Cisco 7800 Series Administration Manual page 29

About the Cisco IP Phone
Network Protocol
Link Layer Discovery
Protocol-Media Endpoint Devices
(LLDP-MED)
Network Transport Protocol (NTP)
Real-Time Transport Protocol
(RTP)
Real-Time Control Protocol
(RTCP)
Session Initiation Protocol (SIP)
Secure Real-Time Transfer protocol
(SRTP)
Purpose
LLDP-MED is an extension of the
LLDP standard developed for voice
products.
NTP is a networking protocol for
clock synchronization between
computer systems over
packet-switched, variable-latency
data networks.
RTP is a standard protocol for
transporting real-time data, such as
interactive voice and video, over
data networks.
RTCP works in conjunction with
RTP to provide QoS data (such as
jitter, latency, and round trip delay)
on RTP streams.
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.
Cisco IP Phone 7800 Series Administration Guide for Cisco Unified Communications Manager
Network Protocols
Usage Notes
The Cisco IP Phone supports
LLDP-MED on the SW port to
communicate information such as:
• Voice VLAN configuration
• Device discovery
• Power management
• Inventory management
For more information about
LLDP-MED support, see the
LLDP-MED and Cisco Discovery
Protocol white paper at this URL:
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Cisco IP Phones have an NTP client
integrated into the software.
Cisco IP Phones use the RTP
protocol to send and receive
real-time voice traffic from other
phones and gateways.
RTCP is enabled by default.
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.
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