Add SRTP support for Asterisk

After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.

Original patch by mikma, updated for trunk and revised by me.

(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel

Review: https://reviewboard.asterisk.org/r/191/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3

include/asterisk/rtp_engine.h

@@ -71,6 +71,9 @@ extern "C" {
 
 #include "asterisk/astobj2.h"
 #include "asterisk/frame.h"
+#include "asterisk/netsock.h"
+#include "asterisk/sched.h"
+#include "asterisk/res_srtp.h"
 
 /* Maximum number of payloads supported */
 #define AST_RTP_MAX_PT 256
@@ -458,6 +461,11 @@ int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *m
  */
 int ast_rtp_engine_unregister(struct ast_rtp_engine *engine);
 
+int ast_rtp_engine_register_srtp(struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res);
+
+void ast_rtp_engine_unregister_srtp(void);
+int ast_rtp_engine_srtp_is_registered(void);
+
 #define ast_rtp_glue_register(glue) ast_rtp_glue_register2(glue, ast_module_info->self)
 
 /*!
@@ -1724,6 +1732,9 @@ struct ast_rtp_glue *ast_rtp_instance_get_active_glue(struct ast_rtp_instance *i
  */
 struct ast_channel *ast_rtp_instance_get_chan(struct ast_rtp_instance *instance);
 
+int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *policy);
+struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance);
+
 #if defined(__cplusplus) || defined(c_plusplus)
 }
 #endif