Bootstrap
Protocol (BootP)
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BootP
enables a network device, such as the Cisco IP Phone, to discover certain startup
information, such as its IP address.
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—
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Cisco
Discovery Protocol (CDP)
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CDP is a
device-discovery protocol that runs on all Cisco-manufactured equipment.
A device can use CDP to advertise its existence to other devices and receive information
about other devices in the network.
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The Cisco
IP Phone uses CDP to communicate information such as auxiliary VLAN ID, per
port power management details, and Quality of Service (QoS) configuration
information with the Cisco Catalyst switch.
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Domain Name Server (DNS)
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DNS translates domain names to IP addresses.
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Cisco IP Phones have a DNS client to translate domain names into IP addresses.
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Dynamic
Host Configuration Protocol (DHCP)
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DHCP
dynamically allocates and assigns an IP address to network devices.
DHCP
enables you to connect an IP phone into the network and have the phone become
operational without the need to manually assign an IP address or to
configure additional network parameters.
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DHCP is enabled by default. If disabled, you must manually configure the IP address, subnet mask, and gateway on each phone
locally.
We recommend that you use the DHCP custom option 160, 159.
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Hypertext Transfer Protocol (HTTP)
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HTTP is
the standard protocol for transfer of information and movement of documents across the
Internet and the web.
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Cisco IP
Phones use HTTP for XML services, provisioning, upgrade and for troubleshooting purposes.
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Hypertext Transfer Protocol Secure (HTTPS)
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Hypertext Transfer Protocol Secure (HTTPS) is a combination of
the Hypertext Transfer Protocol with the SSL/TLS protocol to provide encryption
and secure identification of servers.
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Web
applications with both HTTP and HTTPS support have two URLs configured. Cisco
IP Phones that support HTTPS choose the HTTPS URL.
A lock
icon is displayed to the user if the connection to the service is via HTTPS.
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Internet
Protocol (IP)
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IP is a
messaging protocol that addresses and sends packets across the network.
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To
communicate with IP, network devices must have an assigned IP address, subnet,
and gateway.
IP
addresses, subnets, and gateways identifications are automatically assigned if
you are using the Cisco IP Phone with Dynamic Host Configuration Protocol
(DHCP). If you are not using DHCP, you must manually assign these properties to
each phone locally.
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Link
Layer Discovery Protocol (LLDP)
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LLDP is
a standardized network discovery protocol (similar to CDP) that is supported on
some Cisco and third-party devices.
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The
Cisco IP Phone supports LLDP on the PC port.
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Link
Layer Discovery Protocol-Media Endpoint Devices (LLDP-MED)
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LLDP-MED
is an extension of the LLDP standard developed for voice products.
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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:
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper0900aecd804cd46d.shtml
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Network Transport Protocol (NTP)
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NTP is a networking protocol for clock synchronization between computer systems over packet-switched, variable-latency data
networks.
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Cisco IP Phones have an NTP client integrated into the software.
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Real-Time Transport Protocol (RTP)
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RTP is a
standard protocol for transporting real-time data, such as interactive voice
and video, over data networks.
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Cisco IP
Phones use the RTP protocol to send and receive real-time voice traffic from
other phones and gateways.
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Real-Time Control Protocol (RTCP)
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RTCP
works in conjunction with RTP to provide QoS data (such as jitter, latency, and
round trip delay) on RTP streams.
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RTCP is
disabled by default.
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Session Description Protocol (SDP)
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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.
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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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Session
Initiation Protocol (SIP)
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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.
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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.
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Secure
Real-Time Transfer protocol (SRTP)
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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.
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Cisco IP
Phones use SRTP for media encryption.
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Transmission Control Protocol (TCP)
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TCP is a
connection-oriented transport protocol.
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—
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Transport Layer Security (TLS)
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TLS is a
standard protocol for securing and authenticating communications.
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When security is implemented, Cisco IP Phones use the TLS protocol when securely registering with the third-party call control
system.
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Trivial
File Transfer Protocol (TFTP)
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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.
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TFTP
requires a TFTP server in your network, which can be automatically identified
from the DHCP server.
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User
Datagram Protocol (UDP)
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UDP is a
connectionless messaging protocol for delivery of data packets.
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UDP is used only for RTP streams. SIP uses UDP, TCP, and TLS.
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