User guide

Chapter 25: Enterprise VoIP Network Management
MAXCS ACC 6.7.1 Administration Manual 305
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
SIP Early Media
(Applies to SIP protocol
and SIP trunk only)
SIP Early Media allows two SIP devices to communicate before
a SIP call is actually established. It is important for interoper
-
ability with the SIP trunk carrier’s PSTN gateway. If SIP Early
Media is not checked, the caller may not hear the exact ring
-
back tone provided by the CO (the caller may not hear any
ringback tone at all).
SIP Transport
There are several SIP Transport options. Note that security op-
tions TLS and SRTP can be configured for individual IP phone
extensions in the IP Phone Configuration screen. (For more in
-
formation on security settings, see “SIP Transport” in the table
on page 217.) 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).
Note: All SIP trunks must use UDP.
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 applica
-
tion programs at each end have been exchanged. TCP is re-
sponsible for ensuring that a message is divided into the pack-
ets that IP manages and for reassembling the packets back into
the complete message at the other end.
Note: AltiGen phones do not use TCP.
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.
Parameter Description