Cisco CP-7975G Administration Guide - Page 22

Cisco Unified IP Phones use SCCP for call, You can con the Cisco Unified IP Phone - sip

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What Networking Protocols are Used? Chapter 1 An Overview of the Cisco Unified IP Phone Table 1-2 Supported Networking Protocols on the Cisco Unified IP Phone (continued) Networking Protocol Purpose Usage Notes Link Layer Discovery Protocol-Media Endpoint Devices (LLDP-MED) LLDP-MED is an extension of the LLDP standard developed for voice products. The Cisco Unified IP Phone supports LLDP-MED on the SW port to communicate information such as: • Voice VLAN configuration • Device discovery • Power management • Inventory management For more information about LLDP-MED support, see the LLDP-MED and Cisco Discovery Protocol white paper: http://www.cisco.com/en/US/tech/tk652/tk70 1/technologies_white_paper0900aecd804cd4 6d.shtml Real-Time Control Protocol (RTCP) RTCP works with Real-Time Transport Protocol (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 Communications Manager. For more information, see Network Configuration, page 4-32. Real-Time Transport Protocol RTP is a standard protocol for transporting Cisco Unified IP Phones use the RTP protocol (RTP) real-time data, such as interactive voice and to send and receive real-time voice traffic video, over data networks. from other phones and gateways. 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). Cisco Unified IP Phone Administration Guide for Cisco Unified Communications Manager 8.5 1-8 OL-23092-01

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1-8
Cisco Unified IP Phone Administration Guide for Cisco Unified Communications Manager 8.5
OL-23092-01
Chapter 1
An Overview of the Cisco Unified IP Phone
What Networking Protocols are Used?
Link Layer Discovery
Protocol-Media Endpoint
Devices (LLDP-MED)
LLDP-MED is an extension of the LLDP
standard developed for voice products.
The Cisco Unified IP Phone supports
LLDP-MED on the SW port to communicate
information such as:
Voice VLAN configuration
Device discovery
Power management
Inventory management
For more information about LLDP-MED
support, see the
LLDP-MED and Cisco
Discovery Protocol
white paper:
1/technologies_white_paper0900aecd804cd4
6d.shtml
Real-Time Control Protocol
(RTCP)
RTCP works with Real-Time Transport
Protocol (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 Communications Manager. For more
information, see
Network Configuration,
page 4-32
.
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.
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).
Table 1-2
Supported Networking Protocols on the Cisco Unified IP Phone (continued)
Networking Protocol
Purpose
Usage Notes