Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.

As mentioned on the review for this, WebRTC has moved towards choosing
DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds
support for this but makes it available for normal SIP clients as well.

Testing has been done to ensure that this introduces no regressions with
existing behavior and also that it functions as expected.

Review: https://reviewboard.asterisk.org/r/2113/
........

Merged revisions 373229 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/sip/include/sip.h

@@ -36,6 +36,7 @@
 #include "asterisk/security_events.h"
 #include "asterisk/features.h"
 #include "asterisk/http_websocket.h"
+#include "asterisk/rtp_engine.h"
 
 #ifndef FALSE
 #define FALSE    0
@@ -1212,6 +1213,8 @@ struct sip_pvt {
 	struct ast_cc_config_params *cc_params;
 	struct sip_epa_entry *epa_entry;
 	int fromdomainport;                 /*!< Domain port to show in from field */
+
+	struct ast_rtp_dtls_cfg dtls_cfg;
 };
 
 /*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission
@@ -1361,6 +1364,8 @@ struct sip_peer {
 	enum sip_peer_type type; /*!< Distinguish between "user" and "peer" types. This is used solely for CLI and manager commands */
 	unsigned int disallowed_methods;
 	struct ast_cc_config_params *cc_params;
+
+	struct ast_rtp_dtls_cfg dtls_cfg;
 };
 
 /*!