Cisco 7800 Series Administration Manual page 32

Multiplatform phones
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Network Protocols
Network Protocol
Link Layer Discovery Protocol
(LLDP)
Link Layer Discovery
Protocol-Media Endpoint Devices
(LLDP-MED)
Network Transport Protocol (NTP)
Real-Time Transport Protocol
(RTP)
Real-Time Control Protocol
(RTCP)
Session Description Protocol (SDP)
Cisco IP Phone 7800 Series Multiplatform Phones Administration Guide
18
Purpose
LLDP is a standardized network
discovery protocol (similar to CDP)
that is supported on some Cisco and
third-party devices.
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.
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.
About the Cisco IP Phone
Usage Notes
The Cisco IP Phone supports LLDP
on the PC port.
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:
h t t p : / / w w w . c i s c o . c o m / e n / U S / t e c h / t k 6 5 2 / t k 7 0 1 / t e c h n o l o g i e s _ w h i t e _ p a p e r 0 9 0 0 a e c d 8 0 4 c d 4 6 d . s h t m l
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 disabled by default.
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.

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