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asterisk/res/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>
***/
/* 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;
#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_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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