall, configure Network Address
Translation (NAT) in your router. NAT is performed between the public internet and your
private Local Area Network (LAN) by your router. Your remote codec sends IP data packets
to the studio router's public static IP address and the router performs NAT, which forwards
these data packets to the private IP address allocated by the router to your codec. As part of
this process we recommended you:
· Connect to your router using a web-browser.
· Configure it to allocate a static private IP address for each codec.
Important Note: The IP address may change if the codec is allocated a DHCP IP address
by the router and it loses power or is temporarily disconnected from the LAN. This will
cause problems for remote codecs attempting to dial and connect.
5. Ensure your router's firewall is configured with the relevant TCP and UDP IP ports open to
allow data traffic between your codec and the remote codec. The process is fairly simple if
you use the following procedure:
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a. Connect to your router using a web-browser.
b. Navigate to http://portforward.com/english/applications/port_forwarding/Tieline-G5/
default.htm (Note: when configuring a Commander or i-Mix G3 codec at the studio use
http://portforward.com/english/applications/port_forwarding/Tieline-G3/default.htm )
c. Click to select your router manufacturer from the list.
d. Next, click to select your router model from the list.
e. Follow the instructions to complete port forwarding
6. Visit www.portforward.com and download the port checking application to verify your router's
ports are open.
7. Configure the static IP address in your codec using the instructions in the next section. To
allow multiple codecs to share a single public static IP address behind a firewall and route
the calls correctly, your codecs and the firewall need to be configured similarly to the
example diagram which follows. Ensure the port, IP address, Subnet Mask and Gateway
settings in your codecs match those configured in your router.
Port Forwarding to 3 Studio Codecs Sharing a Public Static IP Address
Important Note:
· The most common studio configuration issue is a firewall which blocks the incoming and/
or outgoing TCP and UDP ports, or not configuring NAT and port forwarding correctly. The
following table lists the firewall ports you need to open for each model of Tieline codec if
they are dialing your router at the studio. If the remote codec is also connected to a LAN
with a firewall you may also need to open the ports at the remote end of the link to
connect successfully.
· Some firewalls require symmetric port configuration. The codec supports configuration of
the "send" audio port (codec port at the remote end of the link to which you are sending
audio) and "return" audio port (port used by the local codec to receive audio from the
remote codec).
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Commander G3/i-Mix G3
Firewall Ports
Bridge-IT/Bridge-IT XTRA
Merlin and Genie Codec
Families
TCP
UDP
TCP
UDP
TCP
UDP
IP1 Session IP1 Audio Port: Session Port Audio (Proto): Session Port:
Audio Port
Port: 9002
9000
(Sess): 9002
9000
9002
Stream 1: 9000
IP2 Session IP2 Audio Port: Web-GUI: 80 SIP Session:
Alternative
Audio Port
Port: 9012
9010
5060
Session: 9012 Stream 2: 9010
Toolbox
Toolbox
Alternative
SIP Audio:
Web-GUI: 80
Audio Port
Software: 5550 Software: 5550 Web-GUI: 8080
5004
Stream 3: 9020
SIP Session:
Alternative
Audio Port
5060
Web-GUI: 8080 Stream 4: 9030
SIP Audio:
Audio Port
5004
Stream 5: 9040
Audio Port
Stream 6: 9050
SIP Session:
5060
SIP Audio:
5004, 5006,
5008, 5010,
5012, 5014
Configuring a Static Public or Private IP Address in Bridge-IT (v.2.x firmware)
To enter a static IP Address into the codec for NAT:
1.
2.
3.
4.
5.
Press the HOME
button to return to the Home screen.
Use the navigation buttons to select Settings and press
.
Use the down navigation button to select LAN and press
.
Select Eth1 and navigate to IPv4 mode and press
.
DHCP is enabled by default. Select Static and press
.
6. The Static IP address menu is revealed after DHCP is disabled. Use the navigation buttons to
select v4 Static and press
.
7. Use the numeric KEYPAD to enter the IP address and press
to store the setting. Note:
use the * or # buttons to enter the periods in the IP address and use the RETURN
button
to delete any numbers already entered.
8. Enter changes to the v4 Subnet (Subnet Mask) or v4 Gway (Default Gateway) in the same
way if they are required (check with your network administrator for these settings).
9. After all changes have been made use the navigation buttons to scroll to the top of the menu
and select Apply Setting, then press the OK button to save all changes.
10. From the Home screen select Settings > Unit > Eth1 in the codec menus to ensure the
new static IP address has been entered correctly.
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Configuring a Static IP Address in Commander G3 and i-Mix G3 Codecs
To set up a static IP address in Commander G3 and i-Mix G3 codecs select Menu >
Configuration > Advanced > LAN settings > IP Setup > Setup > Static > IP Address >
[enter IP address] > press OK > Subnet Mask [enter Subnet Mask] > press OK > Gateway
[enter Gateway] > press OK > reboot the codec.
Record IP Address Details
IPv4 Static IP Address
IP Address
Subnet Mask
Default Gateway
IPv6 Mode: Manual
.
.
.
.
.
.
.
.
.
(Bridge-IT, Genie and Merlin codecs only)
IP Address
:
:
:
:
:
:
:
:
:
:
:
:
:
IPv6 Prefix Size
IPv6 Gateway
:
Getting Connected
Once the studio codec is configured you are now ready to receive an incoming call from the remote
codec over the internet. Always dial from the field codec to the studio codec over the internet unless
the remote codec is assigned a public static IP address and you know this address.
If you dial the studio using a cell-phone data network at the remote site you will not normally
experience any firewall or port blocking issues at the remote end of the link using default Tieline
ports.
Troubleshooting: How to Determine Where Firewall Port Blocking is
Occurring
If you find you are unable to either send or receive audio between the studio and remote codecs you
can use Tieline's Link Quality reading to diagnose where ports are being blocked. LQ can be
displayed on the front LCD screen of Tieline's Bridge-IT, Merlin and Genie codecs by selecting
Cxns, then select the connection you want to view and press the OK button. LQ readings are also
displayed on the home screen of all Commander and i-Mix G3 codecs.
Link Quality (LQ) Readings
Send and Return LQ numbers help you to determine if a problem is occurring at either end of a
connection. For example, on an IP connection the Return LQ reading represents the audio being
downloaded from the network locally (i.e. audio data is being sent by the remote codec).
Conversely, the Send LQ reading represents the audio data being sent by the local codec (i.e.
being downloaded by the remote codec). To ensure a stable connection, try to maintain a
reliable reading of 80 or higher for both the Send and Return LQ reading.
Important Note:
· The Return link quality reading is the same as the Local (L) setting displayed on a
G3 codec.
· The Send link quality reading is the same as the Remote (R) setting displayed on a
G3 codec.
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Diagnosing Port Blocking via the Studio Codec LQ
If the studio codec Return LQ reading is 01 then incoming audio from the remote codec is being
blocked by a firewall at either point A or B in the following diagram. If the studio codec Send LQ
reading is 01 then outgoing audio from the studio is being blocked by a firewall at either point C
or D in the following diagram.
Diagnosing Port Blocking via the Remote Codec LQ
If you attach your Tieline codec at the remote site to a LAN with access to the internet you can
often dial and connect to the studio without any problem. It is less likely that a firewall will block
outgoing TCP and UDP ports. However, if there is a firewall at the remote site it may block
incoming data packets from the studio.
The principle is the same at the remote codec for diagnosing blocked ports. If the remote codec
Return LQ reading is 01 then incoming audio from the studio codec is being blocked by a
firewall at either point C or D in the preceding diagram. If the remote codec Send LQ reading is
01 then the outgoing audio from the remote codec is being blocked by a firewall at either point A
or B in the preceding diagram.
Troubleshooting TCP Port Blocking
Error messages on the codec screen can help to diagnose TCP port blocking.
1. "Connection Refused" usually means that the firewall is configured correctly but the
codec is not using the expected port. For example, the firewall is set up to forward via
port 9002 but codec is 'listening' to port 10,000. “Connection Refused” is not normally
shown if the firewall is not configured correctly because a firewall will by design silently
drop any forwarding requests to ports that it doesn’t have open (see next point). Note:
"Connection Refused" will also be displayed if the Commander G3 or i-Mix G3 codec
you are calling is already connected.
2. “Connection Timeout” can mean one of two things:
· The firewall is not configured correctly and the attempted codec connection is being
silently dropped, e.g. a remote codec is dialing to port 9002 but the studio firewall
port forwarding is not configured.
· The UDP port is not port forwarded correctly. Tieline codecs send test data during
connection establishment to make sure that the audio path is configured correctly;
if this process fails then it will also result in a “Connection Timeout”.
How do I determine which end is blocking data flow?
Tieline test codec firewalls have the default Tieline TCP and UDP ports open. You can dial into
these test codecs (or other codecs you know are configured correctly) from your recently
configured studio and remote codecs and use the LQ readings to diagnose whether your studio
or remote codec firewall is blocking your data packets. If one codec connects ok and the other
one doesn't, then you will know which end is likely to be causing the problem. As an example:
1. Dial from site 1 to a Tieline test codec.
2. Dial from site 2 to Tieline test codec.
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If both of these connect successfully then the “outbound” TCP path for session data is OK, and
the inbound UDP audio path is OK.
3. Dial to site 1 from a codec you know is configured correctly.
4. Dial to site 2 from a codec you know is configured correctly.
If either of these calls fail then TCP and/or UDP inbound data is being blocked on the failed
connection (see "Troubleshooting TCP Port Blocking" above).
Testing your Codec
· Visit http://www.tieline.com/Support/Test-lines for a list of test IP codec addresses you can use
to verify your codec is configured correctly.
· See Testing IP Network Connections for more IP test information.
Learning More About IP Networks
For more IP network information please see the section titled Understanding IP Networks which
discusses:
· Private versus public IP addresses.
· Static versus DHCP assigned IP addresses.
· Network Address Translation (NAT), port forwarding and firewalls.
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Understanding IP Networks
Types of IP Addresses Available
Type of IP
Address
Public
Static Public IP
Address
How the IP
Address is
Allocated
Internet Service
Providers (ISPs)
Dynamically
Assigned Public
IP Address
Internet Service
Providers (ISPs)
Private Dynamically
Assigned Private
IP Address
DHCP Server/
Router on your
own private LAN
network.
Static Private IP
Address
LAN
Administrator
Description
ISP’s allocate a static public IP address to allow
network devices to communicate with each other
over the internet. It works like a public telephone
number and will allow your remote codec to call
your studio codec over the Internet.
ISP’s
usually
allocate
dynamically
(automatically) assigned public IP addresses to
allow network devices to communicate with each
other over the Internet. (Not recommended for
studio installations because each time you
connect to your ISP the IP address can change).
A DHCP server-allocated IP address that is
automatically assigned to a device on a LAN to
allow it to communicate with other devices and
the internet. This address can change each time
a device connects.
A network administrator-allocated static address
which is programmed into a device to allow it to
connect to a LAN. Often a security measure to
only allow access to devices approved by a
network administrator.
Obtaining Public IP Addresses
To send audio streams over the public internet you need to use a public IP address assigned to you
by your ISP (Internet Service Provider).
A public IP address is like your public telephone number and allows you to be contacted over the
internet in much the same way people dial your public telephone number. They come in two forms;
dynamic (DHCP) and static. Most ISPs assign a dynamic public IP address by default, which can
often change without you knowing. This is suitable for a quick demo of your Tieline codec, but for a
permanent installation you will need to request a permanent static public IP address.
Once the Static Public IP address is assigned to your internet connection (router) at the studio you
need to create a link between the public IP address and your codec’s private IP address on the
LAN. This is called Network Address Translation.
Depending upon how your network is configured, it may also be possible to simply connect your
Tieline codec directly into your ADSL modem/router and receive a public address from the router.
Private LAN IP Addresses
By default your Tieline codec will normally be automatically assigned a private IP address when you
connect it to a typical router over a LAN.
Private IP Addresses are associated with LANs and normally reside behind a firewall and are not
visible to the internet. They are generally in the ranges: 10.0.0.1 – 10.255.255.255, 169.254.0.0 –
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169.254.255.255, 172.16.0.0 – 172.31.255.255 and 192.168.0.0 – 192.168.255.255 and are
assigned by network DHCP servers and routers.
These IP Addresses are generally assigned for a predefined period (known as a lease) by your
network's DHCP server or router. This IP address will generally expire after the lease period. DHCP
assigned IP Addresses may also change if the device is disconnected for lengthy periods or if power
to the device is turned off and back on. As a result, it is advised that you make this IP address
permanent by assigning it as a Static DHCP IP Address. This will ensure you are able to always
forward incoming audio packets to your codec using the same private IP address at the studio using
port forwarding (see the section on port forwarding for more details). Consult your Network
Administrator if you are unsure how to do this.
Network Address Translation (NAT)
Network Address Translation (NAT) is a method of connecting multiple devices to the internet using
one public IP address.
The best way to explain NAT is to use the example of a phone system at an office that has one
public telephone number and multiple extensions. This type of telephone system allows people to
call you on a single public telephone number and performs the translation and routing of the public
number to a particular private extension. Similarly, in order to receive an IP call from a remote codec
over the public internet, the same network address translation principle applies. NAT and port
forwarding allows a single device, such as a broadband router, to act as an agent between the public
internet and a local private LAN.
The relationship between public and private IP addresses and NAT is displayed in the following
diagram and the following section explains port forwarding configuration in more detail.
Port Forwarding: Tieline TCP and UDP Port Settings
For your Tieline Codec to communicate over the public internet an IP Address alone is not sufficient.
In TCP/IP and UDP networks the codec port is the endpoint of your connection. Ports are doorways
for IP devices to communicate with each other. Picture a house and imagine the front door is the
entry point represented by a public or private IP address. Then you want to get to several codecs in
different rooms of the same house and ports represent the doors to each of those rooms. In principle
this is how port addressing works.
For example, several codecs may dial into your studio using the same public static IP address. In
this situation it is necessary to configure codec 'programs' with audio streams using different audio
ports for discretely routing each incoming and outgoing audio stream. By doing this your studio's
network routers know where IP packets for each audio stream should be routed, i.e. to which codec
and respective audio outputs.
When data packets are received from remote codecs at a particular public IP address, port
information is translated from data packets to ensure the correct packets are sent to the correct
studio codecs. This process is performed by PAT (Port Address Translation), which is a feature of
NAT (Network Address Translation) devices.
Tieline codecs use TCP ports for setting up the communication session and UDP ports for
streaming audio. While TCP ports are generally open, UDP ports are generally blocked by network
devices which contain firewalls and will stop you delivering your audio. Depending on the codecs you
are using, you need to configure your firewall to allow TCP and UDP protocols to pass through the
ports listed in the table below.
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Tips for Creating Reliable IP Connections
The following 10 tips are provided to help obtain the best possible IP connection between two
codecs, without paying for Quality of Service (QoS).
1. Always use the best quality Internet Service Provider (ISP). Tier 1 service providers are best
as their infrastructure actually makes up the internet ‘backbone’. Wikipedia lists the major
service providers that make up the internet backbone at: http://en.wikipedia.org/wiki/
internet_backbone. In Australia Telstra is equivalent to one of these service providers.
2. You will get the best quality connection if both the local (studio) and remote codecs use the
same ISP. This can substantially increase reliability, audio bandwidth and reduce audio
delay. Using the same service provider nationally can give better results than using different
local service providers. This is especially true if one of the service providers is a cheap, lowend domestic service provider, which buys its bandwidth from other ISPs. Second and third
tier providers sublease bandwidth from first tier providers and can result in connection
reliability issues due to multiple switch hops. We also highly recommend using First Tier
ISPs if connecting two codecs in different countries.
3. Sign up for a business plan that provides better performance than domestic or residential
plans. Business plans typically have a fixed data limit per month with an additional cost for
data beyond that limit. In addition, Service Level Agreements (SLA) will often provide better
support and response times in the event of a connection failure. Domestic plans are often
speed-limited or “shaped” when usage exceeds a predefined limit. These plans are cheap but
they are dangerous for streaming broadcast audio.
4. Ensure that the speed of the connection for both codecs is adequate for the job. The
minimum upload speed recommended is 256 kbps for a studio codec and 64 kbps for a field
unit connection.
5. Use good quality equipment to connect your codecs to the internet. (Tieline successfully
uses Cisco® switching and routing equipment.):
· If you are using a DSL or ADSL connection make sure you purchase a high quality modem
that can easily meet your speed requirements. This is especially important if you are over 4
kms from an exchange.
· If you have multiple codecs connected to a local area network (LAN) please ensure that your
network infrastructure is designed for media streaming and not domestic usage. Tieline has
tested several cheap 8-port switches that lose more packets between local computers than
an international IP connection between Australia and the USA!
· If using a wireless connection ensure that the antenna signal strength received is strong. The
type of antenna used and the amount of output gain also affects connection quality.
Important Note: You should be able to stream audio between two codecs on your LAN
and get ‘link quality’ readings of S99R99. If you see anything less than this then you
should get a network engineer to investigate the issue.
6. Once your internet connection is installed at the studio check that the connection
performance is approximately what you ordered and are paying for. A connection can perform
below advertised bit rates if:
· There is an error in ISP configuration;
· There is an error in modem configuration;
· There is a poor quality line between the studio and the exchange;
· There are too may phones or faxes connected to the phone line; or
· Line filters have been connected incorrectly.
You can test your internet connection speed by connecting a PC to the internet and using http://
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www.speedtest.net/index.php. If the bandwidth detected is low then something is wrong. Get it fixed
before going live!
7. Use a dedicated DSL/ADSL line for your codecs. Do not share a link with PCs or company
networks. The only exception to this rule is if an organisation has network equipment and
engineers that can implement and manage quality of service (QoS) on its network.
8. Use UDP as the preferred audio transport protocol. TCP generally results in lower bit rates
and random drop-outs of audio over the internet. Only use TCP if UDP is blocked by firewalls
and you are unable to connect.
9. When using UDP ensure the total bit rate (audio bit rate plus header bit rate) is no more than
80% of the ISP connection rate. IP headers require around 20 kbps in addition to the audio
bit rate. For example, with a 64 kbps connection the audio bit rate should be (64-20) x 0.8 =
31.2 kbps or lower. For TCP we suggest a limit of 50% or less.
10.Wireless IP connections can easily become congested and result in packet loss and audio
drop-outs. It is very difficult to guarantee connection quality when there is no way of knowing
how many people are sharing the same wireless connection.
Important Note: Be careful when using cell-phone connections at special events where
thousands of people have mobile phones. This can result in poor quality connections
and audio drop-outs if cell-phone base stations are overloaded.
IP Connection Checklist
Complete the following check list and aim for a score of at least 8 out of 10 before going live.
20.4
Number
Check
1
Using a reputable Tier1 ISP that’s part of internet backbone.
2
The same ISP is being used for both codec connections.
3
The ISP Plan is a Business Plan or equivalent.
4
The ISP connection speed is adequate.
5
Equipment is high quality and suitable for media streaming.
6
The ISP connection speed has been tested and is suitable.
7
The ISP connection is not shared with other PCs or devices.
8
UDP is being used as the audio transport protocol.
9
No more than 80% of ISP connection bandwidth is being used.
10
There are no wireless connections being used.
Result
Testing IP Network Connections
There are a few very simple tools that you can use to test whether a codec can be reached over an
IP network.
· Visit http://www.speedtest.net/ to test the upload and download speed of your IP connections and
identify your public IP address.
· Visit www.portforward.com and download the port checking application to verify your router's ports
are open. Note: Using a port scanner to test a codec will be unsuccessful if you try to scan and
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the port is already in use, i.e. the codec is connected.
· Visit www.subnetonline.com and use an online port scanner to check for open and closed TCP
ports. This site also has numerous other software tools, including an online ping web-tool for IPv4,
plus TraceRoute and TracePath software tools.
Ping the Codec
A ping test can be used to test whether it is possible to reach a codec or any device over an IP
network. A ping test measures:
· The round-trip time of packets.
· Any packet loss.
There are two types of ping tests:
1. Short test: sends 4 packets and delivers statistics.
i. Point to the start menu on your PC and click once.
ii. In the search text box type Run and press Enter.
iii. Type CMD in the Run dialog text box and click OK.
iv. Type ping and the IP address of the codec you are pinging (i.e. ping 192.168.0.159) and
press the Enter key on your keyboard.
v. The round trip time of the packets is displayed, as well as any packet loss.
2. Long test: sends packets continuously until stopped.
i. Point to the start menu on your PC and click once.
ii. In the search text box type Run and press Enter.
iii. Type CMD in the Run dialog text box and click OK.
iv. Type ping, the IP address of the codec you are pinging, and then -t (i.e. ping
203.36.205.163 -t) and press the Enter key on your keyboard.
v. Let the test run for several minutes and then press CTRL C.
vi. The round trip time of the packets is displayed, as well as any packet loss for the period of
time that the test occurred.
Trace the Route of IP Packets
Another utility available on your PC is traceroute. This tool can be used to determine the route and
number of hops that data packets are taking to their destination (codec). This is useful because the
more routers that packets traverse, the more latency your connection will have, and the less reliable
it will be.
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i. Point to the start menu on your PC and click once.
ii. In the search text box type Run and press Enter.
iii. Type CMD in the Run dialog text box and click OK.
iv. Type tracert, the IP address of the codec you are contacting (i.e. tracert 203.36.205.163)
and press the Enter key on your keyboard.
20.5
Software Licences
This product uses a combination of proprietary and open-source software programs.
Some of the software included in this product contains copyrighted software that is
licensed under various open-source licenses (e.g. GNU General Public License v2,
GNU Lesser GPL v2.1). A detailed list of open source licenses used in this product
is included in the user manual. This can be downloaded from the Help Panel in the
Web Browser Interface or from the Tieline website <http://tieline.com>. You may
request a copy for the open source software on DVD by contacting our support team
on +61 (0)8 9249 6688. Tieline Pty Ltd will charge a small handling fee for
distribution of this software.
Some of the open source software of this product is based on the works of the
Gentoo project and is not directed, managed, sold or supported by Gentoo
Foundation, Inc. The Gentoo name is a trademark of Gentoo Foundation, Inc.
Open Source GPL compatible Licenses:
o Some of the open-source software in the product is licensed under GPL version 3.
A copy of the license can be obtained at http://www.gnu.org/licenses/gpl.html.
o Some of the open-source software in the product is licensed under GPL version 2.
A copy of the license can be obtained at http://www.gnu.org/licenses/old-licenses/
gpl-2.0.html.
o Some of the open-source software in the product is licensed under LGPL version 3.
A copy of the license can be obtained at http://www.gnu.org/licenses/lgpl.html.
o Some of the open-source software in the product is licensed under LGPL version
2.1. A copy of the license can be obtained at http://www.gnu.org/licenses/oldlicenses/lgpl-2.1.html.
Open Source BSD style Licenses:
• bind:
o Portions: Copyright (c) 1987, 1990, 1993, 1994 The Regents of the University of
California. All rights reserved. Additional clause - All advertising materials
mentioning features or use of this software must display the following
acknowledgment: This product includes software developed by the University of
California, Berkeley and its contributors.
o Portions: Copyright (c) 2004 Masarykova universita (Masaryk University, Brno,
Czech Republic) All rights reserved.
o Portions: Copyright (c) 1997 - 2003 Kungliga Tekniska Högskolan (Royal Institute
of Technology, Stockholm, Sweden). All rights reserved.
o Portions (2 clause BSD license, 3rd clause removed): Copyright (c) 1998 Doug
Rabson. All rights reserved.
o Portions: Copyright ((c)) 2002, Rice University. All rights reserved.
o Portions: Copyright 2000 Aaron D. Gifford. All rights reserved.
o Portions (2 clause BSD license, 3rd clause removed): Copyright (c) 1998 Doug
Rabson. Copyright (c) 2001 Jake Burkholder. All rights reserved.
o Portions: Copyright (C) 1995, 1996, 1997, and 1998 WIDE Project. All rights
reserved.
o Portions: Copyright (c) 2000-2002 Japan Network Information Center. All rights
reserved.
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o idnkit: Copyright (c) 2000-2002 Japan Network Information Center.
reserved.
All rights
o zkt: Copyright (c) 2005 - 2008, Holger Zuleger HZnet. All rights reserved.
• dhcpcd - 2 clause BSD license, clause 3 removed
o Copyright (c) 2006-2011 Roy Marples <roy@marples.name>
• eventlog
o Copyright (c) 2003 BalaBit IT Ltd.
• file - 2 clause BSD license, clause 3 removed
o Copyright (c) Ian F. Darwin 1986, 1987, 1989, 1990, 1991, 1992, 1994, 1995.
o Software written by Ian F. Darwin and others;
o maintained 1994- Christos Zoulas.
o This software is not subject to any export provision of the United States
Department of Commerce, and may be exported to any country or planet.
• glibc
o Code incorporated from 4.4 BSD: Copyright (C) 1991 Regents of the University of
California. All rights reserved.
o Sun RPC support (from rpcsrc-4.0): Copyright (c) 2010, Oracle America, Inc.
• htop
o Copyright (c) 2004-2006 The Trustees of Indiana University and Indiana University
Research and Technology Corporation. All rights reserved.
o Copyright (c) 2004-2005 The Regents of the University of California. All rights
reserved.
o Copyright (c) 2007 Cisco Systems, Inc.
All rights reserved.
o Portions: Copyright (c) 2004-2005 The University of Tennessee and The University
of Tennessee Research Foundation. All rights reserved
o Portions: Copyright (c) 2004-2005 High Performance Computing Center Stuttgart,
University of Stuttgart. All rights reserved.
o Portions: Copyright (c) 2006, 2007 Advanced Micro Devices, Inc. All rights
reserved.
• less - 2 clause BSD license, clause 3 removed
o Copyright (C) 1984-2011 Mark Nudelman
• libpcre
o Basic Library Functions: Copyright (c) 1997-2010 University of Cambridge. All
rights reserved.
o C++ Wrapper Functions: Copyright (c) 2007-2010, Google Inc. All rights reserved
• libuuid
o Copyright (c) 1996, 1997, 1998, 1999, 2007. Theodore Ts'o.
• lighttpd
o Copyright (c) 2004, Jan Kneschke, incremental. All rights reserved.
• net-snmp
o Copyright 1989, 1991, 1992 by Carnegie Mellon University. All rights reserved.
o Derivative Work - 1996, 1998-2000
o Copyright 1996, 1998-2000 The Regents of the University of California. All rights
reserved.
© Tieline Pty. Ltd. 2015
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o Copyright (c) 2001-2003, Networks Associates Technology, Inc. All rights
reserved.
o Portions of this code are copyright (c) 2001-2003, Cambridge Broadband Ltd. All
rights reserved.
o Copyright © 2003 Sun Microsystems, Inc., 4150 Network Circle, Santa Clara,
California 95054, U.S.A. All rights reserved.
o Copyright (c) 2003-2010, Sparta, Inc. All rights reserved.
o Copyright (c) 2004, Cisco, Inc and Information Network. Center of Beijing
University of Posts and Telecommunications. All rights reserved.
o Copyright (c) Fabasoft R&D Software GmbH & Co KG, 2003. oss@fabasoft.com. Author:
Bernhard Penz <bernhard.penz@fabasoft.com>
o Copyright (c) 2007 Apple Inc. All rights reserved.
o Copyright (c) 2009, ScienceLogic, LLC. All rights reserved.
• openrc - 2 clause BSD license, clause 3 removed
o Copyright (c) 2007-2009 Roy Marples <roy@marples.name>
• OpenSSH
o Copyright (c) 1995 Tatu Ylonen <ylo@cs.hut.fi>, Espoo, Finland. All rights
reserved.
o 32-bit CRC compensation attack detector: Copyright (c) 1998 CORE SDI S.A., Buenos
Aires, Argentina. All rights reserved.
o ssh-keyscan: Copyright 1995, 1996 by David Mazieres <dm@lcs.mit.edu>.
o One component of OpenSSH source code: Copyright (c) 1983, 1990, 1992, 1993, 1995.
The Regents of the University of California. All rights reserved.
o Remaining components under 2 clause BSD (clause 3 removed) Copyright holders:
Markus Friedl, Theo de Raadt, Niels Provos, Dug Song, Aaron Campbell, Damien
Miller, Kevin Steves, Daniel Kouril, Wesley Griffin, Per Allansson, Nils Nordman,
Simon Wilkinson
o Parts of portable version under 2 clause BSD (clause 3 removed) Copyright
holders: Ben Lindstrom, Tim Rice, Andre Lucas, Chris Adams, Corinna Vinschen, Cray
Inc., Denis Parker, Gert Doering, Jakob Schlyter, Jason Downs, Juha Yrjölä, Michael
Stone, Networks Associates Technology, Inc., Solar Designer, Todd C. Miller, Wayne
Schroeder, William Jones, Darren Tucker, Sun Microsystems, The SCO Group, Daniel
Walsh, Red Hat, Inc.
o Parts of openbsd-compat: Copyright holders: Todd C. Miller, Theo de Raadt, Damien
Miller, Eric P. Allman, The Regents of the University of California, Constantin S.
Svintsoff.
• OpenSSL: crypto/blowfish, crypto/des
o Copyright (C) 1995-1997 Eric Young (eay@cryptsoft.com).
o Clause 3: All advertising materials mentioning features or use of this software
must display the following acknowledgement: This product includes software
developed by Eric Young (eay@cryptsoft.com).
• strace:
o Copyright (c) 1991, 1992 Paul Kranenburg <pk@cs.few.eur.nl>
o Copyright (c) 1993 Branko Lankester <branko@hacktic.nl>.
o Copyright (c) 1993 Ulrich Pegelow <pegelow@moorea.uni-muenster.de>.
o Copyright (c) 1995, 1996 Michael Elizabeth Chastain <mec@duracef.shout.net>.
o Copyright (c) 1993, 1994, 1995, 1996 Rick Sladkey <jrs@world.std.com>.
o Copyright (C) 1998-2001 Wichert Akkerman <wakkerma@deephackmode.org>..
© Tieline Pty. Ltd. 2015
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o All rights reserved.
• util-linux: text-utils
o Copyright (c) 2000-2001 Gunnar Ritter. All rights reserved.
Redistribution and use in source and binary forms, with or without modification,
are permitted provided that the following conditions are met:
1. Redistributions of source code must retain the above copyright notice, this list
of conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright notice, this
list of conditions and the following disclaimer in the documentation and/or other
materials provided with the distribution.
3. Neither the name of the <ORGANIZATION> nor the names of its contributors may be
used to endorse or promote products derived from this software without specific
prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND ANY
EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL,EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
Open Source MIT style Licenses:
• glibc: DNS resolver taken from BIND 4.9.5
o Portions Copyright (C) 1993 by Digital Equipment Corporation.
• ncurses
o Copyright (c) 1998-2010,2011 Free Software Foundation, Inc.
o install-sh : 1994 X Consortium
• OpenSSH
o Portions of code under MIT-style license to the copyright holders: Free Software
Foundation, Inc.
Permission is hereby granted, free of charge, to any person obtaining a copy of
this software and associated documentation files (the "Software"), to deal in the
Software without restriction, including without limitation the rights to use, copy,
modify, merge, publish, distribute, sublicense, and/or sell copies of the Software,
and to permit persons to whom the Software is furnished to do so, subject to the
following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED,
INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,FITNESS FOR A
PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION
OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,OUT OF OR IN CONNECTION WITH THE
SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
Open Source ISC style Licenses:
• bind
o Copyright (C) 2004-2011 Internet Systems Consortium, Inc. ("ISC")
o Copyright (C) 1996-2003 Internet Software Consortium.
© Tieline Pty. Ltd. 2015
Bridge-IT Manual
o Portions: Copyright (C) 1996-2001
193
Nominum, Inc.
o Portions: Copyright (C) 1995-2000 by Network Associates, Inc.
o Portions: Copyright (C) 2002 Stichting NLnet, Netherlands, stichting@nlnet.nl.
o Dynamically Loadable Zones (DLZ) contributer: Rob Butler.
o Portions: Copyright (c) 1993 by Digital Equipment Corporation.
O Portions: Copyright (c) 1999-2000 by Nortel Networks Corporation.
O Portions: Copyright (C) 2004 Nominet, Ltd.
O Portions: Copyright RSA Security Inc.
O Portions: Copyright (c) 1996, David Mazieres <dm@uun.org>, Copyright (c) 2008,
Damien Miller <djm@openbsd.org>
• expat
o Copyright (c) 1998, 1999, 2000 Thai Open Source Software Center Ltd and Clark
Cooper.
o Copyright (c) 2001, 2002, 2003, 2004, 2005, 2006 Expat maintainers.
• libffi
o Copyright (c) 1996-2011
Anthony Green, Red Hat, Inc and others.
• OpenSSH
o Portions of code under ISC-style license to the copyright holders: Internet
Software Consortium, Todd C. Miller, Reyk Floeter,
Chad Mynhier.
• popt
o Copyright (c) 1998 Red Hat Software.
• vixie-cron
O Copyright 1988,1990,1993 by Paul Vixie. All rights reserved.
o Copyright (C) 2004-2011 Internet Systems Consortium, Inc. ("ISC")
o Copyright (C) 1997,2000 by Internet Software Consortium, Inc.
THE SOFTWARE IS PROVIDED "AS IS" AND THE COPYRIGHT HOLDERS AND CONTRIBUTORS
DISCLAIM ALL WARRANTIES WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED
WARRANTIES OF MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE COPYRIGHT HOLDERS
AND CONTRIBUTORS BE LIABLE FOR ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL
DAMAGES OR ANY DAMAGES WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS,
WHETHER IN AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT
OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
Open Source UCB License:
• util-linux
This product includes software developed by the University of California, Berkeley
and its contributors.
Copyright (c) 1989 The Regents of the University of California.
All rights reserved.
THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS
OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT
SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
© Tieline Pty. Ltd. 2015
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CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH
DAMAGE.
Open Source OpenSSL License:
• OpenSSL
o "This product includes software developed by the OpenSSL Project for use in the
OpenSSL Toolkit (http://www.openssl.org/)"
o "This product includes cryptographic software written by Eric Young
(eay@cryptsoft.com)"
Copyright (c) 1998-2011 The OpenSSL Project.
All rights reserved.
Redistribution and use in source and binary forms, with or without modification,
are permitted provided that the following conditions are met:
1. Redistributions of source code must retain the above copyright notice, this list
of conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright notice, this
list of conditions and the following disclaimer in the documentation and/or other
materials provided with the distribution.
3. All advertising materials mentioning features or use of this software must
display the following acknowledgment: "This product includes software developed by
the OpenSSL Project for use in the OpenSSL Toolkit. (http://www.openssl.org/)"
4. The names "OpenSSL Toolkit" and "OpenSSL Project" must not be used to endorse or
promote products derived from this software without prior written permission. For
written permission, please contact openssl-core@openssl.org.
5. Products derived from this software may not be called "OpenSSL" nor may
"OpenSSL" appear in their names without prior written permission of the OpenSSL
Project.
6. Redistributions of any form whatsoever must retain the following acknowledgment:
"This product includes software developed by the OpenSSL Project for use in the
OpenSSL Toolkit (http://www.openssl.org/)"
THIS SOFTWARE IS PROVIDED BY THE OpenSSL PROJECT ``AS IS'' AND ANY EXPRESSED OR
IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT
SHALL THE OpenSSL PROJECT OR ITS CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
Original SSLeay License:
This product includes cryptographic software written by Eric Young
(eay@cryptsoft.com). This product includes software written by Tim Hudson
(tjh@cryptsoft.com).
Copyright (C) 1995-1998 Eric Young (eay@cryptsoft.com)
All rights reserved.
This package is an SSL implementation written by Eric Young (eay@cryptsoft.com).
The implementation was written so as to conform with Netscapes SSL.
This library is free for commercial and non-commercial use as long as the following
© Tieline Pty. Ltd. 2015
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conditions are aheared to. The following conditions apply to all code found in
this distribution, be it the RC4, RSA, lhash, DES, etc., code; not just the SSL
code. The SSL documentation included with this distribution is covered by the same
copyright terms except that the holder is Tim Hudson (tjh@cryptsoft.com).
Copyright remains Eric Young's, and as such any Copyright notices in the code are
not to be removed.
If this package is used in a product, Eric Young should be given attribution as the
author of the parts of the library used. This can be in the form of a textual
message at program startup or in documentation (online or textual) provided with
the package.
Redistribution and use in source and binary forms, with or without modification,
are permitted provided that the following conditions are met:
1. Redistributions of source code must retain the copyright notice, this list of
conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright notice, this
list of conditions and the following disclaimer in the and/or other materials
provided with the distribution.
3. All advertising materials mentioning features or use of this software must
display the following acknowledgement: "This product includes cryptographic
software written by Eric Young (eay@cryptsoft.com)". The word 'cryptographic' can
be left out if the rouines from the library being used are not cryptographic
related :-).
4. If you include any Windows specific code (or a derivative thereof) from the apps
directory (application code) you must include an acknowledgement: "This product
includes software written by Tim Hudson (tjh@cryptsoft.com)"
THIS SOFTWARE IS PROVIDED BY ERIC YOUNG ``AS IS'' AND ANY EXPRESS OR IMPLIED
WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT
SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH
DAMAGE.
Open Source netperf License:
Copyright (C) 1993 Hewlett-Packard Company
ALL RIGHTS RESERVED.
THE SOFTWARE AND DOCUMENTATION IS PROVIDED "AS IS". HEWLETT-PACKARD COMPANY DOES
NOT WARRANT THAT THE USE, REPRODUCTION, MODIFICATION OR DISTRIBUTION OF THE
SOFTWARE OR DOCUMENTATION WILL NOT INFRINGE A THIRD PARTY'S INTELLECTUAL PROPERTY
RIGHTS. HP DOES NOT WARRANT THAT THE SOFTWARE OR DOCUMENTATION IS ERROR FREE. HP
DISCLAIMS ALL WARRANTIES, EXPRESS AND IMPLIED, WITH REGARD TO THE SOFTWARE AND THE
DOCUMENTATION. HP SPECIFICALLY DISCLAIMS ALL WARRANTIES OF MERCHANTABILITY AND
FITNESS FOR A PARTICULAR PURPOSE. HEWLETT-PACKARD COMPANY WILL NOT IN ANY EVENT BE
LIABLE FOR ANY DIRECT, INDIRECT, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES
(INCLUDING LOST PROFITS) RELATED TO ANY USE, REPRODUCTION, MODIFICATION, OR
DISTRIBUTION OF THE SOFTWARE OR DOCUMENTATION.
Open Source Info-ZIP license:
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Copyright (c) 1990-2001 Info-ZIP.
All rights reserved.
For the purposes of this copyright and license, "Info-ZIP" is defined as the
following set of individuals:
Mark Adler, John Bush, Karl Davis, Harald Denker, Jean-Michel Dubois, Jean-loup
Gailly, Hunter Goatley, Ian Gorman, Chris Herborth, Dirk Haase, Greg Hartwig,
Robert Heath, Jonathan Hudson, Paul Kienitz, David Kirschbaum, Johnny Lee, Onno van
der Linden, Igor Mandrichenko, Steve P. Miller, Sergio Monesi, Keith Owens, George
Petrov, Greg Roelofs, Kai Uwe Rommel, Steve Salisbury, Dave Smith, Christian
Spieler, Antoine Verheijen, Paul von Behren, Rich Wales, Mike White
This software is provided "as is," without warranty of any kind, express or
implied. In no event shall Info-ZIP or its contributors be held liable for any
direct, indirect, incidental, special or consequential damages arising out of the
use of or inability to use this software.
Permission is granted to anyone to use this software for any purpose, including
commercial applications, and to alter it and redistribute it freely, subject to the
following restrictions:
1. Redistributions of source code must retain the above copyright notice,
definition, disclaimer, and this list of conditions.
2. Redistributions in binary form must reproduce the above copyright notice,
definition, disclaimer, and this list of conditions in documentation and/or other
materials provided with the distribution.
3. Altered versions--including, but not limited to, ports to new operating systems,
existing ports with new graphical interfaces, and dynamic, shared, or static
library versions--must be plainly marked as such and must not be misrepresented as
being the original source. Such altered versions also must not be misrepresented
as being Info-ZIP releases--including, but not limited to, labeling of the altered
versions with the names "Info-ZIP" (or any variation thereof, including, but not
limited to, different capitalizations), "Pocket UnZip," "WiZ" or "MacZip" without
the explicit permission of Info-ZIP. Such altered versions are further prohibited
from is representative use of the Zip-Bugs or Info-ZIP e-mail addresses or of the
Info-ZIP URL(s).
4. Info-ZIP retains the right to use the names "Info-ZIP," "Zip," "UnZip," "WiZ,"
"Pocket UnZip," "Pocket Zip," and "MacZip" for its own source and binary releases.
© Tieline Pty. Ltd. 2015
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197
Compliances and Certifications
FCC Compliance Notice
This equipment has been tested and found to comply with the limits for a class A digital device,
pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection
against harmful interference when the equipment is operated in a commercial environment. This
equipment generates, uses and can radiate radio frequency energy and, if not installed and used in
accordance with the instruction manual, may cause harmful interference to radio communications.
Operation of this equipment in a residential area may cause harmful interference, in which case the
user will be required to correct the interference at his/her own expense. There is no guarantee,
however, that interference will not occur in a particular installation. If this equipment does cause
harmful interference to radio or television reception, which can be determined by turning the
equipment off and on, the user is encouraged to try to correct the interference by one or more of the
following measures:
· Reorient or relocate the receiving antenna.
· Increase the separation between the equipment and receiver.
· Connect the equipment to an outlet on a circuit different from that to which the receiver is
connected.
· Consult the dealer or an experienced radio or TV technician for help.
CAUTION:
Changes or modifications to this unit not expressly approved by the party responsible for
compliance could void the user’s authority to operate this equipment.
Declaration of Conformity
The Tieline Bridge-IT IP codec meets the requirements of directives for CE and C-Tick certifications.
Technical documentation required by the conformity assessment procedure is kept at the head
office of Tieline Technology; 1/25 Irvine Drive, Malaga, Western Australia 6090.
EN 55 022 Statement
This is to certify that Tieline Bridge-IT is shielded against the generation of radio interference in
accordance with the application of EN 55 022: 2006 Class A. Technical documentation required by
the conformity assessment procedure is kept at the head office of Tieline Technology; 1/25 Irvine
Drive, Malaga, Western Australia 6090.
Canadian Department of Communications Radio Interference Regulations
This digital apparatus (Tieline Bridge-IT) does not exceed the Class A limits for radio-noise
emissions from digital apparatus as set out in the Radio Interference Regulations of the Canadian
Department of Communications.
Règlement sur le brouillage radioélectrique du ministère des
Communications
Cet appareil numérique (Tieline Bridge-IT) respecte les limites de bruits radioélectriques visant les
appareils numériques de classe A prescrites dans le Règlement sur le brouillage radioelectrique du
ministère des Communications du Canada.
Safety of Electrical and Electronic Products and Components
The IECEE CB Scheme is an international system for mutual acceptance of CB test reports and
© Tieline Pty. Ltd. 2015
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certificates covering the safety of electrical and electronic products and components. The IEC CB is
a multilateral scheme among participating countries and certification organisations, based on the
use of international (IEC) standards.
This product has been tested by an independent certifying company and has been certified to
comply with IEC 60950-1(ed.2);am1
20.7
Trademarks and Credit Notices
1. Windows is a registered trademark of Microsoft Corporation in the United States and/or other
countries.
2. Windows XP and Vista are either trademarks or registered trademarks of Microsoft Corporation in
the United States and/or other countries.
3. Firefox is a registered trademark of Mozilla Corporation in the United States and/or other
countries.
4. Solaris is a trademark of Sun Microsystems Inc. in the United States and/or other countries.
5. Linux is the registered trademark of Linus Torvalds in the U.S. and other countries.
6. Google™ is the registered trademark of Google Inc in the United States and/or other countries.
7. iPod® is a registered trademark of Apple Inc., registered in the U.S. and other countries.
8. Java is a trade mark Sun Microsystems Inc. in the United States and/or other countries.
9. Other product names mentioned within this document may be trademarks or registered
trademarks, or a trade name of their respective owner.
© Tieline Pty. Ltd. 2015
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199
Specifications
Input/Output Specifications
Analog Audio Inputs
Analog
Audio
Outputs
AES3 In
AES3 Out
Headphones
Control Ports In/Out
2 x Female XLR (Channel 1 mic/line; channel 2 line only)
2 x Male XLR
1 x female XLR (Channel 1 in; shared with Ch1 analog input)
1 x male XLR
1 x 6.35mm (1/4") Jack on rear panel
Two relay inputs and two opto-isolated outputs for machine control via
Phoenix connector
Input High Impedance > 5K ohm
Audio
Impedance
Output Impedance
<50 ohm Balanced
Clipping Level
+22dBu (input and outputs)
24 bit A/D & D/A
Converters
Frequency Response 20Hz to 20kHz
Total
Harmonic <0.0039% at +16dBu, or -88dBu unweighted
Distortion
Signal
To
Noise >90dB at +22dBu, unweighted
Ratio
Sample Frequencies
IP
Sample 16kHz, 32kHz, 44.1kHz, 48kHz
Frequencies
Algorithms
IP
Opus, Tieline Music, Tieline MusicPLUS, G.711, G.722, MPEG Layer 2
IP (Pro version only) AAC-LD, AAC-ELD, AAC-LC, AAC-HE, AAC-HEv2
IP (Pro version only) 16/24 bit Enhanced apt-X
IP (uncompressed)
Linear PCM16
Data and Control Interfaces
USB
LAN
Serial
USB 2.0 (Type B) host port on the rear panel
10/100 base T RJ45 connector
RS232 up to 115kbps with or without CTS/RTS flow control via female DB9
connector, can be used as a proprietary data channel
Front Panel Interfaces
Display
SD/SDHC Card Slot
Keypad
Navigation
General
128 x 64 monochrome LCD
Supports SDHC Flash Cards
20 button keypad
5 button keypad
Dimensions
Weight
Power Consumption
Operating
Temperature
Humidity Operating
Range
Internal Battery
8.5” x 5.9” x 1.75” (216mm x 150mm x 44mm)
2.07lb/940g
12V DC, 400mA
0°C to 50°C (32°F to 122°F)
© Tieline Pty. Ltd. 2015
20%
(0 to 35°C), non-condensing
Panasonic CR2032, 3V coin type 20mm
200
Bridge-IT Manual
Index
-BBackup
configuring
48
how it works
48
-AAES/EBU
audio levels
24
input and output
24
program input settings
sample rate
24
sample rate conversion
-C24
24
AES/EBU - HTML5 Toolbox
output sample rate
135
Compliances
197
Configuration
web-GUI software
AES/EBU - Java Toolbox
output sample rate
69
AES3
audio levels
24
input and output
24
program input settings
sample rate
24
sample rate conversion
24
24
69
Algorithm
latency
158
programming of 158
sample rates
158
types
158
Applications, codec
9
Audio levels
adjustment
20
ch1 mic/line level audio
20
ganging inputs
20
IGC
20
IGC Auto Level
20
intelligent gain control
20
metering
20
phantom power
20
quick adjustment of levels
20
Auto Reconnect
operation
36
programming of
36
51
Configuration files - HTML5 Toolbox
backup
145
restore
145
AES3 - HTML5 Toolbox
output sample rate
135
AES3 - Java Toolbox
output sample rate
Certifications
197
Codec
applications
9
features
9
introduction
9
Configuration files - Java Toolbox
backup
104
restore
104
Connecting
30
default algorithm profiles
45
dialing
30
first steps
30, 30
hanging up
35
how to connect
30
preparing to connect
30
speed dialing
37
Connection
link quality
33
statistics
33
Connections
AES3
13
analog
13
DC power
13
digital
13
headphone output
13
LAN
13
opto-isolated outputs
13
rear panel
13
relay inputs
13
RS-232
13
USB 2.0 slave
13
Contrast Adjustment
11
© Tieline Pty. Ltd. 2015
Index
restore via web-GUI
Control ports
169
Control Ports - HTML5 Toolbox
configuration
150
opto-isolated inputs
150
relay outputs
150
Control Ports - Java Toolbox
GPIO port programming
programming
109
Features
9
Features, codec
FEC
how it works
Programming
109
Dial and disconnect - Java Toolbox
dial an audio stream
97
disconnect an audio stream
97
Dialing
default algorithm profiles
hanging up
35
how to connect
30
speed dialing
37
45
131
-E169
-F175
Factory defaults - HTML5 Toolbox
restore via web-GUI 144
Factory defaults - Java Toolbox
© Tieline Pty. Ltd. 2015
11
-H140
Hanging up a connection
35
Headphones
monitoring
26
mono connections
26
output levels
26
return program audio
26
stereo connections
26
HTML5 Toolbox Quick Connect
dial & disconnect connections
126
IP setup
126
ISDN setup
126
launching
126
load & unload a program
126
POTS setup
126
PPM monitoring
126
Dim LCD Screen
adjustment
11
Factory default settings
restoration of, via codec
154
Ganging inputs
20
GPIOs - HTML5 Toolbox
150
GPIOs - Java Toolbox
109
GUI ports
172
55
Default ports
172
Dial & disconnect - HTML5 Toolbox
dial a program or audio stream
140
disconnect a program or audio stream
unload a program
140
Encode/Decode Direction
167
167
-G-
Data
bidirectional encoding
169
unidirectional encoding
169
DSCP
9
Front Panel Controls
-DDefault password
new web-GUI password
103
Firmware upgrades - HTML5 Toolbox
Forward error correction
FEC
167
how it works
167
Programming
167
Controls
11
Country settings
30
Credit notices
198
201
HTML5 Toolbox Web-GUI
configure connections
131
configure Ethernet ports
131
configure QoS
131
configure VLANs
131
dial a program or audio stream
140
disconnect a program or audio stream
DNS settings
131
DSCP
131
IPv4 address configuration
131
140
202
Bridge-IT Manual
HTML5 Toolbox Web-GUI
IPv6 address configuration
QoS
131
unload a program
140
configure Ethernet ports
65
configure QoS
65
DNS settings
65
IPv4 address configuration
65
IPv6 address configuration
65
lock programs
100
unlock programs
100
131
-I-
Jitter buffer
automatic
164
fixed
164
programming of 164
IGC
20
Inputs
adjusting input levels
20
audio metering
20
ch1 mic/line level audio
20
ganging
20
IGC
20
IGC Auto Level
20
intelligent gain control
20
phantom power
20
quick adjustment of levels
20
-KKeypad
button descriptions
11
function button descriptions
-L-
Inputs - HTML5 Toolbox
analog
135
digital AES3
135
ganging
135
lock settings
135
renaming
135
setting levels
135
Language selection
Codec menus
27
Web-GUI 57
Language selection - HTML5 Toolbox
LCD Screen
auto dim adjustment
11
contrast adjustment
11
Inputs - Java Toolbox
analog
69
digital AES3
69
ganging
69
lock settings
69
renaming
69
setting levels
69
Intelligent gain control
20
Introduction
9
Introduction to the HTML5 Toolbox
Introduction to the web-GUI 57
IP address
details
155
DHCP
155
programming
155
static
155
IP overheads
158
-JJava Toolbox Web-GUI
configure connections
11
115
License installation - HTML5 Toolbox
146
License installation - Java Toolbox
106
Licenses
checking for
176
installation of 176
updates
176
115
Licensing
checking for
176
installation of 176
updates
176
Link quality
33
monitoring
33
Lock programs
49
Lock programs - HTML5 Toolbox
143
Logs
clear history via web-GUI 175
Logs - HTML5 Toolbox
send logs to Tieline
147
view event logs
147
65
Logs - Java Toolbox
© Tieline Pty. Ltd. 2015
Index
Logs - Java Toolbox
clear log history
104
send logs to Tieline
107
view event logs
107
LQ
Navigation
how to
Navigation buttons
-M-
Opto-isolators
169
Overview
manual
5
5
-P-
Manual Conventions
5
Master panel - Java Toolbox
multi-unicast dialing
79
Menus
codec menus
Panel lock
Password
default
15
Monitoring
auto select
26
connection statistics
33
headphone outputs
26
headphones
26
input audio only
26
link quality
33
monitor input
26
packet arrivals
33
STL silence detection mode
Multicasts - Java Toolbox
about
84
client program config
88
multicast server versus client
server program configuration
Multiple unicasts - Java Toolbox
configuration
79
dialing
79
Multi-unicasts - Java Toolbox
configuration
79
dial & disconnect
98
dialing
79
Navigating menus
how to
15
© Tieline Pty. Ltd. 2015
50
57
Peer-to-Peer cxns - Java Toolbox
configuration
73
dialing
73
RS232 data enable
73
26
Multicasting
front panel configuration
38, 41
multicast client programs
41
multicast server programs
38
-N-
11
-O-
33
Manual
conventions
overview
5
15
84
84
Phantom power
20
Ports
172
Profiles
45
Programming
aac
158
algorithms
158
apt-X Enhanced
158
check IP details
155
DHCP IP addresses
155
FEC
167
forward error correction
167
G.711
158
G.722
158
IP addresses
155
jitter buffer
164
linear audio
158
MPEG
158
Music
158
MusicPlus
158
routine tasks
155
static IP addresses
155
Programs
about multicasts
28
about multi-unicasts
28
about peer-to-peer calls
28
audio streams
28
deleting
45
dial multiple connections
37
203
204
Bridge-IT Manual
Programs
dialing
35
disconnect multiple connections
how do they work
28, 30
lock
49
multiple unicast
30
point-to-point
30
session data
28
unicast
30
unlock
49
what are they
30
programs
175
user settings
175
37
-QQoS
DSCP
173
programming
103
Reset defaults - HTML5 Toolbox
factory default settings
144
programs
144
user settings
144
Restore factory default settings
via codec menus
175
Programs - HTML5 Toolbox
backup and restore
145
Programs - Java Toolbox
backup and restore
104
config multi-unicast dialing
79
config of point-to-point programs
delete
100
editing
100
monitor multi-unicasts
79
view settings
100
web-GUI dialing
73
Reset - Java Toolbox
factory default settings
programs
103
user settings
103
73
RS232
programming via codec
169
RS232 - HTML5 Toolbox
baud rates via web-GUI
flow control via web-GUI
148
148
RS232 - Java Toolbox
baud rates via web-GUI
flow control via web-GUI
108
108
Rules
Explained
62
Rules - HTML5 Toolbox
control port configuration
Rules - Java Toolbox
GPIO port programming
109
invalid rules
109
opto-isolated outputs
109
programming control ports
109
173
Quality of Service
configuration
131
DSCP
131, 173
programming
173
-S-
Quick start
dialing
30
first steps
30, 30
how to connect
30
Sample rate
158
SD/SDHC card
backup, failover
48
failover, how it works
48
sampling rate and settings
-RRear Panel Connections
13
Redialling connections
36
Relay closures
169
Relays
pin outs
169
programming
169
Reset
factory default settings
150
175
48
SIP
dialing SIP addresses
44
peer-to-peer connections
43
SIP - HTML5 Toolbox
configure SIP settings
137
SDP
137
session description protocol
137
SIP server configuration
137
SIP - Java Toolbox
configure SIP programs
93
© Tieline Pty. Ltd. 2015
Index
SIP - Java Toolbox
configuring SIP settings
91
SDP
91
session description protocol
91
SIP server configuration
91
-WWarnings & safety information
digital phone systems
6
thunderstorms and lightning
SIP ports
172
Software - Java Toolbox
upgrades
113
Web Browser
Using the web-GUI
Software upgrades - HTML5 Toolbox
Specifications
199
Speed dialing
37
154
-TTCP port settings
172
Time-to-live
174
ToolBox Web-GUI options
about HTML5 Quick Connect
about HTML5 Toolbox
51
about Java Toolbox
51
compatibility
51
connecting to a codec
51
prerequisites
51
Trademarks
198
Troubleshooting
IP connection tips
TTL
51
186
174
-UUDP port settings
172
Unicasts - Java Toolbox
configuration
73
dialing
73
Unlock Programs
49
Unlock programs - HTML5 Toolbox
Upgrades - Java Toolbox
software
113
USB
connecting a PC
installing drivers
56
56
-VVLAN Configuration
© Tieline Pty. Ltd. 2015
131
205
143
6
57
Web-GUI
connecting over a LAN
52
connection pane
57
description
57
help pane
57
installing USB drivers
56
internet connections
52
LAN troubleshooting
52
launching via a USB connection
master pane
57
PC LAN settings
52
port selection
52
rules pane
57
statistics pane
57
system pane
57
Web-GUI - HTML5 Toolbox
change theme
115
connections panel
115
firmware panel
115
help panels
115
inputs panel
115
licensing panel
115
master panels
115
modules panel
115
network panel
115
opening panels
115
options panel
115
PPMs panel
115
program loader panel
115
reset panel
115
rules panel
115
settings panels
115
SIP panel
115
skin selection
115
statistics panel
115
theme selection
115
Web-GUI - Java Toolbox
dial an audio stream
97
disconnect an audio stream
97
57