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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We
recommend that you use DHCP custom option 150. With this method, you configure
the TFTP server IP address as the option value. For additional supported DHCP
configurations, see the documentation for your particular Cisco Unified Communications Manager release.
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Cisco
Audio Session Tunneling (CAST)
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The CAST
protocol allows IP phones and associated applications behind the phone to
discover and communicate with the remote endpoints without requiring changes to
the traditional signaling components like Cisco Unified Communications Manager
and gateways. The CAST protocol allows separate hardware devices to synchronize
related media and it allows PC applications to augment nonvideo-capable phones
to become video enabled using the PC as the video resource.
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The Cisco IP Phone uses CAST as an interface between CUVA and Cisco Unified Communications Manager using the Cisco IP Phone
as a SIP proxy.
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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
IPPhone 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, gateway, and a TFTP server on each phone locally.
We
recommend that you use DHCP custom option 150. With this method, you configure
the TFTP server IP address as the option value. For additional supported DHCP
configurations, see the documentation for your particular Cisco Unified Communications Manager release.
Note
|
If you cannot use option 150, use DHCP option 66.
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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.
Note
|
IP phones can be HTTPS clients; they cannot be HTTPS servers.
|
|
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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IEEE
802.1X
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The IEEE
802.1X standard defines a client-server-based access control and authentication
protocol that restricts unauthorized clients from connection to a LAN through
publicly accessible ports.
Until
the client is authenticated, 802.1X access control allows only Extensible
Authentication Protocol over LAN (EAPOL) traffic through the port to which the
client is connected. After authentication is successful, normal traffic can
pass through the port.
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The
Cisco IP Phone implements the IEEE 802.1X standard through support for the
following authentication methods: EAP-FAST and EAP-TLS.
When
802.1X authentication is enabled on the phone, you should disable the PC port
and voice VLAN.
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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 IPPhone with Dynamic Host Configuration Protocol
(DHCP). If you are not using DHCP, you must manually assign these properties to
each phone locally.
The
Cisco IP Phones support IPv6 address. For more information, see the documentation for your particular Cisco Unified
Communications Manager release.
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Link
Layer Discovery Protocol (LLDP)
|
LLDP is
a standardized network discovery protocol (similar to CDP) that is supported on
some Cisco and third-party devices.
|
The
Cisco IPPhone supports LLDP on the PC port.
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Link
Layer Discovery Protocol-Media Endpoint Devices (LLDP-MED)
|
LLDP-MED
is an extension of the LLDP standard developed for voice products.
|
The
Cisco IPPhone 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)
|
RTP is a
standard protocol for transporting real-time data, such as interactive voice
and video, over data networks.
|
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)
|
RTCP
works in conjunction with RTP to provide QoS data (such as jitter, latency, and
round trip delay) on RTP streams.
|
RTCP is
enabled by default.
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Session
Initiation Protocol (SIP)
|
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.
|
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)
|
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
Phones use SRTP for media encryption.
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Transmission Control Protocol (TCP)
|
TCP is a
connection-oriented transport protocol.
|
Cisco IP
Phones use TCP to connect to Cisco Unified Communications Manager and to access
XML services.
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Transport Layer Security (TLS)
|
TLS is a
standard protocol for securing and authenticating communications.
|
When security is implemented, Cisco IP Phones use the TLS protocol when securely registering with the Cisco Unified Communications
Manager. For more information, see the documentation for your particular Cisco Unified Communications Manager release.
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Trivial
File Transfer Protocol (TFTP)
|
TFTP
allows you to transfer files over the network.
On the
Cisco IPPhone, TFTP enables you to obtain a configuration file specific to the
phone type.
|
TFTP
requires a TFTP server in your network, which can be automatically identified
from the DHCP server. If you want a phone to use a TFTP server other than the
one specified by the DHCP server, you must manually assign the IP address of
the TFTP server by using the Network Setup menu on the phone.
For more
information, see the documentation for your particular Cisco Unified Communications Manager release.
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User
Datagram Protocol (UDP)
|
UDP is a
connectionless messaging protocol for delivery of data packets.
|
UDP is used only for RTP streams. SIP uses UDP, TCP and TLS.
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