Specifications

1-6
Cisco Unified IP Phone 7906G and 7911G for Cisco Unified Communications Manager 8.0
OL-21033-01
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 Control
Protocol (RTCP
RTCP works 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 by using Cisco Unified Communications
Manager. For more information, see Network Configuration,
page 4-28.
Secure Real-Time
Transport Protocol
(SRTP)
SRTP is available in addition to
RTP. SRTP adds security by
encrypting media streams during
data transport.
For SRTP to work, the phone or phones being called must
also support SRTP or else those phones cannot decrypt the
secure media stream.
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 Communications Manager 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
Communications Manager 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 Communications Manager.
For more information, refer to 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