Cisco 7941G Administration Manual page 29

Cisco Unified IP Phone
Network Protocol
Real-Time Transport
Protocol (RTP)
Real-Time Control
Protocol (RTCP)
Session Description
Protocol (SDP)
Session Initiation
Protocol (SIP)
Skinny Client Control
Protocol (SCCP)
Transmission Control
Protocol (TCP)
Purpose
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 Quality of Service (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 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.
SCCP includes a message set that
allows communications between call
control servers and endpoint clients
such as IP Phones. SCCP is proprietary
to Cisco.
TCP is a connection-oriented transport
protocol.
Cisco Unified IP Phone 7941G, 7941G-GE, 7942G, 7961G, 7961G-GE, and 7962G Administration Guide for Cisco Unified
Network Protocols
Usage Notes
Cisco Unified IP Phones use the RTP
protocol to send and receive real-time
voice traffic from other phones and
gateways.
RTCP is disabled by default, but you can
enable it on a per phone basis by using
Cisco Unified Communications Manager.
SDP capabilities, such as codec types,
DTMF detection, and comfort noise, are
normally configured on a global basis by
Cisco Unified Communications Manager
or 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
signal 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.
You can configure the Cisco Unified IP
Phones to use either SIP or Skinny Client
Control Protocol (SCCP). Cisco Unified
IP Phones do not support the SIP protocol
when the phones are operating in IPv6
address mode.
Cisco Unified IP Phones use SCCP for
call control. You can configure the Cisco
Unified IP Phone to use either SCCP or
Session Initiation Protocol (SIP).
Cisco Unified IP Phones use TCP to
connect to Cisco Unified
Communications Manager and to access
XML services.
Communications Manager 9.0 (SCCP and SIP)
9