Cisco 8832 Administration Manual page 29

About the Cisco IP Conference Phone
Network Protocol
Internet Protocol (IP)
Link Layer Discovery Protocol (LLDP)
Link Layer Discovery Protocol-Media
Endpoint Devices (LLDP-MED)
Real-Time Transport Protocol (RTP)
Real-Time Control Protocol (RTCP)
Session Initiation Protocol (SIP)
Purpose
IP is a messaging protocol that addresses
and sends packets across the network.
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.
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.
Cisco IP Conference Phone 8832 Administration Guide for Cisco Unified Communications Manager
Supported Network Protocols
Usage Notes
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 phone with Dynamic
Host Configuration Protocol (DHCP). If
you are not using DHCP, you must
manually assign these properties to each
phone locally.
The phones support IPv6 address. For more
information, see the documentation for your
particular Cisco Unified Communications
Manager release.
The phone supports LLDP on the PC port.
The 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 s : / / 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
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.
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