Cisco 6941 Administration Guide - Page 23

SCCP. Cisco Unified IP Phones do not support the, Cisco Unified IP Phones use TCP to connect

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Chapter What Networking Protocols are Used? Table 1-4 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/tk701/tech nologies_white_paper0900aecd804cd46d.shtml Real-Time Transport Protocol (RTP) RTP is a standard protocol for transporting Cisco Unified IP Phones use the RTP protocol to real-time data, such as interactive voice send and receive real-time voice traffic from other and video, over data networks. phones and gateways. Real-Time Control 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 by using Cisco Unified CM. 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 Phone 6921, 6941, 6945, and 6961 use SCCP, version 20 for call control. Secure Real-Time Transfer protocol (SRTP) 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. Cisco Unified IP Phones use SRTP for media encryption. 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. Cisco Unified IP Phone 6921, 6941, 6945, and 6961 Administration Guide for Cisco Unified Communications Manager 8.5 (SCCP and SIP) OL-23769-01 1-11

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1-11
Cisco Unified IP Phone 6921, 6941, 6945, and 6961 Administration Guide for Cisco Unified Communications Manager 8.5 (SCCP and SIP)
OL-23769-01
Chapter
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:
nologies_white_paper0900aecd804cd46d.shtml
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 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 by using Cisco Unified CM.
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 Phone 6921, 6941, 6945, and 6961
use SCCP, version 20 for call control.
Secure Real-Time
Transfer protocol (SRTP)
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.
Cisco Unified IP Phones use SRTP for media
encryption.
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.
Table 1-4
Supported Networking Protocols on the Cisco Unified IP Phone (continued)
Networking Protocol
Purpose
Usage Notes