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: http://svn.digium.com/svn/asterisk/trunk@268894 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 26 insertions, 0 deletions

diff --git a/main/global_datastores.c b/main/global_datastores.c
index 5b9630257..cddad6ec1 100644
--- a/main/global_datastores.c
+++ b/main/global_datastores.c
@@ -84,3 +84,29 @@ const struct ast_datastore_info dialed_interface_info = {
 	.destroy = dialed_interface_destroy,
 	.duplicate = dialed_interface_duplicate,
 };
+
+static void secure_call_store_destroy(void *data)
+{
+	struct ast_secure_call_store *store = data;
+
+	ast_free(store);
+}
+
+static void *secure_call_store_duplicate(void *data)
+{
+	struct ast_secure_call_store *old = data;
+	struct ast_secure_call_store *new;
+
+	if (!(new = ast_calloc(1, sizeof(*new)))) {
+		return NULL;
+	}
+	new->signaling = old->signaling;
+	new->media = old->media;
+
+	return new;
+}
+const struct ast_datastore_info secure_call_info = {
+	.type = "encrypt-call",
+	.destroy = secure_call_store_destroy,
+	.duplicate = secure_call_store_duplicate,
+};