Operation Manual

AT-RG213 Residential VoIP Gateway - SIP Software Reference Manual
SIP Software Release 6-0-0
J613-M0524-00
46
• Callee and calling ʺnumberʺ delivery, where numbers can be
any (preferably unique) naming scheme
• Personal mobility, i.e., the ability to reach a called party under
a single, location-independent address even when the user
changes terminals
• Terminal-type negotiation and selection: a caller can be given
a choice how to reach the party, e.g., via Internet telephony,
mobile phone, an answering service, etc.
• Terminal capability negotiation
• Caller and callee authentication
• Blind and supervised call transfer
• Invitations to multicast conferences
When a user wants to call another user, the caller initiates the call with an
invite request. The request contains enough information for the called party
to join the session. If the client knows the location of the other party it can
send the request directly to their IP address. If not the client can send it to a
locally configured SIP network server. If that server is a proxy server it will
attempt to resolve the called user’s location and send the request to them.
There are many ways it can do this, such as searching the DNS or accessing
databases. Alternatively, the server may be a redirect server that may return
the called user location to the calling client for it to try directly. During the
course of locating a user, one SIP network server can, of course, proxy or
redirect the call to additional servers until it arrives at one that definitely
knows the IP address where the called user can be found.
Once found, the request is sent to the user, and from there several options
arise. In the simplest case, the user’s telephony client receives the request—
that is, the user’s phone rings. If the user takes the call, the client responds to
the invitation with the designated capabilities* of the client software and a
connection is established. If the user declines the call, the session can be
redirected to a voice mail server or to another user.
ʺDesignated capabilitiesʺ refers to the functions that the user wants to
invoke. The client software might support videoconferencing, for example,
but the user may only want to use audio conferencing. Regardless, the user
can always add functions—such as videoconferencing, white-boarding, or a
third user—by issuing another invite request to other users on the link.
SIP has two additional significant features. The first is a stateful SIP proxy
server’s ability to split or ʺforkʺ an incoming call so that several extensions
can be rung at once. The first extension to answer takes the call. This feature
is handy if a user is working between two locations (a lab and an office, for
example), or where someone is ringing both a boss and their secretary.
The second significant feature is SIP’s unique ability to return different
media types. Take the example of a user contacting a company. When the
SIP server receives the client’s connection request, it can return to the
customer’s phone client via a Web Interactive Voice Response page (IVR or
could use the term Interactive Web Response or IWR), with the extensions of
the available departments or users provided on the list. Clicking the
appropriate link sends an invitation to that user to set up a call.