Cisco 6901 Standard 매뉴얼 - 페이지 13

{카테고리_이름} Cisco 6901 Standard에 대한 매뉴얼을 온라인으로 검색하거나 PDF를 다운로드하세요. Cisco 6901 Standard 28 페이지. Unified ip phone
Cisco 6901 Standard에 대해서도 마찬가지입니다: 데이터시트 (5 페이지), 관리 매뉴얼 (10 페이지), 사용자 설명서 (15 페이지), 관리 매뉴얼 (12 페이지), 사용자 설명서 (50 페이지), 매뉴얼 (10 페이지)

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Cisco 6901 Standard 매뉴얼
Cisco Unified IP Phone
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)
Skinny Client Control Protocol
(SCCP)
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.
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.
Network Protocols
Usage notes
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:
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Cisco Unified IP Phones use the
RTP protocol to send and receive
real-time voice traffic from other
phones and gateways.
By default, the phones have RTCP
disabled, but you can enable it on
each individual phone 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.
You can configure the Cisco
Unified IP Phone to use either SIP
or Skinny Client Control Protocol
(SCCP).
Cisco Unified IP Phone 6901 and
6911 use SCCP, version 20 for call
control.
Cisco Unified IP Phone
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