AEQ
TALENT
21
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 number of the IP port
where that channel decoder expects the audio to arrive at. Minimum value 1,024.
Maximum value 65,534. Default value: 5004
•
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 "
Smart
RTP
" 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
"
Smart
RTP
"
.
4.2.2. PROXY SIP.
This type of connection is selected when the TALENT is used working together with an external
SIP server that will provide connection with remote unit through any network, even Internet,
without caring about IP addresses. Both units (local and remote) must be registered in the same
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
<unit_name>
, for instance: "
phxtalent_1231" or "phoenixMaster
".
o
<unit_name>@<realm_SIP_server>
, for instance:
"
" o "
".
o
<unit_name>@< SIP_server _IP>
, for instance:
"
" o "
" (where
232.168.1.2. is the AEQ´s SIP server "
sip.aeq.es
")
o
<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: "
[email protected]: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.
