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Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
OL-4277-01
Chapter 1
Overview of the Cisco IP Phone Models 7905G and 7912G
SIP Overview
SIP Overview
Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF)
standard for real-time calls and conferencing over Internet Protocol (IP). SIP is an
ASCII-based, application-layer control protocol that can be used to establish,
maintain, and terminate calls between two or more endpoints.
Like other VoIP protocols, SIP provides signaling and session management
within a packet telephony network. Signaling allows call information to be carried
across network boundaries. Session management controls the attributes of an
end-to-end call.
This section includes these topics:
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SIP Functions
SIP does the following:
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Determines the location of the target endpoint—SIP supports address
resolution, name mapping, and call redirection.
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Determines the 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 the
media capabilities that can be supported by all endpoints.
•
Determines the availability of the target endpoint—If a call cannot be
completed because the target endpoint is unavailable, SIP determines
whether the called party is already on the phone or did not answer in the
allotted number of rings. It then returns a message indicating why the target
endpoint was unavailable.
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Establishes a session between the originating and target endpoint—If the call
can be completed, SIP establishes a session between the endpoints. SIP also
supports mid-call changes such as adding another endpoint to the conference
and changing media characteristic or codec.
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Transfers and terminates calls—SIP supports the transfer of calls from one
endpoint to another. During call transfer, SIP simply establishes a session
between the transferee and a new endpoint (specified by the transferring
