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The following modules are in development:

	Generic assignment of RTP proxy for packet forwarding when local and
	remote caller is not directly routable.  This can include briding when
	both source and destination are offsite or on different subnets.  This
	will enable calls between local users and remote users when either or
	both parties are behind NAT.  It is assumed a block of rtp ports (and 
	the sip port) will be "port forwarded" to sipwitch.  All proxying is 
	transparent and hence directly usable for secure calling with ZRTP.

	This module is meant to eventually offer generic support for premise 
	routers when used by providers to offer sip/voip service to a subscriber.
	It offers rtp proxying and routing based on the assumption that all calls
	will be handed off to an external voip provider and automatic rtp
	proxy bridging between a subscribers local subnet and an isp.  In theory
	this would be deployed in an isp supplied premise router to enable a 
    local user to subscribe a series of local softphone/sip devices with a 
    remote voip service provider.

	Allows for shell scripts to be executed when things happen in sipwitch,
	such as registration/deregistration of sip users.

	Publishes sipwitch as a zeroconf sip service.  Can only be built with
	avahi only so far.

The following additional modules are currently planned:

	Stun server plugin module that uses sipwitch registration database for
	authentication.  For when sipwitch is deployed in public hosting
	scenarios.  May include an rtp proxy mode to facilate calls between
	users who may be behind NAT and also were unable to "stun"...

	Will be used to register sipwitch with multiple sip service providers
	and manage provider routing rulesets.

	Will be used to generate inter-nodal refers when multiple sipwitch
	locations are used with each location having a "block" of users &
	extension numbers.

	Will be used for future gateway handling and destination routing, rather 
	than originally planned internal one.

	Support of hotelling of user id's and telecenter routing with remote 
	provider.  To be used in conjunction with handoff to remote bayonne
	server for completing calling card operations and audio prompts.
	Telecenter NAT/RTP proxying behavior is similar to subscriber.

	For using sipwitch as a front-end "secure" telephone switch with a
	backend insecure B2BUA style IP-PBX.  All secure stations register
	with sipwitch and sipwitch then forwards (re-registers) with the IP-PBX.
	All destinations not reachable in the secure domain are referred to
	(forwarded to) the insecure IP-PBX to handle.