1-7
Cisco Unified IP Phone 7931G Administration Guide for Cisco Unified Communications Manager 8.0 (SCCP and SIP)
OL-20798-01
Chapter 1 An Overview of the Cisco Unified IP Phone
What Networking Protocols Are Used?
Real-Time Transport
Protocol (RTP)
RTP is a standard protocol for transporting
real-time data, such as interactive voice and
video, over data networks.
Cisco Unified IP Phones use the RTP
protocol to send and receive real-time voice
traffic from other phones and gateways.
Real-Time Transport
Protocol (RTCP)
RTCP works in conjunction with RTP to
provide QoS data (such as jitter, latency, and
round trip delay) on RTP streams.
RTCP is disabled by default, but you can
enable it on a per phone basis using Cisco
Unified CM. For more information, see
Network Configuration Menu, page 4-6.
Session Initiation Protocol
(SIP)
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.
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.
Skinny Client Control
Protocol (SCCP)
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.
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).
Session Description Protocol
(SDP)
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.
SDP capabilities, such as codec types,
DTMF detection, and comfort noise, are
normally configured on a global basis by
Cisco Unified CM or Media Gateway in
operation. Some SIP endpoints may allow
these parameters to be configured on the
endpoint itself.
Transmission Control
Protocol (TCP)
TCP is a connection-oriented transport protocol. Cisco Unified IP Phones use TCP to
connect to Cisco Unified CM and to access
XML services.
Transport Layer Security
(TLS)
TLS is a standard protocol for securing and
authenticating communications.
When security is implemented, Cisco
Unified IP Phones use the TLS protocol
when securely registering with Cisco
Unified CM.
For more information, see the
Cisco Unified Communications Manager
Security Guide.
Table 1-2 Supported Networking Protocols on the Cisco Unified IP Phone (continued)
Networking Protocol Purpose Usage Notes