Draft SIP.edu Requirements for Gateway Calling Line ID
Dennis Baron, MIT - 7/29/04

Background

SIP.edu primarily promotes calls from SIP devices to PBX telephones.
These calls pass through a SIP gateway to the campus PBX.  In the
canonical example the call would come as "From:alice@littleu.edu".
However, the call can come from anywhere in the internet so it could
arrive at the SIP.edu proxy as "From:17476698123@sipphone.com".

In the second phase of a SIP.edu implementation, schools register SIP
UAs with their proxy allowing calls to be received on the UA instead
of on the campus PBX telephone.  Also, while not specified in SIP.edu,
these UAs will usually want to be able to receive calls from the
campus PBX.  To allow for this a PBX extension is assigned and routed
to the SIP gateway and then to the SIP.edu proxy.  If the PBX
extension allows for DID then the SIP UA will also be able to receive
calls from the PSTN via the PBX.  If the gateway receives a calling
party number from the PBX it usually includes it in the "From:" field
so that it can be displayed on the called SIP UA.

In addition to receiving PBX calls, campus SIP UAs will often want to
place calls directly to PBX extensions.  To support this the SIP.edu
proxy forwards the PBX extension numbers to the gateway and on to the
PBX.  Calls to the PSTN via the PBX can also be supported, typically
by dialing 9 plus an E.164 number.  Since there is often a per-minute
cost associated with PSTN calls the SIP.edu proxy may want to restrict
these calls to locally registered or authenticated SIP UAs.

While SIP.edu encourages campuses to use addresses of the form
"From:alice@littleu.edu" this does not provide an E.164 number for the
SIP.edu gateway to use as the calling number when forwarding the call
to the PBX.  For this reason implementations have instead used
"From:17476698123@littleu.edu".  This provides an E.164 number that
the gateway can use for Caller-ID on the PBX and PSTN.  However, this
number cannot be authenticated by the gateway.

Gateway Requirements

1)  PBX/PSTN to SIP Calls

    In order for the called SIP UA to receive a "reply-able" address,
    the gateway SHOULD include the calling party number in the "From:"
    field.  In addition the gateway SHOULD be capable of inserting the
    domain name served by the SIP.edu proxy.  An example would be
    "From:17476698123@bigu.edu".  The gateway MAY also allow digit
    manipulation so that digits may be removed or added.  This would
    allow the gateway to originate calls as "From:917476698123@bigu.edu",
    conforming to a local dialing plan on the SIP.edu proxy.

2)  SIP to PBX/PSTN Calls

    Since the "From:" field may contain an E.164 number that cannot be
    authenticated by the gateway, the gateway SHOULD be capable of
    using another SIP header to extract the calling party number to be
    presented to the PBX.  Since it trusts the SIP.edu proxy the proxy
    can perform authentication of the "From:" field, inserting or
    stripping this header as appropriate.  In addition the SIP.edu
    proxy can generate this field by mapping of the "From:" field to
    to the E.164 number associated with the caller - eg. authenticating
    a call "From:bob@bigu.edu" and inserting a header with Bob's E.164
    number.