AEQ
PHOENIX ALIO
66
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
and in the list view
will change to green .
•
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).