Cisco 6800 Series Administration Manual page 33

Multiplatform phones
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About the Cisco IP Phone
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)
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 Cisco IP Phone, TFTP
enables you to obtain a
configuration file specific to the
phone type.
UDP is a connectionless messaging
protocol for delivery of data
packets.
Cisco IP Phone 6800 Series Multiplatform Phones Administration Guide
Network Protocols
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.
Cisco IP Phones use SRTP for
media encryption.
When security is implemented,
Cisco IP Phones use 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.
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