Usage notes
Purpose
Network protocol
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). Cisco
Unified IP Phones do not
support the SIP protocol
when the phones are
operating in IPv6 address
mode.
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.
Session Initiation Protocol
(SIP)
Cisco Unified IP Phone 8941
and 8945 use SCCP, version
20, for call control.
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.
Skinny Client Control
Protocol (SCCP)
Cisco Unified IP Phones use
SRTP for media encryption.
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.
Secure Real-Time Transfer
protocol (SRTP)
Cisco Unified IP Phones use
TCP to connect to Cisco
Unified Communications
Manager and to access XML
services.
TCP is a connection-oriented transport protocol.
Transmission Control
Protocol (TCP)
Cisco Unified IP Phone 8941 and 8945 Administration Guide for Cisco Unified Communications Manager 10.0 (SCCP
and SIP)
13
Network Protocols