Assigning Codec Profiles To Ip Addresses - Altigen ACM 6.7 Administrator's Manual

Assigning Codec Profiles to IP Addresses

Parameter
DTMF Delivery
(Applies to SIP protocol
only)
SIP Early Media
(Applies to SIP protocol
and SIP trunk only)
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
322
MAXCS ACM 6.7 Administration Manual
Description
Default—If SIP INFO is used to deliver DTMF.
RFC 2833—The DTMF pay load is embedded with RTP. Most 3rd-
party SIP gateways support this standard.
In band—If DTMF tone is delivered over the voice band. It's not
reliable over G.711 codec and will not work over G.729/G.723
codec
SIP Early Media allows two SIP devices to communicate before
a SIP call is actually established. It is important for
interoperability with the SIP trunk carrier's PSTN gateway. If SIP
Early Media is not checked, the caller may not hear the exact
ringback tone provided by the CO (the caller may not hear any
ringback tone at all).
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 security settings, see "
Administrator Guide table on page 223.) 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.
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.
SIP Transport
" in the