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asterisk/res/res_srtp.c
Alexander Traud 4de0e50c32 res_srtp: Set all possible flags while selecting the Crypto Suite. 
The flags of a previous selection could have been set within the
object 'srtp', for example, when the previous selection returned
failure after setting just 'some' flags. Now, not to clutter the
code, all possible flags are cleared first, and then the selected
flags are set as before.

ASTERISK-28903

Change-Id: I1b9d7aade7d5120244ce7e3a8865518cbd6e0eee
2020-05-20 10:46:07 -05:00

1281 lines
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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 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" /* 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;
#ifdef HAVE_SRTP_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
#ifdef HAVE_SRTP_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
#ifdef HAVE_SRTP_GCM
case AST_AES_GCM_128:
crypto_policy_set_aes_gcm_128_16_auth(p);
return 0;
case AST_AES_GCM_256:
crypto_policy_set_aes_gcm_256_16_auth(p);
return 0;
case AST_AES_GCM_128_8:
crypto_policy_set_aes_gcm_128_8_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 rtcp)
{
int res = 0;
int i;
int retry = 0;
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;
#ifdef HAVE_SRTP_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
#ifdef HAVE_SRTP_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
#ifdef HAVE_SRTP_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;
} 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_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;
} 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_verb(1, "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,
);
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