res_srtp.c

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/*
 * 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
 *
 * \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>
***/
 
/*** MAKEOPTS
<category name="MENUSELECT_RES" displayname="Resource Modules">
    <member name="ENABLE_SRTP_AES_192" displayname="Enable AES 192 cipher suite in res_srtp" touch_on_change="res/res_srtp.c">
        <defaultenabled>no</defaultenabled>
        <support_level>option</support_level>
        <depend>srtp</depend>
        <depend>res_srtp</depend>
        <depend>HAVE_SRTP_192</depend>
    </member>
    <member name="ENABLE_SRTP_AES_256" displayname="Enable AES 256 cipher suite in res_srtp" touch_on_change="res/res_srtp.c">
        <defaultenabled>no</defaultenabled>
        <support_level>option</support_level>
        <depend>srtp</depend>
        <depend>res_srtp</depend>
        <depend>HAVE_SRTP_256</depend>
    </member>
    <member name="ENABLE_SRTP_AES_GCM" displayname="Enable AES GCM cipher suite in res_srtp" touch_on_change="res/res_srtp.c">
        <defaultenabled>no</defaultenabled>
        <support_level>option</support_level>
        <depend>srtp</depend>
        <depend>res_srtp</depend>
        <depend>HAVE_SRTP_GCM</depend>
    </member>
</category>
***/
 
/* See https://docs.asterisk.org/Deployment/Secure-Calling/ */
 
#include "asterisk.h"                   /* for NULL, size_t, memcpy, etc */
 
#include <math.h>                       /* for pow */
 
#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/astobj2.h"           /* for ao2_t_ref, etc */
#include "asterisk/frame.h"             /* for AST_FRIENDLY_OFFSET */
#include "asterisk/logger.h"            /* for ast_log, ast_debug, etc */
#include "asterisk/module.h"            /* for ast_module_info, etc */
#include "asterisk/sdp_srtp.h"
#include "asterisk/res_srtp.h"          /* for ast_srtp_cb, ast_srtp_suite, etc */
#include "asterisk/rtp_engine.h"        /* for ast_rtp_engine_register_srtp, etc */
#include "asterisk/utils.h"             /* for ast_free, ast_calloc */
 
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:
        crypto_policy_set_aes_cm_128_hmac_sha1_80(p);
        return 0;
 
    case AST_AES_CM_128_HMAC_SHA1_32:
        crypto_policy_set_aes_cm_128_hmac_sha1_32(p);
        return 0;
 
#if defined(HAVE_SRTP_192) && defined(ENABLE_SRTP_AES_192)
    case AST_AES_CM_192_HMAC_SHA1_80:
        crypto_policy_set_aes_cm_192_hmac_sha1_80(p);
        return 0;
 
    case AST_AES_CM_192_HMAC_SHA1_32:
        crypto_policy_set_aes_cm_192_hmac_sha1_32(p);
        return 0;
#endif
#if defined(HAVE_SRTP_256) && defined(ENABLE_SRTP_AES_256)
    case AST_AES_CM_256_HMAC_SHA1_80:
        crypto_policy_set_aes_cm_256_hmac_sha1_80(p);
        return 0;
 
    case AST_AES_CM_256_HMAC_SHA1_32:
        crypto_policy_set_aes_cm_256_hmac_sha1_32(p);
        return 0;
#endif
#if defined(HAVE_SRTP_GCM) && defined(ENABLE_SRTP_AES_GCM)
    case AST_AES_GCM_128:
        crypto_policy_set_aes_gcm_128_16_auth(p);
        return 0;
 
    case AST_AES_GCM_128_8:
        crypto_policy_set_aes_gcm_128_8_auth(p);
        return 0;
#endif
#if defined(HAVE_SRTP_GCM) && defined(ENABLE_SRTP_AES_GCM) && defined(ENABLE_SRTP_AES_256)
    case AST_AES_GCM_256:
        crypto_policy_set_aes_gcm_256_16_auth(p);
        return 0;
 
    case AST_AES_GCM_256_8:
        crypto_policy_set_aes_gcm_256_8_auth(p);
        return 0;
#endif
 
    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 flags)
{
    int res = 0;
    int i;
    int rtcp  = (flags & 0x01) >> 0;
    int retry = (flags & 0x02) >> 1;
    struct ast_rtp_instance_stats stats = {0,};
 
tryagain:
 
    if (!srtp->session) {
        ast_log(LOG_ERROR, "SRTP unprotect %s - missing session\n", rtcp ? "rtcp" : "rtp");
        errno = EINVAL;
        return -1;
    }
 
    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 on SSRC %u because of %s\n",
                ast_rtp_instance_get_ssrc(srtp->rtp), srtp_errstr(res));
        } else {
            if ((srtp->warned >= 10) && !((srtp->warned - 10) % 150)) {
                ast_verb(2, "SRTP unprotect failed on SSRC %u because of %s %d\n",
                    ast_rtp_instance_get_ssrc(srtp->rtp), 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 (!srtp->session) {
        ast_log(LOG_ERROR, "SRTP protect %s - missing session\n", rtcp ? "rtcp" : "rtp");
        errno = EINVAL;
        return -1;
    }
 
    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;
    int status;
 
    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 */
    status = srtp_create(&temp->session, &policy->sp);
    if (status != err_status_ok) {
        /* Session either wasn't created or was created and dealloced. */
        temp->session = NULL;
        ast_srtp_destroy(temp);
        ast_log(LOG_ERROR, "Failed to create srtp session on rtp instance (%p) - %s\n",
                rtp, srtp_errstr(status));
        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)
{
    struct ast_srtp *old = *srtp;
    int res = ast_srtp_create(srtp, rtp, policy);
 
    if (!res && old) {
        ast_srtp_destroy(old);
    }
 
    if (res) {
        ast_log(LOG_ERROR, "Failed to replace srtp (%p) on rtp instance (%p) "
                "- keeping old\n", *srtp, rtp);
    }
 
    return res;
}
 
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;
}
 
struct ast_sdp_crypto {
    char *a_crypto;
    unsigned char local_key[SRTP_MAX_KEY_LEN];
    int tag;
    char local_key64[((SRTP_MAX_KEY_LEN) * 8 + 5) / 6 + 1];
    unsigned char remote_key[SRTP_MAX_KEY_LEN];
    int key_len;
};
 
static void res_sdp_crypto_dtor(struct ast_sdp_crypto *crypto)
{
    if (crypto) {
        ast_free(crypto->a_crypto);
        crypto->a_crypto = NULL;
        ast_free(crypto);
 
        ast_module_unref(ast_module_info->self);
    }
}
 
static struct ast_sdp_crypto *crypto_init_keys(struct ast_sdp_crypto *p, const int key_len)
{
    unsigned char remote_key[key_len];
 
    if (srtp_res.get_random(p->local_key, key_len) < 0) {
        return NULL;
    }
 
    ast_base64encode(p->local_key64, p->local_key, key_len, sizeof(p->local_key64));
 
    p->key_len = ast_base64decode(remote_key, p->local_key64, sizeof(remote_key));
 
    if (p->key_len != key_len) {
        ast_log(LOG_ERROR, "base64 encode/decode bad len %d != %d\n", p->key_len, key_len);
        return NULL;
    }
 
    if (memcmp(remote_key, p->local_key, p->key_len)) {
        ast_log(LOG_ERROR, "base64 encode/decode bad key\n");
        return NULL;
    }
 
    ast_debug(1 , "local_key64 %s len %zu\n", p->local_key64, strlen(p->local_key64));
 
    return p;
}
 
static struct ast_sdp_crypto *sdp_crypto_alloc(const int key_len)
{
    struct ast_sdp_crypto *p, *result;
 
    if (!(p = ast_calloc(1, sizeof(*p)))) {
        return NULL;
    }
    p->tag = 1;
    ast_module_ref(ast_module_info->self);
 
    /* default is a key which uses AST_AES_CM_128_HMAC_SHA1_xx */
    result = crypto_init_keys(p, key_len);
    if (!result) {
        res_sdp_crypto_dtor(p);
    }
 
    return result;
}
 
static struct ast_sdp_crypto *res_sdp_crypto_alloc(void)
{
    return sdp_crypto_alloc(SRTP_MASTER_KEY_LEN);
}
 
static int res_sdp_crypto_build_offer(struct ast_sdp_crypto *p, int taglen)
{
    int res;
 
    /* Rebuild the crypto line */
    ast_free(p->a_crypto);
    p->a_crypto = NULL;
 
    if ((taglen & 0x007f) == 8) {
        res = ast_asprintf(&p->a_crypto, "%d AEAD_AES_%d_GCM_%d inline:%s",
            p->tag, 128 + ((taglen & 0x0300) >> 2), taglen & 0x007f, p->local_key64);
    } else if ((taglen & 0x007f) == 16) {
        res = ast_asprintf(&p->a_crypto, "%d AEAD_AES_%d_GCM inline:%s",
            p->tag, 128 + ((taglen & 0x0300) >> 2), p->local_key64);
    } else if ((taglen & 0x0300) && !(taglen & 0x0080)) {
        res = ast_asprintf(&p->a_crypto, "%d AES_%d_CM_HMAC_SHA1_%d inline:%s",
            p->tag, 128 + ((taglen & 0x0300) >> 2), taglen & 0x007f, p->local_key64);
    } else {
        res = ast_asprintf(&p->a_crypto, "%d AES_CM_%d_HMAC_SHA1_%d inline:%s",
            p->tag, 128 + ((taglen & 0x0300) >> 2), taglen & 0x007f, p->local_key64);
    }
    if (res == -1 || !p->a_crypto) {
        ast_log(LOG_ERROR, "Could not allocate memory for crypto line\n");
        return -1;
    }
 
    ast_debug(1, "Crypto line: a=crypto:%s\n", p->a_crypto);
 
    return 0;
}
 
static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, int key_len, unsigned long ssrc, int inbound)
{
    if (policy_res.set_master_key(policy, master_key, key_len, NULL, 0) < 0) {
        return -1;
    }
 
    if (policy_res.set_suite(policy, suite_val)) {
        ast_log(LOG_WARNING, "Could not set remote SRTP suite\n");
        return -1;
    }
 
    policy_res.set_ssrc(policy, ssrc, inbound);
 
    return 0;
}
 
static int crypto_activate(struct ast_sdp_crypto *p, int suite_val, unsigned char *remote_key, int key_len, struct ast_rtp_instance *rtp)
{
    struct ast_srtp_policy *local_policy = NULL;
    struct ast_srtp_policy *remote_policy = NULL;
    struct ast_rtp_instance_stats stats = {0,};
    int res = -1;
 
    if (!p) {
        return -1;
    }
 
    if (!(local_policy = policy_res.alloc())) {
        return -1;
    }
 
    if (!(remote_policy = policy_res.alloc())) {
        goto err;
    }
 
    if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_LOCAL_SSRC)) {
        goto err;
    }
 
    if (set_crypto_policy(local_policy, suite_val, p->local_key, key_len, stats.local_ssrc, 0) < 0) {
        goto err;
    }
 
    if (set_crypto_policy(remote_policy, suite_val, remote_key, key_len, 0, 1) < 0) {
        goto err;
    }
 
    /* Add the SRTP policies */
    if (ast_rtp_instance_add_srtp_policy(rtp, remote_policy, local_policy, 0)) {
        ast_log(LOG_WARNING, "Could not set SRTP policies\n");
        goto err;
    }
 
    ast_debug(1 , "SRTP policy activated\n");
    res = 0;
 
err:
    if (local_policy) {
        policy_res.destroy(local_policy);
    }
 
    if (remote_policy) {
        policy_res.destroy(remote_policy);
    }
 
    return res;
}
 
static int res_sdp_crypto_parse_offer(struct ast_rtp_instance *rtp, struct ast_sdp_srtp *srtp, const char *attr)
{
    char *str = NULL;
    char *tag = NULL;
    char *suite = NULL;
    char *key_params = NULL;
    char *key_param = NULL;
    char *session_params = NULL;
    char *key_salt = NULL;       /* The actual master key and key salt */
    char *lifetime = NULL;       /* Key lifetime (# of RTP packets) */
    char *mki = NULL;            /* Master Key Index */
    int found = 0;
    int key_len_from_sdp;
    int key_len_expected;
    int tag_from_sdp;
    int suite_val = 0;
    unsigned char remote_key[SRTP_MAX_KEY_LEN];
    int taglen;
    double sdes_lifetime;
    struct ast_sdp_crypto *crypto;
    struct ast_sdp_srtp *tmp;
 
    str = ast_strdupa(attr);
 
    tag = strsep(&str, " ");
    suite = strsep(&str, " ");
    key_params = strsep(&str, " ");
    session_params = strsep(&str, " ");
 
    if (!tag || !suite) {
        ast_log(LOG_WARNING, "Unrecognized crypto attribute a=%s\n", attr);
        return -1;
    }
 
    /* RFC4568 9.1 - tag is 1-9 digits */
    if (sscanf(tag, "%30d", &tag_from_sdp) != 1 || tag_from_sdp < 0 || tag_from_sdp > 999999999) {
        ast_log(LOG_WARNING, "Unacceptable a=crypto tag: %s\n", tag);
        return -1;
    }
 
    if (!ast_strlen_zero(session_params)) {
        ast_log(LOG_WARNING, "Unsupported crypto parameters: %s\n", session_params);
        return -1;
    }
 
    /* On egress, Asterisk sent several crypto lines in the SIP/SDP offer
       The remote party might have choosen another line than the first */
    for (tmp = srtp; tmp && tmp->crypto && tmp->crypto->tag != tag_from_sdp;) {
        tmp = AST_LIST_NEXT(tmp, sdp_srtp_list);
    }
    if (tmp) { /* tag matched an already created crypto line */
        unsigned int flags = tmp->flags;
 
        /* Make that crypto line the head of the list, not by changing the
           list structure but by exchanging the content of the list members */
        crypto = tmp->crypto;
        tmp->crypto = srtp->crypto;
        tmp->flags = srtp->flags;
        srtp->crypto = crypto;
        srtp->flags = flags;
    } else {
        crypto = srtp->crypto;
        crypto->tag = tag_from_sdp;
    }
 
    ast_clear_flag(srtp, AST_SRTP_CRYPTO_TAG_8);
    ast_clear_flag(srtp, AST_SRTP_CRYPTO_TAG_16);
    ast_clear_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
    ast_clear_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
    ast_clear_flag(srtp, AST_SRTP_CRYPTO_AES_192);
    ast_clear_flag(srtp, AST_SRTP_CRYPTO_AES_256);
    ast_clear_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
 
    if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_80")) {
        suite_val = AST_AES_CM_128_HMAC_SHA1_80;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
        key_len_expected = 30;
    } else if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_32")) {
        suite_val = AST_AES_CM_128_HMAC_SHA1_32;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
        key_len_expected = 30;
#if defined(HAVE_SRTP_192) && defined(ENABLE_SRTP_AES_192)
    } else if (!strcmp(suite, "AES_192_CM_HMAC_SHA1_80")) {
        suite_val = AST_AES_CM_192_HMAC_SHA1_80;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
        key_len_expected = 38;
    } else if (!strcmp(suite, "AES_192_CM_HMAC_SHA1_32")) {
        suite_val = AST_AES_CM_192_HMAC_SHA1_32;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
        key_len_expected = 38;
    /* RFC used a different name while in draft, some still use that */
    } else if (!strcmp(suite, "AES_CM_192_HMAC_SHA1_80")) {
        suite_val = AST_AES_CM_192_HMAC_SHA1_80;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
        key_len_expected = 38;
    } else if (!strcmp(suite, "AES_CM_192_HMAC_SHA1_32")) {
        suite_val = AST_AES_CM_192_HMAC_SHA1_32;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_192);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
        key_len_expected = 38;
#endif
#if defined(HAVE_SRTP_256) && defined(ENABLE_SRTP_AES_256)
    } else if (!strcmp(suite, "AES_256_CM_HMAC_SHA1_80")) {
        suite_val = AST_AES_CM_256_HMAC_SHA1_80;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
        key_len_expected = 46;
    } else if (!strcmp(suite, "AES_256_CM_HMAC_SHA1_32")) {
        suite_val = AST_AES_CM_256_HMAC_SHA1_32;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
        key_len_expected = 46;
    /* RFC used a different name while in draft, some still use that */
    } else if (!strcmp(suite, "AES_CM_256_HMAC_SHA1_80")) {
        suite_val = AST_AES_CM_256_HMAC_SHA1_80;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
        key_len_expected = 46;
    } else if (!strcmp(suite, "AES_CM_256_HMAC_SHA1_32")) {
        suite_val = AST_AES_CM_256_HMAC_SHA1_32;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME);
        key_len_expected = 46;
#endif
#if defined(HAVE_SRTP_GCM) && defined(ENABLE_SRTP_AES_GCM)
    } else if (!strcmp(suite, "AEAD_AES_128_GCM")) {
        suite_val = AST_AES_GCM_128;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_16);
        key_len_expected = AES_128_GCM_KEYSIZE_WSALT;
    /* RFC contained a (too) short auth tag for RTP media, some still use that */
    } else if (!strcmp(suite, "AEAD_AES_128_GCM_8")) {
        suite_val = AST_AES_GCM_128_8;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_8);
        key_len_expected = AES_128_GCM_KEYSIZE_WSALT;
#endif
#if defined(HAVE_SRTP_GCM) && defined(ENABLE_SRTP_AES_GCM) && defined(ENABLE_SRTP_AES_256)
    } else if (!strcmp(suite, "AEAD_AES_256_GCM")) {
        suite_val = AST_AES_GCM_256;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_16);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
        key_len_expected = AES_256_GCM_KEYSIZE_WSALT;
    /* RFC contained a (too) short auth tag for RTP media, some still use that */
    } else if (!strcmp(suite, "AEAD_AES_256_GCM_8")) {
        suite_val = AST_AES_GCM_256_8;
        ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_8);
        ast_set_flag(srtp, AST_SRTP_CRYPTO_AES_256);
        key_len_expected = AES_256_GCM_KEYSIZE_WSALT;
#endif
    } else {
        ast_debug(3, "Unsupported crypto suite: %s\n", suite);
        return -1;
    }
 
    while ((key_param = strsep(&key_params, ";"))) {
        unsigned int n_lifetime;
        char *method = NULL;
        char *info = NULL;
 
        method = strsep(&key_param, ":");
        info = strsep(&key_param, ";");
        sdes_lifetime = 0;
 
        if (strcmp(method, "inline")) {
            continue;
        }
 
        key_salt = strsep(&info, "|");
 
        /* The next parameter can be either lifetime or MKI */
        lifetime = strsep(&info, "|");
        if (!lifetime) {
            found = 1;
            break;
        }
 
        mki = strchr(lifetime, ':');
        if (mki) {
            mki = lifetime;
            lifetime = NULL;
        } else {
            mki = strsep(&info, "|");
        }
 
        if (mki && *mki != '1') {
            ast_log(LOG_NOTICE, "Crypto MKI handling is not supported: ignoring attribute %s\n", attr);
            continue;
        }
 
        if (lifetime) {
            if (!strncmp(lifetime, "2^", 2)) {
                char *lifetime_val = lifetime + 2;
 
                /* Exponential lifetime */
                if (sscanf(lifetime_val, "%30u", &n_lifetime) != 1) {
                    ast_log(LOG_NOTICE, "Failed to parse lifetime value in crypto attribute: %s\n", attr);
                    continue;
                }
 
                if (n_lifetime > 48) {
                    /* Yeah... that's a bit big. */
                    ast_log(LOG_NOTICE, "Crypto lifetime exponent of '%u' is a bit large; using 48\n", n_lifetime);
                    n_lifetime = 48;
                }
                sdes_lifetime = pow(2, n_lifetime);
            } else {
                /* Decimal lifetime */
                if (sscanf(lifetime, "%30u", &n_lifetime) != 1) {
                    ast_log(LOG_NOTICE, "Failed to parse lifetime value in crypto attribute: %s\n", attr);
                    continue;
                }
                sdes_lifetime = n_lifetime;
            }
 
            /* Accept anything above ~5.8 hours. Less than ~5.8; reject. */
            if (sdes_lifetime < 1048576) {
                ast_log(LOG_NOTICE, "Rejecting crypto attribute '%s': lifetime '%f' too short\n", attr, sdes_lifetime);
                continue;
            }
        }
 
        ast_debug(2, "Crypto attribute '%s' accepted with lifetime '%f', MKI '%s'\n",
            attr, sdes_lifetime, mki ? mki : "-");
 
        found = 1;
        break;
    }
 
    if (!found) {
        ast_log(LOG_NOTICE, "SRTP crypto offer not acceptable: '%s'\n", attr);
        return -1;
    }
 
    key_len_from_sdp = ast_base64decode(remote_key, key_salt, sizeof(remote_key));
    if (key_len_from_sdp != key_len_expected) {
        ast_log(LOG_WARNING, "SRTP descriptions key length is '%d', not '%d'\n",
            key_len_from_sdp, key_len_expected);
        return -1;
    }
 
    /* on default, the key is 30 (AES-128); throw that away (only) when the suite changed actually */
    /* ingress: optional, but saves one expensive call to get_random(.) */
    /*  egress: required, because the local key was communicated before the remote key is processed */
    if (crypto->key_len != key_len_from_sdp) {
        if (!crypto_init_keys(crypto, key_len_from_sdp)) {
            return -1;
        }
    } else if (!memcmp(crypto->remote_key, remote_key, key_len_from_sdp)) {
        ast_debug(1, "SRTP remote key unchanged; maintaining current policy\n");
        return 0;
    }
 
    if (key_len_from_sdp > sizeof(crypto->remote_key)) {
        ast_log(LOG_ERROR,
            "SRTP key buffer is %zu although it must be at least %d bytes\n",
            sizeof(crypto->remote_key), key_len_from_sdp);
        return -1;
    }
    memcpy(crypto->remote_key, remote_key, key_len_from_sdp);
 
    if (crypto_activate(crypto, suite_val, remote_key, key_len_from_sdp, rtp) < 0) {
        return -1;
    }
 
    if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_32)) {
        taglen = 32;
    } else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_16)) {
        taglen = 16;
    } else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_8)) {
        taglen = 8;
    } else {
        taglen = 80;
    }
    if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_256)) {
        taglen |= 0x0200;
    } else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_192)) {
        taglen |= 0x0100;
    }
    if (ast_test_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME)) {
        taglen |= 0x0080;
    }
 
    /* Finally, rebuild the crypto line */
    if (res_sdp_crypto_build_offer(crypto, taglen)) {
        return -1;
    }
 
    ast_set_flag(srtp, AST_SRTP_CRYPTO_OFFER_OK);
    return 0;
}
 
static const char *res_sdp_srtp_get_attr(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32)
{
    int taglen;
 
    if (!srtp) {
        return NULL;
    }
 
    /* Set encryption properties */
    if (!srtp->crypto) {
        if (AST_LIST_NEXT(srtp, sdp_srtp_list)) {
            srtp->crypto = res_sdp_crypto_alloc();
            ast_log(LOG_ERROR, "SRTP SDP list was not empty\n");
        } else {
            const int len = default_taglen_32 ? AST_SRTP_CRYPTO_TAG_32 : AST_SRTP_CRYPTO_TAG_80;
            const int attr[][3] = {
            /* This array creates the following list:
             * a=crypto:1 AES_CM_128_HMAC_SHA1_ ...
             * a=crypto:2 AEAD_AES_128_GCM ...
             * a=crypto:3 AES_256_CM_HMAC_SHA1_ ...
             * a=crypto:4 AEAD_AES_256_GCM ...
             * a=crypto:5 AES_192_CM_HMAC_SHA1_ ...
             * something like 'AEAD_AES_192_GCM' is not specified by the RFCs
             *
             * If you want to prefer another crypto suite or you want to
             * exclude a suite, change this array and recompile Asterisk.
             * This list cannot be changed from rtp.conf because you should
             * know what you are doing. Especially AES-192 and AES-GCM are
             * broken in many VoIP clients, see
             * https://github.com/cisco/libsrtp/pull/170
             * https://github.com/cisco/libsrtp/pull/184
             * Furthermore, AES-GCM uses a shorter crypto-suite string which
             * causes Nokia phones based on Symbian/S60 to reject the whole
             * INVITE with status 500, even if a matching suite was offered.
             * AES-256 might just waste your processor cycles, especially if
             * your TLS transport is not secured with equivalent grade, see
             * https://security.stackexchange.com/q/61361
             * Therefore, AES-128 was preferred here.
             *
             * If you want to enable one of those defines, please, go for
             * CFLAGS='-DENABLE_SRTP_AES_GCM' ./configure && sudo make install
             */
                { len, 0, 30 },
#if defined(HAVE_SRTP_GCM) && defined(ENABLE_SRTP_AES_GCM)
                { AST_SRTP_CRYPTO_TAG_16, 0, AES_128_GCM_KEYSIZE_WSALT },
#endif
#if defined(HAVE_SRTP_256) && defined(ENABLE_SRTP_AES_256)
                { len, AST_SRTP_CRYPTO_AES_256, 46 },
#endif
#if defined(HAVE_SRTP_GCM) && defined(ENABLE_SRTP_AES_GCM) && defined(ENABLE_SRTP_AES_256)
                { AST_SRTP_CRYPTO_TAG_16, AST_SRTP_CRYPTO_AES_256, AES_256_GCM_KEYSIZE_WSALT },
#endif
#if defined(HAVE_SRTP_192) && defined(ENABLE_SRTP_AES_192)
                { len, AST_SRTP_CRYPTO_AES_192, 38 },
#endif
            };
            struct ast_sdp_srtp *tmp = srtp;
            int i;
 
            for (i = 0; i < ARRAY_LEN(attr); i++) {
                if (attr[i][0]) {
                    ast_set_flag(tmp, attr[i][0]);
                }
                if (attr[i][1]) {
                    ast_set_flag(tmp, attr[i][1]);
                }
                tmp->crypto = sdp_crypto_alloc(attr[i][2]); /* key_len */
                tmp->crypto->tag = (i + 1); /* tag starts at 1 */
 
                if (i < ARRAY_LEN(attr) - 1) {
                    AST_LIST_NEXT(tmp, sdp_srtp_list) = ast_sdp_srtp_alloc();
                    tmp = AST_LIST_NEXT(tmp, sdp_srtp_list);
                }
            }
        }
    }
 
    if (dtls_enabled) {
        /* If DTLS-SRTP is enabled the key details will be pulled from TLS */
        return NULL;
    }
 
    /* set the key length based on INVITE or settings */
    if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_80)) {
        taglen = 80;
    } else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_32)) {
        taglen = 32;
    } else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_16)) {
        taglen = 16;
    } else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_TAG_8)) {
        taglen = 8;
    } else {
        taglen = default_taglen_32 ? 32 : 80;
    }
    if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_256)) {
        taglen |= 0x0200;
    } else if (ast_test_flag(srtp, AST_SRTP_CRYPTO_AES_192)) {
        taglen |= 0x0100;
    }
    if (ast_test_flag(srtp, AST_SRTP_CRYPTO_OLD_NAME)) {
        taglen |= 0x0080;
    }
 
    if (srtp->crypto && (res_sdp_crypto_build_offer(srtp->crypto, taglen) >= 0)) {
        return srtp->crypto->a_crypto;
    }
 
    ast_log(LOG_WARNING, "No SRTP key management enabled\n");
    return NULL;
}
 
static struct ast_sdp_crypto_api res_sdp_crypto_api = {
    .dtor = res_sdp_crypto_dtor,
    .alloc = res_sdp_crypto_alloc,
    .build_offer = res_sdp_crypto_build_offer,
    .parse_offer = res_sdp_crypto_parse_offer,
    .get_attr = res_sdp_srtp_get_attr,
};
 
static void res_srtp_shutdown(void)
{
    ast_sdp_crypto_unregister(&res_sdp_crypto_api);
    ast_rtp_engine_unregister_srtp();
    srtp_install_event_handler(NULL);
#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;
    }
 
    if (ast_sdp_crypto_register(&res_sdp_crypto_api)) {
        ast_log(AST_LOG_WARNING, "Failed to register SDP SRTP crypto API\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,
);