Yealink W60P Administrator's Guide

Introduction
xi
VoIP Principle
VoIP
VoIP (Voice over Internet Protocol) is a technology using the Internet Protocol instead of
traditional Public Switch Telephone Network (PSTN) technology for voice communications.
It is a family of technologies, methodologies, communication protocols, and transmission
techniques for the delivery of voice communications and multimedia sessions over IP networks.
The H.323 and Session Initiation Protocol (SIP) are two popular VoIP protocols that are found in
widespread implementation.
H.323
H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T)
that defines the protocols to provide audio-visual communication sessions on any packet
network. The H.323 standard addresses call signaling and control, multimedia transport and
control, and bandwidth control for point-to-point and multi-point conferences.
It is widely implemented by voice and video conference equipment manufacturers, is used
within various Internet real-time applications such as GnuGK and NetMeeting and is widely
deployed by service providers and enterprises for both voice and video services over IP
networks.
SIP
SIP (Session Initiation Protocol) is the Internet Engineering Task Force’s (IETF’s) standard for
multimedia conferencing over IP. It 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 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
attributes of an end-to-end call.
SIP provides capabilities to:
 Determine the location of the target endpoint -- SIP supports address resolution, name
mapping, and call redirection.
 Determine media capabilities of the target endpoint -- Via Session Description Protocol
(SDP), SIP determines the “lowest level” of common services between endpoints.
Conferences are established using only media capabilities that can be supported by all
endpoints.
 Determine the availability of the target endpoint -- A call cannot be completed because
the target endpoint is unavailable, SIP determines whether the called party is already on
the DECT IP phone or does not answer in the allotted number of rings. It then returns a
message indicating why the target endpoint is unavailable.
 Establish a session between the origin and target endpoint -- The call can be completed,

Yealink W60P Specifications

General

Mounting	Desk/Wall
Handset type	Wireless handset
Product type	IP Phone
Product color	Black, Silver
Wall mountable	Yes
Standby time	400 h
AC adapter input voltage	100 - 240 V
Battery technology	Lithium
Headset connectivity	3.5 mm
USB 2.0 ports quantity	USB 2.0 ports have a data transmission speed of 480 Mbps, and are backwards compatible with USB 1.1 ports. You can connect all kinds of peripheral devices to them.
Ethernet LAN (RJ-45) ports	1
Maximum indoor range	50 m
Maximum outdoor range	300 m
Lines quantity	- lines
Phonebook capacity	500 entries
Number of handles included	1 pc(s)
Display diagonal	2.4 \
Display resolution	240 x 320 pixels
Voice codecs	AMR-WB, G.722, OPUS
Number of melodies	9
Security algorithms	SRTP, SSL/TLS
Networking standards	IEEE 802.1Q, IEEE 802.1p, IEEE 802.3af
Ethernet LAN data rates	10, 100 Mbit/s
Ethernet LAN interface type	Fast Ethernet
Volume control	Digital
Keypad number of keys	12
Number of function keys	6
Operating temperature (T-T)	-10 - 50 °C
Operating relative humidity (H-H)	10 - 95 %
Harmonized System (HS) code	85171800

Weight and Dimensions

Weight	7500 g
Package depth	95 mm
Package width	196 mm
Package height	205 mm
Package weight	8300 g
Base dimensions	100 x 25.1 x 130 mm
Handset dimensions (WxDxH)	53 x 20.3 x 175 mm