Cisco 8831 Administration Manual page 27

Cisco Unified IP Conference Phone 8831
Networking 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)
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.
Cisco Unified IP Conference Phone 8831 Administration Guide for Cisco Unified Communications Manager 9.0
Network protocols
Usage Notes
The Cisco Unified IP Conference
Phone 8831 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 Unified IP Phones use the RTP
protocol to send and receive real-time
voice traffic from other phones and
gateways.
RTCP is disabled by default, but you
can enable it on a per phone basis by
using Cisco Unified Communications
Manager.
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.
Cisco Unified IP Phones use SRTP
for media encryption.
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