AN 0B
AEQ PHOENIX AUDIOCODECS. APPLICATION NOTE 0-B
Connecting AEQ Phoenix Mobile units through Internet via
exclusive DSL or cable modem with DHCP router
(Adapted to Phoenix Mobile user interface)
The most convenient and reliable way to connect an IP codec to a remote one through
Internet is by means of a dedicated, exclusive Internet access (DSL, cable modem) as
described below. No other equipments (PC, etc) are connected to this same access.
The modem itself usually has an embedded DHCP server that automatically assigns
valid IP address / mask, gateway and DNS server to the IP system(s) connected to it.
This application note also describes the recommended procedure when AEQ SIP
Proxy is used, where both equipments involved in the communication are registered,
easing the task of finding the remote IP address when it is variable or unknown.
This document shows the specific steps to make this connection possible in the most
general way possible, although some verification procedures have been omitted in
order to ease the reading, and some particular situations may be easier to solve that
the presented one.
For a detailed explanation of every step, and for details on how to configure other
options that may be more convenient for other cases, the reading of the rest of
application notes, as well as User Manual’s chapter 4, is recommended.
1. CONNECTING AND CONFIGURING THE EQUIPMENT
The physical connection will be done to a LAN port.
•
Connect the network cable to the LAN1 port at the back of the AEQ Phoenix
Mobile. Connect the other end to the Ethernet port to the router LAN port.
•
Connect at least a microphone and a pair of headphones to the front audio XLR
connectors of the Phoenix Mobile
•
Connect the power supply cable to the AEQ Phoenix Mobile. If it is equipped
with a battery pack and mains power is not available, you can omit this step.
•
Press the ON button at the left side of AEQ Phoenix Mobile. Verify that the main
display is illuminated.
•
Verify, at the back of the unit, that the green LED at the right of the Ethernet
port starts blinking indicating that the IP connection is active.
•
Configuring the Ethernet port and assignation of Program channel:
Make sure that we are selecting “Program” in the main screen (the one
representing faders and vumeters). At the lower center area you should read
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“Program”. Also, by pressing MENU we enter the Wizard menu that should
have a PROGRAM legend at the top center area. If COORDINATION appears
instead in any of both places, just press Channel key and channel will be shifted
to PROGRAM.
Main screen
Wizard screen
From main screen press Menu to access the Wizard menu. Make sure that
Select Interface shows “IP (Ether)”. Otherwise, please select this.
Highlight the Configure Interface, line using the down arrow key, press Select
and configure the following:
DHCP ON
The following parameters will be automatically configured:
Local IP address, for example 192.168.1.22
Gateway IP address, for example 192.168.1.1
Network Mask, for example 255.255.255.0
DNS Server IP address, for example 192.168.1.1
Also set: SIP Proxy: ON
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2. OTHER POINTS TO CHECK
We are now going to quickly overview the rest of configuration menus where it is
recommended to check that every parameter is correctly configured.
Coding Algorithm
At the Wizard menu Sel.Codec should be G722, for a start, although you can
change this at a later stage for better quality. In order to change the coding
algorithm by another pre-selected one, just select the Sel.codec line, press
Select and select one within the new window shown. In order to choose the
particular coding modes in each list shown in the Sel.codec menu, go to Menu
ADVANCED
6. Link profiles
Select
Options.
The audio routing will be most likely correct by default to send Mic 1 audio to
the transmission and send the received stream to HP1 headphone output. You
can, however, make sure of this from Menu
ADVANCED
3. Audio Settings
Select.
Configure the rest of parameters after reading the manual according to your particular
needs.
PROXY SIP Configuration
You can make sure that this is correct from the main screen: Menu
ADVANCED
5. Communications
3. SIP config
1. Parameters:
Check that User Name is filled with a user identifier that is previously registered
at the AEQ SIP server (for example: phoenix_123).
STUN Protocol
2 Activate ON
This is important and recommended, as it will ease connecting to other
equipment that is behind a router in a local network.
3 Server: At the moment of writing this document, the following STUN server is
recommended: “stun.voxgratia,org” (IP=83.103.82.85)
If the DNS Server field in the communications menu hasn’t been configured,
you must substitute the STUN name by its IP.
4 Port : 3478
Press “back” key in order to go to 3. SIP config
2. Proxy and check:
1 Activate ON
2 Address: sip.aeq.es.
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If the DNS Server field in the communications menu hasn’t been configured,
you must substitute the SIP server name (“sip.aeq.es”) by its IP,
82.165.157.20 at the moment of writing this document.
3 USER must have the same text value as USER NAME from the former menu,
followed by “@sip.aeq.es”, for example: [email protected]
4 Check that a correct password is typed in. (“aeq” by default)
5 Check that REALM is “sip.aeq.es”.
6 Check that DOMAIN is “sip.aeq.es”.
7 Check that EXPIRES = 60
Press back to go to 3. SIP config, then select 4. Call Settings and check that
Media PORT is 5004.
3. REGISTERING AT THE AEQ SIP SERVER
If everything is correctly configured, after these steps the unit should have registered at
the AEQ SIP server and the top line of every screen should display the current time.
In case it is not registered, if the equipment power is cycled, the sequence of legends
ad the top line should be:
<Current time>
Initializing
Finding Nat Type
Register Error
If this happens, just double-check that the configuration is correct and, if necessary,
ask one of our technical assistance services for support.
If the unit registers correctly, the sequence will be:
<Current time>
Initializing
Finding Nat Type
<Current time>
4. SENDING A TEST CALL TO PHOENIX MASTER
We are going to go through the correct configuration of the system and prepare it to
make a test call to a unit called [email protected] that AEQ maintains for our
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customers and that continuously transmits audio at the public IP address
212.170.163.189 .
Check that a set of headphones are connected to the front HP1 connection.
Check that AEQMaster_IP entry is listed in the Call Book and launch the call. In order
to do so…
From main screen: press Menu. Observe the “Select contact” line. If it is:
Select contact:
AEQMaster_IP
…everything is ready and the call is sent only by pressing the green key.
If the line “Select contact” presents a different name than AEQMaster_IP at its right, just
select this line and press “select” key. The call book will be presented. Just select
AEQMaster_IP from the list, press OK to close the list and then finally press the green
key.
If the contact was not already created in the Call Book, you can create it from the
menu:
Main screen
Menu
ADVANCED
2. Contacts. With the help of the user manual,
create a new contact with the following parameters:
Contact: AEQMaster_IP
Module Type: IP
URI: [email protected]
(when PROXY SIP = ON , the unit understands that, if “@sip.aeq.es” is omitted,
the destination is registered in the same server as this unit.
Speed Dial (n): (leave it as it was).
•
The top line at the screen displays (usually very fast) the status of the call, and
the different phases it goes through: Calling, Synchronizing and finally
Connected.
•
The second line should show AEQMaster_IP
•
The third line should show G722 or the current audio coding scheme.
If the remote unit is busy, the top line will show “Busy (486)”
•
Verify that the LED at the side of “TX PROG SYNC” key is lit green, as an
indication that the communication has been successfully established and
synchronized. If the key above HP1 is activated (illuminated) and the
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corresponding fader is open, the audio received from Phoenix Master should be
heard at the headphones.
Once the connection is OK, you can retry as many times as needed varying the
selected audio coding scheme in order to get the best possible quality allowed by the
network (Wizard Menu
Sel Codec.)
The list of available coding algorithms can be created in Menu
Link Profiles
Select
ADVANCED
6.
5. SENDING A CALL TO ANOTHER PHOENIX OR N/ACIP COMPATIBLE CODEC
The test call sent to AEQMaster_IP has been automatically answered. In order to make
a manual call to other codec, you must configure the second one the same way as the
first unit, provided that it is also connected directly to a DHCP modem-router. If it is not,
just follow the corresponding Internet access (AN0C to AN0F) application note.
Once the reporter equipment connects properly to AEQMaster_IP, you can create for
convenience new entries in the call book: Menu
ADVANCED
2. Contacts , with
the name and URI of the remote unit, the same way it was done for Phoenix Master in
the previous chapter, and then easily make calls to them.
You can of course also send the calls without having previously created a Call Book
entry from the main screen, by pressing Menu to go to the Wizard. Select the Call to
line, press Select and “URI” is selected. Then type the desired destination URI in the
line below (select “Enter URI”, then press OK and type the URI in). For this
communications mode based on a SIP server, you only need to introduce the name of
the unit (for example, “Phoenix_123”).
The call indications (optical and also acoustical, if enabled) will be activated at the
destination codec, that will either wait for the user to answer the call or automatically
off-hook (if this mode is enabled), and then establish the connection. The LED next to
TX PROG should become green lit and audio should be heard through the
headphones.
The three first screen lines should be:
Connected
<equipment name>@sip.aeq.es
<coding scheme>
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For more information and other configuration options, please read the User Manual and
Application Notes corresponding to other scenarios. If you still have doubts, please
don’t hesitate to contact one of our assistance services:
[email protected]
[email protected]
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