Assigning Codec Profiles To Ip Addresses - Altigen MaxACD Administrator Manual

Chapter 19: Enterprise VoIP Network Management
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 VoIP devices.
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.
To set IP address ranges and assign codec profiles to them, in Enterprise Manager click
the IP Codec tab.
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MaxACD Administrator 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.
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.
TLS—Secures SIP signaling messages using Transport Layer
Security.
TLS/SRTP—Adds Secure RTP to Transport Layer Security to
secure SIP-associated media.
(If this option is chosen, the voice stream always goes through
the server.)
Persistent TLS/SRTP—Persistent TLS/SRTP for SIP signaling
messages.