Cisco 8961 Administration Manual page 48
Unified communications manager 10.0 sip
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Network protocols
Network protocol
Real-Time Control Protocol
(RTCP)
Session Description Protocol
(SDP)
Session Initiation Protocol (SIP)
Transmission Control Protocol
(TCP)
Transport Layer Security (TLS)
Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager
10.0 (SIP)
24
REVIEW DRAFT - CISCO CONFIDENTIAL
Purpose
RTCP works in conjunction with RTP to provide 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 all endpoints in the conference support.
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.
TCP is a connection-oriented transport protocol.
TLS is a standard protocol for securing and authenticating
communications.
Cisco Unified IP Phone
Usage notes
RTCP for audio calls is disabled
by default. RTCP for video calls
(including both audio streams and
video streams in the video call) is
enabled by default. You can enable
or disable RTCP on individual
phones from the Cisco Unified
Communications Manager
Administration.
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 configuration of these
parameters on the endpoint itself.
Like other VoIP protocols, SIP
addresses the functions of signaling
and session management within a
packet telephony network.
Signaling allows transportation of
call information across network
boundaries. Session management
provides the ability to control the
attributes of an end-to-end call.
Cisco Unified IP Phones support
the SIP protocol when the phones
are operating in IPv6 address, IPv4
address, or dual-stack mode.
Cisco Unified IP Phones use TCP
to connect to Cisco Unified
Communications Manager and to
access XML services.
Upon security implementation,
Cisco Unified IP Phones use the
TLS protocol when securely
registering with Cisco Unified
Communications Manager.
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