Cisco 7931G Manual page 11

Cisco Unified IP Phone 7931G
Networking Protocol
Real-Time Transport
Protocol (RTCP)
Session Initiation
Protocol (SIP)
Skinny Client Control
Protocol (SCCP)
Session Description
Protocol (SDP)
Transmission Control
Protocol (TCP)
Purpose
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.
SCCP includes a messaging set that
allows communications between call
control servers and endpoint clients
such as IP Phones. SCCP is proprietary
to Cisco Systems.
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.
TCP is a connection-oriented transport
protocol.
Cisco Unified IP Phone 7931G Administration Guide for Cisco Unified Communications Manager 9.0 (SCCP and SIP)
Network Protocols
Usage notes
RTCP is disabled by default, but you
can enable it on a per phone basis using
Cisco Unified Communications
Manager. For more information, see
Network Configuration Menu
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.
You can configure the Cisco Unified IP
Phone to use either SIP or Skinny Client
Control Protocol (SCCP).
The Cisco Unified IP Phone does not
support the SIP protocol when the
phones are operating in IPv6 address
mode.
The Cisco Unified IP Phone uses SCCP
for call control. You can configure the
Cisco Unified IP Phone to use either
SCCP or Session Initiation Protocol
(SIP).
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.
The Cisco Unified IP Phone uses TCP
to connect to Cisco
Unified Communications Manager and
to access XML services.
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