Cisco 6800 Series Administration Manual page 189

Technical Details
Network Protocol
Session Description
Protocol (SDP)
Session Initiation
Protocol (SIP)
Secure Real-Time
Transfer protocol
(SRTP)
Transmission
Control Protocol
(TCP)
Transport Layer
Security (TLS)
Trivial File Transfer
Protocol (TFTP)
User Datagram
Protocol (UDP)
External Devices
We recommend that you use good-quality external devices that are shielded against unwanted radio frequency
(RF) and audio frequency (AF) signals. External devices include headsets, cables, and connectors.
Purpose
SDP is the portion of the SIP protocol that
determines which parameters are available
during a connection between two
endpoints. Conferences are established by
using only the SDP capabilities that all
endpoints in the conference support.
SIP is the Internet Engineering Task Force
(IETF) standard for multimedia
conferencing over IP. SIP is an
ASCII-based application-layer control
protocol (defined in RFC 3261) that can
be used to establish, maintain, and
terminate calls between two or more
endpoints.
SRTP is an extension of the Real-Time
Protocol (RTP) Audio/Video Profile and
ensures the integrity of RTP and Real-Time
Control Protocol (RTCP) packets providing
authentication, integrity, and encryption
of media packets between two endpoints.
TCP is a connection-oriented transport
protocol.
TLS is a standard protocol for securing and
authenticating communications.
TFTP allows you to transfer files over the
network.
On the base station, TFTP enables you to
obtain a configuration file specific to the
phone type.
UDP is a connectionless messaging
protocol for delivery of data packets.
Usage Notes
SDP capabilities, such as codec types,
DTMF detection, and comfort noise, are
normally configured on a global basis by
a Third-Party Call Control System or a
Media Gateway in operation. Some SIP
endpoints may allow configuration of these
parameters on the endpoint itself.
Like other VoIP protocols, SIP is designed
to address the functions of signaling and
session management within a packet
telephony network. Signaling allows call
information to be carried across network
boundaries. Session management provides
the ability to control the attributes of an
end-to-end call.
Handsets and base stations use SRTP for
media encryption.
—
When security is implemented, the base
station uses the TLS protocol when
securely registering with the third-party
call control system.
TFTP requires a TFTP server in your
network, which can be automatically
identified from the DHCP server.
UDP is used only for RTP streams. SIP
uses UDP, TCP, and TLS.
Cisco IP DECT 6800 Series Administration Guide
External Devices
181