Assigning Codec Profiles To Ip Addresses - Altigen ACM 6.5 Administration Manual

Assigning Codec Profiles to IP Addresses

Parameter
SIP Transport
Assigning Codec Profiles to IP Addresses
You can specify what codec profile to use when connecting to the following VoIP devices:
IP phones on the LAN
•
a remote IP phone over WAN
•
a remote AltiGen system over WAN
•
SIP Trunk service provider over WAN
•
multiple gateways on the LAN
•
The codec profile assigned in the IP Device Range table (shown below) supersedes the
codec profile defined in the IP dialing table if the IP address is duplicated in both tables.
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MAXCS ACM 6.5 Administration Manual
Description
There are several SIP Transport options. Note that
security options TLS and SRTP can be configured for
individual IP phone extensions in the IP Phone
Configuration screen. (For more information on
SIP Transport
security settings, see "
on page 236.) Extension-level configuration takes
precedence over a codec profile that is assigned in
Enterprise Manager. See the next section.
UDP—User Datagram Protocol is a communications
protocol that offers a limited amount of service when
messages are exchanged between computers in a
network that uses the Internet Protocol (IP).
TCP—Transmission Control Protocol is a set of rules
(protocol) used along with the Internet Protocol (IP)
to send data in the form of message units between
computers over the Internet. TCP is known as a
connection-oriented protocol, which means that a
connection is established and maintained until such
time as the message or messages to be exchanged
by the application programs at each end have been
exchanged. TCP is responsible for ensuring that a
message is divided into the packets that IP manages
and for reassembling the packets back into the
complete message at the other end.
AltiGen phones do not use TCP.
Note:
TLS—Secures SIP signaling messages using
Transport Layer Security. (Does not work for IP
devices behind NAT; UDP will be used, instead.)
TLS/SRTP—Adds Secure RTP to Transport Layer
Security to secure SIP-associated media. (Does not
work for IP devices behind NAT; UDP will be used,
instead.)
(If this option is chosen, the voice stream always
goes through the server.)
Persistent TLS/SRTP—Persistent TLS/SRTP for
SIP signaling messages.
" in the table