This is basically a clone of the original SIP plugin, with the key difference being that it uses libre (http://creytiv.com/re.html) instead of Sofia SIP for its internal stack. As such, it provides an alternative for those who don't want to, or can't, use the Sofia-based SIP plugin. The API it exposes is exactly the same, meaning it should be pretty straightforward to switch from one plugin to another on the client side. The configuration file looks exactly the same as well. As such, you can mostly refer to the SIP plugin documentation for both.
All requests you can send in the SIPre Plugin API are asynchronous, which means all responses (successes and errors) will be delivered as events with the same transaction.
The supported requests are register , unregister , call , accept, info , message , dtmf_info , recording , hold , unhold and hangup . register can be used, as the name suggests, to register a username at a SIP registrar to call and be called, while unregister unregisters it; call is used to send an INVITE to a different SIP URI through the plugin, while accept is used to accept the call in case one is invited instead of inviting; hold and unhold can be used respectively to put a call on-hold and to resume it; info allows you to send a generic SIP INFO request, while dtmf_info is focused on using INFO for DTMF instead; message is the method you use to send a SIP message to the other peer; recording is used, instead, to record the conversation to one or more .mjr files (depending on the direction you want to record); finally, hangup can be used to terminate the communication at any time, either to hangup (BYE) an ongoing call or to cancel/decline (CANCEL/BYE) a call that hasn't started yet.
Actual API docs: TBD. For the time being, refer to the Sofia SIP plugin documentation, as while some of the features listed there may not be available in the SIPre plugin as of now, all of the messages are supposed to be formatted exactly the same way.
