2010-06-08 05:29:08 +00:00
|
|
|
/*
|
|
|
|
* Asterisk -- An open source telephony toolkit.
|
|
|
|
*
|
|
|
|
* Copyright (C) 2005, Mikael Magnusson
|
|
|
|
*
|
|
|
|
* Mikael Magnusson <mikma@users.sourceforge.net>
|
|
|
|
*
|
|
|
|
* See http://www.asterisk.org for more information about
|
|
|
|
* the Asterisk project. Please do not directly contact
|
|
|
|
* any of the maintainers of this project for assistance;
|
|
|
|
* the project provides a web site, mailing lists and IRC
|
|
|
|
* channels for your use.
|
|
|
|
*
|
|
|
|
* This program is free software, distributed under the terms of
|
|
|
|
* the GNU General Public License Version 2. See the LICENSE file
|
|
|
|
* at the top of the source tree.
|
|
|
|
*
|
|
|
|
* Builds on libSRTP http://srtp.sourceforge.net
|
|
|
|
*/
|
|
|
|
|
|
|
|
/*! \file res_srtp.c
|
|
|
|
*
|
|
|
|
* \brief Secure RTP (SRTP)
|
|
|
|
*
|
|
|
|
* Secure RTP (SRTP)
|
|
|
|
* Specified in RFC 3711.
|
|
|
|
*
|
|
|
|
* \author Mikael Magnusson <mikma@users.sourceforge.net>
|
|
|
|
*/
|
|
|
|
|
|
|
|
/*** MODULEINFO
|
|
|
|
<depend>srtp</depend>
|
|
|
|
***/
|
|
|
|
|
|
|
|
/* The SIP channel will automatically use sdescriptions if received in a SDP offer,
|
|
|
|
and res_srtp is loaded. SRTP with sdescriptions key exchange can be activated
|
|
|
|
in outgoing offers by setting _SIPSRTP_CRYPTO=enable in extension.conf before executing Dial
|
|
|
|
|
|
|
|
The dial fails if the callee doesn't support SRTP and sdescriptions.
|
|
|
|
|
|
|
|
exten => 2345,1,Set(_SIPSRTP_CRYPTO=enable)
|
|
|
|
exten => 2345,2,Dial(SIP/1001)
|
|
|
|
*/
|
|
|
|
|
|
|
|
#include "asterisk.h"
|
|
|
|
|
|
|
|
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
|
|
|
|
|
|
|
#include <srtp/srtp.h>
|
|
|
|
|
|
|
|
#include "asterisk/lock.h"
|
|
|
|
#include "asterisk/sched.h"
|
|
|
|
#include "asterisk/module.h"
|
|
|
|
#include "asterisk/options.h"
|
|
|
|
#include "asterisk/rtp_engine.h"
|
|
|
|
|
|
|
|
struct ast_srtp {
|
|
|
|
struct ast_rtp_instance *rtp;
|
|
|
|
srtp_t session;
|
|
|
|
const struct ast_srtp_cb *cb;
|
|
|
|
void *data;
|
|
|
|
unsigned char buf[8192 + AST_FRIENDLY_OFFSET];
|
|
|
|
unsigned int has_stream:1;
|
|
|
|
};
|
|
|
|
|
|
|
|
struct ast_srtp_policy {
|
|
|
|
srtp_policy_t sp;
|
|
|
|
};
|
|
|
|
|
|
|
|
static int g_initialized = 0;
|
|
|
|
|
|
|
|
/* SRTP functions */
|
|
|
|
static int ast_srtp_create(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy);
|
|
|
|
static void ast_srtp_destroy(struct ast_srtp *srtp);
|
|
|
|
static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy);
|
|
|
|
|
|
|
|
static int ast_srtp_unprotect(struct ast_srtp *srtp, void *buf, int *len, int rtcp);
|
|
|
|
static int ast_srtp_protect(struct ast_srtp *srtp, void **buf, int *len, int rtcp);
|
|
|
|
static void ast_srtp_set_cb(struct ast_srtp *srtp, const struct ast_srtp_cb *cb, void *data);
|
|
|
|
static int ast_srtp_get_random(unsigned char *key, size_t len);
|
|
|
|
|
|
|
|
/* Policy functions */
|
|
|
|
static struct ast_srtp_policy *ast_srtp_policy_alloc(void);
|
|
|
|
static void ast_srtp_policy_destroy(struct ast_srtp_policy *policy);
|
|
|
|
static int ast_srtp_policy_set_suite(struct ast_srtp_policy *policy, enum ast_srtp_suite suite);
|
|
|
|
static int ast_srtp_policy_set_master_key(struct ast_srtp_policy *policy, const unsigned char *key, size_t key_len, const unsigned char *salt, size_t salt_len);
|
|
|
|
static void ast_srtp_policy_set_ssrc(struct ast_srtp_policy *policy, unsigned long ssrc, int inbound);
|
|
|
|
|
|
|
|
static struct ast_srtp_res srtp_res = {
|
|
|
|
.create = ast_srtp_create,
|
|
|
|
.destroy = ast_srtp_destroy,
|
|
|
|
.add_stream = ast_srtp_add_stream,
|
|
|
|
.set_cb = ast_srtp_set_cb,
|
|
|
|
.unprotect = ast_srtp_unprotect,
|
|
|
|
.protect = ast_srtp_protect,
|
|
|
|
.get_random = ast_srtp_get_random
|
|
|
|
};
|
|
|
|
|
|
|
|
static struct ast_srtp_policy_res policy_res = {
|
|
|
|
.alloc = ast_srtp_policy_alloc,
|
|
|
|
.destroy = ast_srtp_policy_destroy,
|
|
|
|
.set_suite = ast_srtp_policy_set_suite,
|
|
|
|
.set_master_key = ast_srtp_policy_set_master_key,
|
|
|
|
.set_ssrc = ast_srtp_policy_set_ssrc
|
|
|
|
};
|
|
|
|
|
|
|
|
static const char *srtp_errstr(int err)
|
|
|
|
{
|
|
|
|
switch(err) {
|
|
|
|
case err_status_ok:
|
|
|
|
return "nothing to report";
|
|
|
|
case err_status_fail:
|
|
|
|
return "unspecified failure";
|
|
|
|
case err_status_bad_param:
|
|
|
|
return "unsupported parameter";
|
|
|
|
case err_status_alloc_fail:
|
|
|
|
return "couldn't allocate memory";
|
|
|
|
case err_status_dealloc_fail:
|
|
|
|
return "couldn't deallocate properly";
|
|
|
|
case err_status_init_fail:
|
|
|
|
return "couldn't initialize";
|
|
|
|
case err_status_terminus:
|
|
|
|
return "can't process as much data as requested";
|
|
|
|
case err_status_auth_fail:
|
|
|
|
return "authentication failure";
|
|
|
|
case err_status_cipher_fail:
|
|
|
|
return "cipher failure";
|
|
|
|
case err_status_replay_fail:
|
|
|
|
return "replay check failed (bad index)";
|
|
|
|
case err_status_replay_old:
|
|
|
|
return "replay check failed (index too old)";
|
|
|
|
case err_status_algo_fail:
|
|
|
|
return "algorithm failed test routine";
|
|
|
|
case err_status_no_such_op:
|
|
|
|
return "unsupported operation";
|
|
|
|
case err_status_no_ctx:
|
|
|
|
return "no appropriate context found";
|
|
|
|
case err_status_cant_check:
|
|
|
|
return "unable to perform desired validation";
|
|
|
|
case err_status_key_expired:
|
|
|
|
return "can't use key any more";
|
|
|
|
default:
|
|
|
|
return "unknown";
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static struct ast_srtp *res_srtp_new(void)
|
|
|
|
{
|
|
|
|
struct ast_srtp *srtp;
|
|
|
|
|
|
|
|
if (!(srtp = ast_calloc(1, sizeof(*srtp)))) {
|
|
|
|
ast_log(LOG_ERROR, "Unable to allocate memory for srtp\n");
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
return srtp;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
|
|
struct ast_srtp_policy
|
|
|
|
*/
|
|
|
|
static void srtp_event_cb(srtp_event_data_t *data)
|
|
|
|
{
|
|
|
|
switch (data->event) {
|
|
|
|
case event_ssrc_collision:
|
|
|
|
ast_debug(1, "SSRC collision\n");
|
|
|
|
break;
|
|
|
|
case event_key_soft_limit:
|
|
|
|
ast_debug(1, "event_key_soft_limit\n");
|
|
|
|
break;
|
|
|
|
case event_key_hard_limit:
|
|
|
|
ast_debug(1, "event_key_hard_limit\n");
|
|
|
|
break;
|
|
|
|
case event_packet_index_limit:
|
|
|
|
ast_debug(1, "event_packet_index_limit\n");
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static void ast_srtp_policy_set_ssrc(struct ast_srtp_policy *policy,
|
|
|
|
unsigned long ssrc, int inbound)
|
|
|
|
{
|
|
|
|
if (ssrc) {
|
|
|
|
policy->sp.ssrc.type = ssrc_specific;
|
|
|
|
policy->sp.ssrc.value = ssrc;
|
|
|
|
} else {
|
|
|
|
policy->sp.ssrc.type = inbound ? ssrc_any_inbound : ssrc_any_outbound;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static struct ast_srtp_policy *ast_srtp_policy_alloc()
|
|
|
|
{
|
|
|
|
struct ast_srtp_policy *tmp;
|
|
|
|
|
|
|
|
if (!(tmp = ast_calloc(1, sizeof(*tmp)))) {
|
|
|
|
ast_log(LOG_ERROR, "Unable to allocate memory for srtp_policy\n");
|
|
|
|
}
|
|
|
|
|
|
|
|
return tmp;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void ast_srtp_policy_destroy(struct ast_srtp_policy *policy)
|
|
|
|
{
|
|
|
|
if (policy->sp.key) {
|
|
|
|
ast_free(policy->sp.key);
|
|
|
|
policy->sp.key = NULL;
|
|
|
|
}
|
|
|
|
ast_free(policy);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int policy_set_suite(crypto_policy_t *p, enum ast_srtp_suite suite)
|
|
|
|
{
|
|
|
|
switch (suite) {
|
|
|
|
case AST_AES_CM_128_HMAC_SHA1_80:
|
|
|
|
p->cipher_type = AES_128_ICM;
|
|
|
|
p->cipher_key_len = 30;
|
|
|
|
p->auth_type = HMAC_SHA1;
|
|
|
|
p->auth_key_len = 20;
|
|
|
|
p->auth_tag_len = 10;
|
|
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
|
|
return 0;
|
|
|
|
|
|
|
|
case AST_AES_CM_128_HMAC_SHA1_32:
|
|
|
|
p->cipher_type = AES_128_ICM;
|
|
|
|
p->cipher_key_len = 30;
|
|
|
|
p->auth_type = HMAC_SHA1;
|
|
|
|
p->auth_key_len = 20;
|
|
|
|
p->auth_tag_len = 4;
|
|
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
|
|
return 0;
|
|
|
|
|
|
|
|
default:
|
|
|
|
ast_log(LOG_ERROR, "Invalid crypto suite: %d\n", suite);
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static int ast_srtp_policy_set_suite(struct ast_srtp_policy *policy, enum ast_srtp_suite suite)
|
|
|
|
{
|
|
|
|
return policy_set_suite(&policy->sp.rtp, suite) | policy_set_suite(&policy->sp.rtcp, suite);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int ast_srtp_policy_set_master_key(struct ast_srtp_policy *policy, const unsigned char *key, size_t key_len, const unsigned char *salt, size_t salt_len)
|
|
|
|
{
|
|
|
|
size_t size = key_len + salt_len;
|
|
|
|
unsigned char *master_key;
|
|
|
|
|
|
|
|
if (policy->sp.key) {
|
|
|
|
ast_free(policy->sp.key);
|
|
|
|
policy->sp.key = NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (!(master_key = ast_calloc(1, size))) {
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
memcpy(master_key, key, key_len);
|
|
|
|
memcpy(master_key + key_len, salt, salt_len);
|
|
|
|
|
|
|
|
policy->sp.key = master_key;
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int ast_srtp_get_random(unsigned char *key, size_t len)
|
|
|
|
{
|
|
|
|
return crypto_get_random(key, len) != err_status_ok ? -1: 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void ast_srtp_set_cb(struct ast_srtp *srtp, const struct ast_srtp_cb *cb, void *data)
|
|
|
|
{
|
|
|
|
if (!srtp) {
|
|
|
|
return;
|
|
|
|
}
|
|
|
|
|
|
|
|
srtp->cb = cb;
|
|
|
|
srtp->data = data;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Vtable functions */
|
|
|
|
static int ast_srtp_unprotect(struct ast_srtp *srtp, void *buf, int *len, int rtcp)
|
|
|
|
{
|
|
|
|
int res = 0;
|
|
|
|
int i;
|
|
|
|
struct ast_rtp_instance_stats stats = {0,};
|
|
|
|
|
|
|
|
for (i = 0; i < 2; i++) {
|
|
|
|
res = rtcp ? srtp_unprotect_rtcp(srtp->session, buf, len) : srtp_unprotect(srtp->session, buf, len);
|
|
|
|
if (res != err_status_no_ctx) {
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (srtp->cb && srtp->cb->no_ctx) {
|
|
|
|
if (ast_rtp_instance_get_stats(srtp->rtp, &stats, AST_RTP_INSTANCE_STAT_REMOTE_SSRC)) {
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
if (srtp->cb->no_ctx(srtp->rtp, stats.remote_ssrc, srtp->data) < 0) {
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
} else {
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
if (res != err_status_ok && res != err_status_replay_fail ) {
|
|
|
|
ast_debug(1, "SRTP unprotect: %s\n", srtp_errstr(res));
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
return *len;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int ast_srtp_protect(struct ast_srtp *srtp, void **buf, int *len, int rtcp)
|
|
|
|
{
|
|
|
|
int res;
|
|
|
|
|
|
|
|
if ((*len + SRTP_MAX_TRAILER_LEN) > sizeof(srtp->buf)) {
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
memcpy(srtp->buf, *buf, *len);
|
|
|
|
|
|
|
|
if ((res = rtcp ? srtp_protect_rtcp(srtp->session, srtp->buf, len) : srtp_protect(srtp->session, srtp->buf, len)) != err_status_ok && res != err_status_replay_fail) {
|
|
|
|
ast_debug(1, "SRTP protect: %s\n", srtp_errstr(res));
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
*buf = srtp->buf;
|
|
|
|
return *len;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int ast_srtp_create(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy)
|
|
|
|
{
|
|
|
|
struct ast_srtp *temp;
|
|
|
|
|
|
|
|
if (!(temp = res_srtp_new())) {
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (srtp_create(&temp->session, &policy->sp) != err_status_ok) {
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
temp->rtp = rtp;
|
|
|
|
*srtp = temp;
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void ast_srtp_destroy(struct ast_srtp *srtp)
|
|
|
|
{
|
|
|
|
if (srtp->session) {
|
|
|
|
srtp_dealloc(srtp->session);
|
|
|
|
}
|
|
|
|
|
|
|
|
ast_free(srtp);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy)
|
|
|
|
{
|
|
|
|
if (!srtp->has_stream && srtp_add_stream(srtp->session, &policy->sp) != err_status_ok) {
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
srtp->has_stream = 1;
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int res_srtp_init(void)
|
|
|
|
{
|
|
|
|
if (g_initialized) {
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (srtp_init() != err_status_ok) {
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
srtp_install_event_handler(srtp_event_cb);
|
|
|
|
|
|
|
|
return ast_rtp_engine_register_srtp(&srtp_res, &policy_res);
|
|
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Exported functions
|
|
|
|
*/
|
|
|
|
|
|
|
|
static int load_module(void)
|
|
|
|
{
|
|
|
|
return res_srtp_init();
|
|
|
|
}
|
|
|
|
|
|
|
|
static int unload_module(void)
|
|
|
|
{
|
|
|
|
ast_rtp_engine_unregister_srtp();
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2010-07-20 19:35:02 +00:00
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Secure RTP (SRTP)",
|
2010-06-08 05:29:08 +00:00
|
|
|
.load = load_module,
|
|
|
|
.unload = unload_module,
|
2010-07-20 19:35:02 +00:00
|
|
|
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
|
2010-06-08 05:29:08 +00:00
|
|
|
);
|