AEQ
PHOENIX MERCURY
13
-
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, “phxme_231” for
instance.
o
Pwd:
the Pasword 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 MERCURY 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 3.3 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 3.4.
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. Usually recommended value 5004.
o
Adaptive
/
Fixed
and
Adaptive buffer max:
this option allows you to
configure the type and maximum size of reception buffer. See section 3.4.
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 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 signalling protocol. That signalling 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.
3.2.2. 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.
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,
“
”).
When the correspondent
SIP port
is not the 5060 (SIP Standard port) the identifier must include
the used port. For instance: “
phxme_ [email protected]:5061
”
.
