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This seemed a good place to talk about those few things that do work
in the initial release:


The basic registry function allows one or more devices to register under a
single id.  Multi-device registration can be used for example if one wants
to assign a laptop softphone under the same user id/pwd as the desktop phone
and then have both ring when the target id is dialed.

Registry items can have a logical (email) name and also optionally bound to
an extension #.  The latter is for the convenience of places that use SIP
phone instruments rather than softphone applications.  This means a given
station can be dialed by it's extension # and by a named uri.  The latter is
used mostly for external (public) access to a user's extension if allowed.

There is a class of service system that supports whether a given registry
entry supports multi-target registrations, allows external access to uri's,

The registry supports peer authentication mode and external validation
where external SIP devices and applications can use sip witch as a SIP 
authentication resource, to query whether a given uri is active, etc.


At the time of writing, basic call processing is being completed.  Each call is
managed as a seperate session, and one or more targets can be invited into a
call and rang concurrently.  ring/busy/failure states are handled, and all SIP
responses are mapped into success/ring/busy or failure.  In theory when for
the first target ua to respond with a 200 OK to accept a call and offers it's
sdp, then any other invited targets are released and a single joined session is
formed passing the sdp back up the originating call leg.

Unanswered calls, when all targets have refused to answer, are reported to the
originating ua as "declined" calls.  If all invited targets report busy, then
the originating ua is notified "busy here".  We do not distinguish between
temporary busy, busy everywhere, etc, but report a consolidated busy state to
the target.

At present both registered ua's and "remote" uri's can be dialed, either by
ext# if they have one or by uri name.  Display management is part of the
server, and the display is regenerated with a custom "from" message for each
call leg as needed by the server. 

Routing rules and plugins can also be used to push local "uri's" onto remote
servers in multi-nodal calling.  When a call is redirected to another sipwitch
node, the authentication field is pre-set.  This means all nodes must share
both common authentication information and the "same" SIP realm. When a
registerable user agent appears on a different switch than the one it is
registered for, a complete temporary registry entry is created on the target
switch.  This allows correct line appearence (ua display of the from line) 
for internodal calls as if they were local extensions.  At present, outbound
redirections are counted as "external" calls, but inbound internodals are
treated as if local (temporary) extensions.


SIP info messages are hoped between connected sessions.  There is no proper
error responses yet if info's cannot be handled, so they are all acknowledged
as 200 OK even if info actually could not be processed.

Basic instant messaging support over SIP exists.  However, it can only
successfully deliver to registered targets, and does not queue messages for
offline users for later delivery yet.  However, queing support and delivery
to external targets will be added.

What we will never do:

sipwitch does NOT operate as a "transparent" proxy and is not meant to do so.
Each call leg is a separately managed session of the server with all sip
message content artificially generated as appropriate and needed by the server
itself.  All SDP and media actions happen between the peer devices.  

There is no media processing in sipwitch and no desire to add the latency,
connection states, and conversion issues that would engender.  This also allows
sipwitch to be used intercept-free with secure telephone units such as ZRTP
clients and without licensing issues associated with patent encumbered codecs.
If the UA's support it, video as well as audio can be used since again sipwitch
has no effect on nor participates in peer media processing.

There is no media application processing in sipwitch.  The intent is to use
sipwitch with multi-protocol gateway servers mapped through sipwitch routing
plans to interconnect with the pstn, the bastardized sip offered by tethered
"service providers", with iax nodes, etc.  Applications like ivr, voice mail,
etc, can be provided by an external SIP capable application server such as
GNU Bayonne.