User Manual

Yealink Skype for Business HD IP Phones Administrator Guide
xii
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 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,