AEQ PHOENIX ALIO
Portable IP Audiocodec that is easy to configure and use
Optimized for OPUS encoding algorithms
USER’S MANUAL
ED. 12/17
V. 1.0 - 20/12/2017
Firmware Versions:
Software Version:
CPU 6.80 / DSP 3.35 / FPGA 5.54 or higher
AEQ ControlPHOENIX 2.2.1.20 or higher
CONTENTS
1. INTRODUCTION....................................................................................................................... 5
1.1. General description. ........................................................................................................... 5
1.2. Technical characteristics.................................................................................................... 5
1.3. Available encoding algorithms. .......................................................................................... 6
1.4. Block diagram. ................................................................................................................... 7
1.4.1. Internal diagram. ..................................................................................................... 7
1.4.2. Audio matrix. ........................................................................................................... 8
1.5. Compatibility with other AEQ codecs............................................................................... 10
1.6. Compatibility with third-party codecs. .............................................................................. 10
2. PHYSICAL DESCRIPTION OF THE UNIT............................................................................. 11
2.1. Description of the right panel and connections................................................................ 11
2.1.1. Headphone 1 and 2 outputs.................................................................................. 11
2.1.2. Line inputs ("LINE IN"). ......................................................................................... 11
2.1.3. Line outputs ("LINE OUT").................................................................................... 12
2.2. Description of the back panel and connections. .............................................................. 12
2.2.1. Microphone inputs ("MIC1", "MIC2", "MIC3" and "MIC4"). ................................... 12
2.2.2. Ethernet port. ........................................................................................................ 13
2.2.3. Power supply......................................................................................................... 13
2.2.3.1. Using an external powerbank as an UPS. ............................................. 13
3. USER INTERFACE DESCRIPTION. MANUAL CONTROL. ................................................. 15
3.1. OLED Screen. .................................................................................................................. 16
3.2. Navigation / Channel encoder ("NAVI / Ch"). .................................................................. 17
3.3. Vumeters.......................................................................................................................... 17
3.4. Alphanumeric keyboard and call buttons......................................................................... 18
3.5. Inputs control.................................................................................................................... 19
3.6. Outputs control................................................................................................................. 20
3.7. Function keys. .................................................................................................................. 21
3.7.1. MODE keys ("IP" and "SIP"). ................................................................................ 21
3.7.2. "AUTO" key. .......................................................................................................... 22
3.7.2.1. SmartRTP / AutoAnswer. ........................................................................ 22
3.7.2.2. Auto Hang-Up. ........................................................................................ 22
3.7.2.3. Permanent call. ....................................................................................... 22
3.7.3. "CODEC" key. ....................................................................................................... 22
3.7.4. "HELP" key............................................................................................................ 23
3.7.5. "MENU" key. ......................................................................................................... 24
3.7.5.1. "ETHERNET" submenu. ......................................................................... 24
3.7.5.2. "COMMUNICATIONS" submenu. ........................................................... 25
3.7.5.3. "MAINTENANCE" submenu.................................................................... 26
4. CONFIGURATION AND OPERATION FROM REMOTE CONTROL SOFTWARE.............. 28
4.1. Individual codec control window. ..................................................................................... 28
4.1.1. CONFIG Menu. ..................................................................................................... 29
4.1.2. Mixer control window. ........................................................................................... 32
4.1.3. Vumeters window.................................................................................................. 33
4.2. Connection modes ........................................................................................................... 34
4.2.1. RTP Point to Point (RAW)..................................................................................... 34
4.2.2. PROXY SIP........................................................................................................... 35
4.2.3. DIRECT SIP. ......................................................................................................... 37
4.2.4. Sending audio to multiple destinations: Broadcast, Multicast and Multi-unicast. . 38
4.3. NAT TRAVERSAL. .......................................................................................................... 41
4.3.1. Operation without NAT: "OFF (there is no NAT)". ................................................ 42
4.3.2. Manual NAT: "MANUAL (router configuration)". ................................................... 42
4.3.3. AUTO 1 (local network audio)............................................................................... 43
4.3.4. AUTO 2 (local network audio)............................................................................... 43
4.3.5. AUTO 3 (audio over internet)................................................................................ 43
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AEQ PHOENIX ALIO
4.3.6. AUTO 4 (audio over internet)................................................................................ 44
4.4. FEC modes and reception buffer configuration. .............................................................. 44
4.5. Coding algorithm selection............................................................................................... 45
4.6. Ethernet Port configuration. ............................................................................................. 46
4.7. SNMP configuration. ........................................................................................................ 46
4.8. Password configuration.................................................................................................... 48
4.9. User interface lock. .......................................................................................................... 49
5. QUICK USER’S GUIDE. LOCAL OPERATION. .................................................................... 51
5.1. Connecting the unit. ......................................................................................................... 51
5.1.1. Power supply......................................................................................................... 51
5.1.2. Audio. .................................................................................................................... 51
5.1.3. Communications interface. ................................................................................... 51
5.2. Turning the unit on. .......................................................................................................... 51
5.3. Establishing a communication. ........................................................................................ 51
5.3.1. Establishing an IP communication in RTP mode using SmartRTP. ..................... 52
5.3.1.1. Ending an IP communication in RTP mode. ........................................... 53
5.3.2. Establishing an IP call in PROXY SIP mode. ....................................................... 54
5.3.2.1. Ending an IP call in PROXY SIP mode................................................... 55
5.3.2.2. Receiving and accepting an IP call in PROXY SIP mode....................... 55
5.3.3. Establishing an IP call in DIRECT SIP mode........................................................ 56
5.3.3.1. Ending an IP call in DIRECT SIP mode. ................................................. 57
5.3.3.2. Receiving and accepting an IP call in DIRECT SIP mode...................... 57
6. QUICK START GUIDE. REMOTE CONTROL. ...................................................................... 59
6.1. Equipment connections.................................................................................................... 59
6.1.1. Power supply......................................................................................................... 59
6.1.2. Communications interface. ................................................................................... 59
6.2. Turning the unit on. .......................................................................................................... 59
6.3. Setting up a computer to control the unit. ........................................................................ 59
6.4. Audio. ............................................................................................................................... 60
6.5. Establishing an IP communication. .................................................................................. 60
6.5.1. Establishing an IP communication in RTP mode using SmartRTP. ..................... 60
6.5.1.1. Ending an IP communication in RTP mode. ........................................... 62
6.5.2. Establishing an IP call in PROXY SIP mode. ....................................................... 62
6.5.2.1. Ending an IP call in PROXY SIP mode................................................... 65
6.5.2.2. Receiving and accepting an IP call in PROXY SIP mode....................... 65
6.5.3. Establishing an IP call in DIRECT SIP mode........................................................ 66
6.5.3.1. Ending an IP call in DIRECT SIP mode. ................................................. 68
6.5.3.2. Receiving and accepting an IP call in DIRECT SIP mode...................... 68
7. CONTROL TERMINAL OVER WEB BROWSER. ................................................................. 70
7.1. Upgrading system firmware. ............................................................................................ 70
7.2. Configuring the MAC address associated with the Ethernet interface. ........................... 71
7.3. Technical Assistance Service and on-line manuals......................................................... 72
7.4. Saving and loading configurations. .................................................................................. 72
7.5. Screen capture................................................................................................................. 73
7.6. Status menu. .................................................................................................................... 73
7.7. SNMP............................................................................................................................... 73
7.8. Remotely rebooting the equipment. ................................................................................. 74
8. TECHNICAL SPECIFICATIONS ............................................................................................ 75
9. A.E.Q. WARRANTY................................................................................................................ 76
APPENDIX A: GENERAL CHARACTERISTICS OF ENCODING MODES. ............................. 77
APPENDIX B: PROTOCOLS ASSOCIATED WITH IP COMMUNICATIONS. .......................... 79
B1. Circuit switching versus packet switching. ....................................................................... 79
B1.1. Circuit switching..................................................................................................... 79
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AEQ PHOENIX ALIO
B1.1.1. Advantages:............................................................................................. 79
B1.1.2. Disadvantages: ........................................................................................ 80
B1.2. Packet switching. ................................................................................................... 80
B1.2.1. Switching modes...................................................................................... 80
B1.2.2. Advantages:............................................................................................. 80
B1.2.3. Disadvantages: ........................................................................................ 81
B2. IP protocol......................................................................................................................... 81
B2.1. IP addressing......................................................................................................... 82
B2.2. Unicast vs. Multicast. ............................................................................................. 82
B3. RTP protocol..................................................................................................................... 83
B3.1. Default PHOENIX ALIO configuration. .................................................................. 84
B4. SIP protocol. ..................................................................................................................... 84
B4.1. Working modes...................................................................................................... 84
B4.2. Possible work scenarios. ....................................................................................... 86
B4.3. PHOENIX ALIO default SIP configuration. ............................................................ 86
B5. STUN protocol. ................................................................................................................. 87
APPENDIX C: PORTS USED BY PHOENIX EQUIPMENT....................................................... 89
APPENDIX D: APPLICATION NOTES GUIDE.......................................................................... 90
D1. Internet connection using standard cable access. ........................................................... 90
Application note AN0A. ................................................................................................... 90
Application note AN0B. ................................................................................................... 90
Application note AN0C. ................................................................................................... 90
Application note AN0D. ................................................................................................... 90
Application note AN0E. ................................................................................................... 90
Application note AN0F. ................................................................................................... 90
Application note AN0G.................................................................................................... 90
Application note AN0H. ................................................................................................... 90
D2. Special applications using different kinds of Internet physical accesses, or point to point
connections. ..................................................................................................................... 90
Application note AN1....................................................................................................... 90
Application note AN2....................................................................................................... 90
Application note AN3....................................................................................................... 90
Application note AN4....................................................................................................... 90
Application note AN4B. ................................................................................................... 90
Application note AN5....................................................................................................... 90
Application note AN6....................................................................................................... 90
APPENDIX E: ADDITIONAL INFORMATION............................................................................ 91
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AEQ PHOENIX ALIO
1. INTRODUCTION.
1.1. General description.
AEQ PHOENIX ALIO is a stereo IP audiocodec for mobile applications. It is easy to configure
and operate, and integrates a digital mixer with 4 analog inputs.
It features independent bass and treble control for each input, adapting the characteristics of
each speaker’s voice, or correcting the defects of external signals.
It has been specifically designed for sports reporting applications, but it has also been optimized
to make it easy to use in the most varied scenarios, even musical events.
Its IP connectivity allows the user to choose among a wide variety of connection modes:
dedicated networks, DSL, cable – modem, fiber optic, WiFi, Wig-MAX, Inmarsat satellite, etc.
The application notes listed at the end of this manual provide information about how to operate
the unit in each case. AN-5B stands out among them, describing a simple and effective way to
connect the audiocodec to 3G and 4G networks, as well as to WiFi hotspots through a router
and modem that can be provided by AEQ or obtained locally.
PHOENIX ALIO is optimized for OPUS encoding algorithms, but it is also compatible with other
AEQ and third-party audiocodecs, as it also features AEQ LD Extend modes and the mandatory
algorithms according to EBU-TECH 3326 specification from EBU N/ACIP work group.
When connecting to another AEQ codec, users can take advantage of an exclusive set of tools
that makes initating communications and control of the unit a simple task:
•
The "SmartRTP" proprietary call-initiation protocol that simplifies connection to compatible
codecs.
•
"AEQ ControlPHOENIX", remote control Software that allows for the remote operation and
adjustment of the unit from your station. ControlPhoenix allows you to control everything
related to the call initiation process and also the adjustment of all audio parameters and the
local audio routing of ALIO.
•
"HELP" function, that allows the journalist to use the system to request for assistance from
the station when facing an unexpected situation.
By default, PHOENIX ALIO offers a stereo or mono channel for the program signal with its
corresponding return. A second bidirectional, mono or stereo channel for coordination or
redundancy purposes can be activated by purchasing its license (that activation can be
accomplished from "AEQ ControlPHOENIX" application and is detailed in section 6.4 of the
application user’s manual).
PHOENIX ALIO is powered from mains. Optionally, it can be equipped with an UPS that is
mains-charged and can provide more than 120 minutes of autonomy, and can also feed a
3G/4G router – modem set and even charge mobile devices.
There is a quick start guide available in chapter 5 of this document. However, it is
strongly recommended to carefully read this manual and the "AEQ ControlPHOENIX"
user’s guide before using the unit.
1.2. Technical characteristics.
-
4 female XLR-3 microphone inputs. Low noise preamplifier and switchable Phantom
power supply. 2 KΩ input impedance.
2 female XLR-3 line inputs, with 9 KΩ impedance, 0dBu, nominal level, max. 20 dBu.
2 male XLR-3 outputs. Output impedance: < 100 Ω, nominal level, 0dBu, max. 20 dBu.
2 - ¼" Jack stereo headphone outputs, with volume control and TX/RX balance
adjustment from front panel.
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AEQ PHOENIX ALIO
Communications interface:
-
IP Ethernet port interface, RJ45 connector.
nd
Two independent links can be established (when an optional 2 channel is purchased)
using the same interface.
Satellite: an external satellite interface can be connected to the IP interface.
3G / 4G telephony: a 3G or 4G modem can be connected to the IP interface.
Wireless data links: a "wireless bridge", WiMax or WiFi antenna can be connected to
the IP interface.
Other features:
-
Front user panel with keyboard and rotary encoders.
OLED graphic display.
2 stereo LED VU-meters.
Operating temperature range: -10 to +45 º C (14 to 114 º F).
Dimensions: 242 x 210 x 60 mm (9.5 x 8.3 x 2.4 inches).
Power supply: 15V DC (9 to 18V DC).
Power consumption: 12W max.
External adapter: universal 90-263V input.
Optional UPS with 12V output for the audiocodec and two USB ports for router supply
and/or mobile devices charging.
AEQ offers the "SmartRTP" call initiation protocol in order to greatly simplify the
operation of the audiocodec.
AEQ also offers two SIP servers free of charge as a standard service for the users of
Phoenix ALIO. One of them is configured as main and the other is provided as a backup.
More information can be found in Appendix B.
1.3. Available encoding algorithms.
•
OPUS with Fs= 48 KHz, mono, stereo, with 4 mono and 3 stereo presets. Bit rates
between 12 and 192 Kbps, very low delay and audio bandwidth between 6 and 20 KHz.
•
G711 A-law, u-law (64 Kbps, low delay, 3.5 KHz audio bandwidth).
•
G722 (64 Kbps, low delay, 7 KHz audio bandwidth).
•
AEQ-LD with Fs=16, 32 or 48 KHz, mono or stereo. Available bit rates between 64 and
384 Kbps, audio bandwidth between 7 and 19 KHz.
•
MPEG 1 and 2 - LII, with Fs between 16 and 48 KHz, mono, stereo, dual channel and
Joint stereo. Binary bit rate between 64 and 384 Kbps. Audio bandwidth between 10.5
and 16.5 KHz.
•
PCM (linear): very low delay, transparent quality. Fs=48 or 32 KHz with 12, 16, 20 or 24
bits/sample, mono or stereo (bit rates between 576 and 2304 Kbps), audio bandwidth
between 16 and 20 KHz.
Additional encoding modes can be considered according to each customer’s specific needs,
such as:
• AAC-LC*:
o Mode: Mono / stereo / MS stereo. Sampling frequency: 24 / 32 / 48 KHz
o Bitrate: 32 / 64 / 96 / 128 / 192 / 256 Kbps
• AAC-LD*:
o Mode: Mono / stereo / MS stereo. Sampling frequency: 48 KHz
o Bitrate: 32 / 64 / 96 / 128 / 192 / 256 Kbps
* Operation with two AAC simultaneously not supported.
Please contact AEQ Sales Department or authorized dealers for more information.
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AEQ PHOENIX ALIO
1.4. Block diagram.
1.4.1. Internal diagram.
The unit’s internal design is organized in several functional modules that make Phoenix ALIO
audiocodec a complete IP communications platform. The unit is provided with a professional
quality analog audio input and output system and a versatile audio matrix with processing
capabilities. A simplified view of the distribution of these modules into the equipment’s
motherboard is presented below, as well as a brief description of each module’s functionality:
Internal Phoenix ALIO layout detail
-
-
-
The audio part (ANALOG I/O) integrates 4 microphone preamplifiers featuring
programmable digital gain as well as the line level balanced inputs and outputs together
with two stereo headphone outputs able to drive both high and low impedance sets.
The power supply section (POWER) converts the DC input (12V/1A) to the different DC
voltages required inside the unit.
The FRONT PANEL is connected to the CPU, and consists in an OLED display where
the control and configuration menus are displayed, as well as a set of keys and
indicators associated to the operation of the unit such as the alphanumeric keyboard,
rotary encoders and four high-resolution LED vumeters.
The CPU is a high-performance and low-power ARM processor in charge of several
tasks such as the user interface, configuration of the other programmable elements
(DSP, FPGA, audio processor, preamplifiers, etc.) and the management of IP
communications, etc.
The DSP (FIXED POINT DSP) is a high-performance fixed-point processor that carries
out the encoding and decoding of up to two stereo channels using different
compression algorithms and as later described in this manual.
The audio matrix (AUDIO MATRIX & MIXER) is implemented using a new-generation
low power FPGA, with 6 inputs + 2 stereo receiving buses / 6 outputs + 2 stereo
transmitting buses. The FPGA can perform any crosspoints combination with great
dynamic range and is controlled by the CPU. It also relies on a specialized coprocessor (AUDIO PROCESSOR) that provides individual low and high frequency
adjustment for each input.
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AEQ PHOENIX ALIO
-
The network interface (NET I/F) is an Ethernet 10/100 Mbps interface that allows for
both audio transmission / reception and remote control of the unit through a single port.
The functionality of this port can also be for the connection of standard equipment to
access Wi-Fi, 3G/4G and satellite networks as described in specific application notes
available on AEQ’s website aeq.eu or aeqbroadcast.com.
1.4.2. Audio matrix.
The different audio sources are first converted to digital format (24 bits / 48 KHz sampling rate)
to be processed by a digital audio matrix. Once the outputs are obtained, they are converted
back to analog format:
Audio input/output structure
The following figure shows the internal buses for transmission, reception and CUE of both
communications channels that are calculated in the audio matrix, so the user can better
understand what can be done with the different audio sources and what can be listened at the
different outputs.
As can be seen, each input (4 microphones or 3 microphones + LINE IN) can be routed to any
of the transmit buses (program or coordination) or to CUE pre-listen bus. At the same time, we
can select what to listen at any of the outputs (LINE OUT + 2 stereo headphones), Program or
Coordination buses (choosing the mix level between send and return directions) or the local
CUE pre-listen bus.
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AEQ PHOENIX ALIO
Audio matrix structure
The assignment of mono and stereo signals is performed automatically (for example, if MIC 2 is
routed to the transmit bus of a channel where a stereo coding algorithm has been selected, the
unit will make a crosspoint to both L and R. However, if the line input is sent to a mono channel,
a crosspont creating a L + R sum (attenuated by 6dB) will be accomplished. Just in the same
way, if we want to listen to a mono-encoded channel in the headphones, the signal will be
routed to both sides.
The send buses are accessed through back-lit keys and their colour also denotes to what
SEND-bus the corresponding audio input routed (RED=Program, GREEN=Coordination,
off=CUE). The same colour convention is used for the outputs. The user can select what bus to
monitor by simply pressing on the corresponding rotary encoder. The control of the listening
level is done with the associated rotary encoder and is also visible through level bars on the
OLED display. TX / RX mixing level is controlled with potentiometers associated to each output.
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AEQ PHOENIX ALIO
1.5. Compatibility with other AEQ codecs.
PHOENIX ALIO offers the possibility to connect to other AEQ codecs.
It is compatible with Phoenix MERCURY, STUDIO, VENUS, VENUS V2, VENUS V3,
STRATOS and MOBILE.
The "SmartRTP" mode and OPUS encoding algorithm can be used with Phoenix MERCURY,
STUDIO, VENUS, VENUS V2, VENUS V3 and STRATOS updated to firmware version 5.20 or
above.
1.6. Compatibility with third-party codecs.
PHOENIX ALIO is a portable IP audiocodec compatible with EBU-TECH 3326 technical
specification from EBU N/ACIP workgroup. This technical specification was developed to
guarantee compatibility between equipment from different manufacturers in applications for
professional-quality audio contribution over IP networks. Therefore it is possible to connect
PHOENIX ALIO with any codec from other manufacturer over IP provided that this unit has
been developed according to N/ACIP (please check third-party codecs technical specifications).
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AEQ PHOENIX ALIO
2. PHYSICAL DESCRIPTION OF THE UNIT.
Before anything else, it is necessary to become familiar with the connectors and other elements
present in the back, right and front panels of the unit in order to understand the wiring and
installation required for the PHOENIX ALIO.
2.1. Description of the right panel and connections.
A
B
2.1.1. Headphone 1 and 2 outputs ("HP1" y "HP2").
C
A
¼" Headphone Jack. Unbalanced connection.
¼" Jack pinout
2.1.2. Line inputs ("LINE IN").
B
XLR-3 female connector. Balanced connection.
(Connectors as seen from the soldered
side)
L input - Female
R input - Female
XLR-3 pinout
Pin 1
Pin 2
Pin 3
Ground
+ Input
- Input
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AEQ PHOENIX ALIO
2.1.3. Line outputs ("LINE OUT").
C
XLR-3 Male connector. Balanced connection.
(Connector as seen by the soldered
side)
L output - Male
R ouput - Male
XLR-3 pinout
Pin 1
Pin 2
Pin 3
Ground
+ Output
– Output
2.2. Description of the back panel and connections.
C
B
A
2.2.1. Microphone inputs ("MIC1", "MIC2", "MIC3" and "MIC4").
A
XLR-3 female connector. Balanced connection.
(Connector as seen by the soldered side)
Input - Female
Male plug - cable
XLR-3 pinout
Pin 1
Pin 2
Pin 3
Ground
+ Input
- Input
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AEQ PHOENIX ALIO
All microphone inputs (MIC1, MIC2, MIC3 and MIC4) feature low-noise preamplifiers and are
able to provide Phantom supply (+12 V DC / 10 mA). These can be enabled from each input’s
menu (see section 3.5), in order to offer compatibility with both dynamic or condenser
microphones. The range of the preamplifier gain is wide range (0 to 65dB) making it suitable for
a large range of microphone models available on the market.
2.2.2. Ethernet port.
B
PHOENIX ALIO features one Ethernet port. Using this port, the unit can be connected to a LAN
or WAN network and send / receive audio over IP. This port is also used to configure and
administrate the unit from one or more computers using the remote control software.
Please refer to "AEQ ControlPHOENIX" application manual.
The connector is a standard RJ45 10/100 BT, type, with the following pin assignment:
RJ45 connector pinout
2.2.3. Power supply.
C
PHOENIX ALIO can be powered by an external, specifically designed 15V DC (or 12V DC)
power supply. The unit can be connected to the provided charger either directly or using an
optional UPS in cascade. It cannot be connected directly to a vehicle battery without connecting
the mentioned optional UPS or an equivalent voltage stabilizer.
The power cable termination is fitted with a special connector featuring a locking mechanism to
prevent accidental disconnections.
2.2.3.1. Using an external powerbank as an UPS.
An external, small and portable battery has been
homologated as an optional accessory to Phoenix
ALIO. It can operate as an UPS and also provide a
certain degree of portability to the unit, as it can
provide supply for full operation during about 4 hours
when fully charged.
The recommended model is XT-16000QC2 from XT
Power.
In order to use this powerbank as an UPS, a specifically designed adapter must be used. This
adapter can be purchased from AEQ and allows for charging of the battery at the same time as
the PHOENIX ALIO gets power using the same AC/DC power adapter supplied with the unit
(15V DC). In case that a mains cut happens, no operation interruption will be produced as long
as the battery has enough charge.
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AEQ PHOENIX ALIO
Configure the powerbank's output voltage to 12V by means of its lateral key and check this
configuration in its display, where also the percentage of charge and the current charge status
are displayed. If it is deeply discharged, please provide charge for some minutes without ALIO
connected until at least a 10% percertage of charge is displayed.
This device also features additional USB ports. One of them remains free even when using
AEQ UPS adapter, and can be used to charge phones (producing a logical reduction in the
duration of the battery) or to power an external 4G modem-router. Please check application
note AN-5 for more details.
NOTE: If this powerbank is purchased locally, please contact AEQ to obtain the special UPS
adapter for ALIO.
IMPORTANT NOTE: There is another homologated powerbank model (MP-10000, also from
XT Power) with a different adapter associated and that requires to use the 12V DC power
supply of PHOENIX ALIO. Please make sure that you're using the right combination of
powerbank, adapter and power supply. In case of doubt, please consult the Technical
Assistance Service (sat@aeq.es).
CAUTION: Due to risk of fire or explosion, avoid exposure of the powerbank to
shocks, temperatures above 45ºC, liquid pouring, etc.
The unit should be opened ONLY by qualified personnel.
(Please read the manufacturer’s recommendations for more details)
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AEQ PHOENIX ALIO
3. USER INTERFACE DESCRIPTION. MANUAL CONTROL.
Configuration and operation of the Phoenix ALIO unit can be done either using the equipment
front panel controls featuring an OLED screen and associated controls and indicators, or
remotely using the "AEQ ControlPHOENIX" application (Control and Configuration software
for AEQ Phoenix STRATOS, STUDIO, MERCURY, VENUS, VENUS V2, VENUS V3 and ALIO
audiocodecs). This chapter describes the first mode.
Functional areas or sections of the user interface are described next:
CONTROL SURFACE
A
The control surface has been designed bearing in
mind that, quite often, it is not possible to send a
qualified technician to accomplish an outside
broadcast. It can be totally controlled from the station,
and also locally operated by choosing among several
simple options, that are further simplified when
connected to another AEQ Phoenix audiocodec. A
more detailed configuration can be carried out by
setting up the parameters under the MENU key.
OLED display
C
B
Vumeters
Navigation /
Channel
encoder
D
Alphanumeric keyboard
and call buttons
E
Input controls: Mic and line
level adjustments with
"ON" channel buttons
F
Output controls: listening
level for headphones 1 and
2 and line output with TX /
RX balance control
6 FUNCTION KEYS:
o IP: RTP / Smart RTP modes
o SIP: N/ACIP compatible mode
o AUTO: SmartRTP, auto answer, etc.
o CODEC: codecs list
o HELP: remote support
o MENU: advanced options
G
USER INTERFACE LOCK: From CPU version 6.80 and alter, it is possible to lock certain
functions of the user panel from the control software. These functions are disabled in groups
and, this way, improper operation of the unit by users with little experience can be avoided. In
order to get more details about which functions can be locked, check section 4.9 of this manual.
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AEQ PHOENIX ALIO
3.1. OLED Screen.
A
High-contrast OLED screen with wide viewing angle. 256 x 64 pixels resolution in gray scale.
When Phoenix ALIO is powered-up, after a delay of approx. 15 seconds (that doesn’t indicate
any malfunction in the unit), two welcome screens will appear sequentially; the first one shows
AEQ logo and the second one displays the audiocodec model:
Detail of welcome screens
After a while, the display changes to show the MAIN STATUS screen, where the configuration
of the different inputs and outputs is presented:
Detail of the MAIN STATUS screen
This screen is divided in 7 columns.
From left to right, the first 4 correspond to the 4 active inputs.
The first three ones are always MIC1, MIC2 and MIC3. The fourth can correspond to MIC4 or
LINE IN. This can be selected through the MIC4/LINE menu or through the "#" shortcut key in
the numeric keyboard (the label under the level bar will change accordingly between "MIC4" and
"LINE"). Finally, the three last columns correspond to the outputs (HP1, HP2 and OUT, as
indicated under each level bar).
Each bar represents a relative mix level or output volume.
Each MIC input can also display a "PH" legend above the level bar, indicating that the input has
its corresponding 12V phantom power supply activated.
Also, the input names can appear highlighted as MIC1 in the above example screen. This
indicates that a process (equalization) is applied to the input signal.
Above each output level bar there is a label showing "PGM", "COOR" or "CUE". These are
indicating what program bus is being monitored on each of the outputs.
This display can also display the different operation and configuration menus that are accessed
and browsed through the navigation ("NAVI / Ch") rotary encoder and the rest of keys and
encoders. Details will be provided later on in this chapter. The display also provides detailed
information about active calls, if any. In case that any function of the user interface is locked and
you try to access to this function, the display will show the "LOCKED - ACTION DISABLED"
indication for a few seconds (see section 4.9 of this manual).
To save power and if the unit is "idle", the intensity of the display is dimmed after a while;
normal brightness will be immediately recovered when any control is touched.
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AEQ PHOENIX ALIO
3.2. Navigation / Channel encoder ("NAVI / Ch").
B
This rotary encoder allows the user to browse through the different menus of
the OLED multifunction screen. Turning it changes the selection among the
options presented, moving the highlighting up and down. When the encoder
is turned clockwise, the option selected is moved down, and turning it anticlockwise, the option selected is moved up.
Pressing its button is equivalent to ENTER, validating the highlighted
selection at that moment.
This rotary encoder has a second use. If the the second, optional channel (coordination COOR)
is activated and no menu is being presented on the display, pressing the encoder will make the
user interface change between PGM and COORD.
Tha modification of any of the parameters pertaining to the selected channel does not affect the
selected operation of the other channel; if we are modifying the configuration for program
(PROG), the status for coordination (COORD) will remain the same, with its communication,
operating modes, etc. unaltered.
The active channel is clearly indicated by the "PROG" and "COORD" labels located between
the vumeters. When a call or communication is established in the channel that is not active, the
label will blink while the other remains steadily lit.
3.3. Vumeters.
C
The unit is equipped with two stereo vumeters. From
above, the first one is indicating the level of the TX
signal. The second indicates the level of reception,
RX. If the unit has the second optional coordination
channel activated the vu meters will follow the "NAVI /
Ch" encoder and will display the levels corresponding
to the active channel and with the labels "PROG" and
"COORD" between the vumeters and as explained
above.
Each of the 4 vumeters consists in 2 bars of 20 LEDs each. From right to left, the three first
LEDs are red followed by four orange coloured ones. The remainders of the LEDs are green.
The scale applied has higher resolution around the transition between green and yellow colors.
Please note that the presence of audio is denoted at -36 dBV when the first green LED is
illuminated. Transition from green to yellow is at 0 dBV, from yellow to red is at +7 dBV and,
when all 3 red LEDs are on, displayed level corresponds to +13 dBV or higher.
17
AEQ PHOENIX ALIO
3.4. Alphanumeric keyboard and call buttons.
D
Standard numeric keyboard with 12 keys that,
among many other functions, allows the user to
dial IP addresses and port numbers in when
calling in RTP mode, or to type letters and
symbols when in SIP (Session Initiation
Protocol) mode.
It can also be used to type text: just press each
key repeatedly to switch between the different
letters available for the same key, just like you
would do to type an SMS on a mobile phone.
Depending on the status, the "OK" key (with a green telephone) allows the user to initiate a call
or accept an option within a menu. On the other hand, the "ESC / DEL" key (with a red
telephone) can hang up a communication, delete or go back in a menu (it will also light in red for
around 15 seconds when the unit is turned on, until the welcome screens appear).
When the "OK" key is pressed from the idle screen to make a call, the alphanumeric keyboard
gets illuminated in red when the call is to be made in PROG, or in green when the current
channel is COORD.
Dialing in PROG
Dialing in COORD
The keys are illuminated laterally; this way it is possible to determine whether the channel is
PROG or COORD even by color-blind people
NOTE: The * key shifts between capitals, lower case and numbers. When you are typing, keys
1 and # allows you to enter special characters (@ and "." among them) and key 0 can generate
number "0" or spaces.
The # key is also a shortcut to switch between MIC4 and LINE (provided that we are not typing
into a text menu).
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AEQ PHOENIX ALIO
3.5. Inputs control.
E
Each input has an associated rotary encoder with
pushbutton and an activation/routing key (ON). If the
rotary encoder is turn, the mix gain to the selected bus
is altered for that input, between mute (-infinity) and
+18 dB. The level bar at the display shows an
indication of this level.
The ON buttons allows the user to select which bus the
associated input will be sent to. When it is off, pressing
it will turn the button RED-illuminated, indicating that
the input is sent to PROGRAM transmission channel.
One more pressing and the key will be illuminated in
GREEN, indicating that the channel is sent to COORDINATION transmission. If it is pressed
once again, it will be off again, and the input will only be available locally in the CUE pre-listen
bus.
The selection is cyclic for each input:
PROG
CUE
COOR
When several inputs have their ON button illuminated in the same color, they will be mixed
(each source according to the selected mix level) to the selected bus (PROG, COORD or CUE).
When the encoder button is pressed, access to the associated input’s configuration menu is
provided. Here the user can configure the corresponding gain, equalization (bass/treble),
activation of the 12V Phantom supply and in the case of MIC4 / LINE input, the user will also be
able to select which input is used (microphone 4 or the line input at the right panel).
WARNING: DO NOT turn Phantom power on or off for a microphone unless the Microphone
Gain is set to minimum. Depending on the microphone model and its specifications, the
negative of observing this procedure may result in very high level and high pitched noise that
could be routed to the units communication buses and the headphone outputs.
The corresponding input encoder can be used to navigate within the menu, or alternatively, the
"NAVI / Ch" rotary encoder can also be used. The "BACK" option in each menu or the "ESC /
DEL" key can be used to cancel and go back.
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AEQ PHOENIX ALIO
The "EQ" indication in the main screen (input name highlighted) will become active whenever a
value different to 0 dB is selected for either the bass or treble controls, as previously explained.
The "MIC4/LINE" input menu is slightly different, as there is an additional option to select
between MIC4 and LINE IN:
The fourth input is a line input in this example. Note that the Phantom option becomes
unavailable until the input is switched to MIC4.
NOTE: When browsing an input menu, the user can quickly change to configure another input
by simply pressing the corresponding encoder button. From now on, the modifications made
correspond to the newly selected input.
3.6. Outputs control.
F
The unit has 3 stereo outputs: 2 for headphones, able to drive both high and low impedance
models with ¼" TRS or jack connectors and 1 balanced line output with 2 male XLR connectors.
All of them are located in the right panel of the unit.
Each output has a labeled control section ("HP1", "HP2" and "LINE OUT"), located at the
bottom right area of the front panel.
TX / RX level balance potentiometer: Allows the user to
continuously adjust which ratio of the corresponding output
comes from the channel local transmission (TX, send) and
what from the same channel reception (reception, RX).
When in the central position, equal levels of RX and TX
will be listened to, both with 6dB attenuation.
Output level encoder: Adjusts the listening level of the
corresponding output. It will be displayed on the OLED
with level bars.
Pressing the output level encoder button cyclically selects
the source bus for that output bus, between CUE
(SOURCE LED off), PROG (SOURCE LED illuminated
red) or COORD (SOURCE LED illuminated green). The
display will show a label above the corresponding level
bar:
PROG
CUE
COOR
In the main screen, the display will show the level and source selected for each
output.
Operation is the same for the potentiometers, encoders, SOURCE LEDS and main display for
all HP1, HP2 and LINE OUT outputs.
NOTE: Adjust the volume with caution; excessive listening levels can damage your hearing.
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AEQ PHOENIX ALIO
3.7. Function keys.
G
The units operational keys are completed by 6 illuminated function
keys located at the right area of ALIO front panel. Some of these
keys ("IP", "SIP", "AUTO", "CODEC") configure parameters or
operating modes for the currently active channel only. "HELP" and
"MENU" keys are associated to functions that are common for the
unit.
The detailed description of the function for each key is described
below.
3.7.1. MODE keys ("IP" and "SIP").
These keys are labeled as "IP" and "SIP" and their operation are exclusive (only one of these
can be enabled at any given time). These keys affect the currently selected channel only
(PROG or COORD and provided that the latter is available - licensed). They allow the user to
select how Phoenix ALIO communicates with other equipment.
IP mode: RTP / SmartRTP
When the "IP" key is pressed, the unit will establish all communications in RTP mode, without
using SIP. In order to establish a communication in this mode, the destination IP address must
be known as well as the receiving audio port.
The standard operation, compatible with other manufacturers, requires that both sides of the
communication initiate the call (call the IP and the active audio port of the remote equipment)
and that both have selected the same encoding algorithm for the communication.
However, thanks to the "SmartRTP" functionality from AEQ, the task of making an RTP call is
greatly simplified. When operating Phoenix family of AudioCodecs with this mode activated, it is
only required that the "caller" launches the call. The remote equipment will automatically answer
and send its audio stream to the callers IP and port. Further, the AudioCodec will also detect
and automatically select the encoding algorithm that the calling unit is using to initiate the
communication. The call does not need to be manually accepted and the hang-up event (from
any end) will also be signaled to either unit.
NOTE: The illumination of the key will flash whenever the associated channel is unavailable
because the Ethernet link is down. Once the flashing of the key stops, communications can be
initiated.
Please refer to chapters 3.7.2, 4.2.1 and 5.3.1 for more details.
SIP mode: N/ACIP compatible
SIP (Session Initiation Protocol) is an alternative to RTP for making a call. The SIP key
configures the unit so all communications made with the currently active channels use this
signaling protocol, compatible with other codecs following N/ACIP standard from EBU. It can be
used directly between audiocodecs (DIRECT SIP) or taking advantage of an external Proxy
(PROXY SIP). Its function is to maintain a database of all registered codecs with their IP
addresses and listening ports making it possible to establish a connection between audiocodecs
located in different networks. This allows the user to "forget" about the IP address and port of
the remote unit.
SIP mode will alternatively change between DIRECT SIP and PROXY SIP when pressing the
"SIP" key repeatedly. Depending on the selected mode, the communications menu options
(MENU COMMUNICATIONS SIP) may vary slightly so the user only needs to configure the
required options.
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AEQ PHOENIX ALIO
NOTE: Calls cannot be made when the illumination of the key is flashing. This is indicating that
there are either issues with the physical link or the registration process with a SIP server (when
in PROXY SIP mode). Communications can be established once the key stops flashing.
Please read chapters 4.2.2, 4.2.3, 5.3.2 and 5.3.3 for more details.
3.7.2. "AUTO" key.
This key opens a menu where several automation options are available that simplifies the
operation of the unit and automatic call management. The adjustments made and displayed
apply to the currently active channel only.
The "AUTO" button will be illuminated whenever at least one of the options is activated.
3.7.2.1. SmartRTP / AutoAnswer.
First, we can select whether to activate or not the "SmartRTP" function (available only in RTP
mode), that allows, as explained before, to signal call establishment and hang up from a single
end of the call without using higher level negotiation protocols such as SIP.
When the operating mode for the currently selected channel has been set to SIP, this option is
substituted by "Autoanswer", which makes the unit automatically accept the incoming calls. On
the contrary and if in manual answer mode, incoming calls are signaled on the screen and the
user must press the "OK" key to accept them or the "ESC / DEL" key to reject them.
3.7.2.2. Auto Hang-Up.
Next, the "Auto Hang-Up" option is available in the "AUTO" menu. This option makes the unit
hang up a call whenever the incoming stream can’t be synchronized (either because it is not
received, in this case the call status will be "CONNECTED - NO DATA", or because some
incoming packets are damaged or lost, or because the encoding algorithm is not the one
expected. In this case, the call status will be "CONNECTED - NO SYNC"). In both cases, the
call will be disconnected or hung up once the set out time for synchronization has been
completed – timeout.
This is useful and avoids leaving a unit busy indefinitely or when it is needed to force re-dialling
from the other end (in SIP mode).
3.7.2.3. Permanent call.
This option configures the unit so that it automatically does what is necessary to maintain the
active connection even after temporary line drops or power outages. It is recommended that this
option is activated in one of the communicating audiocodecs only, except when using RTP with
"SmartRTP" mode deactivated. In this case, this option must be enabled at both sides to
guarantee the operational efficiency.
3.7.3. "CODEC" key.
This key allows the selection of:
•
•
A encoding algorithm, among those supported by the unit, for RTP calls
A encoding profile (list of prioritized preferred modes, used when negotiating calls with
other units), for SIP calls.
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AEQ PHOENIX ALIO
The same way as the other function keys (except "HELP" and "MENU"), the selection is
accomplished for the currently active channel: PROG or COORD (the latter only if license is
activated).
As can be noticed in the above sequence of screen captures, when in RTP mode the encoding
algorithms are presented grouped by families (OPUS, G.722, AEQ-LD, PCM, G.711, etc), and
several particular modes are available within each family (A/u law, mono, stereo, different
bitrates, sampling frequencies, etc). When a particular mode is selected, successive calls will be
made using that algorithm, and the remote codec must be configured just in the same way,
unless "SmartRTP" mode is activated: in this case, the encoding algorithm will be automatically
notified to the remote audiocodec.
In SIP mode, however, a list of encoding profiles is presented. Each profile is basically a list with
a name, including one or more precise encoding algorithms, and in order of preference. When a
unit calls another in SIP mode, it proposes this list, and the first one supported by both
audiocodecs will be the one finally adopted.
Management of the existing profiles can only be done by the control software and they remain
stored in the unit. The profiles are stored in the non-volatile part of the ALIO´s memory. From
the front panel of the ALIO the user can only select among a list of stored profiles.
When the user press and hold the "CODEC" key for more than 1 second, an informative screen
will show up describing the currently selected codec or profile for the active channel, without
making any modifications to it. Whenever a SIP call is established, the indicated codec will be
the one that is actually used.
3.7.4. "HELP" key.
This key, that is independent on the active channel, is not following the active channel and it
sends a notification to the remote control software provided that this is in use. This way, a
Phoenix ALIO user that requires assistance or has doubts can ask the operator that is remotely
controlling or monitoring the unit with the "AEQ ControlPHOENIX" software for help.
When the "HELP" key is pressed, a notification will appear on the OLED screen of the ALIO and
the key will start to flash in red. On the remote control software a pop-up notification will appear
asking the operator to get in touch with the ALIO user.
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AEQ PHOENIX ALIO
Once this notification is confirmed by the remote operator, the ALIO’s "HELP" key will stop
flashing and the user will be notified through the OLED display that the remote operator has
acknowledged the call for assistance:
The notification can be cancelled at any moment by pressing the "HELP" key again:
3.7.5. "MENU" key.
This key is not following the active channel and provides access to the configuration menu of
the ALIO. This menu also contains more advanced or less frequently used used options such as
Phoenix ALIO IP configuration, communications parameters (RTP, SIP and NAT parameters),
test options (loops, tones, etc.) or maintenance operations (firmware version, time/date
adjustment, system reboot, buzzer).
3.7.5.1. "ETHERNET" submenu.
The first available option is "ETHERNET" configuration. By entering this submenu (pressing on
the encoder button when the option is highlighted), we can access the IP configuration of the
unit, where we can specify whether it gets the configuration automatically (DHCP ON, requires a
DHCP server active in the local network).
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AEQ PHOENIX ALIO
If DHCP is not used, we need to select the second line in this list and where the current IP
address is shown. Click on the rotary encoder and this will open all the available fields for the
correct IP configuration of the ALIO and allows the user to change the settings for the IP
address, mask, gateway and DNS server:
There are two ways to modify each IP: clicking on the rotary encoder when the IP we want to
change is highlighted will allow us to change the value for each of the four sections of the IP
address between 0 and 255. We can increment or decrease by turning the rotary encoder.
Clicking on the rotary encoder when we have the desired value selected will set the value and
jump to the next section and so on until we have completed the IP address that is required. It is
also possible to enter the numbers with the numeric keypad of the ALIO always completing the
3 digits. For example, if we want to type "6", we should type "0", "0", "6", then confirm each
number clicking on the rotary encoder.
The default IP address for PHOENIX ALIO is 192.168.1.88.
If any of the IP or mask is altered (either by selecting DHCP or manually changing it), the unit
will ask the user for confirmation before rebooting and applying the new configuration when
leaving the "ETHERNET" submenu.
3.7.5.2. "COMMUNICATIONS" submenu.
Advanced parameters related to RTP audio protocol (used in both IP and SIP communication
modes) can be configured here:
•
RTP: Generic parameters that are
standard UDP type RTP (Real Time
protocol is always used, even when
configuration will be always available
mode.
always necessary for IP audio transmission using
Protocol) are configured in this menu. Note that this
a higher level protocol such as SIP is active, so its
and required even if the channel is configured in SIP
The parameters that can be configured in this menu are:
-
Receiving BUFFER type (FIXED or ADAPTIVE size), as well as its SIZE in
milliseconds, between 10 and 2000ms. A fixed size buffer is recommended whenever
possible, unless the opposite is indicated by AEQ for a certain application. The size to
select depends on the network performance, in particular, on the variation of the delay
in packet delivery (jitter). The size specified for the buffer must be higher than the
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AEQ PHOENIX ALIO
maximum variation. Note that a certain network may have a long delay, but quite
constant (low jitter), as is the case in good satellite links. In this case, the buffer size can
still be short.
Large buffer sizes results in unnecessarily long added delays. We recommend to start
with a value around 100 ms, that the user should increase only in case that drops or
occasional losses of synchronization is experienced. Such occasional loss is most likely
due to that the network jitter at times may be larger than those 100ms. In order to
diagnose network jitter, we recommend that the IP statistics available in the Web Server
are used (read section 7.6 of this manual).
•
-
Activation or deactivation of SYMMETRIC RTP mode, that sends the transmitted audio
stream to the same IP and port that the unit is receiving from, independently from the
setting of destination port (this option doesn’t appear when "SmartRTP" mode is active,
as it is no longer necessary).
-
We must also specify the local audio port here (port where the unit expects to receive
audio RTP traffic at). This LOCAL PORT is 5004 by default for program channel, and
5008 for the (optional) coordination channel. It must be a multiple of 4 and can be typed
in with the numeric key-pad (always with 5 digits) or modified with the "NAVI / Ch" rotary
encoder.
IMPORTANT NOTE: When audio local port is changed the unit must be restarted in
order to make and receive calls correctly.
-
Last, we can define here whether to use FEC (Forward Error Correction) or not. If
enabled, the protection level (and consequently, increased overhead) must be specified
among the 4 available options (LOWEST / LOW / MID / HIGH). Please refer to section
4.4 of this manual for more details about FEC.
SIP: Next option in the communications menu corresponds to SIP configuration of the
currently active channel. This option provides to display and modify information about the
configured SIP user: user, password and active SIP PROVIDER, when the channel is
configured in PROXY SIP mode (those parameters need to be setup in advance using
remote control software) or just the user when it’s configured in DIRECT SIP mode.
3.7.5.3. "MAINTENANCE" submenu.
•
FW VERSIONS: This menu can be accessed to check, for informative purposes only, the
currently installed firmware versions (with their dates) of the different programmable devices
within the unit (CPU, DSP and FPGA). The audio processor code is included in the CPU
firmware file.
If, following indications from AEQ SAT, any device version must be updated; this operation
must be done by using the equipment’s Web Server, as explained in section 7.1 of this
manual.
NOTE: Firmware updates don’t modify (unless explicitly stated by AEQ) the current
configuration of the unit and, in particular, its IP setup.
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AEQ PHOENIX ALIO
•
DATE/TIME: It is possible to know the unit’s date and time by selecting this option.
In order to adjust the date, connect a PC and use the "CONFIG"
"Miscellaneous"
"Equipment date and time" menu option in "AEQ ControlPHOENIX" software.
•
TESTS: We can find some test functions under this menu, such as activation of audio loops
for each channel, and test tone insertion (that substitute the corresponding input) for inputs
MIC1 and MIC2, that we can route to the desired channel or CUE in order to make level
adjustments, check audio connectivity, etc.
-
The first and second lines allow for the activation of an audio loop in PROGRAM and
COORD channels. This loop is linear (no compression). Audio entering the TX bus will
be reflected back in the receive (RX) bus. This will also be shown in the corresponding
vumeters and can be useful to measure audio performance, adjust levels, check
connections, etc.
NOTE: LINEAR LOOP will become automatically deactivated when leaving the TEST
menu, to avoid that it is accidentally left active.
-
The third line option enables / disables the substitution of MIC1 input by a 1 KHz tone at
0 dBV level (adjustable between –inf and +18 dBV by means of the MIC encoder). The
mentioned tone will be sent to the bus where MIC1 is routed, and the microphone will
remain muted, if it is connected.
-
The fourth line enables / disables the substitution of MIC2 input by a 2 KHz tone at 0
dBV level (adjustable between –inf and +18 dBV by means of the MIC encoder). The
mentioned tone will be sent to the bus where MIC2 is routed, and the microphone will
remain muted, if it is connected.
•
BUZZER: The unit buzzer signals incoming calls and can be activated or deactivated
through this function (it will sound once as a test whenever its status is changed from OFF
to ON in this menu).
•
DEFAULT SETTINGS: The systems factory settings can be restored at any time through
this option. However, the units IP configuration will remain unchanged avoiding the loss of
connectivity and remote control. Confirmation will be required before reseting the unit’s
adjustments.
IMPORTANT NOTE: The SIP users configuration of the unit will be lost when this option is
activated. It’s important to take note of these parameters before activating it in order to
reconfigure them afterwards.
•
REBOOT: The unit can be reset or rebooted in the event of unexpected or incorrect
performace. Confirmation will be required before rebooting the unit.
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AEQ PHOENIX ALIO
4. CONFIGURATION AND OPERATION FROM REMOTE CONTROL SOFTWARE.
Configuration and operation of Phoenix ALIO can be carried out remotely by means of the
application "AEQ ControlPHOENIX" (AEQ Phoenix STRATOS, STUDIO, MERCURY, VENUS,
VENUS V2, VENUS V3 and ALIO Audiocodecs Configuration and Control Software).
The version that is provided together with the equipment (2.2.1.20 or higher) can control up to 2
units per software instance. If you need to manage more than 2 Phoenix audiocodecs at the
same time, please contact AEQ sales department to purchase a multicodec-license for "AEQ
ControlPHOENIX".
Furthermore, version 2.2.1.20 (or higher) does NOT allow you to control units with CPU version
5.50 or higher and units with previous versions SIMULTANEOUSLY. In that case, these last
ones should be upgraded to version 5.50 or higher.
Please have the user manual of this application at hand. Install and configure it and add the
equipment to the controlled equipment list in order to follow the explanations provided in this
and following chapters step by step. This manual will describe particular ALIO options only, as
well as some important procedures, while the detailed operation is detailed in "AEQ
ControlPHOENIX" user’s manual.
4.1. Individual codec control window.
The individual codec control window is thoroughly described in chapter 6 of "AEQ
ControlPHOENIX" user’s manual. The screen corresponding to Phoenix ALIO is as follows:
The name assigned to the device can be seen in the lower area of the window ("ALIO1" in this
example), as well as the URI corresponding to both channels (URI-1
PROGRAM, URI-2
COORDINATION)
Additionally, a blue link is presented with the equipment’s IP address. By clicking on this link, an
Internet browser will pop up showing Phoenix ALIO Web management window allowing, among
other things, to update firmware and obtain real-time IP traffic statistics when the channel/s
is/are connected.
When the unit has no license for COORDINATION channel activation, the screen will look
slightly different, as the control area for that channel will appear deactivated:
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AEQ PHOENIX ALIO
The left zone shows the general status of both communications channels, PROGRAM (PRG)
and COORDINATION (CRD) (CONNECTED, OK, REGISTERING, etc), as well as the
configured working mode and encoding and the remote equipment’s data (IP address and port
or name) in case it is connected. We can click in any of both areas in order to show a window
that provides all the details of the channel we have clicked on:
•
•
•
•
INTERFACE: indicates the operating mode of the channel (RTP Raw, DIRECT SIP or
Proxy SIP)
Coding: indicates the coding algorithm or profile: OPUS, G711, G722, MPEG L2, <SIP
CODEC PROFILE>… This section also indicates binary rate (128 Kbps for example),
the sampling frequency (48 KHz, for example) and the mode (Mono, Stereo, Dual,
JStereo or MS Stereo).
Status: SIP registering status for IP connections using Proxy SIP mode
("REGISTERING", "REGISTRATION_ERROR", "REGISTERED") or interface status
("OK", "CONNECTED", "CONNECTED_NO_DATA / NO_SYNC").
CONNECTED TO: calling/called equipment’s name or number (identifier, number or
Unknown when ID is not provided) or "not connected" when there is no established
communication.
The lower part of the individual codec window identifies each unit by its given name, IP address
and each channel URIs (identifier for SIP calls) for each of both channels:
On the right part, for both channels, PROGRAM and COORDINATION, we can find the "CALL"
button/indicator and the "SYNC" indicator, as well as two audio presence indicators for both
directions: transmission ("Tx") and reception ("Rx").
4.1.1. CONFIG Menu.
At the right side, "CONFIG" button gives access to a configuration menu with the following
options: "General", "Contacts", "Ethernet", "Miscellaneous" and "Network".
Just click on "CONFIG" button again in order to close this menu.
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AEQ PHOENIX ALIO
The "General" option is the most important of the ones associated to "CONFIG" button: you can
configure the audio routing and levels from/to the equipment, the selected audio encoding
algorithm, the interface to be used (from "INTERFACE" drop-down menu) and access to
"Advanced" channel configuration and IP interface configuration ("I/F setup" button).
If the unit doesn’t have a valid COORDINATION channel activation license, the appearance will
be slightly different, as the control zones for that channel will appear deactivated:
30
AEQ PHOENIX ALIO
The general view offers a graphic display of the audio flow inside the unit. The equipment’s
inputs are shown at the left, entering the input mixer (that can be open by clicking on the "Config
Mix" button –see paragraph 4.1.2).
This mixer outputs three buses: PROGRAM send (depicted in red), COORDINATION send
(depicted in green) and CUE prelisten (represented in blue).
The first bus is sent to the PROGRAM block where several aspects related to communications
(communication type, ports, etc), coding algorithms, etc. can be configured.
The second bus is sent to the COORDINATION channel that, if licensed, allows the same
configuration as PROGRAM.
The CUE bus is routed directly to the outputs block, as explained below.
The output block is presented at the right, where two stereo headphones and a stereo line
output appear. PROGRAM and COORDINATION send and receive buses, as well as CUE,
arrive to this block in order to be able to output or monitor them.
The "OUTPUT MIXER" block allows for the configuration of the assignment of buses to outputs,
level control and send/receive balance adjustment for each one.
The amplifier symbols at the inputs are filled with the color corresponding to the bus they are
being routed to, in order to ease a quick identification of the routing at a glance. Each output, on
the other hand, is colored with the color code of the bus that is being routed to it.
NOTE: The "Config Mix" button opens the complete mixer that controls both the inputs and the
outputs, just the same as the "MIX" button located at the right side of the individual codec
window, as you can see in section 4.1.2.
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AEQ PHOENIX ALIO
Other available options within the "CONFIG" menu are "Contacts" (call book management),
"Ethernet" (IP configuration of the device), "Miscellaneous" (various adjustments) and "Network"
(auxiliary network functions configuration). The details of these menus are described in chapter
6 of "AEQ ControlPHOENIX" user’s manual. The "Miscellaneous" option allows you to activate
the COORDINATION channel, if the corresponding license has been purchased.
Just click on the "CONFIG" button again to close this menu.
4.1.2. Mixer control window.
Also at the right side, the "MIX" button opens a window where all parameters of the ALIO mixer
can be controlled: level and gain for all inputs, input routing, tone adjustments, MIC4 or LINE IN
selection, output routing and level, etc.
Audio can be controlled in real-time from this screen, alternatively or in parallel with the unit
front panel, from the control PC, that can be located either at the side of the unit or remotely.
MIC1, 2 and 3 input channels feature the following controls:
-
Send assignments: allow the user to select which bus the corresponding input is sent to
(PROGRAM, COORDINATION or CUE). These buttons are mutually exclusive, and when
activated, they will be filled in the color assigned to the corresponding bus (red, green or
blue, respectively), as well as the input amplifier symbols in the General view, allowing for a
quick identification of the inputs routing at a glance.
-
Mix fader: allows adjustment of the corresponding input mix to the selected bus between
mute and +18dB. The selected value is shown below the fader, at the right of "PH" button.
-
Tone controls: bass and treble, permitting an adjustment between -12 and +12dB,
individually for each input.
-
"PH" (Phantom) button: activates or deactivates Phantom supply (12V / 10mA) for the
corresponding microphone.
-
Gain: clicking this button opens a window to control the gain of each preamplifier in a range
from 0 to +65dB (for microphone inputs only).
MIC4 and LINE IN inputs are mutually exclusive, what means that, once the bus send
assignment is changed from one of them, the other will become inactive.
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AEQ PHOENIX ALIO
The right area of the Mixer window allows control of the three outputs (2 headphones + LINE
out) in a similar way: the source can be selected for each output from CUE, PROG or COORD
buses. These buttons are mutually exclusive, and when activated, they will be filled in the color
assigned to the corresponding bus (red, green or blue, respectively), as well as in the output
amplifier symbols in the General view, allowing for a quick identification of the outputs outing at
a glance.
Each output’s volume can be varied between 0 dB and mute. The balance between
transmission and reception that the ALIO operator has adjusted on the front panel can be
viewed as well, but only for informative purposes (this adjustment cannot be remotely
controlled).
4.1.3. Vumeters window.
Last, you can access to the equipment’s remote vumeters by clicking on the "VU" button located
below "MIX" one. They will appear into a floating window that you can move to the desired
position. By default, it is possible to open up to 2 vumeters windows in the application (for
example, to check audio transmission between two connected equipments under the
application’s control). If you try to open a third vumeter window, the first one will be closed. Click
on the right top cross in order to close a vumeters window.
The maximum number of vumeters that can be simultaneously displayed can be altered in
the "Remote connection" tab of "General Application Values" (see section 5.1.9.1 of "AEQ
ControlPHOENIX" user’s manual).
The represented vumeters correspond, for both
channels (PRG=PROGRAM, CRD=COORDINATION),
and from left to right, to audio transmitted to the
channel (L and R) and received from it (also L and R).
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AEQ PHOENIX ALIO
4.2. Connection modes
In order to establish an IP communication using PROGRAM or COORD channels, first we need
to choose one of the three available connection modes: "PROXY SIP", "DIRECT SIP" and "RTP
Point to Point (RAW)" from the "INTERFACE" drop-down menu of the desired channel. This is
the same as selecting the communications mode with the front panel "SIP" / "IP" and the use or
not of an external Proxy under MENU
COMMUNICATIONS
SIP.
We can access the IP configuration submenu by clicking on "I/F Setup". This menu is described
in sections 6.1.4.2 and 6.1.4.3 of "AEQ ControlPHOENIX" user’s manual.
It is important to know the details of each type of connection, so they are explained below.
4.2.1. RTP Point to Point (RAW).
This type of connection is selected when the connection over IP will be an RTP type link with
calling of the (IP address:port) to (IP address:port) type. Obviously there is no advanced
signaling protocol in this scenario and you will need to established, set parameters and
disconnect communication from both ends. Audio encoding must be the same (and explicitly
specified) in both ends of the communication.
In order to avoid that hassle, making the calling, hanging up and coding selection tasks easier,
as it will be necessary to do it from one end only, "SmartRTP" can be activated in both involved
audiocodecs (provided that they are AEQ Phoenix compatible with this mode).
If the operating mode required for a contact is "RTP raw", the only valid equipment identifier is:
<IP_address>:<destination port>, for example: "172.26.33.28:5008"
The specified destination port must match the Local port set up for the remote equipment. That
is, in order to make a RTP call, we must know the IP address and local port of the remote unit
even if "SmartRTP" is used.
When you create a Call Book, these fields describing a contact can be
modified in the Call Book that can be accessed from a codec individual
control window through the "Contacts" option in "Configuration" (see
section 5.1.7 of "AEQ ControlPHOENIX" user’s manual).
You can access the IP configuration submenu for RTP Raw mode by clicking on "I/F Setup"
button, and that it is explained in section 6.1.4.3 of "AEQ ControlPHOENIX" user’s manual.
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AEQ PHOENIX ALIO
The parameters to be configured are:
•
FEC mode: this option allows you to configure whether FEC (Forward Error
Correction) is used or not (there is a trade-off for a bigger binary rate). See section
4.4.
•
Local media port: this option allows you to configure the value of the IP port
selected to transmit audio at origin over IP. Minimum value 1,024. Maximum value
65,534. Default values: 5004 (PROG) and 5008 (COORD).
•
Adaptive/Fixed and Adaptive buffer max/Fixed buffer length: this option allows
you to configure the type and maximum size of reception buffer. See section 4.4.
•
Symmetric RTP: when "SmartRTP" mode is not activated, this advanced option at
least allows the user to force the local unit to send audio to the same IP and port
from which it is receiving audio. The destination port specified when making the call
will be ignored when we receive packets from the remote equipment. This option
will allow you to connect to an audiocodec with unknown IP and/or port (because
it’s behind a router with NAT, for instance).
Please notice that "RTP Point to Point" is a complex configuration mode, suitable for permanent
connections that some equipment may not support. Specifically, it can’t be used to connect to
Phoenix Mobile. However, it becomes an easy and effective way to operate with the aid of
"SmartRTP".
4.2.2. PROXY SIP.
This type of connection is selected when the Phoenix ALIO is used working together with an
external SIP server that will provide connection with remote unit through any network, even
Internet, without knowing its IP address. Both units (local and remote) must be registered in SIP
server, which function is to maintain a database with the registered codecs, storing their
connection parameters (IP address, audio ports…) in order to ease the task of making calls
between them even when connected to different networks.
In order to make a call in Proxy SIP mode, you must take into account that for the URI or SIP
identifier of the equipment in question you can use any of the following syntaxes:
o
o
o
o
<unit_name>, for instance: "phxalio_231" o "phoenixMaster"
<unit_name>@<realm_SIP_server>, for instance:
"phxalio_231@sip.aeq.es" or "phoenixMaster@sip.aeq.es"
<unit_name>@< SIP_server _IP>, for instance:
"phxalio_231@232.168.1.2" or "phoenixMaster@232.168.1.2" (where
232.168.1.2. is the AEQ´s SIP server "sip.aeq.es")
<unit_name>@<SIP_server _IP>:<Port>, when the SIP port of SIP server
is not the 5060 (SIP Standard port) the identifier must include the used port,
for instance: phxalio_231@sip.aeq.es:5061
When you create a Call Book, these fields describing a contact can be
modified in the Call Book that can be accessed from a codec individual
control window through the "Contacts" option in "Configuration" (see
section 5.1.7 of "AEQ ControlPHOENIX" user’s manual). In order to call
a same contact using different IP modes (as defined in "INTERFACE"
drop-down menu), different contact entries must be created.
You can access the IP configuration submenu for PROXY SIP mode by clicking on "I/F Setup"
button, and that it is explained in section 6.1.4.2 of "AEQ ControlPHOENIX" user’s manual.
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AEQ PHOENIX ALIO
•
In "SIP Parameters" submenu you can find:
-
-
•
User Name: enables you to edit the name of the unit and how it will be reflected
in the diverse internal menus of the unit. For a start we recommend you not to
change the configured default User Name, "phxalio_231" for instance.
Display Name: editable name. This is the public name of the equipment that
will be used in SIP server, so you can recognize the equipment with this
identifier externally to the system.
Proxy SIP Account: enables you to select a Proxy SIP account from a
previously created and stored list. In case an account is selected, the
parameters described next would be automatically loaded (confirmation is
requested).
Proxy Provider: enables you to select the external SIP server with which the
unit will work from a previously stored list. By default, AEQ server will be
selected.
Authentication: enables you to edit the password and security information for
the user profile associated with the unit in the previously selected SIP server.
By default, the data configured in this field in order to use AEQ server are the
following:
o User: the "User Name" configured in Factory, "phxalio_231" for
instance.
o Pwd: the Password associated to that user.
o Realm: the domain where the SIP Server is located, by default:
sip.aeq.es.
You can find the NAT mode selection at "NAT Traversal" submenu.
NAT Traversal is a set of tools used by the equipment in order to surpass the NAT
(Network Address Translation) performed by some routers. We can select among
several modes depending on the kind of network the unit is connected to.
Phoenix ALIO offers a total of six different operating modes when traversing
devices with NAT (routers, firewalls…). Each one of those modes is suitable for a
different scenario. For instance, when the units that are establishing communication
are in the same local network, the internal working way will be different than when
it’s done through the Internet.
See more details in section 4.3 of this manual.
•
The rest of options to be configured are:
o FEC mode: this option allows you to configure whether FEC (Forward Error
Correction) is used or not (there is a trade-off for a bigger binary rate). See
section 4.4.
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AEQ PHOENIX ALIO
o
o
o
Local media port: this option allows you to configure the value of the IP
port selected to transmit audio at origin over IP. Minimum value 1,024.
Maximum value 65,534. Default values: 5004 (PROG) and 5008 (COORD).
Adaptive/Fixed and Adaptive buffer max/Fixed buffer length: this option
allows you to configure the type and maximum size of reception buffer. See
section 4.4.
Symmetric RTP: this option allows you to force the local unit to send audio
to the same IP and port from which it is receiving audio. The destination
port specified when making the call will be ignored when we receive
packets from the remote equipment. This option will allow you to connect to
equipment with unknown IP and/or port (because it’s behind a router with
NAT, for instance).
Each unit will send audio to the "Local media port" of the remote equipment automatically,
thanks to the SIP signaling protocol. That signaling also accomplishes coding profile negotiation
and call establishment / release from any of both sides of the communication once the remote
equipment has been identified by its IP address and reached.
4.2.3. DIRECT SIP.
This type of connection is selected when you have a connection with SIP protocol in the
signaling phase prior to connection but without the presence of an external SIP server. It is
necessary to know the IP address of the equipment you want to call in advance, but not
necessarily the audio ports.
In order to call in Direct SIP mode, you must take into account that for the URI or SIP identifier
of the equipment the right syntax is "<unit_name>@<unit_IP_address>" type (for instance,
"phxalio_231@172.26.5.57").
When the correspondent SIP port is not the 5060 (SIP Standard port) the identifier must include
the used port. For instance: "phxalio_ 231@172.25.32.11:5061".
When you create a Call Book, these fields describing a contact can be
modified in the Call Book that can be accessed from a codec individual
control window through the "Contacts" option in "Configuration" (see
section 5.1.7 of "AEQ ControlPHOENIX" user’s manual). In order to call
a same contact using different IP modes (as defined in "INTERFACE"
drop-down menu), different contact entries must be created.
You can access the IP configuration submenu for DIRECT SIP mode by clicking on "I/F Setup"
button, and that it is explained in section 6.1.4.2 of "AEQ ControlPHOENIX" user’s manual.
•
In "SIP Parameters" submenu you can find:
-
•
User Name: enables you to edit the name of the unit and how it will be reflected
in the diverse internal menus of the unit. For a start we recommend you not to
change the configured default User Name, "phxalio_231" for instance.
Display Name: editable name. This is the public name of the equipment, so
you can recognize the equipment with this identifier externally to the system.
You can find the NAT mode selection at "NAT Traversal" submenu.
NAT Traversal is a set of tools used by the equipment in order to surpass the NAT
(Network Address Translation) performed by some routers. We can select among
several modes depending on the kind of network the unit is connected to.
Phoenix ALIO offers a total of six different operating modes when traversing
devices with NAT (routers, firewalls…). Each one of those modes is suitable for a
different scenario. For instance, when the units that are establishing communication
are in the same local network, the internal working way will be different than when
it’s done through the Internet.
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AEQ PHOENIX ALIO
See more details in section 4.3 of this manual.
•
The rest of options to be configured are:
o FEC mode: this option allows you to configure whether FEC (Forward Error
Correction) is used or not (there is a trade-off for a bigger binary rate). See
section 4.4.
o Local media port: this option allows you to configure the value of the IP
port selected to transmit audio at origin over IP. Minimum value 1,024.
Maximum value 65,534. Default values: 5004 (PROG) and 5008 (COORD).
o Adaptive/Fixed and Adaptive buffer max/Fixed buffer length: this option
allows you to configure the type and maximum size of reception buffer. See
section 4.4.
o Symmetric RTP: this option allows you to force the local unit to send audio
to the same IP and port from which it is receiving audio. The destination
port specified when making the call will be ignored when we receive
packets from the remote equipment. This option will allow you to connect to
an audiocodec with unknown IP and/or port (because it’s behind a router
with NAT, for instance).
Each unit will send audio to the "Local media port" of the remote equipment automatically,
thanks to the SIP signaling protocol. That signaling also accomplishes coding profile negotiation
and call establishment / release from any of both sides of the communication once the remote
equipment has been identified by its IP address and reached.
4.2.4. Sending audio to multiple destinations: Broadcast, Multicast and Multi-unicast.
It is possible to send the same audio RTP stream to several different destinations in "RTP raw"
mode (see section 4.2.1). There are several possibilities to do so (see "AEQ ControlPHOENIX"
manual):
a) Broadcast: the audio stream can be sent to all the equipments within a local network, only
by specifying a special address in the destination address field. This address is calculated as
the network address with the equipment part filled with 1’s. For instance: if the IP address of our
codec is 192.168.20.3 and network mask is 255.255.255.0, the corresponding broadcast
address is 192.168.20.255. However, if the network mask was, for example, 255.255.0.0, then
the broadcast address would be 192.168.255.255. The audio will be sent to a given port, so the
receiving equipments should have "local media port" set up to this same port so they are able to
receive the RTP stream.
This mode is not recommended for big networks and is usually blocked by the switches and
routers, so its use is restricted to small, well managed local area networks.
b) Multicast: it is also possible to send the audio stream to a special "multicast" address, for
example, 239.255.20.8. If the receiving equipments call to that same IP, they will receive the
audio that is being sent provided that their "local media port" matches the one the transmitter is
sending the packets to. Phoenix ALIO implements IGMP (Internet Group Management Protocol)
in order to subscribe to multicast group. Similarly to broadcast, multicast traffic is usually
blocked by switches and routers, so its use is restricted to local area networks too.
c) Multiple-unicast: Phoenix units can send the same RTP stream to several different IPs by
replication of the encoded audio. This can traverse switches and routers in the same way it
would do if it was a simple (unicast) RTP Raw stream, although it is limited to a certain number
of destination IPs depending on the type of coding algorithm
That parallel streams or "replicas" at nothing more than IP address + port pairs, where audio
copies are to be sent normally. When a contact is created or edited, it is also possible to specify
whether particular replicas use FEC (forward error correction) if it is enabled for that channel, or
disable it for certain streams (because they use stronger links, for example). If the above list is
empty, the audio stream will be sent to the IP + port specified when making the call (main
address in the contact).
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AEQ PHOENIX ALIO
In order to send replicas, a new advanced contact must be created first in the General agenda
where a main IP address and port is specified and a list of additional replicas is provided. In
order to do that, click on the "Contacts" button at the top menu bar, select "New Contact" (make
sure that the agenda selected at the left column is "General" and proceed with the creation of
the new contact:
Give the new contact a name (I.e: "RTP_REPLICA1"), select RTP as the interface mode,
specify the main destination IP address and port (172.26.33.55:5004, respectively, in this
example) and click on the "Advanced" button to specify the selected coding method and add
replicas to the list.
The OPUS MUSIC STEREO 48 KHz / 64 Kbps mode will be used in this example, configuring
the contact to issue replicas to 3 different IP/Port pairs (we have used the "New Replica" button
in order to do that).
Detail of the creation o RTP replicas within a General agenda contact
39
AEQ PHOENIX ALIO
Please check chapter 5.1.7.1 of "AEQ ControPhoenix" User’s manual for more details.
We need to select this contact that includes replicas from the call window when making a call.
Once the available contacts are displayed, select the one we are interested in ("replica3" in this
example), by double-clicking its name, and we will be returned to the call window.
At this moment we are able to check that the specified main IP address and port fields fill the
"Call to" field, and we can also check that the replicas are going to be loaded into the unit, by
clicking on the "Replicas in Contact" before actually making the call:
Once we click on the green "Call" button, the unit will start emitting the audio streams to the
main address and to the specified replica addresses.
ALIO will stop sending replicas as soon as the call is hung. The list will be erased from the unit
and the only way to send them again is to call selecting the same contact.
However, if the Permanent Call mode is activated and there is a mains cut, for example, the unit
WILL send all configured replicas when rebooting.
NOTE 1: When audio is transmitted to several destinations, it can be received from only one of
them (or none) In order to establish which of the units sends the audio back, and only in RTP
Raw mode, a control will appear in the general configuration window that allows for the
activation / deactivation of the transmitted stream to the IP channel. Make sure that only one of
the units connected to the multicast transmitter has this checkbox activated.
40
AEQ PHOENIX ALIO
Channel transmission active
Channel transmission disabled
NOTE 2: Advanced contacts (those allowing specification of the communication mode, coding
algorithm/profile, replicas, etc) can only be stored in the General agenda, that is saved in a
database in the control PC. These contacts can however be copied to the different devices, but
the advanced fields (interface, coding algorithm, SIP account / provider and replicas) will be
lost, so only the contact name and contact data (main IP:port or destination URI) will be stored.
NOTE 3: It is possible to use multiple-unicast transmission with "SmartRTP" active. The
transmitter unit must be the one that generates the calls, and when it hangs up, it will send
notifications to the MAIN destination only (not in the replicas). As a consequence, only this one
will hang-up. If the user needs that all receivers hang-up too, the "Auto Hang-Up" option can be
activated on them defining a reasonable time (let’s say, 5-10 seconds).
If, on the other hand, we need to cut the call from a receiver, only that with transmission
enabled will make the others hang-up (even if it is not the main receiver).
Please read the application notes published by AEQ regarding IP connectivity for more
information on IP communications in particular scenarios.
4.3. NAT TRAVERSAL.
NAT Traversal is a set of tools used by the equipment in order to overcome the issues caused
by NAT (Network Address Translation), performed by some routers. We can select among
several modes depending on the kind of network the unit is connected to.
Phoenix ALIO offers a total of six different operating modes when traversing devices with NAT
(routers, firewalls…). Each one of those modes is suitable for a different scenario. For instance,
when the units that are establishing communication are in the same local network, the internal
working way will be different than when it’s done through the Internet.
Four of the six modes are automatic (AUTO 1 - AUTO 4), another one is manual (MANUAL router configuration) and the last one (OFF - there is no NAT) is used when no devices with
NAT are crossed (the unit is in a local network or connected to the Internet with a single
workstation router). In automatic modes the unit tries to find out its public IP and ports without
the user help, while in manual mode the unit gets those data directly from user (and user gets it
from network administrator).
Due to the technical complexity inherent in most of the parameters involved in this NAT
TRAVERSAL menu and the importance that any modification has in the final operation of the
unit, we recommend that only highly qualified personnel in possession of all the technical
documentation and manuals work on this NAT configuration menu. For additional information,
see APPENDIX B5.
The NAT traversal options of a codec are accessed by following this sequence from the
involved individual codec control window: "Configuration"
"General"
"I/F Setup"
"NAT
Traversal".
Next we will describe the operation without NAT and the other five modes supported by Phoenix
ALIO.
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AEQ PHOENIX ALIO
4.3.1. Operation without NAT: "OFF (there is no NAT)".
The unit uses no mechanism to traverse devices with NAT. This mode will be used only to
operate in the local network (all of the SIP participants are in the same local network, including
the Proxy SIP, if we use it).
4.3.2. Manual NAT: "MANUAL (router configuration)".
This mode will be used when the unit is connected to a local network with shared Internet
access, through a router that will work as NAT (Network Address Translation). In order to use
this mode no DHCP must be used and you need to have access to router configuration (and the
knowledge to do it) or to the Network Administrator that will give us some data to be configured
in the unit and configure the router to open and redirect some IPs and ports (port forwarding).
The procedure is described in detail in the following Application Note (we recommend you to
read it when you decide to use this working mode):
AEQ PHOENIX AUDIOCODECS. APPLICATION NOTE 0 - C
(Connecting AEQ Phoenix units through Internet. Complex scenario configuration. Through
LOCAL network(s), DHCP not used, manual NAT. Making use of AEQ Proxy SIP).
The eight parameters to be configured in the dialog for this mode are:
1. SIP LOCAL IP: read-only parameter that tells you the IP of the IP interface of the
unit as regards SIP, so that the latter can, in turn, convey this to the router or
firewall administrator when it is configured. For instance 172.26.33.35. It can be set
in order to adapt it to network necessities in menu: "Configuration"
"Ethernet".
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AEQ PHOENIX ALIO
2. SIP LOCAL PORT: read-only parameter that tells you the port of the IP interface of
the unit used for SIP signaling, so that the latter can, in turn, convey this to the
router or firewall administrator when it is configured. Before checking the value of
this parameter you should have configured previously whether you want to work
with Proxy or not and restart the unit.
3. SIP PUBLIC IP: parameter that will tell the unit which public IP will correspond to it,
so that it can include the said IP in its SIP messages. The router or firewall
administrator must tell you the value of this parameter. For instance
212.170.180.177
4. SIP PUBLIC PORT: parameter that will tell Phoenix which public port it will have
corresponding to its local SIP port. The router or firewall administrator must tell you
the value of this parameter in order to make the required port-forwarding. For
instance: 8001.
5. RTP LOCAL IP: read-only parameter that tells you the IP of the IP interface of the
unit as regards RTP, so that it can, in turn, convey this to the router or firewall
administrator when it is configured. You will usually configure the same network
interface as for SIP, so it will be the one configured in point number 1: for instance
172.26.33.35
6. RTP LOCAL PORT: read-only parameter that tells you the port of the IP interface
of the unit as regards RTP, so that the latter can, in turn, convey this to the router or
firewall administrator when it is configured. Usually the shown port is 5004.
7. RTP PUBLIC IP: parameter that will tell the unit which public IP will correspond to
the RTP of its IP interface, so that it can send the said IP in its SIP messages. The
router or firewall administrator must tell you the value of this parameter. Usually the
administrator will take out SIP traffic and RTP using the same public IP configure in
point number 3. For instance 212.170.180.177
8. RTP PUBLIC PORT: parameter that will tell the Phoenix ALIO which public port will
correspond to the RTP of its IP interface, so that it can send the said port in its SIP
messages. The router or firewall administrator must tell you the value of this
parameter in order to make the required port-forwarding. For instance: 8002
In the aforementioned note the detailed configuration and need of port-forwarding is explained.
4.3.3. AUTO 1 (local network audio).
This mode will be used primarily when two units that are in the same local network need to
communicate with each other, when the Proxy SIP is on the Internet and it’s the one provided
by AEQ (sip.aeq.es).
4.3.4. AUTO 2 (local network audio).
This mode will be used primarily when two units that are in the same local network need to
communicate with each other, when a Proxy SIP on the Internet is used, it’s not the one
provided by AEQ (sip.aeq.es) and only if the AUTO1 mode doesn’t work properly.
4.3.5. AUTO 3 (audio over internet).
This mode will be used mainly when you wish to put two units in communication with each other
through the Internet, working with no Proxy (DIRECT SIP mode) or using the Proxy SIP
provided by AEQ (sip.aeq.es) (PROXY SIP mode). The two configuration parameters available
on screen for this mode are:
1. STUN SERVER: parameter that tells the unit the STUN server that will be used. On
the Internet there is multitude of public STUN servers. By default, the IP address of
stun.sipgate.net server is configured: 217.10.68.152.
2. STUN PORT: parameter that tells the unit the STUN server port assigned by the
administrator. By default: 3478.
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AEQ PHOENIX ALIO
NOTE: in this mode the Phoenix ALIO behaves in the exact same way as the Phoenix
Mobile unit when it is using a STUN server.
4.3.6. AUTO 4 (audio over internet).
This mode is equivalent to AUTO3 but it will be used the SIP server is not the one provided by
AEQ and there are problems with AUTO3 mode. The configuration parameters are the same as
for AUTO3 (STUN server specification).
4.4. FEC modes and reception buffer configuration.
-
FEC error correction mode. Error correction is performed by sending redundant
information that allows the receiver to recompose the lost data in case of not-perfect
transmissions.
Forward error correction always generates a higher binary rate, and this in turn can
generate more and more losses in very narrow transmission channels, as well as
delays. It is recommended that the communication is started with no FEC (OFF) and,
once established, experiment with the different available modes in case of problems
and check if the results are better with some of them.
•
•
•
•
LOWEST: generates a 40% higher binary rate and produces a 575ms
additional delay.
LOW: generates a 50% higher binary rate and produces a 375ms additional
delay.
MIDDLE: generates a 66% higher binary rate and produces 225ms additional
delay.
HIGH: duals the binary rate producing 125 ms additional delay.
-
Adaptive/Fixed: you can set up the reception buffer as adaptive or fixed. In the first
case, its size will vary according to the network transmission quality. In fixed mode, its
size will be steady according to manual configuration.
-
Adaptive Buffer Max/Fixed buffer length: this is the maximum size of the reception
buffer. When it is defined as adaptive, Phoenix ALIO will start to shorten it from this
value as the network´s transmission quality allows. If it is defined as FIXED, this max
value will remain, as the buffer’s size won’t be varied during the connection. This value
must be set in milliseconds. The longer the buffer is, packet misses will be less likely,
but base delay will also be longer, especially if the buffer is set to FIXED mode.
In order to help you select the best option for each application, we recommend to use a Fixed
buffer, with a low value (around 100ms) in applications where optimal audio quality is the main
concern (mainly when using PCM modes in suitably sized networks). If the received audio
quality is as expected, and the network allows for it, you can continue adjusting the buffer to
lower values in order to minimize delay, until you find that audio is compromised (as the buffer
size reaches the network maximum jitter value). At this point, just increase the buffer a little bit
to have some margin.
In high-quality PCM connections, you can start using highest quality modes (48 KHz, 24 bits,
mono or stereo only if required), and if you can't obtain the desired quality and/or stability (no
noises present) and good delay, you can lower quality progressively until, for example, 16 bit
(CD quality audio).
On the other hand, for applications where lowest possible delay is the main goal, but
transparent audio is not necessary (for example, in voice connections with commentators), it is
better to select the Adaptive Buffer mode, starting from a 1000ms maximum size, approx. If the
network is not too bad, the unit won't increase the buffer to highest values from the network's
jitter value, and it will try to minimize delay continuously. Please not that if the network has very
variable delay, the adjustments required to increase or decrease the buffer size can produce
noticeable artifacts in the received audio, so this method is not recommended for PCM modes
where maximum quality is required, in this case a fixed buffer setting is preferred, as stated
above.
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AEQ PHOENIX ALIO
4.5. Coding algorithm selection.
See section 6.1.3.1 ("Coding selection") at "AEQ ControlPHOENIX" user’s manual.
Although AEQ recommends the use of
OPUS coding algorithms for most uses
(several different modes are available to
match almost any need), a coding
selection window can be accessed by
clicking on the "Select codec" button
located inside the "ENCODER" area of
the general configuration window, where
both OPUS modes and others provided
for compatibility can be found.
Note that the "DECODER" will be
automatically configured for the same
coding algorithm and mode.
However, when the interface
is IP and configured in any of
the SIP modes ("DIRECT
SIP" or "PROXY SIP"),
particular coding algorithms
won’t be selected and
connection "profiles" will be
defined instead, containing
one or more modes. This is
like that because SIP allows
the
participants
in
a
communication to negotiate
the coding algorithm, so the
one to be finally used will be
limited to those includes in
the selected profile.
This possibility allows the configuration of the parameters associated to the coding to be used in
Audio over IP networks basing on SIP protocol (Proxy SIP and Direct SIP modes). This option
simplifies the selection of the algorithm to be used in a communication, because most of the
codecs have several tens of encoding algorithms in order to have the higher compatibility with
other equipments.
When a communication is established using SIP signaling, the codec negotiates the use of the
first compatible encoding algorithm included in a list called SIP CODEC PROFILE. That’s why
you should put these algorithms in order of preference.
Each one of the stored entries includes an alphanumeric identifier and a list of algorithms to
use, organized in order of preference. There several preset profiles in the unit, grouped by
delay, quality… etc.
Profiles can be added, modified or deleted in the "Encoding Profile
Management (SIP)" screen, accessible from the "Tools" menu in
the upper Menu bar, described in paragraph 5.1.8 of "AEQ
ControlPHOENIX" user’s manual.
45
AEQ PHOENIX ALIO
4.6. Ethernet Port configuration.
The "Ethernet config" menu allows you to configure the IP parameters of the Ethernet interface
in the unit.
The parameters to be configured are:
-
Enable DHCP: enables the activation or deactivation of the automatic configuration of
IP addresses, masks and gateways. There must be a DHCP server in the network the
unit is connected to in order to make this option work. When "Enable DHCP" is
validated, the following parameters will be filled automatically; when "Enable DHCP" is
not validated you will be able to change them manually.
-
IP Address: valid IP address associated with that interface.
-
Subnet mask: valid subnet mask associated with that interface.
-
Gateway IP: valid gateway or network gateway address associated with that interface.
-
DNS Server: IP address of the external addresses resolution server, valid in the
geographic zone where codec is placed, or internal server (inside the local network)
authorized to translate alphanumeric URL identifiers into IP addresses.
Once those parameters are configured, and after pressing the "Apply" button, a confirmation
window will appear. After confirming, the equipment reboots and the communication reestablishes in approximately 15 seconds.
If you have any doubts, please consult your IT network technician or directly contact the AEQ or
authorized distributors’ technical support department.
4.7. SNMP configuration.
This unit can be remotely managed using SNMP (Simple Network Management Protocol) using
one of the many client pieces of software available in the market, even for free. SNMP allows
monitoring of the status of several pieces of equipment from very diverse manufacturers and
natures, as well as elaborating reports, generate e-mail alerts, etc.
You can access the configuration menu in "Configuration"
"Network".
46
AEQ PHOENIX ALIO
AEQ PHOENIX Audiocodecs (Mercury, Venus, Studio, Stratos and ALIO) can connect to up to
3 SNMP clients, installed in remote PCs, by simply configuring their IP addresses in the
"SNMP" tab of previous menu. Once one or more SNMP clients are connected and the
corresponding "MIB" descriptive file has been loaded (it can be downloaded from the
equipment's Web Interface: see section 7.8 of this manual), the audiocodec will send/accept
different types of informations to/from each client:
1. Alarms (Traps): Provide information relating to certain equipment's events that can be
considered as alarms. They usually have two possible states: Active or Inactive. The list of
alarms defined for Phoenix ALIO is as follows:
-
phxCh1NoTxAudioAlarm (Audio detection event at Channel 1's input)
phxCh1NoRxAudioAlarm (Audio detection event at Channel 1's output)
phxCh2NoTxAudioAlarm (Audio detection event at Channel 2's input)
phxCh2NoRxAudioAlarm (Audio detection event at Channel 2's output)
phxCh1NoAudioSyncAlarm (Sync event at Channel 1 while connected)
phxCh2NoAudioSyncAlarm (Sync event at Channel 2 while connected)
phxCh1CallEndAlarm (Call ended on Channel 1 due to incoming RTP traffic loss)
phxCh2CallEndAlarm (Call ended on Channel 2 due to incoming RTP traffic loss)
phxOtherAlarm (Other alarms, see name) -> Fail to register in SIP PROXY server,
activation / deactivation of BACKUP interface for Ch1 and/or Ch2.
coldStart alarm (Starting from unit off. This is a standard SNMP alarm, it appears only
one time and has no activation or deactivation).
These alarms are sent whenever they change, but we can choose from the SNMP client which
ones are shown / treated and which not.
The first 4 audio alarms are configurable and they are activated whenever the incoming or
outgoing (depending on the particular alarm) audio level is below a certain threshold for a given
time, parameters that are set either by means of the Control Phoenix software, or using the
following SET commands. For example, alarm "phxCh1NoTxAudioAlarm" will become Active
whenever audio from ch1 input has a level below the threshold defined by
"phxCh1TxAudioThreshold" during a longer time than specified by "phxCh1TxAudioInterval".
2. Configurations (SET): adjustments related to some of the above defined alarms (the SNMP
client will configure them by means of "SET" commands, although in the case of Phoenix units,
they can also be modified by means of "AEQ ControlPHOENIX" remote control software):
-
phxCh1TxAudioThreshold (Audio threshold for channel 1's input)
phxCh1TxAudioInterval (Audio interval for channel 1's input)
phxCh1RxAudioThreshold (Audio threshold for channel 1's output)
phxCh1RxAudioInterval (Audio interval for channel 1's output)
47
AEQ PHOENIX ALIO
-
phxCh2TxAudioThreshold (Audio threshold for channel 2's input)
phxCh2TxAudioInterval (Audio interval for channel 2's input)
phxCh2RxAudioThreshold (Audio threshold for channel 2's output)
phxCh2RxAudioInterval (Audio interval for channel 2's output)
3. Information messages (GET): showing a status, they don't arrive spontaneously or are
activated/deactivated like the Alarms, but they are requested by the SNMP client by means of
"GET" messages:
-
phxCh1Status (Channel 1's status) --> (OK, CONNECTED, CONNECTING,
DISCONNECTING, CALLING, NO LINK, REGISTER ERROR, REGISTERING, CALL
ERROR)
phxCh2Status (Channel 2's status) --> (OK, CONNECTED, CONNECTING,
DISCONNECTING, CALLING, NO LINK, REGISTER ERROR, REGISTERING, CALL
ERROR)
phxCh1TxAudio (Ch1's audio input indicator status). Indicates the status of the virtual
LED indicating audio presence, according to the corresponding configured parameters.
phxCh1RxAudio (Ch1's audio output indicator status). Indicates the status of the virtual
LED indicating audio presence, according to the corresponding configured parameters.
phxCh2TxAudio (Ch2's audio input indicator status). Indicates the status of the virtual
LED indicating audio presence, according to the corresponding configured parameters.
phxCh2RxAudio (Ch2's audio output indicator status). Indicates the status of the virtual
LED indicating audio presence, according to the corresponding configured parameters.
phxCh1OnAir (Ch1's ON AIR option activated or not)
phxCh2OnAir (Ch2's ON AIR option activated or not)
phxCh1Synced (Ch1's audio synchronized or not)
phxCh2Synced (Ch2's audio synchronized or not)
phxCh1BackupInterfaceActive (Whether or not Ch1's backup interface is being used on
a call)
phxCh2BackupInterfaceActive (Whether or not Ch2's backup interface is being used on
a call)
mib2.system (sysUpTime, sysContact, sysDescr, sysServices,…): Standard SNMP
commands indicating things such as equipment's turn on time, etc. For more
information,
please
check
MIB-II
specification
in
RFC1213
(http://tools.ietf.org/html/rfc1213 )
For more information, please consult section 7.7 of this manual and section 6.5.1 of "AEQ
ControlPHOENIX" application manual.
4.8. Password configuration.
Version 2.2.1.1 (or higher) of the application allows you to configure for the unit a numerical
password of up to 4 digits by clicking on "User Password" checkbox (in "Syslog" tab of
"Network management" menu).
That password will be requested whenever you try to
add the unit to the application, whether it’s done by
autodiscovery or manually (once the unit is added, the
password won’t be requested again although you close
and reopen the application, but it will be requested if
you delete the unit and try to add it again or if you add
it from another PC where the application is also
installed).
For more information, please consult section 6.5.3 of "AEQ ControlPHOENIX" application
manual.
48
AEQ PHOENIX ALIO
4.9. User interface lock.
From CPU version 6.80 and alter, it is possible to lock certain functions of the user panel from
from "AEQ ControlPHOENIX" software (version 2.2.1.20 or higher). The only way to recover
local control on the disable functions is to connect to the unit from the application and reenabling them.
When you try to access to a locked function, the display will show the following indication for a
few seconds:
By means of "User Interface permissions" option of "Miscellaneous" window (see section
6.4.8 of "AEQ ControlPHOENIX" user’s manual), the window that allows you to enable/disable
those permissions will be accessed:
All the permissions are active by default. In order to deactivate any of those permissions,
locking the associated function of the unit's user interface, you just have to deselect the
corresponding checkbox and press "Ok". In order to reactivate any of them, you just have to
select the corresponding checkbox and press "Ok".
These settings are stored on the device, so if it is controlled from another PC, they will still
apply.
These are the groups of actions that can be enabled/disabled:
•
•
"Change SIP/RTP mode": allows or forbids the user to change between RTP,
Direct SIP and Proxy SIP operating modes using the "IP" and "SIP" keys.
"Access AUTO menu": allows or forbids the user access to the call automation
functions ("SmartRTP", "AutoAnswer", "Auto Hang-Up", etc.) using the "AUTO" key.
49
AEQ PHOENIX ALIO
•
•
•
•
•
•
•
•
•
•
•
•
•
"Change coding mode / audio profile": allows or forbids the user to change the
encoding mode for next call (or the profile, when operating in SIP modes) using the
"CODEC" key. However, the actual coding for the selected channel can always be
checked by means of a long press on that key.
"Change IP configuration": allows or forbids the user to change the unit’s IP
address, mask, gateway or DNS, as well as changing from manual to DHCP mode
or vice-versa ("ETHERNET" submenu options).
"Modify communications settings (port, buffer, ...)": allows or forbids the user to
alter certain communication parameters such as the kind and size of the receiving
buffer, local RTP port, SIP parameters, etc. ("COMMUNICATIONS" submenu
options).
"Modify maintenance values (date, time, ...)": allows or forbids the user to alter
certain parameters such as date and time, activate loops or test tones, etc.
("MAINTENANCE" submenu options).
"Dial using IP / URI": when activated, the local user can make calls by manually
dialing <IP:Puerto> pair or a URI (when operating in SIP modes). Otherwise, calls
can only be made from the agenda or calls history list (provided that next option is
enabled).
"Dial using Contacts list / Last calls list": when activated, the local user can
make calls by selecting a contact from the agenda or a call from the calls history. If
the former option is also disabled, the user cannot make calls from the front panel
(but he/she will be able to answer incoming calls and, in any case, calls can be sent
from "AEQ ControlPhoenix" application by a remote operator).
"Hang up, reject or accept calls": allows or forbids the usuer to hang up, reject or
accept calls from the unit's interface.
"Change input routing (PROG/COORD/CUE)": allows or forbids the user to
change which bus each input is routed to (PROG / COORD / CUE) by means of the
corresponding "ON" key. The restriction affects all 4 inputs.
"Change input adjustments (preamp gain, tone, ...)": allows or forbids the user to
make adjustments on the configuration of any of the 4 inputs, such as preamp. gain,
equalization, Phantom power supply or shift from line to mic (adjustments accessed
by pressing the corresponding encoder).
"Modify input level": allows or forbids the user to modify the send level of the
inputs by turning the associated rotary encoder.
"Change output routing (PROG/COORD/CUE)": allows or forbids the user to
change the audio source for the headphone / line outputs by pressing the
associated encoder.
"Modify output level": allows or forbids the user to change the headphone and line
output level by turning the associated rotary encoder.
"Modify output TX/RX mix": allows or forbids the user to adjust the ratio of
transmission and reception of the corresponding output by turning the associated
TX/RX potentiometer.
50
AEQ PHOENIX ALIO
5. QUICK USER’S GUIDE. LOCAL OPERATION.
In order to deeply know the operation of Phoenix ALIO unit, it is strongly recommended that
the previous chapters are thoroughly read. In this chapter the basic actions to manually operate
the unit are described. If more detail is needed, please check the information provided in the
previous chapters.
5.1. Connecting the unit.
5.1.1. Power supply.
Power supply to the unit is provided by the provided AC/DC adapter unit or by means of a
homologated UPS. In any case, connection to the unit is made by means of the special latching
connector at the back as described in chapter 2.2.3 of this manual.
5.1.2. Audio.
Chapter 2 in this manual describes the physical connections present in the back and side
panels of the audiocodec in detail, but in a nutshell, this is the simplified procedure:
•
•
•
•
•
•
Connect the required microphones to inputs MIC1 to MIC4 at the back panel.
Select an adequate gain (usually around 40dB) using each input’s configuration menu,
accessible by simply pressing the corresponding encoder button (MIC1-MIC4/LINE).
Activate Phantom supply where necessary (check the microphone manual).
If required, connect the line input source to the XLR connectors at the right side of the
unit labeled as IN (L & R).
Connect one or two headphones to the HP1 and/or HP2 outputs at the right side of the
unit.
If required, connect the line output to the XLR connectors at the right side of the unit
labeled as OUT (L & R).
5.1.3. Communications interface.
•
Connect an Ethernet cable (CAT5 or better) finished in an RJ45 10/100 BT to the LAN
connector provided at the unit’s back panel. The selected cable must be straight when
the connection is made from the unit to a communications device (switch, router…). For
more information about the pinout of this port please check section 2.2.2 of this manual.
5.2. Turning the unit on.
•
•
•
Once the unit is connected to the power supply through its adapter, the OLED screen
will turn on after around 15 seconds*, showing AEQ logo and the audiocodec model
name.
Check that audio routing and levels are correct; if they are not, just operate the controls
to establish the desired configuration.
The unit is ready to be used.
* The "ESC / DEL" key (with a red telephone) will light in red for around 15 seconds when the
unit is turned on, until the welcome screens appear.
5.3. Establishing a communication.
Several operating modes are available, depending on the protocol used for the communication
initiation.
The simplest –yet still very effective- mode, that we recommend for a start, is IP mode (RTP
Raw) with the "SmartRTP" option activated (this option can be found under the AUTO menu),
as it will be adequate for most cases, together with the OPUS family of encoding algorithms.
51
AEQ PHOENIX ALIO
When operating with audiocodecs from other manufacturers, you can use either IP mode
without "SmartRTP" (almost every codec out there can operate in RTP mode) or, alternatively,
a SIP-based mode. When the IP address of the counterpart unit is not known (mobile
connections, dynamic IP, etc) the PROXY SIP mode is recommended, as it relies on an
external server to resolve IP addresses. On the other hand, if IP and ports are well-defined, it is
better to use DIRECT SIP. Select any of the N/ACIP recommended encoding algorithms, such
as G722 or MPEG-1 /2, depending on the quality required, acceptable delay and available
bandwidth.
5.3.1. Establishing an IP communication in RTP mode using SmartRTP.
•
•
•
•
•
•
•
•
•
•
Check that the unit is ON.
Check that the RJ45 cable is connected and latched.
Check that the amber LED, integrated in the RJ45 port, is flashing.
Verify the status of the communications interface. Activate the "IP" button, which will
remain steadily illuminated. If it blinks, there is some problem with the Ethernet
connection and the unit won’t allow you to make calls.
Press the "AUTO" button and make sure that "SmartRTP" option is activated.
Establish the desired audio configuration (inputs routing, mix level, MIC4 / LINE IN
mode, and outputs).
If the coordination channel license is activated, select the channel we want to call from
by means of the "NAVI / Ch" button. The corresponding legend will get illuminated
between the vumeters (PROG or COORD).
Configure the coding algorithm according to the desired quality and network capabilities
by pressing the "CODEC" button. Choosing one of the OPUS modes included in the
unit is recommended, depending on the communication needs (voice, mono music,
stereo music) and network connection quality.
Press the "OK" button. The numeric keyboard will become illuminated in a color that
depends on the active channel: red (PROGRAM) or green (COORDINATION).
Using the "NAVI / Ch" encoder, select whether you want to call from an agenda contact
(CONTACT LIST), select one of the LAST CALLS or manually dial an IP / Port. If
manual mode is selected, go to the "IP" line and fill the numbers moving the encoder
and confirming each one by means of its pushbutton:
Alternatively, you can directly dial the IP address, always typing 3 digits per number,
and confirming each number with the "NAVI / Ch" button.
52
AEQ PHOENIX ALIO
For example: If you want to call to IP 192.26.5.12, just type:
"1","9","2", (NAVI/Ch button),"0","2","6", (NAVI/Ch button),"0","0","5",
button),"0","1","2", (NAVI/Ch button to finish).
•
•
•
•
(NAVI/Ch
Go to next line, "PORT", and select the destination port, either with the encoder (they
change in steps of 4) or typing it (always with 5 digits).
Select the "CALL" option or press the "OK" key again.
Repeat the audio and mode configuration process in the other end. As "SmartRTP"
option is active, the other codec will automatically connect as soon as it starts receiving
RTP traffic, and so there will be no need to dial IP addresses or ports, accept the
incoming call or even configuring the encoding algorithm.
The OLED screen displays the call status, as well as the destination IP and port:
o
CONNECTING: The "OK" key will blink when in this status.
o
CONNECTED / NO SYNC / NO DATA: When the call has been successfully
established, data is received but synchronization to it is not possible, or no data
at all is received, respectively. The "OK" key will remain steadily illuminated.
If status is NO SYNC or NO DATA and auto-hang up option is enabled, the call
will be rejected after the defined time, and the "OK" key illumination will turn off.
Once connected with the remote end, verify that the vumeters in Phoenix ALIO front panel show
the presence of send and received audio, and adjust levels as necessary.
5.3.1.1. Ending an IP communication in RTP mode.
•
In order to finish the communication, just press the "ESC / DEL" key for a longer time,
making sure that the currently selected channel is the one we want to cut. If, as
recommended, "SmartRTP" is activated in both codecs involved, there is no need to
repeat the hang up process in the remote end. The "ESC / DEL" key will blink red
during disconnection, and the display will show the "DISCONNECTING" status. Both
will disappear only when the call has been completely disconnected.
53
AEQ PHOENIX ALIO
5.3.2. Establishing an IP call in PROXY SIP mode.
When, for instance, due to compatibility issues with other manufacturers, SIP signaling is
required, relying on an external server to abstract from involved IP, ports, etc., the PROXY SIP
mode will be used:
•
•
•
•
•
•
•
•
•
•
•
•
Check that the unit is ON.
Check that the RJ45 cable is connected and latched.
Check that the amber LED, integrated in the RJ45 port, is flashing.
Establish the desired audio configuration (inputs routing, mix level, MIC4 / LINE IN
mode, and outputs).
If the coordination channel license is activated, select the channel we want to call from
by means of the "NAVI / Ch" button. The corresponding legend will get illuminated
between the vumeters (PROG or COORD).
Press the "SIP" key until "PROG/COORD CHANNEL - PROXY SIP MODE" is
displayed. Press the "MENU" key and select "COMMUNICATIONS". Under "SIP" menu,
verify that a valid PROVIDER is selected, a valid user is entered and the password is
correct; in case they are not, you can edit them (once all is filled in, confirmation is
provided by pressing the "NAVI / Ch" button, or cancelled with a long pressing of "ESC /
DEL" key; the cursor can be moved using the "NAV / Ch" encoder and characters are
erased with "ESC / DEL" key).
Select a valid SIP PROVIDER (that must have been previously stored in the unit using
the remote control software); the default provider is "AEQ", but up to 8 definitions can
be saved.
Verify the status of the communications. Check that the "SIP" key is steadily illuminated.
If it blinks, there is some problem with the Ethernet connection or the access or
registration on the Proxy server, and the unit won’t allow you to make any calls. If this is
the case, check connectivity and configuration under the ETHERNET, SIP and NAT
menus carefully (all of them accessibly with the "MENU" key).
Select a coding profile according to the desired quality and network capabilities by
pressing the "CODEC" button.
Establish the desired automatic options (Auto Answer, auto-hang up and permanent
call) by means of the "AUTO" button.
Press the "OK" button. The numeric keyboard will become illuminated in a color that
depends on the active channel: red (PROGRAM) or green (COORDINATION).
Using the "NAVI / Ch" encoder, select whether you want to call from an agenda contact
(CONTACT LIST), select one of the LAST CALLS or just type it in manually (URI):
When typing the URI manually, the cursor can be moved by means of the "NAVI / Ch"
encoder and characters at its left can be erased with the "ESC / DEL" key. The (*) key
shifts the writing mode between lower case (abc), capitals (ABC) or numbers (123). This
latest mode is useful to type IP addreses. In alphanumeric mode, the "1" key allows for
the insertion of several symbols (@, -, _, ., : ).
Once the URI is typed in, press the "OK" key to confirm or press the "ESC / DEL" key
for a longer time to cancel.
The URIs must be specified according to one of the following formats:
o
o
o
o
<dest_URI> i.e: phoenixMaster
<dest_URI>@<PROXY_domain> i.e: phoenixMaster@sip.aeq.es
<dest_URI>@<PROXY_IP> i.e: phoenixMaster@232.168.1.2
<dest_URI>@<PROXY_domain>:<SIP port> i.e:
phoenixMaster@sip.aeq.es:5060
54
AEQ PHOENIX ALIO
•
•
•
Select the "CALL" option or press the "OK" key again.
Accept, if necessary, the call in the other end (see 5.3.2.2).
The OLED screen displays the call status, as well as the destination URI address:
o
CONNECTING (depending on the communication interface and the network
status, this one could be an extremely brief status): The "OK" key will blink
during this time.
o
CONNECTED / NO SYNC / NO DATA: When the call has been successfully
established, data is received but synchronization to it is not possible, or no data
at all is received, respectively. The "OK" key will remain steadily illuminated.
If status is NO SYNC or NO DATA and auto-hang up option is enabled, the call
will be rejected after the defined time, and the "OK" key illumination will turn off.
Once connected with the remote end, verify that the vumeters in Phoenix ALIO front panel show
the presence of send and received audio, and adjust levels as necessary.
The LAST CALLS option displays a list with the latest successful OUTGOING or INCOMING
CALLS. Additional information (date, time…) can be obtained by pressing the "NAVI / Ch"
button on each entry. The most recent calls are displayed on top. Both lists can be deleted by
using the DELETE LAST CALLS option.
5.3.2.1. Ending an IP call in PROXY SIP mode.
•
In order to finish the communication, just press the "ESC / DEL" key for a longer time,
making sure that the currently selected channel is the one we want to cut. The "ESC /
DEL" key will blink red during disconnection, and the display will show the
"DISCONNECTING" status. Both will disappear only when the call has been completely
disconnected.
5.3.2.2. Receiving and accepting an IP call in PROXY SIP mode.
If the unit interface is correctly configured and the Autoanswer mode is not active (AUTO
AUTOANSWER OFF), when a call is received:
•
•
•
•
As opposed to "SmartRTP" mode, incoming SIP calls ARE signaled, and unless the
Autoanswer option is enabled, the user can decide whether to accept or reject the call
by means of the "OK" / "ESC/DEL" keys. The unit will emit an acoustic signal. It can,
however, be disabled under the MENU
MAINTENANCE
BUZZER menu. The
"OK" key will simultaneously blink to warn the user.
Information about the caller will appear in the OLED screen, indicating the channel
(PROGRAM or COORDINATION) where the call is coming to.
The user can accept the call by pressing the "OK" key, or reject it by pressing the "ESC
/ DEL" key (assuming that Autoanswer option is not enabled).
If the call is accepted, call status is displayed:
o
CONNECTING (depending on the communication interface and the network
status, this one could be an extremely brief status): The "OK" key will blink
during this time.
o
CONNECTED / NO SYNC / NO DATA: When the call has been successfully
established, data is received but synchronization to it is not possible, or no data
at all is received, respectively. The "OK" key will remain steadily illuminated.
55
AEQ PHOENIX ALIO
If status is NO SYNC or NO DATA and auto-hang up option is enabled, the call
will be rejected after the defined time, and the "OK" key illumination will turn off.
Once connected with the remote end, verify that the vumeters in Phoenix ALIO front panel show
the presence of send and received audio, and adjust levels as necessary.
5.3.3. Establishing an IP call in DIRECT SIP mode.
This mode is simply a variation of PROXY SIP, using the same signaling and call establishment
protocol, but without the aid of an external server, so that the destination IP needs to be known.
Note that since "SmartRTP" mode is offered in Phoenix audiocodecs, the DIRECT SIP mode
doesn’t offer obvious functional advantages, so its use will be typically relegated to cases where
operation against third-party or older units is required.
•
•
•
•
•
•
•
•
•
•
•
Check that the unit is ON.
Check that the RJ45 cable is connected and latched.
Check that the amber LED, integrated in the RJ45 port, is flashing.
Establish the desired audio configuration (inputs routing, gains, mix levels, MIC4 / LINE
IN mode, and outputs).
If the coordination channel license is activated, select the channel we want to call from
by means of the "NAVI / Ch" button. The corresponding legend will get illuminated
between the vumeters (PROG or COORD).
Press the "SIP" key until "PROG/COORD CHANNEL - DIRECT SIP MODE" is
displayed. Press the "MENU" key, select "COMMUNICATIONS" and verify that the
involved channel has been assigned a correct URI name. You can edit that name
(confirmation is provided by pressing the "NAVI / Ch" button, or cancelled with a long
pressing of "ESC / DEL" key; the cursor can be moved using the "NAV / Ch" encoder
and characters are erased with "ESC / DEL" key).
Verify the status of the communications. Check that the "SIP" key is steadily illuminated.
If it blinks, there is some problem with the Ethernet connection and the unit won’t allow
you to make any calls. If this is the case, check connectivity and configuration under the
ETHERNET and NAT menus carefully (all of them accessibly with the "MENU" key).
Select a coding profile according to the desired quality and network capabilities by
pressing the "CODEC" button.
Establish the desired automatic options (Auto Answer, auto-hang up and permanent
call) by means of the "AUTO" key.
Press the "OK" button. The numeric keyboard will become illuminated in a color that
depends on the active channel: red (PROGRAM) or green (COORDINATION).
Using the "NAVI / Ch" encoder, select whether you want to call from an agenda contact
(CONTACT LIST), select one of the LAST CALLS or just type it in manually (URI):
When typing the URI manually, the cursor can be moved by means of the "NAVI / Ch"
encoder and characters at its left can be erased with the "ESC / DEL" key. The (*) key
shifts the writing mode between lower case (abc), capitals (ABC) or numbers (123). This
latest mode is useful to type IP addreses. In alphanumeric mode, the "1" key allows for
the insertion of several symbols (@, -, _, ., : ).
Once the URI is typed in, press the "OK" key to confirm or press the "ESC / DEL" key
for a longer time to cancel.
The URIs must be specified according to one of the following formats:
56
AEQ PHOENIX ALIO
o
o
•
•
•
<dest_URI>@<dest_IP> i.e: phoenixMaster@172.26.33.15
<dest_URI>@<dest_IP>:<SIP_port> i.e:
phoenixMaster@172.26.33.15:5061
Select the "CALL" option or press the "OK" key again.
Accept, if necessary, the call in the other end (see 5.3.3.2).
The OLED screen displays the call status, as well as the destination URI address:
o
CONNECTING (depending on the communication interface and the network
status, this one could be an extremely brief status): The "OK" key will blink
during this time.
o
CONNECTED / NO SYNC / NO DATA: When the call has been successfully
established, data is received but synchronization to it is not possible, or no data
at all is received, respectively. The "OK" key will remain steadily illuminated.
If status is NO SYNC or NO DATA and auto-hang up option is enabled, the call
will be rejected after the defined time, and the "OK" key illumination will turn off.
Once connected with the remote end, verify that the vumeters in Phoenix ALIO front panel show
the presence of send and received audio, and adjust levels as necessary.
The LAST CALLS option displays a list with the latest successful OUTGOING or INCOMING
CALLS. Additional information (date, time…) can be obtained by pressing the "NAVI / Ch"
button on each entry. The most recent calls are displayed on top. Both lists can be deleted by
using the DELETE LAST CALLS option.
5.3.3.1. Ending an IP call in DIRECT SIP mode.
•
In order to finish the communication, just press the "ESC / DEL" key for a longer time,
making sure that the currently selected channel is the one we want to cut. The "ESC /
DEL" key will blink red during disconnection, and the display will show the
"DISCONNECTING" status. Both will disappear only when the call has been completely
disconnected.
5.3.3.2. Receiving and accepting an IP call in DIRECT SIP mode.
If the unit interface is correctly configured and the Autoanswer mode is not active (AUTO
AUTOANSWER OFF), when a call is received:
•
•
•
•
As opposed to "SmartRTP" mode, incoming SIP calls ARE signaled, and unless the
Autoanswer option is enabled, the user can decide whether to accept or reject the call
by means of the "OK" / "ESC/DEL" keys. The unit will emit an acoustic signal. It can,
however, be disabled under the MENU
MAINTENANCE
BUZZER menu. The
"OK" key will simultaneously blink to warn the user.
Information about the caller will appear in the OLED screen, indicating the channel
(PROGRAM or COORDINATION) where the call is coming to.
The user can accept the call by pressing the "OK" key, or reject it by pressing the "ESC
/ DEL" key (assuming that Autoanswer option is not enabled).
If the call is accepted, call status is displayed:
57
AEQ PHOENIX ALIO
o
CONNECTING (depending on the communication interface and the network
status, this one could be an extremely brief status): The "OK" key will blink
during this time.
o
CONNECTED / NO SYNC / NO DATA: When the call has been successfully
established, data is received but synchronization to it is not possible, or no data
at all is received, respectively. The "OK" key will remain steadily illuminated.
If status is NO SYNC or NO DATA and auto-hang up option is enabled, the call
will be rejected after the defined time, and the "OK" key illumination will turn off.
Once connected with the remote end, verify that the vumeters in Phoenix ALIO front panel show
the presence of send and received audio, and adjust levels as necessary.
58
AEQ PHOENIX ALIO
6. QUICK START GUIDE. REMOTE CONTROL.
In order to gain a complete knowledge of the operation of Phoenix ALIO, we recommend
reading the previous chapters and "AEQ ControlPHOENIX" user’s manual carefully. The
paragraphs below describe the basic actions you will need to take for remote operation of the
equipment by means of "AEQ ControlPHOENIX" application.
6.1. Equipment connections.
6.1.1. Power supply.
Power supply to the unit is provided by the provided AC/DC adapter unit or by means of a
homologated UPS. In any case, connection to the unit is made by means of the special latching
connector at the back as described in chapter 2.2.3 of this manual.
6.1.2. Communications interface.
Connect an Ethernet cable (CAT5 or better) finished in an RJ45 10/100 BT to the LAN
connector provided at the unit’s back panel. The selected cable must be straight when the
connection is made from the unit to a communications device (switch, router…). For more
information about the pinout of this port please check section 2.2.2 of this manual.
6.2. Turning the unit on.
•
•
•
Once the unit is connected to the power supply through its adapter, the OLED screen
will turn on after around 15 seconds*, showing AEQ logo and the audiocodec model
name.
Check that audio routing and levels are correct; if they are not, just operate the controls
to establish the desired configuration.
The unit is now ready to be remotely controlled.
* The "ESC / DEL" key (with a red telephone) will light in red for around 15 seconds when the
unit is turned on, until the welcome screens appear.
6.3. Setting up a computer to control the unit.
Connect to the same network a computer with "AEQ ControlPHOENIX" software installed with
version 2.2.1.20 or higher. Follow what’s indicated in chapter 3 of the application’s user manual.
Check that your PHOENIX ALIO is automatically discovered after launching the application,
according to chapter 4.1 of the software user’s manual. Accept the unit and, if you don’t find it,
please check that the network parameters of both the unit and the computer belong to the same
network having in mind the default IP address of the unit (IP: 192.168.1.88, Mask:
255.255.255.0, GWAY=DNS: 192.168.1.100). Keep in mind that, in order to find it with version
2.2.1.20 (or higher), the unit’s CPU version must be 5.50 or higher.
The codec is ready to be controlled when the individual codec control window appears, showing
"OK" label in PRG and CRD status areas and, below, the equipment’s given name ("ALIO1" in
the example below).
59
AEQ PHOENIX ALIO
IMPORTANT NOTE: If more new codecs will be controlled in the same network, you need to
change their IP address one by one as you add them in order to avoid conflicts in the network,
as they will also have the same default IP addresses. Go to "Configuration"
"Ethernet" to
access the dialog that allows you to change the IP parameters of the unit.
6.4. Audio.
Chapter 2 in this manual describes the physical connections present in the back and side
panels of the audiocodec in detail, but in a nutshell, this is the simplified procedure:
•
•
•
•
•
•
Connect the required microphones to inputs MIC1 to MIC4 at the back panel.
Select an adequate gain (usually around 40 dB) using each input’s configuration menu,
accessible by simply pressing the corresponding encoder button (MIC1-MIC4/LINE).
Activate Phantom supply where necessary (check the microphone manual).
If required, connect the line input source to the XLR connectors at the right side of the
unit labeled as IN (L & R).
Connect one or two headphones to the HP1 and/or HP2 outputs at the right side of the
unit.
If required, connect the line output to the XLR connectors at the right side of the unit
labeled as OUT (L & R).
6.5. Establishing an IP communication.
As explained in previous chapters, several operating modes are available, depending on the
protocol used for the communication initiation.
In order to ease the task, AEQ has developed the proprietary "SmartRTP" protocol, that allows
for the establishment of a communication without having to dial from both ends and without
worrying about the coding algorithms matching, as the calling end will provide all the necessary
signaling to its counterpart, so it knows what IP address and port to send the return audio to.
This functionality is, in some sense, similar to the one provided by DIRECT SIP, but without
renouncing to RTP inherent simplicity (all other modes are based on RTP, indeed) and without
the need for additional special control ports.
In order to activate "SmartRTP", press the "AUTO" key and enable this option.
6.5.1. Establishing an IP communication in RTP mode using SmartRTP.
•
•
•
Ensure that the equipment is powered up and controlled by the software.
Establish the appropriate audio configuration (mixer).
Check that there is incoming audio to the channel (PROG or COORD) that we are going
to use to establish the communication: the "Tx"indicator in the individual codec control
window
, in the general configuration screen
will change to green.
and in the list view
•
Go to general configuration screen
and configure "INTERFACE" as "RTP
raw".
•
Enter "I/F Setup" and select the "Local media port": local IP port through which the RTP
audio is received. Ensure that the remote unit, when calling, sends audio to that port
(see section 4.2.1).
60
AEQ PHOENIX ALIO
The same screen allows you to configure the type and size of the receiving buffer and
FEC parameters as a function of the IP network quality so we have the shortest delay
while audio cuts are minimized or eliminated in poor quality networks (see paragraph
4.4 of this manual in order to select the optimal buffer configuration depending on your
application).
•
Return to the general configuration screen, check that the selected coding algorithm in
the green "ENCODER" area corresponding to that channel (PROG or COORD) is
correct, or otherwise click on "Select codec" to change it.
•
Decide whether you will use the advanced automatic connection options or not:
o
o
o
•
"SmartRTP": Activate this option.
"Auto hang-up": Automatic hang-up whenever audio packets are missed for a given
time.
"Permanent call": The device will do what’s necessary to keep the call up, even
after a line drop or mains cut. When "SmartRTP" mode is active, it is
recommended that this option is enabled in the calling end only.
Return to the individual codec control window and click on "CALL"
button of the desired channel, showing then the call screen:
Enter the IP address of the remote unit, either manually or getting it from the buttons:
By clicking here, the last completed calls
("Calls") the contacts stored in the call book
("Eq. Contacts") or the reachable IP units
("Quick Dest.") are shown respectively, but
only those configured with formats compatible
with channel and communication type.
•
Repeat the audio and mode configuration in the other end. As "SmartRTP" is enabled,
the other codec will automatically connect as soon as it starts receiving audio traffic, so
it won’t be necessary to set the coding mode, dial IP addresses or ports, etc. nor even
accept the incoming call.
61
AEQ PHOENIX ALIO
•
•
Press the green "Call" button on the screen, to make the call.
You can monitor the status of the call on the screen:
o
o
o
o
•
•
CALLING
CONNECTING
SYNCHRONIZING
CONNECTED [-NO_DATA / -NO_SYNC]
Verify that the "SYNC" LED beneath the "CALL" button is lighted in green to indicate
that the communication has been successfully established.
Once the connection has been established with the remote end, confirm the presence
of transmitted and received audio by checking the audio presence indicators "Tx" and
"Rx".
NOTE: In order to make calls to multiple destinations, please consult section 4.2.4 of this
manual.
6.5.1.1. Ending an IP communication in RTP mode.
•
To finalize the communication, simply press the "CALL" button in the individual codec
control window and then confirm. As "SmartRTP"mode is active, it won’t be necessary
to manually terminate the call at the remote end.
6.5.2. Establishing an IP call in PROXY SIP mode.
•
•
•
Ensure that the equipment is powered up and controlled by the software.
Establish the appropriate audio configuration (mixer).
Check that there is incoming audio to the channel (PROG or COORD) that we are going
to use to establish the communication: the "Tx" indicator in the individual codec control
window
, in the general configuration screen
will change to green .
•
Go to general configuration screen
and configure "INTERFACE" as "SIP,
Proxy based".
•
Enter "I/F Setup" and click on "SIP Parameters".
and in the list view
62
AEQ PHOENIX ALIO
Check the SIP server configuration ("Proxy
Provider"). Select one that is already configured from
the list, for example the default "AEQ", or otherwise
go to "Manage Providers", create a "New provider"
and fill in the following fields: name, port, address of
the server (either its IP or URL), SIP interface and, if
necessary, check the "Register" field. In this case,
you also need to re-write the "Authentication" data in
the channel’s SIP parameters so they match those in
the new server, as the ones by default are only for
"AEQ". Note that, in case you specify the SIP Server
by its URL instead of its IP address, the DNS Server
must be correctly configured and reachable (at
"Configuration"
"Ethernet").
•
You can also select a SIP account from a previously created and stored list by means of
the drop-down menu "Proxy SIP Accounts". See section 5.1.7.1 of "AEQ
ControlPHOENIX" user’s manual.
Select the working mode to traverse NAT devices ("NAT Traversal") that is more
adequate for the network the unit is connected to.
NOTE: It is recommended that you follow Application Notes 0-A or 0-C, according to the
type of equipment’s connection.
At "I/F Setup" fill in the "Local media port" (where the unit expects to receive RTP audio
traffic at). If you enable Symmetric RTP mode, the unit will send audio to the same port
where it is receiving it from. This is sometimes useful to overcome NAT routers.
The same screen allows you to configure the type and size of the receiving buffer and
FEC parameters as a function of the IP network quality so we have the shortest delay
while audio cuts are minimized or eliminated in poor quality networks (see paragraph
4.4 of this manual in order to select the optimal buffer configuration depending on your
application).
•
Return to the general configuration screen, check that the
selected encoding profile in the green "ENCODER" area
corresponding to that channel (PROG or COORD) is
correct, or otherwise click on "Select codec" to change it.
There are several pre-defined profiles containing several
particular algorithms each one, with preference ordering. They can be edited and more
profiles can be added. The called unit will accept the call using the first coding algorithm
that it supports from the list (independently of the profile selected in that unit at that
time).
•
Decide whether you will use the advanced automatic connection options or not:
o
o
o
o
"Autoanswer": Automatic call answering for all incoming calls, or only those
corresponding to a predefined caller.
"Auto hang-up": Automatic hang-up whenever audio packets are missed for a given
time.
"Permanent call": The device will do what’s necessary to keep the call up, even
after a line drop or mains cut.
"Apply audio profile to incoming calls": allows you to filter the SIP calls depending
on the encoding profile of the receiver.
63
AEQ PHOENIX ALIO
•
Return to the individual codec control window and click on
"CALL" button corresponding to that channel, showing then the
call screen:
Enter the IP address of the remote unit, either
manually or getting it from the buttons:
By clicking here, the last completed calls ("Calls") the contacts stored in the call book
("Eq. Contacts") or the reachable IP units ("Quick Dest.") are shown respectively, but
only those configured with formats compatible with channel and communication type.
•
It is mandatory that the called unit URI is specified in any of the following formats,
adequate for Proxy SIP communications:
o
o
o
o
<equipment’s_name>, for example: "phxalio_231" or "phoenixMaster"
<equipment’s_name>@<SIP_realm>, for example:
"phxalio_231@sip.aeq.es" or phoenixMaster@sip.aeq.es"
<equipment’s_name>@<SIP_server_IP>, for example:
"phxalio_231@232.168.1.2" or "phoenixMaster@232.168.1.2" (where
232.168.1.2. is AEQ SIP Server ("sip.aeq.es") IP address )
<equipment’s_name>@<SIP_server>:<Port>, when SIP port is not 5060 (the
one used by default in SIP SERVER mode). For example:
"phxalio_231@sip.aeq.es:5061"
VERY IMPORTANT: <equipment’s_name> must not be longer than 19 characters.
•
Press the green "Call" button on the screen, to make the call.
•
Answer the call on the other unit, if "Autoanswer" option is not active.
•
You can observe the changes of status from OK to CONNECTED and synchronized:
64
AEQ PHOENIX ALIO
•
Verify that the "SYNC" LED beneath the "CALL" button corresponding to that channel
is lighted in green to indicate that the communication has been successfully
established.
•
Send audio from one equipment to another verifying that the "Tx" and "Rx"
audio presence indicators change to green.
•
If the unit is registering in SIP server and the call is being made but no audio comes
through, please check "NAT TRAVERSAL" configuration (see section 4.3).
6.5.2.1. Ending an IP call in PROXY SIP mode.
•
In order to hang call, just click on "CALL" button corresponding to that channel in one of
both codecs’ individual control window. A confirmation message will appear and the call
will be disconnected after acceptance.
6.5.2.2. Receiving and accepting an IP call in PROXY SIP mode.
If the IP interface is correctly configured and automatic answer mode is OFF, when you receive
a call:
•
•
•
•
•
•
•
The unit and the application will provide acoustic warning. This can be disabled (for the
unit) at "Configuration"
"Miscellaneous"
"Buzzer and test".
The "CALL" button red LED corresponding to that channel at the individual codec
control window in the remote control software corresponding to the unit that is receiving
a call will blink at the same time to warn the user.
In addition, if "Autoanswer" option is not active, an incoming call window will appear
showing the URI identifier of the caller unit:
The call will be accepted by clicking on the individual codec control window "CALL"
button corresponding to that channel or alternatively on the "Accept" button in the
incoming calls window.
The screen will show the status of the call:
o CONNECTING
o SYNCHRONIZING
o CONNECTED [-NO_SYNC / -NO_DATA]
Verify that the "SYNC" LED beneath the "CALL" button corresponding to that channel is
lighted in green to indicate that the communication has been successfully established.
Once connected to the remote end, verify that the "Tx" and "Rx" audio presence
indicators change to green.
65
AEQ PHOENIX ALIO
6.5.3. Establishing an IP call in DIRECT SIP mode.
•
•
•
Ensure that the equipment is powered up and controlled by the software.
Establish the appropriate audio configuration (mixer)
Check that there is incoming audio to the channel (PROG or COORD) we are going to
use to establish the communication: the "Tx" indicator in the individual codec control
window
, in the general configuration screen
will change to green .
and in the list view
•
Go to general configuration screen
and configure "INTERFACE" as
"DIRECT SIP".
•
Enter "I/F Setup" and click on "SIP Parameters". Check that "User Name" and "Display
name" are configured. User name and IP address constitute the equipment’s required
connection information.
•
Select the working mode to traverse NAT devices ("NAT Traversal") that is more
adequate for the network the unit is connected to.
NOTE: It is recommended that you follow Application Notes 0-A or 0-C, according to the
type of equipment’s connection.
•
At "I/F Setup" fill in the "Local media port" (where the unit expects to receive RTP audio
traffic at). If you enable Symmetric RTP mode, the unit will send audio to the same port
where it is receiving it from. This is sometimes useful to overcome NAT routers.
The same screen allows you to configure the type and size of the receiving buffer and
FEC parameters as a function of the IP network quality so we have the shortest delay
while audio cuts are minimized or eliminated in poor quality networks (see paragraph
4.4 of this manual in order to select the optimal buffer configuration depending on your
application).
66
AEQ PHOENIX ALIO
•
Return to the general configuration screen, check that the
selected encoding profile in the green "ENCODER" area
corresponding to that channel (PROG or COORD) is
correct, or otherwise click on "Select codec" to change it.
There are several pre-defined profiles containing several
particular algorithms each one, with preference ordering. They can be edited and more
profiles can be added. The called unit will accept the call using the first coding algorithm
that it supports from the list (independently of the profile selected in that unit at that
time).
•
Decide whether you will use the advanced automatic connection options or not:
o
o
o
o
•
"Autoanswer": Automatic call answering for all incoming calls, or only those
corresponding to a predefined caller.
"Auto hang-up": Automatic hang-up whenever audio packets are missed for a
given time.
"Permanent call": The device will do what’s necessary to keep the call up, even
after a line drop or mains cut.
"Apply audio profile to incoming calls": allows you to filter the SIP calls depending
on the encoding profile of the receiver.
Return to the individual codec control window and click on
"CALL" button corresponding to that channel, showing then the
call screen:
Enter the URI of the remote unit, either manually
or getting it from the buttons:
By clicking here, the last completed calls ("Calls") the contacts stored in the call book
("Eq. Contacts") or the reachable IP units ("Quick Dest.") are shown respectively, but
only those configured with formats compatible with channel and communication type.
67
AEQ PHOENIX ALIO
•
It is mandatory that the called unit URI is specified in any of the following formats,
adequate for Direct SIP communications:
"<equipment’s_name>@<equipment’s_IP_address>" (for example:
phxalio_11@192.168.1.89).
If the SIP port of the other end is not 5060 (standard SIP port), the identifier must
include the port in use:
For example: "phxalio_11@192.168.1.89:5061".
VERY IMPORTANT: <equipment’s_name> must not be longer than 19 characters
•
•
•
Press the green "Call" button on the screen, to make the call.
Answer the call on the other unit, if "Autoanswer" option is not active.
You can observe the changes of status from OK to CONNECTED and synchronized
when calling from PROGRAM:
•
Verify that the "SYNC" LED beneath the "CALL" button is lighted in green to indicate
that the communication has been successfully established.
•
Send audio from one equipment to another verifying that the "Tx" and "Rx"
audio presence indicators change to green.
6.5.3.1. Ending an IP call in DIRECT SIP mode.
•
In order to hang call, just click on "CALL" button of the connected channel in one of
both codecs. A confirmation message will appear and the call will be disconnected after
acceptance.
6.5.3.2. Receiving and accepting an IP call in DIRECT SIP mode.
If the IP interface is correctly configured and automatic answer mode is OFF, when you receive
a call:
•
•
•
The unit and the application will provide acoustic warning. This can be disabled (for the
unit) at "Configuration"
"Miscellaneous"
"Buzzer and test".
The "CALL" button red LED of the called channel at the individual codec control window
in the remote control software corresponding to the unit that is receiving a call will blink
at the same time to warn the user.
In addition, if "Autoanswer" option is not active, an incoming call window will appear
showing the URI identifier of the caller unit:
68
AEQ PHOENIX ALIO
•
•
•
•
The call will be accepted by clicking on the individual codec control window "CALL"
button of the called channel or, alternatively, on the "Accept" button in the incoming
calls window.
The screen will show the status of the call:
o CONNECTING
o SYNCHRONIZING
o CONNECTED [-NO_SYNC / -NO_DATA]
Verify that the "SYNC" LED beneath the "CALL" button corresponding to that channel is
lighted in green to indicate that the communication has been successfully established.
Send audio from one equipment to another verifying that the "Tx" and "Rx" audio
presence indicators change to green.
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AEQ PHOENIX ALIO
7. CONTROL TERMINAL OVER WEB BROWSER.
PHOENIX ALIO audiocodec includes a WebServer that enables you to execute numerous
functions remotely over the Ethernet interface included in the back panel of the unit by means of
a standard web browser (compatibility is guaranteed with Internet Explorer running on Microsoft
Windows operating system).
7.1. Upgrading system firmware.
PHOENIX ALIO is supplied from factory with the latest firmware versions available. However,
firmware versions with new features may be released in the future, making it necessary to
upgrade the equipment to be able to make use of these new functionalities.
Because the upgrading process must be handled with caution, we recommend having it done
by an authorized distributor, or under the instructions of the AEQ Technical Assistance Service.
If questions or problems arise, please consult via electronic mail (sat@aeq.es).
IMPORTANT NOTE: If the CPU of the equipment is upgraded, configuration of the unit, and in
particular IP configuration won’t be modified unless expressly stated by AEQ SAT. In that case,
the user should take note of all important codec parameters before upgrading in order to
reconfigure them afterwards.
The entire PHOENIX ALIO firmware versions upgrading process is done through the IP
interface of the unit, with the aid of a standard web browser (compatibility is guaranteed with
Internet Explorer running on Microsoft Windows operating system). By default, user and
password is "aeq".
IMPORTANT NOTE: the recommended order for upgrading is: MICRO (CPU), DSP and
FPGA. The process is iterative.
To upgrade the firmware, you must follow the steps detailed below:
1. From MENU
ETHERNET, check the IP address associated with the Ethernet
interface.
2. Power down the PHOENIX ALIO.
3. Connect PHOENIX ALIO to the PC from which you are going to perform the upgrading
process using a crossed cable.
4. Power up the PHOENIX ALIO.
5. Open the Internet Explorer web browser and, in the address bar, enter HTTP://<IP
address obtained in point 1>. Press ENTER and the main screen will be displayed.
Main screen detail
6. To upgrade the codec, click on the UPGRADE option.
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AEQ PHOENIX ALIO
7. A user ID and password are requested (by default, both are aeq). After you have
correctly entered these two items, the firmware upgrading screen will be displayed.
Firmware upgrading screen detail
8. Check to see whether the versions displayed are the same as the firmware that is
currently in effect. If they do not match, upgrade the firmware as indicated below.
9. Select the module you want to upgrade in "Upgrade" column. NOTE: Each upgrading
file is specifically designed to upgrade a specific module within the unit: CPU, DSP or
FPGA.
10. In "Select file" enter the access route to the upgrade file containing the new firmware
version, using the "Seleccionar archivo" button.
11. Press the "Upgrade" button in the lower part of the screen.
12. Wait for on-screen confirmation that the operation has been successfully completed.
13. In the Internet browser, go to the UPGRADE section and ensure that all the firmware
versions installed in your codec are now the correct ones.
14. Power the unit down.
7.2. Configuring the MAC address associated with the Ethernet interface.
From this menu the MAC address associated with the Ethernet interface can be edited;
because of the consequences this action could have, the addresses should only be edited if the
codec use situation requires it. The editing should be performed by highly qualified personnel or
under the supervision of AEQ authorized technical services, and always in possession of the
required network information.
ETHERNET, check the IP address associated with the Ethernet
1. From MENU
interface.
2. Power down the PHOENIX ALIO.
3. Connect PHOENIX ALIO to the PC from which you are going to perform the upgrading
process using a crossed cable.
4. Power up the PHOENIX ALIO.
5. Open the Internet Explorer web browser and, in the address bar, enter HTTP://<IP
address obtained in point 1>. Press ENTER and the main screen will be displayed.
Selecting the MAINTENANCE option will enable you to modify the MAC address of the
Ethernet interface of the unit.
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AEQ PHOENIX ALIO
MAC change screen detail
6. Modify the value in the MAC field associated with the desired Ethernet interface.
7. Press the "Apply" button.
8. In the Internet browser, go to the MAINTENANCE section and ensure that the MAC
address is now the correct one.
9. Power the unit down.
7.3. Technical Assistance Service and on-line manuals.
Clicking on the "Support" tab in the upper part of the screen will take you to AEQ website,
where you will find all the information you need to directly contact the AEQ Technical
Assistance Service, as well as all the technical information and manuals regarding the unit.
7.4. Saving and loading configurations.
In the MAINTENANCE section, in the lower part of the screen, you will see the
CONFIGURATION MEMORY option, from which you can save the current configuration of the
unit (by means of "DOWNLOAD" button) or load a configuration previously created and saved
(by selecting the corresponding file and pressing then the "Save configuration" button). The
extension of the files used in this process is .AFU. The "Reset configuration" button allows you
to restore the default configuration of the unit.
Configurations screen detail
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AEQ PHOENIX ALIO
7.5. Screen capture.
The unit allows you to capture the screen displayed in the multifunction screen at any moment
in order to save it into a PC in BMP image format. This function may be useful in order to
prepare documents or report problems or doubts to SAT.
In order to capture a screen, open the web browser and write the following text in address bar:
http://<unit_IP_address>/GrabScreen
A new screen will appear showing the image. Place the pointer on the image, press the right
mouse button and select "Save Image as" (the message may vary depending on the web
browser). The saved image has a resolution of 132x68 pixels, 4 bit/pixel.
Screen capture detail
7.6. Status menu.
By means of IP Status menu you can monitor some statistical parameters regarding the
connection status of IP channels. Some of these parameters are: transmission and reception
buffers status, Jitter, lost packets… Channel 0 corresponds to PROGRAM and Channel 1
corresponds to COORDINATION.
IP Status screen detail
7.7. SNMP.
This unit can be remotely managed using SNMP (Simple Network Management Protocol) using
one of the many client pieces of software available in the market, even for free. SNMP allows
monitoring of the status of several pieces of equipment from very diverse manufacturers and
natures, as well as elaborating reports, generate e-mail alerts, etc.
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AEQ PHOENIX ALIO
In order to add and equipment to the list of units managed by the client, it is necessary to have
access to its "MIB" file (Management Information Base), that describes its SNMP capabilities
(alarms it can generate, accepted commands, manufacturer information, etc.).
The MIB file corresponding to the unit can be downloaded from the Web interface without
installation of any additional software. In order to do so, in the MAINTENANCE section, you can
access the link "Download MIB" under the "SNMP" section.
SNMP screen detail
If you follow that link, the text file will appear. Now you just need to right-click on it and select
"Save as…" and browse a suitable destination folder (see the manual of the selected SNMP
client).
For more information, please consult section 4.7 of this manual and section 6.5.1 of "AEQ
ControlPHOENIX" application manual.
7.8. Remotely rebooting the equipment.
Sometimes a remote unit can work abnormally or become unreachable by the remote control
software as a result of a communications error or in the unit itself. A method has been
developed in order to remotely reboot the audiocodec, so normal operation is recovered.
Inside MAINTENANCE, at the bottom of the screen, you can find "SYSTEM REBOOT" section.
By clicking on "Reboot" button, an information dialog will appear warning that the equipment is
being rebooted and it will be disconnect for some seconds after acceptance ("Resetting
system... please wait a few seconds before connecting again").
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AEQ PHOENIX ALIO
8. TECHNICAL SPECIFICATIONS *
Audio input and output
4 dynamic or condenser MIC inputs
Commutable Phantom 12V 10mA supply. Dynamic
range: >90 dB.
2 x female XLR. 9 Kohm input impedance.
Electronically balanced. Line level.
2 x male XLR. Output impedance < 100 ohm.
Electronically balanced. Line level.
2 ¼" stereo jacks for high or low impedance
headphones with volume control
Stereo analog line input
Stereo analog line output
Headphone output
Audio characteristics
Nominal output level
Max output level
Max input level
Output impedance
Distortion
+0 dBV
+20 dB above nominal, THD < 0.5%
+20 dBV (+22 dBu)
< 100 ohm
< 0.03 % (linear loop)
Dynamic range
Crosstalk
Binary rate
Operating modes
Frequency response
A/D & D/A conversion
Analog I/O: A/D & D/A converter type
Communications interfaces
IP interface
Satellite links
3G/4G modem
Backup, permanent call, etc.
Coding algorithms
Control and data interface
LAN 10 / 100 base T
General characteristics
Weight
Dimensions
Display
(depends on chose encoding algorithm)
> 100 dB (linear, no compression)
< -70 dB
20 to 2304 Kbps
Mono, stereo, Dual, Jstereo, MS stereo
+/-0.5 dB @ 20 Hz-20 KHz
24 bit Sigma-Delta 48 KHz sampling frequency
Ethernet RJ45 10/100Mbps port
An external satellite terminal can be connected to the
IP interface (see application note AN2)
An external 3G/4G USB modem can be connected to
the IP interface by means of an homologated
portable router (see application note AN5)
Menu selectable
OPUS, G.711, G.722, MPEG Layer 2, PCM…
(See APPENDIX A)
RJ45 connector
2 Kgs
W x D x H (mm): 210 x 240 x 60
1 x OLED 128x64 pixels. 16 grayscale levels.
2 x stereo vumeters, 20 multi-color LEDs
* Characteristics are subject to changes without previous notice.
Safety regulations
CE Marking.
Electromagnetic compatibility according to EU directives EN 50081-1, EN 50052-2.
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AEQ PHOENIX ALIO
9. A.E.Q. WARRANTY.
AEQ warrants that this product has been designed and manufactured under a certified Quality
Assurance System. AEQ therefore warrants that the necessary test protocols to assure the
proper operation and the specified technical characteristics of the product have been followed
and accomplished.
This includes that the general protocols for design and production and the particular ones for
this product are conveniently documented.
1.- The present guarantee does not exclude or limit in any way any legally recognized right of
the client.
2.- The period of guarantee is defined to be twelve natural months starting from the date of
purchase of the product by the first client. To be able to apply to the established in this
guarantee, it is compulsory condition to inform the authorized distributor or –to its effect- an
AEQ Sales office or the Technical Service of AEQ within thirty days of the appearance of the
defect and within the period of guarantee, as well as to facilitate a copy of the purchase invoice
and serial number of the product.
It will be equally necessary the previous and expressed conformity from the AEQ Technical
Service for the shipment to AEQ of products for their repair or substitution in application of the
present guarantee.
As a consequence, returns of equipment that does not comply with these conditions will not be
accepted.
3.- AEQ will at its own cost repair the faulty product once returned, including the necessary
labour to carry out such repair, whenever the failure is caused by defects of the materials,
design or workmanship. The repair will be carried out in any of the AEQ authorized Technical
Service Centres. This guarantee does not include the freight charges of the product to or from
such Authorized Technical Service Centre.
4.- No Extension of the Guarantee Period for repaired product shall be applied. Nor shall a
Substituted Products in application of this Guarantee be subject to Guarantee Period Extension.
5.- The present guarantee will not be applicable in the following situations:
Improper use or Contrary use of the product as per the User or Instruction Manual; violent
manipulation; exhibition to humidity or extreme thermal or environmental conditions or sudden
changes of such conditions; electrical discharges or lightning; oxidation; modifications or not
authorized connections; repairs or non-authorized disassembly of the product; spill of liquids or
chemical products.
6.- Under no circumstances, whether based upon this Limited Guarantee or otherwise, shall
AEQ, S.A. be liable for incidental, special, or consequential damages derived from the use or
from the impossibility of using the product.
AEQ shall not be liable for loss of information in the disks or data support that have been altered
or found to be inexact, neither for any accidental damage caused by the user or other persons
manipulating the product.
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AEQ PHOENIX ALIO
APPENDIX A: General characteristics of encoding modes.
OPUS:
OPUS is a completely open and very versatile coding algorithm. Its performance is unrivaled for
voice and audio transmission. It was standarized by Internet Engine Engineering Task Force
(IETF) as RFC 6716, and combines Skype’s SILK codec technology with Xiph.Org's CELT.
This algorithm allows for an excellent audio quality with high compression rate and very low
delay. Phoenix family audiocodecs feature 7 selected OPUS modes covering nearly every
transmission need, from voice to high quality stereo music, with bitrates between 12 and 192
Kbps and audio bandwidth between 6 and 20 KHz. The receiver can automatically adapt to the
particular OPUS mode selected in the transmitting end.
G.711:
ITU encoding standard for processing audio signals in the human voice frequency band,
through the compression of digital audio samples obtained at 8 KHz, and typically used in
telephone systems.
Bandwidth: 3.5 KHz
For further information on this subject, consult:
http://www.itu.int/rec/T-REC-G.711/e
G.722:
ITU encoding standard, based on ADPCM algorithms, for processing audio signals in the
human voice frequency band, through the compression of digital audio samples obtained at 16
KHz, for greater audio quality and clarity.
This is the internationally accepted mode for two-way communication because of its low delay,
which is why it is the most used standard in commentator and sports broadcasting applications.
Bandwidth: 7 KHz.
For further information on this subject, consult:
http://www.itu.int/rec/T-REC-G.722/e
MPEG LAYER II:
Well known, widely accepted encoding mode that is used when the delay is not important, since
MPEG modes always have a greater delay than G.722 modes. There are 64 Kbps encoding
modes with sampling rates of 48, 32 or 24 KHz, and 128 Kbps encoding modes with sampling
rates of 32 and 48 KHz.
Bandwidth: 10 KHz to 15 KHz.
For further information on this subject, consult: ISO/IEC 11172-3 and ISO/IEC 13818-3.
AEQ LD+:
AEQ proprietary mode (based on the previous AEQ-LD Extend mode) that combines the low
delay offered by G.722 with the greater bandwidth of the MPEG modes, optimizing these two
aspects.
PCM (12/16/20/24 bits):
Linear audio without any compression process.
For further information on this subject, consult
http://www.digitalpreservation.gov/formats/fdd/fdd000016.shtml
Other different encoding modes can be taken into account according to specific needs of each
client.
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AEQ PHOENIX ALIO
Available coding algorithms list in Phoenix ALIO. Please contact us to check availability of other
algorithms.
CODEC
RATE (Kbps)
Fs (KHz)
Mode
OPUS
12
20
32
64
64
128
192
48
48
48
48
48
48
48
VOICE RB
VOICE
MUSIC MONO RB
MUSIC MONO
MUSIC STEREO RB
MUSIC STEREO
MUSIC STEREO HQ
Approx audio BW
(KHz)
6
8
20
20
20
20
20
G.711 A-law
64
8
MONO
4
G.711 µ-law
64
8
MONO
4
G.722
64
16
MONO
7.5
64
128
192
128
256
384
64
128
192
256
384
64
128
384 a 768
768 a 1536
576 a 1152
1152 a 2304
16
32
48
16
32
48
32 / 48
32 / 48
32 / 48
32 / 48
48
16 / 24
24
32
32
48
48
MONO
MONO
MONO
STEREO
STEREO
STEREO
MONO
M / ST /JST
M / ST / JST / D
ST
STEREO
MONO
MONO
MONO
STEREO
MONO
STEREO
7.5
15
19
7.5
15
19
10.5
10.5/16.5
10.5/16.5
16.5
16.5
8/12
12
16
16
20
20
AEQ-LD
MPEG-1 Layer
II
MPEG-2 Layer
II
PCM
(12,16,20,24
bit/sample)
Other different encoding modes can be taken into account according to specific needs of each
client.
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AEQ PHOENIX ALIO
APPENDIX B: Protocols associated with IP communications.
Communication over IP networks differs notably from the communications traditionally used to
date in broadcast environments, whether they are POTS or ISDN, in that IP networks do not
have dedicated resources or qualities of service implemented in most systems, with the
associated problems this involves in terms of communication signaling, establishment,
maintenance and cleardown.
This set of problems originates in the technical characteristics that are intrinsic to the definition
and operation of communications systems based on IP protocols. The EBU-TECH 3326
standard developed by the N/ACIP working group provides certain tools for attempting to
simplify work by making use of many protocols associated with IP communication, and which
will be described below.
N/ACIP
• Signaling, understood as connection initiating and ending procedures, as well as
negotiation of connection parameters (encoding algorithms, ports, etc.)
o SDP (Session Description Protocol) to describe the parameters of the
connection
o SAP (Session Announcement Protocol) for multicast type unidirectional links
o SIP (Session Initiation Protocol) simulates the working system in traditional
telephone networks
• Transport: defines the transport protocols over IP networks
o RTP (Real-Time Transport Protocol) over UDP and IPv4
o RTCP (Real-Time Control Transport Protocol) for synchronization and active
retrieval functions
o IP ports defined: 5004 / 5008 (RTP for PROG / COORD, respectively) and 5005
/ 5009 (RTCP)
While this appendix is not intended to be a reference document for all the relevant technical
matters, it should at least serve to give its readers an initial contact with these subjects that will
ease the assimilation of the new working method over IP networks for the Phoenix ALIO user
and, as a result, the use of this equipment. The user interested in expanding his or her
knowledge of some or all of these subjects is encouraged to turn to the extensive, excellent
technical material currently available regarding the IP realm and the technologies associated
with it.
B1. Circuit switching versus packet switching.
The communications systems traditionally used in the broadcast environment for applications
with portable codecs have been mostly telephone or ISDN networks—that is, circuit switching
networks; Phoenix ALIO, on the other hand, uses a packet switching network in its IP interface.
B1.1. Circuit switching.
In a circuit switching network, the switching equipment must establish a physical path between
the communication media prior to the connection between users. This path remains active
during the communication between the users, and is cleared down or released when the
communication ends. Example: Switched telephone network.
Its operation passes through the following stages: request, establishment, file transfer and
connection cleardown.
B1.1.1. Advantages:
•
•
The transmission is made in real time.
Dedicated resources. The nodes that are involved in the communication use the
established circuit exclusively as long as the session lasts.
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AEQ PHOENIX ALIO
•
•
•
Once the circuit has been established, the parties can communicate with each other at
the highest speed that the medium allows, without having to share the bandwidth nor
the use time.
The circuit is fixed. Because a physical circuit is specifically dedicated to the
communication session in question, once the circuit is established there are no losses
of time for calculation and decision-making regarding routing through the intermediate
nodes. Each intermediate node has a single route for the incoming and outgoing
packets that belong to a specific session, which means it is impossible for the packets
to be disordered.
Simplicity in the management of intermediate nodes. Once the physical circuit has been
established, no further decisions need to be made to route the data from origin to
destination.
B1.1.2. Disadvantages:
•
•
•
•
Delay in initiating communication. A time interval is required to make the connection,
which entails a delay in the transmission of the information.
Blockage of resources. No use is made of the circuit during the moments when there is
no transmission between the parties. Bandwidth is wasted while the parties are not
communicating with each other.
The circuit is fixed. The communication route is not readjusted; it is not adapted at each
opportunity to the least costly path between the nodes. Once the circuit has been
established, no use is made of the alternative, less expensive pathways that may
become available during the session.
Poor fault tolerance. If an intermediate node fails, the entire circuit crashes. The
connections then have to be re-established from zero.
B1.2. Packet switching.
The sender divides the message to be sent into an arbitrary number of packets of the same
size, to which a header and the originating and destination addresses are added, as well as
control data that will then be transmitted through different communication media between
temporary nodes until they reach their destination. This switching method is the one that is used
in today's IP networks. It has emerged to optimize transmission capacity through existing lines.
The temporary nodes store the packets in queues in their memories, which need not be very
large.
B1.2.1. Switching modes.
•
•
Virtual circuit: Each packet is routed through the same virtual circuit as the preceding
ones. Therefore the order of arrival of the packets to their destination is controlled and
ensured.
Datagram: Each packet is routed independently from the rest. Thus the network cannot
control the path followed by the packets, nor ensure the order in which they reach their
destination.
B1.2.2. Advantages:
•
•
•
•
In case of error in a packet, only that packet will be resent, without affecting other
packets that arrived without errors.
Interactive communication. Limiting the maximum packet size ensures that no user can
monopolize a transmission line for very long (microseconds), which means that packet
switching networks can handle interactive traffic.
Packet switching increases network flexibility and profitability.
The pathway a communication takes can be altered from one moment to the next (for
example, in case one or more of the routers breaks down).
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AEQ PHOENIX ALIO
•
Theoretically, priorities can be assigned to the packets in a given communication. Thus,
a node can select, from its queue of packets waiting to be transmitted, the ones that
have higher priority.
B1.2.3. Disadvantages:
•
•
•
•
Greater complexity of the intermediate switching devices, which need to have higher
speed and greater calculating capacity to determine the appropriate route for each
packet.
Packet duplication. If a packet takes too long to reach its destination, the receiving
device may conclude that it has been lost, in which case it will send a packet
retransmission request to the sender, which gives rise to the arrival of duplicate
packets.
If the routing calculations account for an appreciable percentage of the transmission
time, the channel throughput (useful information / transmitted information) decreases.
Variations in the mean transit delay of a packet in the network. Parameter known as
jitter.
B2. IP protocol.
The Internet Protocol (IP) is a non-connection oriented protocol used both by the origin and the
destination in data transmission over a switched packet network.
The data in an IP-based network are sent in blocks known as packets or datagrams (in the IP
protocol these terms are used interchangeably). In particular, in IP there is no need for
configuration before a device attempts to send packets to another with which it has not
communicated previously.
The Internet Protocol provides an unreliable datagram service called UDP (User Datagram
Protocol), also known as "best effort", a phrase that expresses good intentions but offers few
guarantees. IP does not offer any mechanism to determine whether a packet reaches its
destination, and only provides security (by means of checksums) to cover its headers, and not
the transmitted data. For example, since it gives no guarantee that the packet will reach its
destination, it could arrive damaged, in the wrong order with respect to other packets,
duplicated, or simply not arrive. If reliability is needed, it is provided by transport layer protocols
such as TCP (Transport Control Protocol).
Reliability over TCP is obtained through the use of retransmissions. Real-time applications such
as an audio link, with the timing requirements inherent in the information contained in the link,
do not offer any useful guarantee. Since the data that are not received, and whose
retransmission is requested of the sender by the receiver, will in most cases arrive out of order,
they will end up as useless information that will have served only to overload the network. For
all these reasons, the protocol selected to serve as a communication substrate in real-time
applications is UDP.
UDPDatagram
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AEQ PHOENIX ALIO
Transport over IP protocols, independently of the reliability they offer, add new functionalities to
the basic ones provided by IP, such as packet numbering to facilitate, on the receiving end, the
detection of losses (although not their correction) and of disorder in the information received;
and the advent of the port concept as an identifier of different logic connections over the same
IP interface.
For complete information on IP protocol, we recommend consulting:
http://tools.ietf.org/html/rfc791
http://www.iana.org/assignments/port-numbers
B2.1. IP addressing.
An IP address is a number that logically and hierarchically identifies an interface of a device in
a network that uses the IP protocol. The format used is X.X.X.X, where each X represents a
group of eight bits translated into decimal form—that is, whose minimum value is 0.0.0.0 and
whose maximum value is 255.255.255.255.
IP addresses are classified in two major groups: static and dynamic.
• It is typical for a user to connect to the Internet from his or her home using an IP
address. This address may change when the user reconnects, and this manner of
assigning IP addresses is called a dynamic IP address (normally abbreviated as
dynamic IP).
• The Internet sites that, by nature, need to be continuously connected generally have a
static IP address (as with the dynamic address, a similar abbreviated form is used:
static or fixed IP)—that is, an address that does not change over time.
Another possible IP address classification can be made according to address validity:
•
•
Public: IP addresses that are valid in the entire Internet network. Currently, due to the
poor management that has traditionally been applied to the available IP addresses, they
are a scarce, highly costly resource.
Private: addresses that are only valid in a closed section of the IP network, typically
corporate and not subject to free access, with only one point of connection to the
Internet, called a gateway, constituted by a router.
B2.2. Unicast vs. Multicast.
Unicast is the transmission of information from a single sender to a single receiver. It is
distinguished from multicast (transmission to certain specific recipients—more than one—from a
single sender), broadcast (in which the recipients are all the stations in the network) and
anycast (transmission to a single recipient—any unspecified recipient).
The unicast method is the one currently being used on the Internet, and is applied for both live
and on-demand transmissions. The multicast method can only be used in corporate
environments, despite some isolated efforts to introduce it on the Internet, and is applied only
for live transmissions.
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AEQ PHOENIX ALIO
Graphical comparison: Unicast vs. Multicast
The effect that unicast transmission has on network resources is accumulative consumption.
Each user who connects to a multimedia transmission consumes as many kilobits per second
as the content encoding will permit.
B3. RTP protocol.
RTP are the initials of Real-time Transport Protocol. It is a transport level protocol used for the
transmission of information in real time, as occurs with audio and video. Normally it is paired
with RTCP (RTP Control Protocol) and is located on UDP.
The IP ports defined for its use are 5004 (RTP) and 5005 (RTCP) for PROG and 5008/5009
for COORD.
The functions of the RTP/RTCP protocol are:
•
•
•
Management of the reception buffer in order to minimize the jitter effect introduced by
the network.
Recovery of the reference clock based on information inserted by the transmitting
equipment.
Test tools to permit the user to verify the bandwidth, the delay and estimated jitter in the
connection.
RTP Header
For complete information on RTP/RTCP protocol, we recommend consulting:
http://tools.ietf.org/html/rfc1889
http://tools.ietf.org/html/rfc1890
http://tools.ietf.org/html/rfc3550
http://tools.ietf.org/html/rfc3551
http://tools.ietf.org/html/rfc3711
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AEQ PHOENIX ALIO
B3.1. Default PHOENIX ALIO configuration.
PHOENIX ALIO is an IP audiocodec that operates by using RTP over UDP in IP version 4.
By default, PHOENIX ALIO is supplied from the factory with the following IP ports defined: 5004
for RTP and the next one (5005 in this case) for RTCP, for the PROGRAM channel, and 5008 +
5009 for the COORDINATION channel (if licensed).
The RTP port values can be modified from its internal menu, and RTCP ports will be
automatically assigned accordingly.
B4. SIP protocol.
Session Initiation Protocol (SIP) is a protocol developed by the IETF MMUSIC Working Group
with the intention of establishing the standard for initiating, modifying and ending interactive
user sessions involving multimedia elements such as video, voice and instant messaging.
SIP is used simply to initiate and terminate voice and video calls. Once the communication is
established, the exchange of voice / video information is conducted only over RTP.
One of the objectives of SIP was to contribute a set of processing functions to apply to calls and
capacities present in the public switched telephone network. Thus, it implemented typical
functions that a common telephone terminal offers, such as: calling a number, making a
telephone ring when called, hearing a dial tone or busy tone. The implementation and
terminology in SIP are different.
SIP requires proxy servers and register elements to give a practical service. Although two SIP
terminals can communicate with each other without the mediation of SIP infrastructures through
the use of URIs of the name@IP-address type (which is why SIP is defined as a point-to-point
protocol), this approach is impracticable for a public service because of the problems inherent in
IP addressing, where obtaining static public addresses is nearly impossible and extremely
costly.
To simplify the operation of the unit, AEQ offers, at no additional cost, the services of its
2 own SIP servers (one of them working as main server and the other one as backup
server), although it cannot guarantee its operation 100% of the time, nor be held responsible for
the inconveniences that this may produce for the end user. The unit leaves the factory
preconfigured with the parameters required to work with the resources of any of these 2 SIP
servers.
SIP makes use of elements called proxy servers to help route the requests toward the user’s
current location, authenticate users to give them service, enable call routing policies to be
implemented, and contribute added capabilities to the user.
SIP also contributes register functions that enable the user to inform the proxy servers of his or
her location.
For complete information on the SIP protocol, we recommend consulting:
http://tools.ietf.org/html/rfc3261
B4.1. Working modes
With the PROXY SIP option activated in the Phoenix ALIO, when the unit is started up it will
automatically connect and register itself in the SIP Proxy server configured in its memory,
indicating its name (URI: name@domain) and position (IP address).
To establish any communication, the unit that wishes to establish the connection will search the
SIP Proxy server register for the information regarding the called device and will redirect the
call—in a way that is transparent to the user—toward the real physical place where the device is
located.
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SIP protocol operation diagram. Phase 1: Registration. Phase 2: Search for the called device in the SIP
server database. Phase 3: Establishment of the connection
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This working method, supported by external SIP servers, enables the physical position of a
device to be made independent from its logic identifier and, through the use of the SIP protocol,
makes it unnecessary to know more data regarding the called device than its URI.
During the establishment of the communication phase, the encoding algorithm is negotiated
simultaneously, based on the Link Profiles / SIP Codec Profiles defined in each of the devices
at the two ends of the connection circuit.
B4.2. Possible work scenarios.
Depending on the type of network to which the PHOENIX ALIO is connected, the codec will
have one or another type of IP address available to it.
•
•
•
Static public IP addresses offer the ideal situation, since they guarantee that the IP
interface of the codec will always be assigned to a fixed address (regardless of whether
it is turned off and then powered up again) and directly accessible to the rest of the
network users. Phoenix ALIO operates perfectly with an associated SIP server and
equipment identifiers of the name@domain type (PROXY SIP), and even without an
associated SIP server with a URI of the name@IP-address type (DIRECT SIP) if the
device on the opposite end of the communication circuit also has an IP address of the
same type.
This situation corresponds to use an Internet access by means of a single workstation
router (just one piece of equipment connected) and to hire a fixed IP.
Dynamic public IP addresses, corresponding to use an Internet access by means of a
single workstation router and a dynamic IP (the most usual). Allows the use of URIs of
the name@domain (PROXY SIP) or name@IP-address (DIRECT SIP) type, but it is
advisable always to work with an associated SIP server (PROXY SIP), since the IP
address assigned to the equipment may change each time the user powers up the unit.
Private IP addresses, both static and dynamic, corresponding to connections in a local
network with several workstations, that access to the Internet through a router with
NAT. Those do not allow the use of URIs of the name@IP-address type because the IP
address of the identifier is not public, and is only valid in the section of the network to
which it has been assigned; it lacks a universal meaning. In this case the use of an
associated SIP server and a STUN server is imperative to get past the NAT (Network
Address Translation) implemented in the router that acts as an interface between the
private network and the public one. See section NAT TRAVERSAL (4.3).
B4.3. PHOENIX ALIO default SIP configuration.
To simplify operating the unit, AEQ offers, at no additional cost, the services of 2 own
SIP servers.
PHOENIX ALIO is supplied from the factory with both SIP servers preconfigured (SYSTEM
SIP PROVIDERS menu) defined as "AEQ" and "AEQ 2" with the following configuration:
•
•
•
•
•
PROXY SIP "AEQ" Host: sip.aeq.es
PROXY SIP "AEQ 2" Host: sip2.aeq.es
PROXY SIP "AEQ" and "AEQ 2" Port: 5060
PROXY SIP "AEQ" and "AEQ 2" Domain: sip.aeq.es
PROXY SIP "AEQ" and "AEQ 2" Register Expires: 60 min.
PHOENIX ALIO is supplied preconfigured with 2 users registered in both servers:
•
•
PROXY SIP "AEQ" and "AEQ 2" Authentication Users: phxalio_XXX y phxalio_XXY
(where Y=X+1)
PROXY SIP "AEQ" and "AEQ 2" Authentication Passwords: 8 digits alphanumeric
passwords
This configuration is the right one for working with any of both AEQ’s SIP servers.
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B5. STUN protocol.
STUN (Simple Transversal of UDP over NATs) is a network protocol of the client /server type
that allows NAT clients to find their public IP address, the type of NAT where it is located and
the Internet port associated with the local port through NAT. This information is used to
configure a UDP communication between two hosts located behind NAT routers.
NAT (Network Address Translation) is a mechanism used by IP routers to exchange packets
between two networks that assign each other incompatible addresses. It consists of converting,
in real time, the addresses used in the transported packets. It is also necessary to edit the
packets to enable the operation of protocols that include address information within the protocol
conversation. It is most commonly utilized to enable the use of private addresses and still
provide connectivity with the rest of the Internet.
PHOENIX ALIO includes a STUN client that sends a request to a STUN server. The STUN
server then informs the client of its public IP and which port has been opened by NAT to permit
incoming traffic to enter the client’s network. This information enables the Phoenix ALIO to
identify its position within the SIP server. This protocol is used in "AUTO3" and "AUTO4" NAT
TRAVERSAL modes (see section 4.3).
Registration in a STUN server
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Notification from the server of the corresponding NATs detected
The response further enables the STUN client to determine the type of NAT being used, since
different NAT types handle incoming UDP packets in different ways. STUN supports three of the
four main existing types of NAT: Full Cone, Restricted Cone and Port Restricted Cone. It does
not, however, support Symmetric NAT, also known as bidirectional NAT, although Phoenix
ALIO allows it to be detected and reports its presence to the user.
Once the client has discovered its public address, it can advise its peers of that address.
STUN is useful as a complement to protocols like SIP. SIP uses UDP packets to signal sound,
video and text traffic over the Internet, but does not enable communication to be established
when the devices at the ends of the communication circuit are behind NAT routings. This is why
STUN is customarily used in these applications, to permit communication to be established.
The connection with the STUN server is normally made through port 3478 by means of UDP.
The STUN server can then provide the client with an alternate IP and communication port.
For complete information on the STUN protocol, we recommend consulting:
http://tools.ietf.org/html/rfc3489
AEQ always has a PHOENIX unit available for test at "phoenixMaster@sip.aeq.es" URI
and its 2 SIP servers are also available at sip.aeq.es and sip2.aeq.es and with warranty
that both work according to the official standard.
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APPENDIX C: Ports used by PHOENIX equipment.
When Phoenix unit is installed in a private IP network and you want to establish communication
with other units through that network router+firewall (gateway), three indications related to the
ports used by the unit must be taken into account:
1 - Output permissions in router+firewall.
Phoenix unit will send packets to different servers and/or other units (each one will use a
different port). Therefore, firewall will have to allow that packets from Phoenix unit IP are sent
towards the following ports:
Protocol
UDP
UDP
UDP
Port number
53
162
5004-5010
(may change)
UDP
3478-3479
UDP
5060-5062
UDP
50000-60000
TCP
4422
Usage
DNS (domain name server) protocol
SNMP Traps port
RTP protocol (audio packets going towards the remote
unit, ports depend on the remote unit network, not
Phoenix one)
STUN protocol (server for getting the public IP of the
private network)
SIP protocol (SIP signalling packets, port depens on the
remote unit network, not Phoenix network)
Vumeters protocol (only when remote control PC is not
in Phoenix network)
Remote control protocol (only when Phoenix must be
connected to PC and PC is not in its network. Port
depends on PC, not on Phoenix)
2 - Input permissions in router+firewall.
Phoenix unit will have to be able to receive packets from units installed out of the private
network. Therefore, firewall will have to allow that packets sent to Phoenix unit IP to the
following ports are received:
Protocol
UDP
UDP
UDP
Port number
161
5004-5010
(may change)
5060-5062
TCP
80
TCP
4422
Usage
SNMP protocol (monitoring)
RTP protocol (audio packets towards the remote unit,
ports depend on Phoenix configuration)
SIP protocol (SIP signaling packets, port depends on
the remote unit network, not Phoenix network)
HTTP protocol (only when PC must be connected to
Phoenix web server for a firmware upgrading, for
instance)
Remote control protocol (only when PC must be
connected to Phoenix and PC is not in its network)
3 - When router+firewall uses NAT translation between private and public addresses, then a
Port Forwarding must be made in the router for each one of the ports described in section 2 and
for each one of the IPs of Phoenix units installed in that private network. In that case, remote
unit will send its packets towards router IP (it’s a public IP) and the ports configured in router by
means of Port Forwarding (these are public ports), not to each Phoenix ports (these are private
ports).
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APPENDIX D: Application notes guide.
This index tries to give users guidance on selecting the most advisable application note in order
to connect two audiocodecs of Phoenix family, depending on its requirements and working
environment. Each application note describes the way to configure each of the audiocodecs.
When both ends are different (for instance, at one end there’s a Phoenix Mobile and at the other
end a Phoenix ALIO), different application notes should be followed in order to configure each
one.
All notes are available in electronic format in the CD furnished with the unit.
D1. Internet connection using standard cable access.
Application note AN0A.
Phoenix Studio, Venus or Mercury audiocodec directly connected to Internet by means
of a dedicated cable-Modem with DHCP. SIP call using AEQ SIP Proxy.
Application note AN0B.
Same case as AN0A, but adapted to Phoenix Mobile.
Application note AN0C.
Phoenix Studio, Venus or Mercury audiocodec connected to a LAN (together with other
IP equipments), connected to Internet by means of a router with NAT that can be
configured (or we have access to the Network Manager). SIP call using AEQ SIP Proxy.
Application note AN0D.
Same case as AN0C, but adapted to Phoenix Mobile.
Application note AN0E.
Same case as AN0C, but calling in DIRECT SIP with no SIP Proxy involved.
Application note AN0F.
Same case as AN0E, but adapted to Phoenix Mobile.
Application note AN0G.
IP Audio distribution at large international events using Phoenix Studio audiocodec.
Application note AN0H.
AoIP distribution and contribution system with Phoenix Venus audiocodecs for radio
broadcasting companies.
D2. Special applications using different kinds of Internet physical accesses, or point to
point connections.
Application note AN1.
Connecting a Phoenix (Studio, Venus, Mercury or Mobile) to Internet through a PC via a
WiFi network.
Application note AN2.
Connecting two Phoenix Mobile units using a BGAN satellite link.
Application note AN3.
Connecting two audiocodecs (Phoenix Studio - Phoenix Studio & Phoenix Mobile Phoenix Studio) using a private WiMAX network.
Application note AN4.
Connecting two Phoenix Studio units using a dedicated point-to-point IP radio link.
Application note AN4B.
Same case as AN4, but also for Venus and Mercury and configuring the units via
remote control software.
Application note AN5.
Connecting Phoenix IP audiocodecs to 3G/4G networks.
Application note AN6.
Connection and remote control of Phoenix ALIO audiocodecs from the studio and
simultaneous usage as a phone set.
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AEQ PHOENIX ALIO
APPENDIX E: Additional information.
NOTE: This equipment complies with the limits for a Class A digital device, pursuant to
part 15 of the FCC Rules. These limits are designed to provide reasonable protection
against harmful interference when the equipment is operated in a commercial
environment. This equipment generates, uses, and can radiate radio frequency energy
and, if not installed and used in accordance with the instruction manual, may cause
harmful interference to radio communications. Operation of this equipment in a residential
area is likely to cause harmful interference in which case the user will be required to
correct the interference at his own expense.
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AEQ PHOENIX ALIO