Cisco 3951 Administration Manual page 29

Chapter 1 An Overview of the Cisco Unified IP Phone
Table 1-1 Supported Networking Protocols on the Cisco Unified IP Phone (continued)
Networking Protocol
Internet Protocol (IP)
Real-Time Transport
(RTP)
Session Description
Protocol (SDP)
Session Initiation
Protocol (SIP)
OL-10969-01
Purpose
IP is a messaging protocol that
addresses and sends packets across
the network.
RTP is a standard protocol for
transporting real-time data, such as
interactive voice and video, over data
networks.
SDP is the portion of the SIP protocol
that determines which parameters are
available during a connection
between two endpoints. Conferences
are established using only the SDP
capabilities that are supported by all
endpoints in the conference.
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 Unified SIP Phone 3951 Administration Guide for Cisco Unified CallManager 5.1
What Networking Protocols Are Used?
Usage Notes
To communicate using 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 Unified IP Phone with
Dynamic Host Configuration
Protocol (DHCP). If you are not
using DHCP, you must manually
assign these properties to each phone
locally.
Cisco Unified IP Phones use the RTP
protocol to send and receive
real-time voice traffic from other
phones and gateways.
SDP capabilities, such as codec
types, DTMF detection, and comfort
noise are normally configured on a
global basis by the
Cisco Unified CallManager or the
Media Gateway in operation. Some
SIP endpoints may allow these
parameters to be configured on the
endpoint itself.
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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