Sonifex AVN-TB6D, AVN-TB6 Handbook

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Manufacturers of Audio Products for AV,
Installed Sound, Broadcast Radio & Broadcast TV
AVN-TB6
6 Button Talkback Intercom (Freestanding)
HANDBOOK
AVN-TB6D
6 Button Talkback Intercom (Desktop)
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This handbook is for use with the following product:
AVN-TB6 & AVN-TB6D 6 Button Talkback Intercom
AW10890A,
Stock Code: 30-371
Revision 1.01, July 2018
©Sonifex Ltd, 2018
All Rights Reserved
Sonifex Ltd, 61, Station Road, Irthlingborough,
Northants, NN9 5QE, England.
Tel: +44 (0)1933 650 700
Fax: +44 (0)1933 650 726
Website: http://www.sonifex.co.uk
Information in this document is subject to change without notice and does not represent a
commitment on the part of the vendor. Sonifex Ltd shall not be liable for any loss or damage
whatsoever arising from the use of information or any error contained in this manual.
No part of this manual may be reproduced or transmitted in any form or by any means, electronic
or mechanical, including photocopying, recording, information storage and retrieval systems, for any
purpose other than the purchaser’s personal use, without the express written permission of Sonifex
Ltd. Unless otherwise noted, all names of companies, products and persons contained herein are
part of a completely fictitious adaptation and are designed solely to document the use of Sonifex
product.
Made in the UK by
For the latest Sonifex handbook information please visit the Sonifex website at www.sonifex.co.uk
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Contents & Figures
Contents
Product Warranty - 2 Year Extended i
Sonifex Warranty & Liability Terms & Conditions i
1. Definitions i
2. Warranty i
Unpacking Your Product ii Repairs & Returns iii
Conformity iii
Safety & Installation of Mains Operated Equipment iii
Voltage Setting Checks iii Fuse Rating iii Power Cable & Connection iii WEEE Directive iv Atmosphere/Environment iv
1. AVN-TB6 & AVN-TB6D 6 Button Talkback Intercom 1
Introduction 1
2. Installation 3
Network Connection 3 Microphone 4 GPIO 4 Power Supply 4
3. Front Panel Controls, Connections & Indicators 5
Status Indicators 6
AoIP 6 Ethernet 6 Talk 6 AGC 6 Clock 6 AC PSU 6 DC PSU 6
Channel 1 to 6 Intercom Call Buttons 7 Listen Button 7 Page Button 7 Rotary Encoder Control 7 Reset Switch 7 Mon Button 7 User Button 7 LS Mute Button 7 Mic Mute Button 7 Microphone Input Connector 7
+48V Phantom Power Indicator 8 Headset Connector (AVN-TB6D only) 8 Headphone Output Connector 8 Power Switch 8
4. Rear Panel Connections 9
Headset Connector (AVN-TB6 only) 10 GPI/O 10 Ethernet 1GB & AoIP 1GB Network Interfaces 10 AoIP SFP Interface 10 12V DC Input 10 Mains Fuse 11 Mains AC Input 11
5. Quick Start Guide 12
6. Intercom Calls 13
Making a Call 13 Receiving a Call 13 During a Call 13 Ending a Call 13 Group Calls 13 Call Signal 13 Auto Answer 14 Listen Button 14 Call Functions 14
7. Intercom Calls with IFB 15
8. Non-Intercom Calls 16
9. Page 17
Making a Page 17 Receiving a Page 17 During a Page 17 Ending a Page 17
10. Volume Control 18
Speaker & Headphone Volume 18 Call Channel Volume 18 Monitor Volume 18 Call Signal Volume 18
11. Network Address Announcement 19
12. Embedded Web Server 20
Information Tab 20
Device Information Web Page 20 Audio Assignments Web Page 22
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Contents & Figures
GPIO Assignments Web Page 23
Configuration Tab 24
Network Web Page 25 AoIP Stream Settings Web Page 26 PTP Profiles Web Page 27 Channels - Basic Settings Web Page 29 Channels - Advanced Settings Web Page 30 Group Talk Web Page 31 Front Panel - Display Settings Web Page 32 Front Panel - Colour Settings Web Page 32 Front Panel - User Button Settings Web Page 33 Audio - Microphone Settings Web Page 34 Audio - Monitor & Speaker Settings Web Page 35 Page Settings Web Page 36 GPIO Settings Web Page 37 Virtual GPIO Settings Web Page 38 Miscellaneous Settings Web Page 39
System Tab 39
13. Ember+ Interface 42
Connecting to the Ember+ Interface 42 GPIO 42
Type Parameter 43 Input Function Parameter 43 Input Action Parameter 43 Output Trigger Parameter 43 Status Parameter 43 Set Output Function 43
Relay 43
Trigger Parameter 43 Status Parameter 43 Active Function 43
Virtual GPIO Control 44
Type Parameter 44 Input Function Parameter 44 Input Driver Name Parameter 44 Output Trigger Parameter 44 Status Parameter 44 Set Output Function 44
Parameter Value Specification 45
Identity 45 Network 45 AoIP 45 PTP 45 Display 46 Mic 46 Channels 46 Group Talk 47 Page 47 GPIO 47 Relay 48 User 49 Monitors 49 Speaker 49 Power Button 49 AC PSU Status 49 DC PSU Status 49
14. Technical Specifications 50
15. Additional Information 52
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Contents & Figures
Figures
Fig 1-1: The AVN-TB6 Front Panel 1 Fig 1-2: The AVN-TB6D Front Panel 1 Fig 2-1: Preferred Network Topology 3 Fig 3-1: AVN-TB6 and AVN-TB6D Front Panels 5 Fig 4-1: AVN-TB6 and AVN-TB6D Rear Panels 9 Table 6-1: Call Functions 14 Fig 12-1: Upper Section of Device Information Web Page 20 Fig 12-2: Lower Section of Device Information Web Page 21 Fig 12-3: Audio Assignments Web Page 22 Fig 12-4: Upper Section of GPIO Assignments Web Page 23 Fig 12-5: Lower Section of GPIO Assignments Web Page 24 Fig 12-6: Network Web Page 25 Fig 12-7: AoIP Stream Settings Web Page 26 Fig 12-8: Upper Section of PTP Profiles Web Page 27 Fig 12-9: Lower Section of PTP Profiles Web Page 27 Fig 12-10: Channels - Basic Settings Web Page 29 Fig 12-11: Channels - Advanced Settings Web Page 30 Fig 12-12: Group Talk Web Page 31 Fig 12-13: Front Panel - Display Settings Web Page 32 Fig 12-14: Front Panel - Colour Settings Web Page 32 Fig 12-15: Front Panel - User Button Settings Web Page 33 Fig 12-16: Audio - Microphone Settings Web Page 34 Fig 12-17: Audio - Monitor & Speaker Settings Web Page 35 Fig 12-18: Page Settings Web Page 36 Fig 12-19: GPIO Settings Web Page 37 Fig 12-20: Virtual GPIO Settings Web Page 38 Fig 12-21: Miscellaneous Settings Web Page 39 Fig 12-22: Update Firmware Section of System Web Page 39
Fig 12-23: Save Configuration Section of System Web Page 40 Fig 12-24: Load Configuration Section of System Web Page 40 Fig 12-25: Save System Log Section of System Web Page 41 Fig 13-1: Ember+ Root Structure 42 Fig 13-2: Ember+ GPIO Branch Structure 42 Fig 13-3: Ember+ Relay Node Structure 43 Fig 13-4: Ember+ Virtual GPIO Branch Structure 44 Table 15-1: DSCP Names & Their Corresponding IP Precedence 52
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Warranty Registration
Sonifex Limited 61 Station Road Irthlingborough Northamptonshire NN9 5QE United Kingdom Tel: +44 (0)1933 650 700 Fax: +44 (0)1933 650 726 Email: [email protected] Internet: www.sonifex.co.uk
Register Online for an Extended 2 Year Warranty
As standard, Sonifex products are supplied with a 1 year back to base warranty.
If you register the product online, you can increase your product warranty to 2 years and we can also keep you informed of any product design improvements or modifications.
Product:
Serial No:
To register your product, please go online to www.sonifex.co.uk/register
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i
Warranty
Product Warranty - 2 Year Extended
As standard, Sonifex products are supplied with a 1 year back to base warranty. In order to register the date of purchase and so that we can keep you informed of any product design improvements or modifications, it is important to complete the warranty registration online. Additionally, if you register the product on the Sonifex website, you can increase your product warranty to 2 years. Go to the Sonifex website at: www.sonifex.co.uk/ register to apply for your 2 year warranty.
Sonifex Warranty & Liability Terms & Conditions
1. Definitions
‘the Company’ means Sonifex Ltd and where relevant includes companies within the same group of companies as Sonifex Limited.
‘the Goods’ means the goods or any part thereof supplied by the Company and where relevant includes: work carried out by the Company on items supplied by the Purchaser; services supplied by the Company; and software supplied by the Company.
‘the Purchaser’ means the person or organisation who buys or has agreed to buy the Goods.
‘the Price’ means the Price of the Goods and any other charges incurred by the Company in the supply of the Goods.
‘the Warranty Term’ is the length of the product warranty which is usually 12 months from the date of despatch; except when the product has been registered at the Sonifex website when the Warranty Term is 24 months from the date of despatch.
‘the Contract’ means the quotation, these Conditions of Sale and any other document incorporated in a contract between the Company and the Purchaser.
This is the entire Contract between the parties relating to the subject matter hereof and may not be changed or terminated except in writing in accordance with the provisions of this Contract. A reference to the consent, acknowledgement, authority or agreement of the Company means in writing and only by a director of the Company.
2. Warranty
a. The Company agrees to repair or (at its discretion) replace Goods
which are found to be defective (fair wear and tear excepted) and which are returned to the Company within the Warranty Term provided that each of the following are satisfied:
i. notification of any defect is given to the Company immediately
upon its becoming apparent to the Purchaser;
ii. the Goods have only been operated under normal operating
conditions and have only been subject to normal use (and in particular the Goods must have been correctly connected and must not have been subject to high voltage or to ionising radiation and must not have been used contrary to the Company’s technical recommendations);
iii. the Goods are returned to the Company’s premises at the
Purchaser’s expense;
iv. any Goods or parts of Goods replaced shall become the
property of the Company;
v. no work whatsoever (other than normal and proper
maintenance) has been carried out to the Goods or any part of the Goods without the Company’s prior written consent;
vi. the defect has not arisen from a design made, furnished or
specified by the Purchaser;
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Warranty
vii. the Goods have been assembled or incorporated into other
goods only in accordance with any instructions issued by the Company;
viii. the defect has not arisen from a design modified by the
Purchaser;
ix. the defect has not arisen from an item manufactured by a person
other than the Company. In respect of any item manufactured by a person other than the Company, the Purchaser shall only be entitled to the benefit of any warranty or guarantee provided by such manufacturer to the Company.
b. In respect of computer software supplied by the Company the
Company does not warrant that the use of the software will be uninterrupted or error free.
c. The Company accepts liability:
(i) for death or personal injury to the extent that it results from the
negligence of the Company, its employees (whilst in the course of their employment) or its agents (in the course of the agency);
(ii) for any breach by the Company of any statutory undertaking as
to title, quiet possession and freedom from encumbrance.
d. Subject to conditions (a) and (c) from the time of despatch of
the Goods from the Company’s premises the Purchaser shall be responsible for any defect in the Goods or loss, damage, nuisance or interference whatsoever consequential economic or otherwise or wastage of material resulting from or caused by or to the Goods. In particular the Company shall not be liable for any loss of profits or other economic losses. The Company accordingly excludes all liability for the same.
e. At the request and expense of the Purchaser the Company will test
the Goods to ascertain performance levels and provide a report of
the results of that test. The report will be accurate at the time of the test, to the best of the belief and knowledge of the Company, and the Company accepts no liability in respect of its accuracy beyond that set out in Condition (a).
f. Subject to Condition (e) no representation, condition, warranty or
other term, express or implied (by statute or otherwise) is given by the Company that the Goods are of any particular quality or standard or will enable the Purchaser to attain any particular performance or result, or will be suitable for any particular purpose or use under specific conditions or will provide any particular capacity, notwithstanding that the requirement for such performance, result or capacity or that such particular purpose or conditions may have been known (or ought to have been known) to the Company, its employees or agents.
g. (i) To the extent that the Company is held legally liable to the
Purchaser for any single breach of contract, tort, representation or other act or default, the Company’s liability for the same shall not exceed the price of the Goods.
(ii) The restriction of liability in Condition (g)(i) shall not apply to
any liability accepted by the Seller in Condition (c).
h. Where the Goods are sold under a consumer transaction (as defined
by the Consumer Transactions (Restrictions on Statements) Order
1976) the statutory rights of the Purchaser are not affected by these Conditions of Sale.
Unpacking Your Product
Each product is shipped in protective packaging and should be inspected for damage before use. If there is any transit damage take pictures of the product packaging and notify the carrier immediately with all the relevant details of the shipment. Packing materials should be kept for inspection and also for if the product needs to be returned.
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CE Conformity
The product is shipped with the following equipment so please check to ensure that you have all of the items below. If anything is missing, please contact the supplier of your equipment immediately.
Item Quantity
Product unit 1
IEC mains lead fitted with moulded mains plug 1
Handbook 1
If you require a different power lead, please let us know when ordering the product.
Repairs & Returns
Please contact Sonifex or your supplier if you have any problems with your Sonifex product. Email [email protected] for the repair/ upgrade/returns procedure, or for support & questions regarding the product operation.
Conformity
The products in this manual comply with the essential requirements of the relevant European health, safety and environmental protection legislation.
The technical justification file for this product is available at Sonifex Ltd.
The declaration of conformity can be found at: http://www.sonifex.co.uk/declarations
Safety & Installation of Mains Operated Equipment
There are no user serviceable parts inside the equipment. If you should ever need to look inside the unit, always disconnect the mains supply before removing the equipment covers. The cover is connected to earth by means of the fixing screws. It is essential to maintain this earth/ ground connection to ensure a safe operating environment and provide electromagnetic shielding.
Voltage Setting Checks
Ensure that the machine operating voltage is correct for your mains power supply by checking the box in which your product was supplied. The voltage is shown on the box label. The available voltage settings are 115V, or 230V. Please note that all products are either switchable between 115V and 230V, or have a universal power supply.
Fuse Rating
The product is supplied with a single fuse in the live conducting path of the mains power input. For reasons of safety it is important that the correct rating and type of fuse is used. Incorrectly rated fuses could present a possible fire hazard, under equipment fault conditions. The active fuse is fitted on the outside rear panel of the unit.
Power Cable & Connection
An IEC power connector is supplied with the product which has a moulded plug attached.
The mains plug or IEC power connector is used as the disconnect device. The mains plug and IEC power connector shall remain readily operable to disconnect the apparatus in case of a fault or emergency.
The mains lead is automatically configured for the country that the product is being sent to, from one of:
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Safety & Installation
Territory Voltage IEC Lead Type Image
UK & Middle East 230V UK 3 pin to IEC lead
Europe 230V
European Schuko round 2 pin to IEC lead
USA, Canada and South America
115V 3 flat pin to IEC lead
Australia & New Zealand
230V Australasian 3 flat pin to IEC lead
Connect the equipment in accordance with the connection details and before applying power to the unit, check that the machine has the correct operating voltage for your mains power supply.
This apparatus is of a class I construction. It must be connected to a mains socket outlet with a protective earthing connection.
Important note: If there is an earth/ground terminal on the rear panel of the product then it must be connected to Earth.
WEEE Directive
The Waste Electrical and Electronic Equipment (WEEE) Directive was agreed on 13 February 2003, along with the related Directive 2002/95/EC on Restrictions of the use of
certain Hazardous Substances in electrical and electronic equipment (RoHS). The Waste Electrical and Electronic Equipment Directive (WEEE) aims to minimise the impacts of electrical and electronic equipment on the environment during their life times and when they become waste. All products manufactured by Sonifex Ltd have the WEEE directive label placed on the case. Sonifex Ltd will be happy to give you information about local organisations that can reprocess the product when it reaches its “end of use”, or alternatively all products that have reached “end of use” can be returned to Sonifex and will be reprocessed correctly free of charge.
Atmosphere/Environment
This apparatus should be installed in an area that is not subject to excessive temperature variation (<0°C, >50°C), moisture, dust or vibration.
This apparatus shall not be exposed to dripping or splashing, and no objects filled with water, such as vases shall be placed on the apparatus.
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Introduction 1
1. AVN-TB6 & AVN-TB6D 6 Button Talkback Intercoms
Introduction
The AVN-TB6 and AVN-TB6D are 6 channel talkback intercom control units from the Sonifex range of IP based AVN (Audio/Video/Network) products. The AVN-TB6 is a freestanding version, which can be converted into a 1U rackmount using the AVN-TB6RK rack-kit. The AVN-TB6D is a desktop version that can be fitted flush into a work surface if required. Both models offers similar functionality.
Fig 1-1: The AVN-TB6 Front Panel
Fig 1-2: The AVN-TB6D Front Panel
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1 Introduction
These units provide broadcast quality audio communication between studios, offices and different areas in a facility or building complex, using RAVENNA/AES67 as the transport mechanism, allowing simple CAT 5 cabling and expansion.
RAVENNA (of which AES67 is a subset) allows for the distribution of audio across a network. The AVN range use RAVENNA as the communication method providing compatibility with other AES67 systems.
Each of the 6 channels on the AVN-TB6 and AVN-TB6D can be configured to provide communications with other remote networked intercom units or AoIP streams, and an independently configurable page function can contact selected units with priority over standard intercom calls if required.
There is a monitor channel that can route the audio from an AoIP source to the headphones and speaker. This could be used to take an IFB feed or an off-air transmission signal or simply to listen to any audio source.
A user configurable GPIO system, with 10 physical ports and 10 virtual ports, can be used to control operational functions on local or networked units, or drive outputs as selected states change, and a voltage free relay contact can be used to operate external equipment.
A built-in web server provides complete configuration control of the units and also allows for firmware updates and configuration backup. An Ember+ interface also gives access to the configuration options as well as providing remote control and monitoring of the GPIO and virtual GPIO ports.
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Installation 2
2. Installation
Network Connection
The AVN-TB6 and AVN-TB6D have 2 separate gigabit network ports:
Upper Port: Configuration and control interface Lower Port: Audio over IP (AoIP) interface
In an ideal installation, the lower port should be connected to a dedicated network switch that handles audio and audio related traffic only. All other intercom units, AoIP sources and destinations, and grandmaster clocks should be connected to this network. The upper port can then be connected to a separate network to provide access to the embedded web server. With this setup, potentially heavy network traffic on a company LAN does not affect the performance of the AoIP network.
The AVN-TB6 and AVN-TB6D and any other Sonifex talkback intercoms can be used with a single network connection using the lower (AoIP) port as this also provides access to the embedded web server. The talkback intercom units, AoIP sources and destinations, and grandmaster clocks should still ideally be connected to a dedicated network switch.
AVN-TB6D
AVN-TB6
AVN-TB10AR
AVN-GMC
Lower
AoIP Port
Upper
Ethernet Port
Lower
AoIP Port
Upper
Ethernet Port
Lower
AoIP Port
Upper
Ethernet Port
Gigabit Ethernet Switch
Other AoIP Sources/Destinations Main Network Switch
Fig 2-1: Preferred Network Topology
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2 Installation
Microphone
These units have connections for a microphone via a 3 pin XLR on the front panel, or a headset via a 5 pin XLR. The headset connector is located on the rear panel of the AVN-TB6. On the AVN-TB6D, the headset connector is located on the front panel. Only one microphone or headset should be connected. If phantom power is required, this can be enabled via the webserver. See Audio - Microphone Settings web page on page 33 for details.
GPIO
The GPIO connector provides 10 ports, individually configurable as either inputs or outputs, via a 15 way ‘D’ type connector. Connection details are shown on page 10 Outputs are open collector driven, and inputs are active low. See GPIO Settings web page on page 37 for details.
Power Supply
The units can be powered from the mains via the supplied IEC lead, or by using the optional external 12V DC power supply, part number AVN-DC150. Please contact Sonifex Ltd sales for more details. Both power supplies can be connected and powered at the same time to provide redundancy if required.
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Front Panel Controls, Connections & Indicators 3
3. Front Panel Controls, Connections & Indicators
Fig 3-1: AVN-TB6 and AVN-TB6D Front Panels
Rotary
Encoder
Control
Microphone Input Connector
Loudspeaker
Status
Indicators
Reset
Switch
Phantom Power Indicator
Headphone Output
Power Button
Channel 1 to 6 Intercom
Call Buttons
Page
Button
Listen
Button
Status
Indicators
Microphone
Input Connector
Headset
Connector
Phantom
Power Indicator
Power Button
Headphone
Socket
Channel 1 to 6 Intercom
Call Buttons
Page
Button
Reset
Switch
Rotary Encoder
Control
Loudspeaker
Mon Button User Button
LS Mute Button Mic Mute Button
User Button
Mic Mute
Button
Mon
Button
LS Mute Button
Listen Button
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3 Front Panel Controls, Connections & Indicators
Status Indicators
The RGB status indicators are located on the left side of the front panel.
AoIP
Indicates the status of AoIP services on the network as follows:
Green All transmit AoIP streams are available and at least one external AoIP service is available.
Amber Either the transmit AoIP stream has not started or there are no external AoIP services available.
Off The transmit AoIP stream has not started and there are no external AoIP services available. The indicator will also be off if there is no link on the AoIP network port (lower network connector).
The AoIP network interface is configured via the embedded web server. See the Network web page on page 25.
Ethernet
Indicates the link status of the Ethernet network port (upper network connector) as follows:
Green The link is up.
Off The link is down.
The Ethernet network interface is configured via the embedded web server. See the Network web page on page 25.
Talk
Illuminates green if an intercom call or page is active.
AGC
Indicates the status of the automatic gain control as follows:
Green The AGC is enabled.
Red The audio source level for one of the outputs is too high and is being limited. The monitored outputs are the headphone/headset output and the speaker output.
Clock
Indicates the status of PTP clock as follows:
Green The unit is a PTP slave with a clock offset of less or equal to ±1µs or the unit is the PTP master.
Amber The unit is a PTP slave with a clock offset of between ±1µs and ±10µs.
Flash Amber The unit is in a listening state and is not yet acting as the master or a slave.
Red The unit is a PTP slave with a clock offset of greater than ±10µs.
Flash Red The unit is a PTP slave and no master is available.
AC PSU
Indicates the status of the internal AC-DC power supply as follows:
Green AC-DC generated voltage is within normal operating range of +11V to +13V. This is the normal condition.
Amber AC-DC generated voltage is less than +11V or greater than +13V. This indicates a warn condition.
Red AC-DC generated voltage is less than +10V or greater than +14V. This indicates a fault condition.
DC PSU
Indicates the status of the DC power input as follows:
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Front Panel Controls, Connections & Indicators 3
Green DC voltage is within normal operating range of +11V to +13V. This is the normal condition.
Amber DC voltage is less than +11V or greater than +13V. This indicates a warn condition.
Red Not connected or DC voltage is less than +10V or greater than +14V. This indicates a fault condition if a DC power source is connected.
Channel 1 to 6 Intercom Call Buttons
These buttons initiate an intercom call to a pre-configured target intercom unit or AoIP stream. The channels are configured via the embedded web server. See the Channels - Advanced Settings web page on page 30.
Listen Button
This button is used when making an intercom call to change the default call type. See Listen Button on page 14 for more details.
Page Button
The page button initiates a talk only call to a pre-configured list of target intercom units. The button flashes red when active. The target intercoms are selected via the embedded web server. See the Page Settings web page on page 36
Rotary Encoder Control
The rotary encoder control is used to set the volume levels for the headphones, speaker and the input levels for the various audio inputs. See Volume Control on page 18 for details.
Reset Switch
The recessed reset switch, located below the rotary encoder control, initiates a full hardware reset.
Mon Button
This button activates the monitor channel. The monitor can be set to provide audio from any AoIP source. To configure the monitor, see the Audio - Monitor & Speaker Settings web page on page 34 for details.
User Button
This button is configurable and can be used to control various functions on the unit. To configure this button, see the Front Panel - User Button web page in on on page 32.
LS Mute Button
This button mutes the audio feed to the speaker. The button illuminates red when the mute function is active. The operational mode of the button is user configurable, see the Audio - Monitor & Speaker Settings web page on page 34 for details.
Mic Mute Button
This button can be used to temporarily mute the microphone/headset input. The button illuminates red when active. The operational mode of the button is user configurable, see the Audio - Microphone Settings web page on page 33 for details.
Microphone Input Connector
This 3 pin XLR socket is the connection for the microphone. The connections are as follows:
Pin 1: Ground Pin 2: Microphone Phase Pin 3: Microphone Non-Phase
Phantom power can be supplied if required. To change the mic settings, see the Audio - Microphone Settings web page on page 33 for details.
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3 Front Panel Controls, Connections & Indicators
+48V Phantom Power Indicator
This indicator illuminates red when the phantom power supply is enabled. . To change the phantom power setting, see the Audio - Microphone Settings web page on page 33 for details.
Headset Connector (AVN-TB6D only)
This 5 pin XLR socket is the connection for the headset. The connections are as follows:
Pin 1: Microphone Non-Phase Pin 2: Microphone Phase Pin 3: Ground Pin 4: Headphone Left Pin 5: Headphone Right
Phantom power can be supplied if required. To change the headset settings, see the Audio - Microphone Settings web page on page 33 for details.
Headphone Output Connector
The headphone connection is via a 6.35mm (¼ inch) jack socket. The connections are as follows:
Tip: Headphone Left Ring: Headphone Right Sleeve: Ground
When headphones are connected, the speaker mute is automatically activated.
Power Switch
The power switch turns the unit on and off. When the unit is on, the indicator is green. When the unit is off, the indicator is red. The power switch can be disabled if required, see the Miscellaneous Settings web page on page 39 for details.
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Rear Panel Connections 4
Headset
Connector
Ethernet Network Interface
12V DC
Input
Mains Fuse
Mains AC
Input
AoIP SFP Interface
4. Rear Panel Connections
Fig 4-1: AVN-TB6 and AVN-TB6D Rear Panels
GPI/O
Ethernet Network Interface
AoIP SFP Interface
GPI/O 12V DC
Input
Mains Fuse
Mains AC Input
AoIP
Network
Interface
AoIP
Network
Interface
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4 Rear Panel Connections
Headset Connector (AVN-TB6 only)
This 5 pin XLR socket is the connection for the headset. The connections are as follows:
Pin 1: Microphone Non-Phase Pin 2: Microphone Phase Pin 3: Ground Pin 4: Headphone Left Pin 5: Headphone Right
Phantom power can be supplied if required. To change the headset settings, see the Audio - Microphone Settings web page on page 33 for details.
GPI/O
This 15 pin ‘D’ type socket provides 10 configurable general purpose inputs or outputs and a voltage free switching relay contact. The connections are as follows:
Pin 1 GPIO Port 1 Pin 2 GPIO Port 2 Pin 3 GPIO Port 3 Pin 4 GPIO Port 4 Pin 5 GPIO Port 5 Pin 6 Relay – Normally Open Contact Pin 7 Relay – Normally Closed Contact Pin 8 Relay – Common Pin 9 GPIO Port 6 Pin 10 GPIO Port 7 Pin 11 GPIO Port 8 Pin 12 GPIO Port 9 Pin 13 GPIO Port 10 Pin 14 Fused (200mA) +12VDC Supply Pin 15 Ground
GPIO ports configured as an output are open collector type. GPIO inputs are active low. See the GPIO Settings web page on page 37 for details on how to configure the GPIO and relay settings.
Ethernet 1GB & AoIP 1GB Network Interfaces
These RJ45 connectors are the gigabit network ports. The upper connector is the configuration and control port and the lower connector is the audio over IP (AoIP) port.
AoIP SFP Interface
This SFP (small form-factor pluggable) connector allows an alternative interface type to be used for the AoIP port, for example fibre. If a SFP interface is used, the AoIP network port cannot be used at the same time.
12V DC Input
This 4 pin connector allows an external +12V power supply to be used to power the unit. When looking at the rear of the unit, the pin locations and connections are as follows:
Location Connection Pin 1 Upper Left +12V DC Pin 2 Upper Right +12V DC Pin 3 Lower Left 0V Pin 4 Lower Right 0V
Both the 12V DC input and the mains AC input can be connected at the same time.
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Rear Panel Connections 4
Mains Fuse
This 20mm x 5mm anti-surge mains fuse is rated at 2A.
Mains AC Input
This universally filtered IEC is the mains AC input power supply to the unit.
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5 Quick Start Guide
5. Quick Start Guide
This section outlines the steps required to get the AVN-TB6/AVN-TB6D connected to a network and in a state where it can make and receive intercom calls.
1. Connect the upper Ethernet port to a suitable network to access the web server to configure the initial settings. By default this port is configured to use the static IP address 192.168.0.100.
2. Apply power to the unit via the IEC mains connection or via an external 12V DC power supply.
3. Once the unit is powered, open a web browser and connect to the unit by entering the IP address 192.168.0.100 into the browser address bar. Once connected, the device information web page will be shown.
4. On the web page, select Network from the configuration tab and configure the Ethernet and AoIP ports to the required settings. The IP addresses assigned to the 2 ports must be on different subnets. For example, with the subnet mask on both ports set to 255.255.255.0, if the IP address on the Ethernet port is set to 192.168.0.100, the AoIP port must be set to 192.168.x.y where x is not 0 and y is unique on the chosen subnet.
5. Although not essential at this point, it is a good idea to assign a recognisable intercom name so that other users can easily identify and use services from this unit.
6. To simplify the configuration process, make sure the Auto Multicast option is enabled. This removes the need to set the transmit AoIP multicast IP address as it will be configured automatically based on the AoIP port IP address. Once the settings on the network page have been configured, press the submit button to save them to the connected unit. When the network settings have been changed, the unit will automatically restart. If the settings of the network port the browser is connected to have changed, a new connection for the browser will need to be established using the new IP address. If dynamic address mode has been selected, the active IP address for the network port can
be determined by using the network address announcement function. See Network Address Announcement on page 19 for details.
7. If a dedicated grandmaster clock, such as the AVN-GMC from Sonifex, is going to be used as the master PTP clock for the network audio system, the PTP profile on the AVN-TB6/AVN-TB6D will need to be set to the current setting on the grandmaster device. On the web page, select PTP Profiles from the configuration tab. Select the required profile from the drop down list box and ensure that the domain number for the selected profile matches the grandmaster setting. Press the submit button to save the new settings. If no dedicated grandmaster device is used, the connected intercom units will automatically establish a master clock hierarchy. In this case, the default profile can be used.
8. Connect the lower AoIP port to the audio network.
9. On the web page, select Microphone Settings from the Audio sub tab. Select the required microphone type, with or without phantom power. Now change the microphone or headset gain so that metering LEDs around the front panel rotary control peak in the amber section for normal speech. Press the submit button to save the new settings.
10. The final step is to assign remote intercoms or AoIP streams to the required channels. On the web page, select Basic Settings from the Channels sub tab. Select the required remote intercom or AoIP stream from the Available AoIP Sources list box at the top of the page. This shows a dynamic list of currently available intercom devices including those with IFB, and AoIP streams on the network. To configure a channel for the selected AoIP source, press the corresponding assign button. For sources that are not currently available, the source name can be manually entered in the edit box for each channel. For intercom assignments, a channel on the selected device will need to be assigned to this unit in order to make intercom calls. Press the submit button to save the new settings.
The unit should now have the minimum settings configured to allow intercom calls between devices and AoIP streams.
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Intercom Calls 6
6. Intercom Calls
Each channel button on the front panel can be assigned to a remote intercom unit in order to make and receive voice calls. See the Channels - Advanced Settings web page on page 30 for full details of the available options.
Making a Call
To make a call request, press the channel button assigned to the required remote unit. If the called unit has the auto answer option enabled, the call will be connected immediately. The button status for an active call is described later in this section. If the called unit’s auto answer option is disabled, the selected channel button will slowly flash using the colour assigned to the default function for that channel. For example, if the default function is talk and listen, and the colour assigned to the talk and listen buttons is amber, the button will flash amber. If a call request is made to an intercom that doesn’t have a valid channel assignment for the calling unit, the call will be cancelled. To correct this error, assign the intercom name from the unit attempting to make the call to a channel on the target unit.
Receiving a Call
When an intercom call request is received from a remote unit, the call will be handled by the local channel assigned to the calling intercom unit. The way the call is handled is dependent on the auto answer option setting for the local channel. If the auto answer option is enabled, the call is connected immediately. If the auto answer option is disabled, the channel button that is assigned to the caller will rapidly flash using the colour assigned to the requested call type. For example, if the calling unit has requested a talk only call, the call will be treated locally as listen only. If the local colour assignment for listen buttons is green, the channel button will flash green. If the call signal is enabled, a repeated audible signal will be heard in the headphones or speaker. The call is answered by pressing the flashing channel button and the audio from each unit will be routed depending on the callers selected call type.
During a Call
When an intercom call is connected, the channel button is illuminated using the colour assigned to the call type. The intercom unit that initiates the call always dictates the call type. If the monitor channel is active when an intercom call is connected, the audio routed to the headphones or speaker from the monitor source is automatically muted or dimmed according to the monitor settings.
Ending a Call
An active call can be terminated by either party. To end a call, press the illuminated channel button. The call is ended immediately and the button illumination is turned off. If the monitor channel is active, it will return to the normal state when all calls have ended.
Group Calls
Intercom calls can be made simultaneously to multiple remote units using the group talk function. The group talk set can include any combination of channels, see the Group Talk web page on page 31 for details. Group calls are selected by assigning the group talk function to the user button, see the Front Panel - User Button web page on page 32 for details. The user button, linked to the group talk set, can then be used to make or end a group call.
Call Signal
The call signal provides an audible indication that an incoming call is waiting to be answered. It can be heard in the headphones and through the speaker. If the call signal is not required, it can be turned off. See the Channels - Advanced Settings web page on page 30 for details.
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6 Intercom Calls
Auto Answer
The auto answer option is individually configurable for each channel. An incoming call request on a channel that has this option enabled will connect immediately. To change the auto answer settings, see the Channels ­Advanced Settings web page on page 30 for details.
Listen Button
Each channel has a default function option that selects the type of call that is initiated when the corresponding channel button is pressed. See Channels - Advanced Settings web page on page 30 for details. The front panel listen button can be used to change the default call function to listen only, talk and listen or talk forced listen. To override the default function for a channel, press the listen button. The first press will select listen only, and the button will illuminate using the colour assigned listen buttons. A second press will select talk and listen, and the button will illuminate using the colour assigned to talk and listen buttons. A third press will select talk forced listen, and the button will flash alternately using the colours assigned to both talk and listen buttons. A fourth press will cancel the listen function and the button illumination will turn off. Once the listen button indicates the required call function, press a channel button to call the assigned unit. If a channel button is not pressed within 3 seconds of the last listen button
press, the listen function is cancelled.
Call Functions
The following table shows the corresponding call functions between the calling and receiving unit. The default channel button colours for the call type is also shown.
Calling Unit Receiving Unit Call Type Default Channel
Button Colour
Call Type Default Channel
Button Colour
Talk Only Red Listen Only Green Listen Only Green Talk Only Red Talk & Listen Amber Talk & Listen Amber Talk Forced Listen
- Talk inactive
- Talk Active
Green with Red Flash Red with Green Flash
Talk Forced Listen Talk Only Talk & Listen
Red Amber
Table 6-1: Call Functions
The Talk Forced Listen call works differently to the standard talk/listen functions. When this type of call is initiated, the call starts as a listen only call. The callers channel button is illuminated using the colour assigned to listen buttons, with a brief flash using the colour assigned to talk buttons. The channel button on the receiving unit is illuminated to indicate a talk call. The calling unit can then repeatedly press the channel button to enable and disable talk. When talk is enabled, the channel button illuminates using the colour assigned to talk buttons, with a brief flash using the colour assigned to listen buttons. The channel button on the receiving unit is illuminated to indicate a talk & listen call when the callers talk is enabled. The receiving unit can end the call by pressing the channel button as per other call functions. The calling unit ends the call by briefly holding the channel button until the illumination has turned off.
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Intercom Calls with IFB 7
7. Intercom Calls with IFB
Other models in the AVN Talkback range, such as the AVN-TB10AR, AVN­TB20AR and the AVN-TB20AD, can generate specific IFB AoIP streams. These IFB streams from remote units can by assigned to the channel buttons on the AVN-TB6 and AVN-TB6D by enabling the channel IFB mode. See the Channels - Advanced Setting web page on page 30 for more details. With IFB mode enabled, the selected IFB feed from the remote unit will be heard in the headphones and through the speaker when the channel is idle, and the channel button will flash briefly in the colour assigned to listen buttons. Once a call is initiated and connected, the remote unit will modify the audio content of the IFB stream and the receiver will hear the required interrupt audio. Intercom calls with IFB are controlled in exactly the same way as normal intercom calls.
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8 Non-Intercom Calls
8. Non-Intercom Calls
The AVN-TB6 and AVN-TB6D allow any AoIP stream to be assigned to the channel buttons. When a call is initiated to a non AVN Talkback device, the unit does not attempt to communicate with the remote device for call control. It simply routes the AoIP stream assigned to the channel to the headphones and loudspeaker and unmutes the microphone audio routed to the microphone AoIP transmit stream. If simply call indication is required between devices, the GPIO ports on the unit can be used to provide control and tally functions. See the GPIO Settings web page on page 37 for details on how to configure the GPIO and relay settings.
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Page 9
9. Page
The page function provides an easy method of making a high priority talk only call to a preselected group of target intercoms. Up to 20 intercom units can be selected as page targets and each unit can be configured to treat received page calls as a priority over normal intercom calls. See the Page Settings web page on page 36 for details.
Making a Page
To make a page call, press the page button. The button will slowly flash green if at least one of the configured page targets is able to receive the page. A configured target may not be receiving the page for a number of reasons. These include:
• The target unit may be switched off or not connected to the network.
• The target unit may be already involved in a page call, either outgoing or incoming.
• The target unit may be involved in an intercom call and the page priority option on that unit is disabled, meaning that the intercom call takes priority.
Receiving a Page
When a page is received and accepted, the page button fast flashes red. If the page priority option is disabled, and an intercom call is currently active, the page will be ignored.
During a Page
During a page call, the microphone audio signal from the paging unit will be heard by the units that accepted the page. The page targets do not need to respond to the page to hear the audio. Any current intercom calls on the units receiving the page will be suspended and any active monitor channels on the paging and receiving units will be muted for the duration of the page call.
Ending a Page
A page call can only be terminated by the unit that initiated it. To end a page call, press the green flashing page button. The page is ended immediately and the button stops flashing. Any suspended intercom calls on the target page units are resumed and any active monitor channels on the paging and receiving units are unmuted.
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10 Volume Control
10. Volume Control
The rotary encoder control on the front panel is used to set the volume levels of the speaker, the headphones and the input levels for the various audio sources.
Speaker & Headphone Volume
The volume control automatically switches from speaker to headphones when a headphone jack plug is connected. To adjust the volume, turn the rotary encoder control. The LEDs around the rotary control show the current level between minimum (muted) and maximum (+12dB gain). There are 21 LEDs and they are illuminated in a 1, 2, 1, 2 pattern giving 41 discrete level steps. The LEDs will return to showing the microphone signal level if no adjustments have been made for 1.5 seconds, or if the rotary control button is pressed.
Call Channel Volume
The volume levels of the audio from remote units assigned to the call channel buttons can be individually adjusted to compensate for varying talk levels. This can be done during a call if required. To adjust the channel volume, press the rotary encoder control which will then flash to indicate that volume adjustment mode has been initiated. This mode will automatically cancel if no further action is taken within 3 seconds. Then select a channel to adjust by pressing the associated channel button. The rotary encoder control and the selected channel button will now flash using the colour assigned to the default function for that channel. The LEDs around the rotary control show the current level in a similar way to the speaker and headphone volume. Turn the rotary encoder control to adjust the volume level. The volume adjustment mode will automatically cancel if no adjustments have been made for 1.5 seconds, or if the rotary control button is pressed.
Monitor Volume
The volume level of the audio from the selected monitor source can be adjusted to compensate for varying levels. This can be done regardless of whether the monitor is currently active or not. To adjust the monitor volume, press the rotary encoder control which will then flash to indicate that volume adjustment mode has been initiated. This mode will automatically cancel if no further action is taken within 3 seconds. Then press the monitor button. The rotary encoder control and the monitor button will now flash green. The LEDs around the rotary control show the current level in a similar way to the speaker and headphone volume. Turn the rotary encoder control to adjust the monitor level. The volume adjustment mode will automatically cancel if no adjustments have been made for 1.5 seconds, or if the rotary control button is pressed.
Call Signal Volume
The volume level of the call signal routed to the speaker and headphones can be adjusted to provide a comfortable mix level with other active audio sources. The call signal volume control automatically switches from speaker to headphones when a headphone jack plug is connected. To adjust the call signal volume, press and hold the rotary encoder control until it starts to flash blue. The LEDs around the rotary control show the current level in a similar way to the speaker and headphone volume. Turn the rotary encoder control to adjust the call signal level. The volume adjustment mode will automatically cancel if no adjustments have been made for 1.5 seconds, or if the rotary control button is pressed.
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Network Address Announcement 11
11. Network Address Announcement
These units have a network address announcement mode which can be used to determine the current settings for either of the 2 network ports:
1. Ensure that volume control mode is not currently active i.e. the LEDs around the rotary encoder control are showing the microphone signal level. Also, ensure no calls or pages are active.
2. Press the rotary encoder control button 6 times within 2 seconds. If the button is not pressed 6 times within a 2 second period starting from the first press, the network address announcement mode will not start.
3. Once the network address announcement mode has been initiated, the rotary encoder control and channel buttons 1, 2, 4 and 5 will flash red.
4. Press channel 1 for the Ethernet port IP address, channel 2 for the Ethernet port sub-net mask, channel 4 for the AoIP port IP address and channel 5 for the AoIP port sub-net mask. Once a selection has been made, the chosen details will be announced through the loudspeaker or the headphones. If no selection is made within 3 seconds, the network address announcement mode will be cancelled, and the rotary encoder control and channel buttons will return to the non-illuminated state.
5. Once the chosen details have been announced, the unit return to normal operating mode.
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12 Embedded Web Server
12. Embedded Web Server
The AVN-TB6 and AVN-TB6D have embedded web servers which provide easy access to all the configuration options through a web browser. It also gives access to system information and allows the firmware to be easily updated when new releases are made available.
model label. This makes it easier to identify the connected intercom unit, especially when configuring multiple units at the same time. The right hand side of each page has a brief help section that describes the content of each section.
Information Tab
The pages under the Information tab provide device information as well as a summary of the audio and GPIO assignments.
Device Information Web Page
The top part of this page shows details of the connected unit as shown:
To access the embedded web server, the unit needs to be connected to a network via either of the 2 network ports on the rear panel. The upper port is the general access Ethernet port and the lower port is the audio over IP (AoIP) port. By default, both ports are set to static address mode with the upper port IP address set to
192.168.0.100 and the lower port IP address set to 192.168.1.100. If the network address mode for the port to be used has been set to dynamic, the unit will attempt to acquire an IP address from a DHCP server or auto configuration if no DHCP server is found. The current IP address of these ports can be determined by using the network address announcement function. See Network Address Announcement on page 19 for details. If a port has yet to acquire an IP address or has failed to link, the corresponding IP address and subnet mask be announced as
0.0.0.0. Once the IP address of the required port is known, simply type this into the address bar of a web browser. The Device Information page of the connected unit will be displayed. This is the default page and will always be displayed first when connecting to the embedded web server. Each page of the web server shows the intercom name in the upper right hand corner, under the
Fig 12-1: Upper Section of Device Information Web Page
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Embedded Web Server 12
This information shows the current status of the unit as well as the software versions of the various modules running on the unit. When contacting Sonifex technical support, it is important to provide the information shown on this page.
The lower half of the page shows the status of the Precision Time Protocol (PTP) clock, as well as the configuration of the network ports as shown:
Fig 12-2: Lower Section of Device Information Web Page
The network IP addresses and subnet masks shown are the actual values currently in use.
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12 Embedded Web Server
Two additional pages under the information tab, provide a configuration summary of the audio and GPIO assignments.
Audio Assignments Web Page
The Audio Assignments page shows the current channel and monitor configuration as shown:
Fig 12-3: Audio Assignments Web Page
Each channel is listed with the intercom name or AoIP stream name assignment if any. Channels that are assigned to remote sources indicate whether that source is currently online. If it is online, the assignment is shown in black and the AoIP port IP address of the source is also shown. If the target assignment is an AVN talkback device, an AVN logo is shown next to the channel number. Unavailable sources or un­assigned channels are shown in grey.
The Monitor Assignment shows the current configuration and status of the monitor channel. If the configured AoIP stream is currently available, the assignment is shown in black. Unavailable streams are shown in gr ey.
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GPIO Assignments Web Page
The GPIO Assignments page shows the current GPIO, virtual GPIO and relay configuration as shown:
Fig 12-4: Upper Section of GPIO Assignments Web Page
Each physical GPIO can be configured as an input, output or it can be disabled. Any configuration errors that will stop the GPIO functioning are shown in red. For example, GPIO 6 has been assigned as an input that activates a virtual output. However, none of the virtual GPIO have been set to an output that is driven by GPI6. At the bottom of the GPIO assignments is the list of functions that trigger the relay – if any.
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12 Embedded Web Server
Each virtual GPIO can also be configured as an input, output or it can be disabled. As with the GPIO assignments, configuration errors are shown in red. Virtual inputs require a remote virtual output driver to trigger the input. The driver details are shown below the virtual input description and will be grey if the driver service is not available. If the service is available, it will be shown in black.
Fig 12-5: Lower Section of GPIO Assignments Web Page
Configuration Tab
The pages under the Configuration tab are split into groups and all of the unit’s configuration settings can be edited here. Each page has a SUBMIT button that sends the changes to the connected unit. The SUBMIT button remains disabled until changes have been made on the page. Once the SUBMIT button has been pressed, a busy indicator will be shown and the page will be temporarily disabled until the changes have been written.
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Network Web Page
The Network page shows the current configuration of the Ethernet and AoIP network ports. The intercom name is also set here.
Intercom Name – The intercom name is the user friendly name that identifies the unit on the network. It is a good idea to assign a user name or location as this is easily recognised by other users. The default intercom name is made from the device ID and the 7 digit product serial number i.e. AVN-TB6-1234567. The intercom name can only contain letters, numbers and hyphens although it cannot start or end with a hyphen.
Address Mode – Each network port has its own independent address mode which determines how the port obtains its IP address. When set to dynamic, the unit will attempt to acquire an IP address automatically from either a DHCP server or via auto configuration if no DHCP server is found. The actual IP address will be shown on the device information page. When static mode is used, the IP address and subnet mask values entered will be assigned to the corresponding network port.
Static IP Address – This is the IP address that will be assigned to the corresponding network port when static address mode is selected. It is important to ensure that this IP address is not currently in use on the network. These values are not used when the address mode is dynamic.
Static Subnet Mask – This is the subnet mask that will be used for the corresponding network port when static address mode is selected. These values are not used when the address mode is dynamic.
Fig 12-6: Network Web Page
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Auto Multicast – When this option is enabled and the address mode of the AoIP port is set to static, the AoIP transmit multicast IP address, shown on the AoIP Streams Web Page, is automatically configured based on the AoIP port IP address. If the IP address of the AoIP port is changed, the multicast address is updated. This simplifies configuration and ensures the multicast address is unique on the network.
If any of the network configuration options are changed, the unit will automatically restart to implement the new settings. If the IP address of the network port that is currently being used to access the web server is changed, a new connection will need to be made once the unit has restarted. Otherwise, the new page will be shown automatically once the restart is complete.
Network Defaults Intercom Name: AVN-TB6-xxxxxxx or AVN-TB6D-xxxxxxx
Where xxxxxxx is the product serial number Ethernet Port: Address Mode: Static Static IP Address: 192.168.0.100 Static Subnet Mask: 255.255.255.0 Audio over IP Port: Address Mode: Static Static IP Address: 192.168.1.100 Static Subnet Mask: 255.255.255.0 Auto Multicast: Enabled
AoIP Stream Settings Web Page
The AoIP Stream Settings page shows the current configuration of the AoIP options. The single transmit source on the AVN-TB6 and AVN-TB6D is for the microphone/headset input.
Fig 12-7: AoIP Stream Settings Web Page
AoIP Stream Name Prefix – The AoIP stream name prefix forms part of the AoIP stream name which will be displayed on remote devices when searching for AoIP streams. This stream name is comprised of the AoIP stream name prefix, set here for the microphone transmit source, and the intercom name. For example, if the default microphone AoIP stream name prefix “Mic” is used, and the intercom name is set to “DEMO-USER”, the advertised AoIP stream name would be “Mic@DEMO-USER”.
DSCP – The DSCP or Differentiated Services Code Point value is used by DiffServ to control the precedence of outgoing AoIP data packets. Network hardware can use this value to ensure audio data has higher priority over other network traffic. See Table 15-1 DSCP Names & Their Corresponding IP Precedence on page 51 for more information.
Multicast IP Address – The multicast IP address is the IPv4 multicast destination address for the transmit source. Remote network devices that are configured to receive this source will use this IP address to access the
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Embedded Web Server 12
audio data. It is important to ensure that the IP address entered here is unique on the network as AoIP stream sources cannot share the same multicast IP address. Multicast IP addresses should be within the range
224.0.0.0 to 239.255.255.255. If the Auto Multicast option on the Network
page is enabled and the AoIP port address mode is set to static, this multicast IP address is automatically configured and cannot be changed.
AoIP Stream Settings Defaults
Mic AoIP Source: AoIP Stream Name Prefix: Mic DSCP: AF41 Multicast IP Address: 239.0.1.100
PTP Profiles Web Page
The PTP Profiles page shows the currently selected PTP profile. It also allows the parameters in each profile to be edited.
Fig 12-8: Upper Section of PTP Profiles Web Page
Active Profile – There are 3 PTP profiles available: Default, AES67 Media and Custom. This drop down list selects which of the profiles will be active.
DSCP – The DSCP or Differentiated Services Code Point value is used by DiffServ to control the precedence of outgoing PTP packets. Network hardware can use this value to ensure PTP data has higher priority over other network traffic. See Table 13-1 – DSCP Names & Their Corresponding IP Precedence on page 51 for more information.
Fig 12-9: Lower Section of PTP Profiles Web Page
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Profile Select – This drop down list selects the profile to view and change settings. All of the available profiles can be edited.
Reset to Defaults – Selecting this checkbox and pressing the submit button resets the selected profile to the default settings for that profile.
Delay Mechanism – This selects the method for calculating and maintaining the clock synchronisation communication delay between master and slave devices. Both P2P (Peer to Peer) and E2E (End to End) methods are supported. The best method for a particular installation will depend on the type of network hardware used. If all the switches are known to be IEEE 1588 capable, P2P should be used. Otherwise it is best to use E2E.
Announce Interval – This is the rate at which the announce message is sent, one per the selected time. The announce message forms part of the Best Master Clock Algorithm (BMCA) and contains the properties of the clock which sends it. If a device receives an announce message from a better clock, it will enter the slave state.
Announce Timeout – This is the time the unit will wait for an announce message and is a multiple of announce intervals. If no announce messages are received within the timeout interval, the unit will assume the role of grandmaster.
Sync Interval – This is the rate at which the sync message is sent, one per the selected time. The sync and follow up messages are sent from the master to the slave to determine the difference in clock frequency. This information is used in conjunction with the network delay to synchronise the clocks.
Delay Request Interval – This defines the rate at which a slave clock sends delay request messages to the master. The delay request message allows a slave device to calculate the network delay from the slave to the master. This option is only valid when using the E2E delay mechanism.
Peer Delay Request Interval – This is the rate at which a device exchanges peer delay measurement messages. This allows each unit to track the network delays between itself and its connected neighbours. This option is only valid when using the P2P delay mechanism.
Priority 1 & Priority 2 – These values define a precedence setting used by the best master clock algorithm when selecting a grandmaster clock. Priority 2 defines a fine tune setting when multiple clock sources have similar ordering criteria.
Domain – This defines the group of PTP devices that the unit will communicate with.
Slave Only – This option limits the unit to being a slave PTP clock only.
PTP Profiles Defaults
Active Profile: Default DSCP: AF PHB Default Profile: Delay Mechanism: E2E Announce Interval: 2 secs Announce Timeout: 3 secs Sync Interval: 1 sec Delay Request Interval: 1 sec Peer Delay Request Interval: 1 sec Priority 1: 128 Priority 2: 128 Domain: 0 Slave Only: Disabled AES67 Media Profile: Delay Mechanism: E2E Announce Interval: 2 secs Announce Timeout: 3 secs
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Sync Interval: 1/8 sec Delay Request Interval: 1 sec Peer Delay Request Interval: 1 sec Priority 1: 128 Priority 2: 128 Domain: 0 Slave Only: Disabled Custom Profile: Delay Mechanism: E2E Announce Interval: 2 secs Announce Timeout: 3 secs Sync Interval: 1 sec Delay Request Interval: 1 sec Peer Delay Request Interval: 1 sec Priority 1: 128 Priority 2: 128 Domain: 0 Slave Only: Disabled
Channels - Basic Settings Web Page
The Basic Settings page is located under the Channels sub tab. It allows the AoIP stream source assignments to be set for all 6 channel buttons on the front panel. The other channel options are shown on the Channels ­Advanced Settings web page – see page 30 for details.
Fig 12-10: Channels - Basic Settings Web Page
Available AoIP Sources – This dynamic drop down list shows the 3 types of AoIP sources that are currently visible on the network and available for selection as a target for a channel. These types are AVN Intercom Devices, AVN Intercom Devices with IFB and AoIP Streams. To make an assignment, select a source from the list and then press the assign button on the required channel.
Channels 1 - 6 – Each row shows the current AoIP source assignment for the corresponding channel. To change the stream name, select a source from the Available AoIP Sources list and press the assign button on the required channel. Alternatively, the stream name can also be entered manually for remote sources that are currently offline.
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Channels - Basic Settings Defaults
Channels 1 – 6: AoIP Source: not set
Channels - Advanced Settings Web Page
The Advanced Settings page is located under the Channels sub tab. It shows the current assignments for each of the 6 channel buttons on the front panel. Other options that control intercom calls are also set here.
Fig 12-11: Channels - Advanced Settings Web Page
Channel Select – This drop down list selects the channel to view and change settings.
AoIP Source – This setting assigns the intercom name or AoIP stream name to connect to when the intercom channel is active. If the source is an intercom device, this name corresponds to the intercom name that is assigned on the network web page. This name can be entered manually, or it can be selected from the dynamic list of available AoIP sources detailed below.
Available AoIP Sources – This dynamic drop down list shows the 3 types of AoIP sources that are currently visible on the network and available for selection as a target for a channel. These types are AVN Intercom Devices, AVN Intercom Devices with IFB and AoIP Streams. Selecting an AoIP source from this list places the name into the AoIP source edit box.
IFB Mode – This option allows the selected channel to be assigned to an IFB AoIP stream from a remote intercom device. This stream type will have a designated audio feed that will be replaced by or mixed with the microphone audio when the channel is active. If IFB mode for the channel is enabled and the AoIP source is an available intercom device, but the configured IFB stream does not exists, a non IFB mode call will be connected instead.
IFB Source – This is the AoIP stream name prefix of the IFB stream on the selected AoIP Source. This name can be entered manually, or it can be selected from the dynamic list of available IFB sources detailed below. This option is only available if IFB mode is enabled.
Available IFB Sources – This dynamic drop down list shows the names of the available IFB streams on the selected AoIP source. If the selected AoIP source is not an IFB enabled intercom device, this list will be empty. This option is only available if IFB mode is enabled.
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Auto Answer – This option allows an incoming call on the selected channel to be answered automatically without the need to press the associated channel button.
Default Function – The default function option sets the call type that is initiated when the corresponding channel button is pressed. Talk & Listen calls allow both the caller and receiver hear each other. In a Talk Only call, only the receiver hears the caller. In a Listen Only call, only the caller hears the receiver. Talk Forced Listen calls allow the caller to hear the receiver, with the option to enable or disable talk to the receiver by repeatedly pressing the channel button. If IFB mode is enabled on the selected channel, Talk Forced Listen is not available. The default function can be overridden when initiating call by using the listen button on the front panel.
Button Mode – Selects the operational mode of the selected channel talk button. In momentary mode, the channel is active whilst the button is pressed. In latching mode, a separate button press is required to enable and disable the channel. In auto mode, the button has a momentary action that will automatically latch on or off if pressed briefly. The button can also be disabled if required. Momentary mode is not available if auto answer is enabled.
Call Signal – This option enables an audible indication that an incoming call is waiting to be answered. The call signal is heard through the loudspeaker, or the headphones if connected. This setting applies to all channels.
Channels - Advanced Settings Defaults Channels 1 – 6: AoIP Source: Not set IFB Mode: Disabled IFB Source: Not set Auto Answer: Disabled Default Function: Talk & Listen Button Mode: Auto Call Signal: Enabled
Group Talk Web Page
The Group Talk page shows the channels that are included in the group talk set.
Fig 12-12: Group Talk Web Page
Channels – Click on this list to select the channels that are to be included in the group talk set. To select multiple channels, hold down the Ctrl key on the keyboard and then make a selection.
Group Talk Defaults
Channels: 1, 2, 3, 4, 5, 6
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Front Panel - Display Settings Web Page
The Display Settings page is located under the Front Panel sub tab. It allows the LED brightness to be adjusted.
Fig 12-13: Front Panel - Display Settings Web Page
LED Brightness – Click on the + or – buttons to change the brightness of the LEDs on the front panel in the range 1 to 7.
Front Panel - Display Settings Defaults
LED Brightness: 4
Front Panel - Colour Settings Web Page
The Colour Settings page is located under the Front Panel sub tab. It shows the colours that are assigned to the intercom call and user buttons as shown:
Fig 12-14: Front Panel - Colour Settings Web Page
Talk Buttons – This setting defines the colour of the channel buttons when a talk only call is active.
Listen Buttons – This setting defines the colour of the channel buttons when a listen only call is active.
Talk & Listen Buttons – This setting defines the colour of the channel buttons when a talk & listen call is active.
User Button – This setting defines the colour of the user assignable function button.
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Front Panel - Colour Settings Defaults
Talk Buttons: Red Listen Buttons: Green Talk & Listen Buttons: Amber User Button: Blue
Front Panel - User Button Settings Web Page
The User Button Settings page is located under the Front Panel sub tab. It shows the configuration of the user button.
Fig 12-15: Front Panel - User Button Settings Web Page
Function – This option sets the function for the user button. Group Talk initiates a group talk call to the members of the group talk set. GPO Control allows a selection of physical and virtual GPO ports to be asserted. The GPO and virtual GPO ports that are under user button control are assigned in the GPIO and virtual GPIO settings pages.
Button Mode – Selects the operational mode of the selected user button. In momentary mode, the selected user function is active whilst the button is pressed. In latching mode, a separate button press is required to enable and disable the selected user function. In auto mode, the button has a momentary action that will automatically latch on or off if pressed briefly. The button can also be disabled if required.
Front Panel - User Button Defaults
User Button: Function: Group Talk Button Mode: Auto
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Audio - Microphone Settings Web Page
The Microphone Settings page is located under the Audio sub tab. The microphone input type and associated options are set here.
Fig 12-16: Audio - Microphone Settings Web Page
Microphone Type – This drop down list selects the required microphone input type and also enables phantom power if required.
Gain Control – This option selects between fixed or auto gain control. When auto gain is selected, the gain levels for the mic and headset inputs are automatically adjusted to compensate for varying talk levels.
Mic Gain – This option sets the input gain for the mic in the range 0dB to +60dB in 3dB steps.
Headset Gain – This option sets the input gain for the headset in the range 0dB to +60dB in 3dB steps.
AGC Target Level – This option sets the required output level for the microphone/headset automatic gain control. This option is available only when the gain control is set to auto gain.
Auto Mute – This option selects whether the microphone or headset AoIP stream is automatically muted when no calls are active. Please note, if this option is not enabled, the audio from the microphone or headset will be available to any other devices on the network.
Mute Button Mode – Selects the operational mode of the mic mute button on the front panel. In momentary mode, the mic mute is active whilst the button is pressed. In latching mode, a separate button press is required to enable and disable the mic mute function. In auto mode, the button has a momentary action that will automatically latch on or off if pressed briefly. The button can also be disabled if required.
Audio - Microphone Settings Defaults
Microphone Type: Normal Mic Gain Control: Fixed Gain Mic Gain: 30 Headset Gain: 30 AGC Target Level: -18dBFS Auto Mute: Enabled Mute Button Mode: Auto
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Audio - Monitor & Speaker Settings Web Page
The Monitor & Speaker Settings page is located under the Audio sub tab and it shows the current assignment for the monitor channel. The loudspeaker options are also set here.
Fig 12-17: Audio - Monitor & Speaker Settings Web Page
AoIP Stream – This setting is the name of the AoIP stream that will be routed to the monitor channel. The stream name is comprised of the AoIP stream name prefix and the intercom name. For example, if the required AoIP stream, called “Input1”, is being transmitted from an intercom unit with the name “DEMO-SOURCE”, the AoIP stream name would be “Input1@ DEMO-SOURCE”. This name can be entered manually, or it can be selected from the dynamic list of available AoIP streams detailed below.
Available AoIP Streams – This dynamic drop down list shows the AoIP streams that are currently visible on the network. Selecting an AoIP stream from this list places the name into the AoIP stream edit box.
Talkback Mode – This setting controls how the audio level of the monitor is modified when an intercom channel is active. There are 3 options, the level of the monitor is either muted, reduced by 75% or reduced by 50%.
Button Mode – Selects the operational mode of the monitor button on the front panel. In momentary mode, the monitor is active whilst the button is pressed. In latching mode, a separate button press is required to enable and disable the monitor. In auto mode, the button has a momentary action that will automatically latch on or off if pressed briefly. The button can also be disabled if required.
Echo Cancellation – This setting enables the acoustic echo cancellation algorithm which removes audio content, originating from the speaker, from the microphone input. This prevents a remote user from hearing a delayed echo of themselves.
Mute Button Mode – This setting selects the operational mode of the speaker mute button on the front panel. In momentary mode, the speaker is muted whilst the button is pressed. In latching mode, a separate button press is required to enable and disable the speaker mute. In auto mode, the button has a momentary action that will automatically latch on or off if pressed briefly. The button can also be disabled if required with the mute permanently on or off.
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Audio - Monitor & Speaker Settings Defaults
AoIP Stream: not set Talkback Mode: Mute Button Mode: Auto Echo Cancellation: Disabled Mute Button Mode: Auto
Page Settings Web Page
The Page Settings page shows the current list of target intercoms for the page function. The page priority setting is also set here.
Fig 12-18: Page Settings Web Page
Page Priority – This option selects whether received page calls have priority over current intercom calls. If this option is enabled, all current intercom calls will be suspended during a received page call. These calls will then resume once the page call has ended. If a page request is received and this option is disabled, the page will be ignored.
Page Targets – This is the list of up to 20 target intercom devices that will be called during a page. To add to the list, select a new device from the available intercom devices select box or manually enter the name in the intercom devices edit and then press the ADD button. To remove one or more targets from the list, highlight the required page targets and press the DEL button. Multiple targets can be selected by clicking the required options whilst holding down the Ctrl key on the keyboard.
Intercom Device – A new page target intercom name can be added to the list by entering the name in this edit box and pressing the ADD button. Alternatively, it can be selected from the dynamic list of available intercom devices detailed below.
Available Intercom Devices – This dynamic drop down list shows the intercom units that are currently visible on the network and available for selection as a page target. Selecting an intercom from this list places the intercom name into the intercom device edit box.
Page Settings Defaults
Page Priority: Enabled Page Targets: none set
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GPIO Settings Web Page
The GPIO Settings page shows the current configuration of the 10 GPIO ports on the rear panel mounted GPIO connector. The functions, inputs and states that control the output relay are also set here.
Fig 12-19: GPIO Settings Web Page
GPIO Select – This drop down list selects the GPIO port to view and change settings.
Type – This option selects whether the selected GPIO port is an input, an output or whether it is disabled. Ports configured as inputs are active low and they have a weak internal pullup. Ports configured as outputs are open collector driven and are therefore active low.
Input Function – This option selects the required function that the selected GPIO input controls. GPIO inputs can be set to control a local function and a virtual GPIO output if required. If no local function control is needed, but control of a virtual GPIO output is required, select Virtual GPO Only. The assignment for which GPIO input triggers a virtual GPIO output is made on the virtual GPIO settings page. This option is only available when the type is set to input.
Input Action – This option selects the operational mode of the selected GPIO input. In momentary mode, the selected function is active whilst the input is asserted low. In latching mode, a separate active low pulse is required to enable and disable the selected function. This option is only available when the type is set to input.
Output Trigger – This option selects the function that triggers the GPIO output. GPIO outputs can be triggered by a local function, a virtual GPIO input and an Ember+ consumer if required. If a local function trigger is not needed, but a virtual GPIO input or Ember+ consumer trigger is required, select Virtual GPI Only or Ember+ Consumer Only. The assignment for which virtual GPIO input triggers a GPIO output is made on the virtual GPIO settings page. This option is only available when the type is set to output.
Relay Trigger – The output relay, which provides both normally open and normally closed contacts on the GPIO connector on the rear panel, can be triggered by multiple functions, inputs and states. When any of the selected triggers become active, the relay is switched on. To select multiple triggers, click on the required options whilst holding down the Ctrl key on the
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keyboard. The CLEAR ALL button provides an easy way to remove all current relay trigger assignments.
GPIO Settings Defaults
GPIO Port 1 – 10: Type: Disabled Input Function: Talk/Listen 1 Input Action: Latching Output Trigger: Talk/Listen 1
Virtual GPIO Settings Web Page
The Virtual GPIO Settings page shows the current configuration of the 10 virtual GPIO ports. These virtual ports provide a method of controlling functions on remote units by linking a virtual output on one unit to a virtual input on another unit.
Fig 12-20: Virtual GPIO Settings Web Page
vGPIO Select – This drop down list selects the virtual GPIO port to view and change settings.
Type – This option selects whether the selected virtual GPIO port is an input, an output or whether it is disabled.
Input Function – This option selects the required function that the selected virtual GPIO input controls. This list includes GPO 1 – 10 which allows a virtual GPIO input to control a physical GPIO output. Please note, if a GPO port is selected, the type option for that GPIO port must be set to output. This option is only available when the type is set to input.
Input Driver Source – This setting is the name of the virtual GPIO output on a remote unit that will drive the virtual GPIO input on this unit. These source names are generated automatically based on the source type and the intercom name. For example, for virtual output port 1 on an intercom unit with the name “DEMO-SOURCE”, the source name would be “VGPO1@ DEMO-SOURCE”. This name can be entered manually, or it can be selected from the dynamic list of available vGPO sources detailed below. This option is only available when the type is set to input.
Available vGPO Sources – This dynamic drop down list shows the virtual GPIO outputs that are currently visible on the network and available for selection as an input driver source. Selecting a source from this list places the virtual GPIO output name into the input driver source edit box. This option is only available when the type is set to input.
Output Trigger – This option selects the function that triggers the virtual GPIO output. This list includes GPI 1 – 10 which allows a physical GPIO input to trigger a virtual GPIO output. Please note, if a GPI port is selected, the type option for that GPIO port must be set to input otherwise the virtual output will not be triggered. This option is only available when the type is set to output.
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Virtual GPIO Settings Defaults
Virtual GPIO Port 1 – 10: Type: Disabled Input Function: Talk/Listen 1 Input Driver Device: not set Output Trigger: Talk/Listen 1
Miscellaneous Settings Web Page
The Miscellaneous Settings page shows general settings for the connected unit.
Fig 12-21: Miscellaneous Settings Web Page
Power Button – This option enables the power button on the front panel of the unit. When enabled, pressing the power button will switch the unit on and off.
AC PSU Status – This option enables the AC PSU status LED on the front panel.
DC PSU Status – This option enables the DC PSU status LED on the front panel.
Auto Save on Page Exit – With this option enabled, any changes made on a page are automatically saved when leaving the page. This option is initially cleared when connecting to a device.
Miscellaneous Settings Defaults
Power Button: Enabled AC PSU Status: Enabled DC PSU Status: Enabled
System Tab
The System Tab allows the firmware on the unit to be updated. It also has options for saving and loading the configuration settings to and from a local file.
Update Firmware
Fig 12-22: Update Firmware Section of System Web Page
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The current firmware version on the connected unit is shown. New versions of firmware will be released as new features are added, and when any bug fixes are completed. Click on the software downloads link to visit the Sonifex web page. If an update is available, download the latest file. The file will have the ‘.SWU’ extension. Click the select button and browse your computer to locate the new firmware file and then click Update. Once the file has been successfully uploaded to the unit, the firmware update process will begin. This takes approximately 1 minute during which time a progress page will be shown. Once the update is complete, the device information page will be shown automatically.
Save Configuration To A File
Fig 12-23: Save Configuration Section of System Web Page
The current configuration of the connected unit can be saved to a local xml file by clicking the SAVE CONFIG button. The filename will be “avn_tb_” followed by the intercom name. It is a good idea to save the configuration settings of all units on the network once they have been setup as this provides a quick and easy way of returning the units to a known working condition.
Load Configuration From A File
Fig 12-24: Load Configuration Section of System Web Page
The current configuration of the connected unit can be overwritten from a previously saved xml file. The settings on the network page will only be updated if the Overwrite Network Settings check box is ticked. Click the select button and browse your computer to locate the required configuration file and then click the LOAD CONFIG button. Once the file has been successfully uploaded, the unit will automatically reboot to load the new configuration settings. This takes approximately 25 seconds during which time a progress page will be shown. If the settings of the network port the browser is connected to have changed, a new connection will need to be made once the unit has restarted. Otherwise, the device information page will be shown automatically.
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Save System Log To A File
Fig 12-25: Save System Log Section of System Web Page
The AVN-TB6 and AVN-TB6D maintain an internal log that records system events and errors. This information may be useful for resolving configuration issues. To download the log, click the SAVE LOG button.
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13 Ember+ Interface
13. Ember+ Interface
The AVN-TB6 and AVN-TB6D have an Ember+ provider interface that exposes all of the available configuration options. It also allows the GPIO, virtual GPIO and the relay to be controlled and monitored by an external Ember+ consumer.
Connecting to the Ember+ Interface
The Ember+ interface is available on both the Ethernet and AoIP network connections using port 9000. Each of the configuration options are arranged in branches grouped by function. Fig 13-1 shows a graphical representation of the root structure.
Fig 13-1: Ember+ Root Structure
The configuration options are presented as either string or integer types. Where possible, integer type options are enumerated to indicate what each of the setting values are. A full list of the options and values associated with each of the Ember+ parameters are on page 45.
GPIO
The GPIO branch of the Ember+ provider interface allows monitoring and control of the GPIO ports and the relay. Fig 13-2 shows the structure of each GPIO child node For further details on the configuration options, refer to page 37.
Fig 13-2: Ember+ GPIO Branch Structure
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Each GPIO node has a similar set of parameters and functions:
Type Parameter
This enumerated integer parameter is the current port type.
Input Function Parameter
This enumerated integer parameter is the function that will be activated when the GPIO port, when configured as an input, is asserted.
Input Action Parameter
This enumerated integer parameter is the operational mode of the GPIO port when it is configured as an input.
Output Trigger Parameter
This enumerated integer parameter is the function that triggers the GPIO port when it is configured as an output.
Status Parameter
The status parameter shown in Fig 13-2 represents the current state of the GPIO port regardless of whether the port is set as an input or an output. A value of 0 indicates the port is inactive and a 1 indicates the port is active. This parameter has an Ember+ stream identifier and a consumer can subscribe to it to automatically receive updates when the status changes.
Set Output Function
The set output leaf shown in Fig 13-2 is a function that allows control of the GPIO port when it is configured as an output. The function exposes 1 argument called enable, which is an integer with a value of non-zero for assert the output, or zero for de-assert the output. The function also returns an integer result which reflects the current state of the GPIO output similar to the status parameter. A return value of -1 indicates there was an error invoking the function. If the GPIO port is configured as an input, invoking this function has no effect.
Relay
The relay node of the GPIO branch of the Ember+ provider interface allows the relay to be controlled by an Ember+ consumer. Fig 13-3 shows the structure of the relay node. For further details on the configuration options, refer to page 37.
Fig 13-3: Ember+ Relay Node Structure
Trigger Parameter
This integer parameter is the combined value of the functions that trigger the operation of the relay. As multiple trigger functions can be selected, the value shown is the sum of the selected items corresponding values.
Status Parameter
The status parameter shown in Fig 13-3 represents the current state of the relay. A value of 0 indicates the relay is inactive and a 1 indicates the relay is active. This parameter has an Ember+ stream identifier and a consumer can subscribe to it to automatically receive updates when the status changes.
Active Function
The active leaf shown in Fig 13-3 is a function that allows control of the relay. The function exposes 1 argument called enable, which is an integer with a value of non-zero to activate the relay, or zero to de-activate the relay. The function also returns an integer result which reflects the current state of the relay similar to the status parameter. A return value of -1 indicates there was an error invoking the function.
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Virtual GPIO Control
The virtual GPIO branch of the Ember+ provider interface allows monitoring and control of the virtual GPIO ports. Fig 13-4 shows the structure of each virtual GPIO child node. For further details on the configuration options, refer to page 38.
Fig 13-4: Ember+ Virtual GPIO Branch Structure
Type Parameter
This enumerated integer parameter is the current virtual GPIO port type.
Input Function Parameter
This enumerated integer parameter is the function that will be activated when the virtual GPIO port, when configured as an input, is asserted.
Input Driver Name Parameter
This string parameter is the name of the virtual output on a remote unit that drives the virtual GPIO port when it is configured as an input.
Output Trigger Parameter
This enumerated integer parameter is the function that triggers the GPIO port when it is configured as an output.
Status Parameter
The status parameter shown in Fig 13-4 represents the current state of the virtual GPIO port regardless of whether the port is set as an input or an output. A value of 0 indicates the port is inactive and a 1 indicates the port is active. This parameter has an Ember+ stream identifier and a consumer can subscribe to it to automatically receive updates when the status changes.
Set Output Function
The set output leaf shown in Fig 13-4 is a function that allows control of the virtual GPIO port when it is configured as an output. To use this function, the output trigger must be set to ember-consumer, and at least one remote unit must have this output selected as the driver for a virtual input. The function exposes 1 argument called enable, which is an integer with a value of non-zero for assert the output, or zero for de-assert the output. The function also returns an integer result which reflects the current state of the virtual GPIO output similar to the status parameter. A return value of -1 indicates there was an error invoking the function. If the virtual GPIO port is configured as an input, invoking this function has no effect.
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Ember+ Interface 13
Parameter Value Specification
This section provides a full list of the possible parameter values for the Ember+ interface. Integer values with the ‘0x’ prefix are in hexadecimal format.
Identity
This is a read only leaf that describes the unit.
Network
Intercom Name (string): Max length = 23 characters Address Mode (integer): 0 = Dynamic, 1 = Static IP Address (string): Max length = 15 characters Subnet Mask (string): Max length = 15 characters Auto Multicast (integer): 0 = Disable, 1 = Enable
AoIP
Stream Name (string): Max length = 12 characters DSCP (integer): Min = 0, Max = 56 IP Address (string): Max length = 15 characters
PTP
DSCP (integer): Min = 0, Max = 56 Profile (integer): 0 = Default 1 = AES67 Media 2 = Custom Domain Number (integer): Min = 0 Max = 127 (Default Profile) Max = 255 (AES67 Media Profile) Max = 255 (Custom Profile) Priority 1 (integer): Min = 0, Max = 255 Priority 2 (integer): Min = 0, Max = 255 Announce Interval (integer): Setting value is 2
integer
Min = 0 (Default Profile) Min = 0 (AES67 Media Profile) Min = -3 (Custom Profile) Max = 4 Announce Timeout (integer): Min = 2, Max = 10 Sync Interval (integer): Setting value is 2
integer
Min = -1, Max = 1 (Default Profile) Min = -4, Max = 1 (AES67 Media Profile) Min = -7, Max = 6 (Custom Profile) Min Delay Request Interval (integer): Setting value is 2
integer
Min and Max values in following read only parameters Min Delay Request Interval Min (integer): Value is read only Greater of 0 or Sync Interval (Default Profile) Greater of -3 or Sync Interval (AES67 Media Profile) Sync Interval (Custom Profile) Min Delay Request Interval Max (integer): Value is read only:
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Lesser of 5 or Sync Interval+5 (Default Profile) Lesser of 5 or Sync Interval+5 (AES67 Media Profile) Sync Interval+5 (Custom Profile) Min Peer Delay Request Interval (integer): Setting value is 2
integer
Min = 0, Max = 5 (Default Profile) Min = 0, Max = 5 (AES67 Media Profile) Min = -1, Max = 5 (Custom Profile) Slave Only (integer): 0 = Disable, 1 = Enable Delay Mechanism (integer): 0 = E2E, 1 = P2P Reset Defaults (integer): 1 = Reset Profile to Default Values
Display
LED Brightness (integer): Min = 1, Max = 7 Talk Colour, Listen Colour, Talk Listen Colour, User 1 Colour (integer): 0 = Red 1 = Green 2 = Amber 3 = Blue 4 = Purple 5 = Light Blue 6 = White
Mic
Type (integer): 0 = Mic 1 = Phantom Mic 2 = Headset 3 = Phantom Headset Gain Control (integer): 0 = Fixed, 1 = Automatic Mic Gain (integer): Actual Gain is value x 3 Min = 0, Max = 20 Headset Gain (integer): Actual Gain is value x 3 Min = 0, Max = 20 AGC Target Level (integer): 0 = -24dBFS 1 = -21dBFS 2 = -18dBFS 3 = -15dBFS 4 = -12dBFS 5 = -9dBFS 6 = -6dBFS Auto Mute (integer): 0 = Disable, 1 = Enable Mute Button Mode (integer): 0 = Auto 1 = Momentary 2 = Latching 3 = Disabled On
4 = Disabled Off
Channels
AoIP Source (string): Max length = 36 characters IFB Mode (integer): 0 = Disable, 1 = Enable IFB Source (string): Max length = 12 characters Auto Answer (integer): 0 = Disable, 1 = Enable Default Function (integer): 0 = Talk & Listen 1 = Talk Only 2 = Listen Only 3 = Talk Forced Listen Button Mode (integer): 0 = Auto 1 = Momentary 2 = Latching
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3 = Disabled Call Signal (integer): 0 = Disable, 1 = Enable
Group Talk
Group Talk 1 (integer): Value is sum of required options 0x1 = Channel 1 0x2 = Channel 2 0x4 = Channel 3 0x8 = Channel 4 0x10 = Channel 5
0x20 = Channel 6
Page
Priority (integer): 0 = Disable, 1 = Enable Target 1, Target 2, Target 3, Target 4, Target 5, Target 6, Target 7, Target 8, Target 9, Target 10, Target 11, Target 12, Target 13, Target 14, Target 15, Target 16, Target 17, Target 18, Target 19, Target 20 (string): Max length = 23 characters
GPIO
Type (integer): 0 = Input 1 = Output 2 = Disabled Input Function (integer): 0 - 5 = Talk/Listen 1 - 6 6 = Page 7 = User Button 1 8 = Monitor 1 9 = Speaker Dim 10 = Speaker Mute 11 = Microphone Mute 12 = Virtual GPO Only Input Action (integer): 0 = Momentary 1 = Latching Output Trigger (integer): 0 - 5 = Talk/Listen 1 - 6 6 = Any Talk/Listen 7 - 12 = Call Request 1 - 6 13 = Any Call Request 14 = Page 15 = User Button 1 16 = Monitor 1 17 = Speaker Dim 18 = Speaker Mute 19 = Microphone Mute 20 = Virtual GPI Only 21 = Ember+ Consumer Only
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Relay
Trigger (integer): Value is sum of required options 0x1 = Talk/Listen 1 0x2 = Talk/Listen 2 0x4 = Talk/Listen 3 0x8 = Talk/Listen 4 0x10 = Talk/Listen 5 0x20 = Talk/Listen 6 0x40 = Call Request 1 0x80 = Call Request 2 0x100 = Call Request 3 0x200 = Call Request 4 0x400 = Call Request 5 0x800 = Call Request 6 0x1000 = Page 0x2000 = User Button 1 0x4000 = Monitor 1 0x8000 = Speaker Dim 0x10000 = Speaker Mute 0x20000 = Mic Mute 0x40000 = GPI 1 0x80000 = GPI 2 0x100000 = GPI 3 0x200000 = GPI 4 0x400000 = GPI 5 0x800000 = GPI 6 0x1000000 = GPI 7 0x2000000 = GPI 8 0x4000000 = GPI 9 0x8000000 = GPI 10 0x10000000 = Virtual GPI 1 0x20000000 = Virtual GPI 2 0x40000000 = Virtual GPI 3 0x80000000 = Virtual GPI 4 0x100000000 = Virtual GPI 5 0x200000000 = Virtual GPI 6
0x400000000 = Virtual GPI 7 0x800000000 = Virtual GPI 8 0x1000000000 = Virtual GPI 9 0x2000000000 = Virtual GPI 10 0x4000000000 = AC PSU Warn 0x8000000000 = DC PSU Warn 0x10000000000 = Ember+ Consumer
vGPIO
Type (integer): 0 = Input 1 = Output 2 = Disabled Input Function (integer): 0 - 5 = Talk/Listen 1 - 6 6 = User Button 1 7 = Monitor 1 8 = Speaker Dim 9 = Speaker Mute 10 = Microphone Mute 11 - 20 = GPO 1 - 10 Input Driver Name (string): Max length = 30 characters Output Trigger (integer): 0 - 5 = Talk/Listen 1 - 6 6 = Any Talk/Listen 7 - 12 = Call Request 1 - 6 13 = Any Call Request 14 = Page 15 = User Button 1 16 = Monitor 1 17 = Speaker Dim 18 = Speaker Mute 19 = Microphone Mute 20 - 29 = GPI 1 - 10 30 = Ember+ Consumer
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User
Function (integer): 0 = Group Talk 1 = GPO Control Button Mode (integer): 0 = Auto 1 = Momentary 2 = Latching 3 = Disabled
Monitors
AoIP Stream (string): Max length = 36 characters Talkback Mode (integer): 0 = Mute 1 = Dim 75% 2 = Dim 50% Button Mode (integer): 0 = Auto 1 = Momentary 2 = Latching 3 = Disabled
Speaker
Echo Cancel (integer): 0 = Disable, 1 = Enable Mute Button Mode (integer): 0 = Auto 1 = Momentary 2 = Latching 3 = Disabled On 4 = Disabled Off
Power Button
Enable (integer): 0 = Disable, 1 = Enable
AC PSU Status
Enable (integer): 0 = Disable, 1 = Enable
DC PSU Status
Enable (integer): 0 = Disable, 1 = Enable
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14 Technical Specifications
14. Technical Specifications
Audio-Over-IP Specification
Open Standards: RAVENNA, AES67
Device Discovery: Bonjour (mDNS / DNS-SD)
Audio Delivery: RTP/UDP over IPv4 multicast
QoS: DiffServ
Stream Management: RTSP/SDP
Control: Web server/Ember+
Format: Linear PCM 24-bit (L24)
Channels Per Stream: 2
Frames Per Packet: 48
Transmit Streams: 1 (fixed)
Sample Rate: 48 kHz
Ember+ Interface Connection
Interface Type: Provider
Network Interface: Ethernet port and AoIP port
Port: 9000
Timing Synchronisation
Profile Support: Default, AES67 Media & Custom profiles
Timing Protocol: PTPv2, IEEE1588-2008
Microphone and Headset Input
Input Impedance: > 2.5kΩ balanced
Gain Range: 0dB to +60dB
0dBFS Line-Up: Adjustable in steps of 3dB from -58dBu to +2dBu
Frequency Response: 20Hz to 20kHz, +0/-0.2dB
Noise: -127dBu, 20kHz BW, Rs=200Ω ref. 76dB gain
Headphone Output
Output Impedance: Drives 150mW into 32Ω to 600Ω headphones
0dBFS Line-Up: +20dBu
Frequency Response: 20Hz to 20kHz, +0/-0.2dB
THD+N: < -108dBFS, -30dBFS, 20Hz to 20kHz, unity gain, 20kHz BW
Noise: -110dBFS, 20kHz BW
Loudspeaker
Power Output: 4W
Volume: Mute to full volume via front panel control
Connections
Microphone: XLR-3 pin female (electronically balanced)
Headphone: ¼ inch (6.35mm) stereo jack socket
Headset: 1 x XLR-5 pin female (electronically balanced input)
GPIO: 15-way ‘D’-type socket
Network: 2 x gigabit Ethernet, RJ45 1 x SFP
Mains AC Input: Universal filtered IEC, continuously rated 85-264VAC, 47-63Hz, 20W
DC Input: 4 pin 7.5A power jack socket, 10-14VDC
Fuse Rating: Anti-surge fuse 2A 20mm x 5mm
Equipment Type
AVN-TB6: 6 channel freestanding talkback intercom control unit with RAVENNA AoIP
AVN-TB6D: 6 channel desktop talkback intercom control unit with RAVENNA AoIP
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Technical Specifications 14
Physical Specification AVN-TB6
Dimensions (Raw): 25.5cm (W) x 19.4cm (D) x 4.4cm (H) (1U)
10.1” (W) x 7.7” (D) x 1.8” (H) (1U)
Dimensions (Boxed): Weight: 33cm (W) x 26cm (D) x 17cm (H) 13” (W) x 10.3” (D) x 6.7” (H)
Accessories
AVN-DC150: 150W DC power supply with KPJX-4S plug
AVN-TB6RK: 1U rack-kit for AVN-TB6RK
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15 Aditional Information
15. Additional Information
DSCP Name DS Field Value IP Precedence (Description)
CS0 0: Best effort
CS1, AF11-AF13 8, 10, 12, 14 1: Priority
CS2, AF21-AF23 16, 18, 20, 22 2: Immediate
CS3, AF31-AF33 24, 26, 28, 30 3: Flash (mainly used for voice
signalling)
CS4, AF41-AF43 32, 34, 36, 38 4: Flash Override
CS5, EF PHB 40, 46 5: Critical (mainly used for RTP)
VOICE ADMIT 44 5: Critical (uses Call Admission
Control)
CS6 48 6: Internetwork Control
CS7 56 7: Network Control
Table 15-1: DSCP Names & Their Corresponding IP Precedence
Table 15-1 shows the corresponding IP precedence for the DSCP names used in the PTP Profiles and AoIP Stream Settings. For more information on the types of DSCP name, see the following documents:
CS: Class Selector (RFC 2474)
AFxy: Assured Forwarding (x= class, y = Drop Precedence) (RFC 2597)
EF: Expedited Forwarding (RFC 3246)
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Notes
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