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IPS 1000 Series VoIP Gateway
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SIP determines the highest level of common services between the end points. Conferences are
established using only the media capabilities that can be supported by all end points.
Determine the availability of the target end point—If a call cannot be completed because the target end
point 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 end point is
unavailable.
Establish a session between the originating and target end point—If the call can be completed, SIP
establishes a session between the end points. SIP also supports mid-call changes, such as the addition
of another end point to the conference or the changing of a media characteristic or Codec.
Handle the transfer and termination of calls—SIP supports the transfer of calls from one end point to
another. During a call transfer, SIP simply establishes a session between the transferee and a new end
point (specified by the transferring party) and terminates the session between the transferee and the
transferring party. At the end of a call, SIP terminates the sessions between all parties.
2.1.1. Components of SIP
SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user agent can
function in one of the following roles:
User agent client (UAC)—A client application that initiates the SIP request.
User agent server (UAS)—A server application that contacts the user when a SIP request is received
and that returns a response on behalf of the user.
Typically, a SIP end point is capable of functioning as both a UAC and a UAS, but functions only as one
or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the UA that
initiated the request.
From an architecture standpoint, the physical components of a SIP network can be grouped into two
categories: clients and servers.