Cisco 7821 Administration Manual page 25

For cisco unified communications manager 10.0 (sip)
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Network protocol
Link Layer Discovery
Protocol-Media Endpoint
Devices (LLDP-MED)
Real-Time Transport
Protocol (RTP)
Real-Time Control
Protocol (RTCP)
Session Initiation Protocol
(SIP)
Secure Real-Time Transfer
protocol (SRTP)
Cisco IP Phone 7821, 7841, and 7861 Administration Guide for Cisco Unified Communications Manager 10.0 (SIP)
Purpose
LLDP-MED is an extension of the LLDP
standard developed for voice products.
RTP is a standard protocol for
transporting real-time data, such as
interactive voice and video, over data
networks.
RTCP works in conjunction with RTP to
provide QoS data (such as jitter, latency,
and round trip delay) on RTP streams.
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.
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.
Supported Network Protocols
Usage notes
The Cisco 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/technologies_white_
paper0900aecd804cd46d.shtml
Cisco IP Phones use the RTP
protocol to send and receive real-time
voice traffic from other phones and
gateways.
RTCP is enabled by default.
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 IP
Phone to use SIP.
Cisco IP Phones use SRTP for media
encryption.
11

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