Merge in the RTP engine API.

This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3

main/asterisk.c

@@ -120,7 +120,6 @@ int daemon(int, int);  /* defined in libresolv of all places */
 #include "asterisk/cdr.h"
 #include "asterisk/pbx.h"
 #include "asterisk/enum.h"
-#include "asterisk/rtp.h"
 #include "asterisk/http.h"
 #include "asterisk/udptl.h"
 #include "asterisk/app.h"
@@ -3579,7 +3578,6 @@ int main(int argc, char *argv[])
 		exit(1);
 	}
 
-	ast_rtp_init();
 	ast_dsp_init();
 	ast_udptl_init();