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

1 files changed, 57 insertions, 0 deletions

diff --git a/include/asterisk/res_srtp.h b/include/asterisk/res_srtp.h
new file mode 100644
index 000000000..7853ead6f
--- /dev/null
+++ b/include/asterisk/res_srtp.h
@@ -0,0 +1,57 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010 FIXME
+ *
+ * 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
+ * \brief SRTP resource
+ */
+
+#ifndef _ASTERISK_RES_SRTP_H
+#define _ASTERISK_RES_SRTP_H
+
+struct ast_srtp;
+struct ast_srtp_policy;
+struct ast_rtp_instance;
+
+struct ast_srtp_cb {
+	int (*no_ctx)(struct ast_rtp_instance *rtp, unsigned long ssrc, void *data);
+};
+
+struct ast_srtp_res {
+	int (*create)(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy);
+	void (*destroy)(struct ast_srtp *srtp);
+	int (*add_stream)(struct ast_srtp *srtp, struct ast_srtp_policy *policy);
+	void (*set_cb)(struct ast_srtp *srtp, const struct ast_srtp_cb *cb, void *data);
+	int (*unprotect)(struct ast_srtp *srtp, void *buf, int *size, int rtcp);
+	int (*protect)(struct ast_srtp *srtp, void **buf, int *size, int rtcp);
+	int (*get_random)(unsigned char *key, size_t len);
+};
+
+/* Crypto suites */
+enum ast_srtp_suite {
+	AST_AES_CM_128_HMAC_SHA1_80 = 1,
+	AST_AES_CM_128_HMAC_SHA1_32 = 2,
+	AST_F8_128_HMAC_SHA1_80     = 3
+};
+
+struct ast_srtp_policy_res {
+	struct ast_srtp_policy *(*alloc)(void);
+	void (*destroy)(struct ast_srtp_policy *policy);
+	int (*set_suite)(struct ast_srtp_policy *policy, enum ast_srtp_suite suite);
+	int (*set_master_key)(struct ast_srtp_policy *policy, const unsigned char *key, size_t key_len, const unsigned char *salt, size_t salt_len);
+	void (*set_ssrc)(struct ast_srtp_policy *policy, unsigned long ssrc, int inbound);
+};
+
+#endif /* _ASTERISK_RES_SRTP_H */