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

channels/sip/srtp.c

@@ -0,0 +1,51 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2006 - 2007, Mikael Magnusson
+ *
+ * Mikael Magnusson <mikma@users.sourceforge.net>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file sip_srtp.c
+ *
+ * \brief SIP Secure RTP (SRTP)
+ *
+ * Specified in RFC 3711
+ *
+ * \author Mikael Magnusson <mikma@users.sourceforge.net>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/utils.h"
+#include "include/srtp.h"
+
+struct sip_srtp *sip_srtp_alloc(void)
+{
+	struct sip_srtp *srtp;
+
+	srtp = ast_calloc(1, sizeof(*srtp));
+
+	return srtp;
+}
+
+void sip_srtp_destroy(struct sip_srtp *srtp)
+{
+	if (srtp->crypto) {
+		sdp_crypto_destroy(srtp->crypto);
+	}
+	srtp->crypto = NULL;
+	ast_free(srtp);
+}