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Peer SIP Call Server - SIP Witch is a call server for the SIP protocol. As a
call server it services call registration for SIP devices and destination
routing through SIP gateways.  SIP Witch does not perform codec operations or
media proxying and thereby enables SIP endpoints to directly peer negotiate
call setting and process peer to peer media streaming even when over multiple
SIP Witch call nodes.  This means SIP Witch operates without introducing
additional media latency or offering a central point for audio capture, unlike
"IP-PBX" solutions such as Asterisk, Yate, CallWeaver, FreeSwitch, etc.

The key differences between SIP Witch and IP PBX packages is that it acts as a
call server only, and not a media server or media proxy.  This means SIP Witch
supports peer to peer endpoint media streaming, which facilitates secure
calling media protocols such as ZRTP, as well as generally intercept free
media communications.  SIP Witch uses destination routing, so unlike IP PBX's
like Asterisk, it does not establish call sessions before routing.  This means
faster and more scalable call handling.

SIP Witch is designed to support network scaling of telephony services, rather
than the heavily compute-bound solutions we find in use today.  This means a
call node has a local authentication/registration database, and this is
mirrored, so that any active call node in a cluster can accept and service a
call.  This allows for the possibility of live failover support in the future
as well.

SIP Witch is not a SIP "router", and does not try to address the same things
as iptel "Ser".  It is being designed to be best used either to create an
on-premise SIP telephone system, telecenter server, or call center, when used
in combination with GNU Bayonne, or as a hosted SIP telephone system.  One
important feature will include direct use of URI routing to support direct
peer to peer calls between service domains over the public internet without
needing mediation of an intermediary "service provider" so that people can
publish and call sip: uri's unconstrained.  This is about freedom to
communicate and the removal of artifical barriers and constraints whether
imposed by monopoly service providers or by governments.