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, 403 insertions, 0 deletions

diff --git a/res/res_srtp.c b/res/res_srtp.c
new file mode 100644
index 000000000..8b753ff87
--- /dev/null
+++ b/res/res_srtp.c
@@ -0,0 +1,403 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2005, 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.
+ *
+ * Builds on libSRTP http://srtp.sourceforge.net
+ */
+
+/*! \file res_srtp.c
+ *
+ * \brief Secure RTP (SRTP)
+ *
+ * Secure RTP (SRTP)
+ * Specified in RFC 3711.
+ *
+ * \author Mikael Magnusson <mikma@users.sourceforge.net>
+ */
+
+/*** MODULEINFO
+         <depend>srtp</depend>
+***/
+
+/* The SIP channel will automatically use sdescriptions if received in a SDP offer,
+   and res_srtp is loaded. SRTP with sdescriptions key exchange can be activated
+  in outgoing offers by setting _SIPSRTP_CRYPTO=enable in extension.conf before executing Dial
+
+  The dial fails if the callee doesn't support SRTP and sdescriptions.
+
+  exten => 2345,1,Set(_SIPSRTP_CRYPTO=enable)
+  exten => 2345,2,Dial(SIP/1001)
+*/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <srtp/srtp.h>
+
+#include "asterisk/lock.h"
+#include "asterisk/sched.h"
+#include "asterisk/module.h"
+#include "asterisk/options.h"
+#include "asterisk/rtp_engine.h"
+
+struct ast_srtp {
+	struct ast_rtp_instance *rtp;
+	srtp_t session;
+	const struct ast_srtp_cb *cb;
+	void *data;
+	unsigned char buf[8192 + AST_FRIENDLY_OFFSET];
+	unsigned int has_stream:1;
+};
+
+struct ast_srtp_policy {
+	srtp_policy_t sp;
+};
+
+static int g_initialized = 0;
+
+/* SRTP functions */
+static int ast_srtp_create(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy);
+static void ast_srtp_destroy(struct ast_srtp *srtp);
+static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy);
+
+static int ast_srtp_unprotect(struct ast_srtp *srtp, void *buf, int *len, int rtcp);
+static int ast_srtp_protect(struct ast_srtp *srtp, void **buf, int *len, int rtcp);
+static void ast_srtp_set_cb(struct ast_srtp *srtp, const struct ast_srtp_cb *cb, void *data);
+static int ast_srtp_get_random(unsigned char *key, size_t len);
+
+/* Policy functions */
+static struct ast_srtp_policy *ast_srtp_policy_alloc(void);
+static void ast_srtp_policy_destroy(struct ast_srtp_policy *policy);
+static int ast_srtp_policy_set_suite(struct ast_srtp_policy *policy, enum ast_srtp_suite suite);
+static int ast_srtp_policy_set_master_key(struct ast_srtp_policy *policy, const unsigned char *key, size_t key_len, const unsigned char *salt, size_t salt_len);
+static void ast_srtp_policy_set_ssrc(struct ast_srtp_policy *policy, unsigned long ssrc, int inbound);
+
+static struct ast_srtp_res srtp_res = {
+	.create = ast_srtp_create,
+	.destroy = ast_srtp_destroy,
+	.add_stream = ast_srtp_add_stream,
+	.set_cb = ast_srtp_set_cb,
+	.unprotect = ast_srtp_unprotect,
+	.protect = ast_srtp_protect,
+	.get_random = ast_srtp_get_random
+};
+
+static struct ast_srtp_policy_res policy_res = {
+	.alloc = ast_srtp_policy_alloc,
+	.destroy = ast_srtp_policy_destroy,
+	.set_suite = ast_srtp_policy_set_suite,
+	.set_master_key = ast_srtp_policy_set_master_key,
+	.set_ssrc = ast_srtp_policy_set_ssrc
+};
+
+static const char *srtp_errstr(int err)
+{
+	switch(err) {
+	case err_status_ok:
+		return "nothing to report";
+	case err_status_fail:
+		return "unspecified failure";
+	case err_status_bad_param:
+		return "unsupported parameter";
+	case err_status_alloc_fail:
+		return "couldn't allocate memory";
+	case err_status_dealloc_fail:
+		return "couldn't deallocate properly";
+	case err_status_init_fail:
+		return "couldn't initialize";
+	case err_status_terminus:
+		return "can't process as much data as requested";
+	case err_status_auth_fail:
+		return "authentication failure";
+	case err_status_cipher_fail:
+		return "cipher failure";
+	case err_status_replay_fail:
+		return "replay check failed (bad index)";
+	case err_status_replay_old:
+		return "replay check failed (index too old)";
+	case err_status_algo_fail:
+		return "algorithm failed test routine";
+	case err_status_no_such_op:
+		return "unsupported operation";
+	case err_status_no_ctx:
+		return "no appropriate context found";
+	case err_status_cant_check:
+		return "unable to perform desired validation";
+	case err_status_key_expired:
+		return "can't use key any more";
+	default:
+		return "unknown";
+	}
+}
+
+static struct ast_srtp *res_srtp_new(void)
+{
+	struct ast_srtp *srtp;
+
+	if (!(srtp = ast_calloc(1, sizeof(*srtp)))) {
+		ast_log(LOG_ERROR, "Unable to allocate memory for srtp\n");
+		return NULL;
+	}
+
+	return srtp;
+}
+
+/*
+  struct ast_srtp_policy
+*/
+static void srtp_event_cb(srtp_event_data_t *data)
+{
+	switch (data->event) {
+	case event_ssrc_collision:
+		ast_debug(1, "SSRC collision\n");
+		break;
+	case event_key_soft_limit:
+		ast_debug(1, "event_key_soft_limit\n");
+		break;
+	case event_key_hard_limit:
+		ast_debug(1, "event_key_hard_limit\n");
+		break;
+	case event_packet_index_limit:
+		ast_debug(1, "event_packet_index_limit\n");
+		break;
+	}
+}
+
+static void ast_srtp_policy_set_ssrc(struct ast_srtp_policy *policy,
+		unsigned long ssrc, int inbound)
+{
+	if (ssrc) {
+		policy->sp.ssrc.type = ssrc_specific;
+		policy->sp.ssrc.value = ssrc;
+	} else {
+		policy->sp.ssrc.type = inbound ? ssrc_any_inbound : ssrc_any_outbound;
+	}
+}
+
+static struct ast_srtp_policy *ast_srtp_policy_alloc()
+{
+	struct ast_srtp_policy *tmp;
+
+	if (!(tmp = ast_calloc(1, sizeof(*tmp)))) {
+		ast_log(LOG_ERROR, "Unable to allocate memory for srtp_policy\n");
+	}
+
+	return tmp;
+}
+
+static void ast_srtp_policy_destroy(struct ast_srtp_policy *policy)
+{
+	if (policy->sp.key) {
+		ast_free(policy->sp.key);
+		policy->sp.key = NULL;
+	}
+	ast_free(policy);
+}
+
+static int policy_set_suite(crypto_policy_t *p, enum ast_srtp_suite suite)
+{
+	switch (suite) {
+	case AST_AES_CM_128_HMAC_SHA1_80:
+		p->cipher_type = AES_128_ICM;
+		p->cipher_key_len = 30;
+		p->auth_type = HMAC_SHA1;
+		p->auth_key_len = 20;
+		p->auth_tag_len = 10;
+		p->sec_serv = sec_serv_conf_and_auth;
+		return 0;
+
+	case AST_AES_CM_128_HMAC_SHA1_32:
+		p->cipher_type = AES_128_ICM;
+		p->cipher_key_len = 30;
+		p->auth_type = HMAC_SHA1;
+		p->auth_key_len = 20;
+		p->auth_tag_len = 4;
+		p->sec_serv = sec_serv_conf_and_auth;
+		return 0;
+
+	default:
+		ast_log(LOG_ERROR, "Invalid crypto suite: %d\n", suite);
+		return -1;
+	}
+}
+
+static int ast_srtp_policy_set_suite(struct ast_srtp_policy *policy, enum ast_srtp_suite suite)
+{
+	return policy_set_suite(&policy->sp.rtp, suite) | policy_set_suite(&policy->sp.rtcp, suite);
+}
+
+static int ast_srtp_policy_set_master_key(struct ast_srtp_policy *policy, const unsigned char *key, size_t key_len, const unsigned char *salt, size_t salt_len)
+{
+	size_t size = key_len + salt_len;
+	unsigned char *master_key;
+
+	if (policy->sp.key) {
+		ast_free(policy->sp.key);
+		policy->sp.key = NULL;
+	}
+
+	if (!(master_key = ast_calloc(1, size))) {
+		return -1;
+	}
+
+	memcpy(master_key, key, key_len);
+	memcpy(master_key + key_len, salt, salt_len);
+
+	policy->sp.key = master_key;
+
+	return 0;
+}
+
+static int ast_srtp_get_random(unsigned char *key, size_t len)
+{
+	return crypto_get_random(key, len) != err_status_ok ? -1: 0;
+}
+
+static void ast_srtp_set_cb(struct ast_srtp *srtp, const struct ast_srtp_cb *cb, void *data)
+{
+	if (!srtp) {
+		return;
+	}
+
+	srtp->cb = cb;
+	srtp->data = data;
+}
+
+/* Vtable functions */
+static int ast_srtp_unprotect(struct ast_srtp *srtp, void *buf, int *len, int rtcp)
+{
+	int res = 0;
+	int i;
+	struct ast_rtp_instance_stats stats = {0,};
+
+	for (i = 0; i < 2; i++) {
+		res = rtcp ? srtp_unprotect_rtcp(srtp->session, buf, len) : srtp_unprotect(srtp->session, buf, len);
+		if (res != err_status_no_ctx) {
+			break;
+		}
+
+		if (srtp->cb && srtp->cb->no_ctx) {
+			if (ast_rtp_instance_get_stats(srtp->rtp, &stats, AST_RTP_INSTANCE_STAT_REMOTE_SSRC)) {
+				break;
+			}
+			if (srtp->cb->no_ctx(srtp->rtp, stats.remote_ssrc, srtp->data) < 0) {
+				break;
+			}
+		} else {
+			break;
+		}
+	}
+
+	if (res != err_status_ok && res != err_status_replay_fail ) {
+		ast_debug(1, "SRTP unprotect: %s\n", srtp_errstr(res));
+		return -1;
+	}
+
+	return *len;
+}
+
+static int ast_srtp_protect(struct ast_srtp *srtp, void **buf, int *len, int rtcp)
+{
+	int res;
+
+	if ((*len + SRTP_MAX_TRAILER_LEN) > sizeof(srtp->buf)) {
+		return -1;
+	}
+
+	memcpy(srtp->buf, *buf, *len);
+
+	if ((res = rtcp ? srtp_protect_rtcp(srtp->session, srtp->buf, len) : srtp_protect(srtp->session, srtp->buf, len)) != err_status_ok && res != err_status_replay_fail) {
+		ast_debug(1, "SRTP protect: %s\n", srtp_errstr(res));
+		return -1;
+	}
+
+	*buf = srtp->buf;
+	return *len;
+}
+
+static int ast_srtp_create(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy)
+{
+	struct ast_srtp *temp;
+
+	if (!(temp = res_srtp_new())) {
+		return -1;
+	}
+
+	if (srtp_create(&temp->session, &policy->sp) != err_status_ok) {
+		return -1;
+	}
+
+	temp->rtp = rtp;
+	*srtp = temp;
+
+	return 0;
+}
+
+static void ast_srtp_destroy(struct ast_srtp *srtp)
+{
+	if (srtp->session) {
+		srtp_dealloc(srtp->session);
+	}
+
+	ast_free(srtp);
+}
+
+static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy)
+{
+	if (!srtp->has_stream && srtp_add_stream(srtp->session, &policy->sp) != err_status_ok) {
+		return -1;
+	}
+
+	srtp->has_stream = 1;
+
+	return 0;
+}
+
+static int res_srtp_init(void)
+{
+	if (g_initialized) {
+		return 0;
+	}
+
+	if (srtp_init() != err_status_ok) {
+		return -1;
+	}
+
+	srtp_install_event_handler(srtp_event_cb);
+
+	return ast_rtp_engine_register_srtp(&srtp_res, &policy_res);
+}
+
+/*
+ * Exported functions
+ */
+
+static int load_module(void)
+{
+	return res_srtp_init();
+}
+
+static int unload_module(void)
+{
+	ast_rtp_engine_unregister_srtp();
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS, "Secure RTP (SRTP)",
+	.load = load_module,
+	.unload = unload_module,
+);