Cisco 8800 Series Manual page 470

Technical Details
Network protocol
Link Layer Discovery Protocol-Media
Endpoint Devices (LLDP-MED)
Real-Time Transport Protocol (RTP)
Real-Time Control Protocol (RTCP)
Session Description Protocol (SDP)
Session Initiation Protocol (SIP)
Transmission Control Protocol (TCP)
Transport Layer Security (TLS)
Cisco IP Phone 8800 Series Multiplatform Phone Administration Guide for Release 11.3(1) and Later
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Purpose
LLDP-MED is an extension of the LLDP
standard for voice products.
RTP is a standard protocol for transporting
real-time data, such as interactive voice,
over data networks.
RTCP works in conjunction with RTP to
provide QoS data (such as jitter, latency,
and round-trip delay) on RTP streams.
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 all endpoints in
the conference support.
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.
TCP is a connection-oriented transport
protocol.
TLS is a standard protocol for securing and
authenticating communications.
Technical Details
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:
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Cisco IP Phones use the RTP protocol to
send and receive real-time voice traffic
from other phones and gateways.
RTCP is disabled by default.
SDP capabilities, such as codec types,
DTMF detection, and comfort noise, are
normally configured on a global basis by
Third-Party Call Control System or Media
Gateway in operation. Some SIP endpoints
may allow configuration of these
parameters on the endpoint itself.
Like other VoIP protocols, SIP addresses
the functions of signaling and session
management within a packet telephony
network. Signaling allows transportation
of call information across network
boundaries. Session management provides
the ability to control the attributes of an
end-to-end call.
Cisco IP Phones support the SIP protocol
when the phones are operating in IPv6-only,
IPv4-only, or in both IPv4 and IPv6.
Cisco IP Phones use TCP to connect to
Third-Party Call Control system and to
access XML services.
Upon security implementation, Cisco IP
Phones use the TLS protocol when securely
registering with Third-Party Call Control
system.