Asterisk/res/res_srtp.c

/*
 * 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>
	<use type="external">openssl</use>
	<support_level>core</support_level>
***/

/* See https://wiki.asterisk.org/wiki/display/AST/Secure+Calling */

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#if HAVE_SRTP_VERSION > 1
# include <srtp2/srtp.h>
# include "srtp/srtp_compat.h"
# include <openssl/rand.h>
#else
# include <srtp/srtp.h>
# ifdef HAVE_OPENSSL
#  include <openssl/rand.h>
# else
#  include <srtp/crypto_kernel.h>
# endif
#endif

#include "asterisk/lock.h"
#include "asterisk/sched.h"
#include "asterisk/module.h"
#include "asterisk/options.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/astobj2.h"

struct ast_srtp {
	struct ast_rtp_instance *rtp;
	struct ao2_container *policies;
	srtp_t session;
	const struct ast_srtp_cb *cb;
	void *data;
	int warned;
	unsigned char buf[8192 + AST_FRIENDLY_OFFSET];
	unsigned char rtcpbuf[8192 + AST_FRIENDLY_OFFSET];
};

struct ast_srtp_policy {
	srtp_policy_t sp;
};

/*! Tracks whether or not we've initialized the libsrtp library */
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 int ast_srtp_replace(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_change_source(struct ast_srtp *srtp, unsigned int from_ssrc, unsigned int to_ssrc);

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,
	.replace = ast_srtp_replace,
	.destroy = ast_srtp_destroy,
	.add_stream = ast_srtp_add_stream,
	.change_source = ast_srtp_change_source,
	.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 int policy_hash_fn(const void *obj, const int flags)
{
	const struct ast_srtp_policy *policy = obj;

	return policy->sp.ssrc.type == ssrc_specific ? policy->sp.ssrc.value : policy->sp.ssrc.type;
}

static int policy_cmp_fn(void *obj, void *arg, int flags)
{
	const struct ast_srtp_policy *one = obj, *two = arg;

	return one->sp.ssrc.type == two->sp.ssrc.type && one->sp.ssrc.value == two->sp.ssrc.value;
}

static struct ast_srtp_policy *find_policy(struct ast_srtp *srtp, const srtp_policy_t *policy, int flags)
{
	struct ast_srtp_policy tmp = {
		.sp = {
			.ssrc.type = policy->ssrc.type,
			.ssrc.value = policy->ssrc.value,
		},
	};

	return ao2_t_find(srtp->policies, &tmp, flags, "Looking for policy");
}

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;
	}

	srtp->policies = ao2_t_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, 5,
		policy_hash_fn, NULL, policy_cmp_fn, "SRTP policy container");
	if (!srtp->policies) {
		ast_free(srtp);
		return NULL;
	}

	srtp->warned = 1;

	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 void policy_destructor(void *obj)
{
	struct ast_srtp_policy *policy = obj;

	if (policy->sp.key) {
		ast_free(policy->sp.key);
		policy->sp.key = NULL;
	}
}

static struct ast_srtp_policy *ast_srtp_policy_alloc()
{
	struct ast_srtp_policy *tmp;

	if (!(tmp = ao2_t_alloc(sizeof(*tmp), policy_destructor, "Allocating policy"))) {
		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)
{
	ao2_t_ref(policy, -1, "Destroying 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: %u\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)
{
#ifdef HAVE_OPENSSL
	return RAND_bytes(key, len) > 0 ? 0: -1;
#else
	return crypto_get_random(key, len) != err_status_ok ? -1: 0;
#endif
}

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;
	int retry = 0;
	struct ast_rtp_instance_stats stats = {0,};

tryagain:

	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 (retry == 0  && res == err_status_replay_old) {
		ast_log(AST_LOG_NOTICE, "SRTP unprotect failed with %s, retrying\n", srtp_errstr(res));

		if (srtp->session) {
			struct ast_srtp_policy *policy;
			struct ao2_iterator it;
			int policies_count;

			/* dealloc first */
			ast_debug(5, "SRTP destroy before re-create\n");
			srtp_dealloc(srtp->session);

			/* get the count */
			policies_count = ao2_container_count(srtp->policies);

			/* get the first to build up */
			it = ao2_iterator_init(srtp->policies, 0);
			policy = ao2_iterator_next(&it);

			ast_debug(5, "SRTP try to re-create\n");
			if (policy) {
				int res_srtp_create = srtp_create(&srtp->session, &policy->sp);
				if (res_srtp_create == err_status_ok) {
					ast_debug(5, "SRTP re-created with first policy\n");
					ao2_t_ref(policy, -1, "Unreffing first policy for re-creating srtp session");

					/* if we have more than one policy, add them */
					if (policies_count > 1) {
						ast_debug(5, "Add all the other %d policies\n",
							policies_count - 1);
						while ((policy = ao2_iterator_next(&it))) {
							srtp_add_stream(srtp->session, &policy->sp);
							ao2_t_ref(policy, -1, "Unreffing n-th policy for re-creating srtp session");
						}
					}

					retry++;
					ao2_iterator_destroy(&it);
					goto tryagain;
				}
				ast_log(LOG_ERROR, "SRTP session could not be re-created after unprotect failure: %s\n", srtp_errstr(res_srtp_create));

				/* If srtp_create() fails with a previously alloced session, it will have been dealloced before returning. */
				srtp->session = NULL;

				ao2_t_ref(policy, -1, "Unreffing first policy after srtp_create failed");
			}
			ao2_iterator_destroy(&it);
		}
	}

	if (!srtp->session) {
		errno = EINVAL;
		return -1;
	}

	if (res != err_status_ok && res != err_status_replay_fail ) {
		/*
		 * Authentication failures happen when an active attacker tries to
		 * insert malicious RTP packets. Furthermore, authentication failures
		 * happen, when the other party encrypts the sRTP data in an unexpected
		 * way. This happens quite often with RTCP. Therefore, when you see
		 * authentication failures, try to identify the implementation
		 * (author and product name) used by your other party. Try to investigate
		 * whether they use a custom library or an outdated version of libSRTP.
		 */
		if (rtcp) {
			ast_verb(2, "SRTCP unprotect failed because of %s\n", srtp_errstr(res));
		} else {
			if ((srtp->warned >= 10) && !((srtp->warned - 10) % 150)) {
				ast_verb(2, "SRTP unprotect failed because of %s %d\n",
					srtp_errstr(res), srtp->warned);
				srtp->warned = 11;
			} else {
				srtp->warned++;
			}
		}
		errno = EAGAIN;
		return -1;
	}

	return *len;
}

static int ast_srtp_protect(struct ast_srtp *srtp, void **buf, int *len, int rtcp)
{
	int res;
	unsigned char *localbuf;

	if ((*len + SRTP_MAX_TRAILER_LEN) > sizeof(srtp->buf)) {
		return -1;
	}

	localbuf = rtcp ? srtp->rtcpbuf : srtp->buf;

	memcpy(localbuf, *buf, *len);

	if ((res = rtcp ? srtp_protect_rtcp(srtp->session, localbuf, len) : srtp_protect(srtp->session, localbuf, len)) != err_status_ok && res != err_status_replay_fail) {
		ast_log(LOG_WARNING, "SRTP protect: %s\n", srtp_errstr(res));
		return -1;
	}

	*buf = localbuf;
	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;
	}
	ast_module_ref(ast_module_info->self);

	/* Any failures after this point can use ast_srtp_destroy to destroy the instance */
	if (srtp_create(&temp->session, &policy->sp) != err_status_ok) {
		/* Session either wasn't created or was created and dealloced. */
		temp->session = NULL;
		ast_srtp_destroy(temp);
		return -1;
	}

	temp->rtp = rtp;
	*srtp = temp;

	ao2_t_link((*srtp)->policies, policy, "Created initial policy");

	return 0;
}

static int ast_srtp_replace(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy)
{
	if ((*srtp) != NULL) {
		ast_srtp_destroy(*srtp);
	}
	return ast_srtp_create(srtp, rtp, policy);
}

static void ast_srtp_destroy(struct ast_srtp *srtp)
{
	if (srtp->session) {
		srtp_dealloc(srtp->session);
	}

	ao2_t_callback(srtp->policies, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, NULL, NULL, "Unallocate policy");
	ao2_t_ref(srtp->policies, -1, "Destroying container");

	ast_free(srtp);
	ast_module_unref(ast_module_info->self);
}

static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy)
{
	struct ast_srtp_policy *match;

	/* For existing streams, replace if its an SSRC stream, or bail if its a wildcard */
	if ((match = find_policy(srtp, &policy->sp, OBJ_POINTER))) {
		if (policy->sp.ssrc.type != ssrc_specific) {
			ast_log(AST_LOG_WARNING, "Cannot replace an existing wildcard policy\n");
			ao2_t_ref(match, -1, "Unreffing already existing policy");
			return -1;
		} else {
			if (srtp_remove_stream(srtp->session, match->sp.ssrc.value) != err_status_ok) {
				ast_log(AST_LOG_WARNING, "Failed to remove SRTP stream for SSRC %u\n", match->sp.ssrc.value);
			}
			ao2_t_unlink(srtp->policies, match, "Remove existing match policy");
			ao2_t_ref(match, -1, "Unreffing already existing policy");
		}
	}

	ast_debug(3, "Adding new policy for %s %u\n",
		policy->sp.ssrc.type == ssrc_specific ? "SSRC" : "type",
		policy->sp.ssrc.type == ssrc_specific ? policy->sp.ssrc.value : policy->sp.ssrc.type);
	if (srtp_add_stream(srtp->session, &policy->sp) != err_status_ok) {
		ast_log(AST_LOG_WARNING, "Failed to add SRTP stream for %s %u\n",
			policy->sp.ssrc.type == ssrc_specific ? "SSRC" : "type",
			policy->sp.ssrc.type == ssrc_specific ? policy->sp.ssrc.value : policy->sp.ssrc.type);
		return -1;
	}

	ao2_t_link(srtp->policies, policy, "Added additional stream");

	return 0;
}

static int ast_srtp_change_source(struct ast_srtp *srtp, unsigned int from_ssrc, unsigned int to_ssrc)
{
	struct ast_srtp_policy *match;
	struct srtp_policy_t sp = {
		.ssrc.type = ssrc_specific,
		.ssrc.value = from_ssrc,
	};
	err_status_t status;

	/* If we find a match, return and unlink it from the container so we
	 * can change the SSRC (which is part of the hash) and then have
	 * ast_srtp_add_stream link it back in if all is well */
	if ((match = find_policy(srtp, &sp, OBJ_POINTER | OBJ_UNLINK))) {
		match->sp.ssrc.value = to_ssrc;
		if (ast_srtp_add_stream(srtp, match)) {
			ast_log(LOG_WARNING, "Couldn't add stream\n");
		} else if ((status = srtp_remove_stream(srtp->session, from_ssrc))) {
			ast_debug(3, "Couldn't remove stream (%u)\n", status);
		}
		ao2_t_ref(match, -1, "Unreffing found policy in change_source");
	}

	return 0;
}

static void res_srtp_shutdown(void)
{
	srtp_install_event_handler(NULL);
	ast_rtp_engine_unregister_srtp();
#ifdef HAVE_SRTP_SHUTDOWN
	srtp_shutdown();
#endif
	g_initialized = 0;
}

static int res_srtp_init(void)
{
	if (g_initialized) {
		return 0;
	}

	if (srtp_init() != err_status_ok) {
		ast_log(AST_LOG_WARNING, "Failed to initialize libsrtp\n");
		return -1;
	}

	srtp_install_event_handler(srtp_event_cb);

	if (ast_rtp_engine_register_srtp(&srtp_res, &policy_res)) {
		ast_log(AST_LOG_WARNING, "Failed to register SRTP with rtp engine\n");
		res_srtp_shutdown();
		return -1;
	}

#ifdef HAVE_SRTP_GET_VERSION
	ast_verb(2, "%s initialized\n", srtp_get_version_string());
#else
	ast_verb(2, "libsrtp initialized\n");
#endif

	g_initialized = 1;
	return 0;
}

/*
 * Exported functions
 */

static int load_module(void)
{
	return res_srtp_init();
}

static int unload_module(void)
{
	res_srtp_shutdown();
	return 0;
}

AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Secure RTP (SRTP)",
	.support_level = AST_MODULE_SUPPORT_CORE,
	.load = load_module,
	.unload = unload_module,
	.load_pri = AST_MODPRI_CHANNEL_DEPEND,
);