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 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>
<support_level>core</support_level>
***/
/* See https://wiki.asterisk.org/wiki/display/AST/Secure+Calling */
#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"
#include "asterisk/astobj2.h"
struct ast_srtp {
struct ast_rtp_instance *rtp;
struct ao2_container *policies;
srtp_t session;
const struct ast_srtp_cb *cb;
void *data;
int warned;
unsigned char buf[8192 + AST_FRIENDLY_OFFSET];
unsigned char rtcpbuf[8192 + AST_FRIENDLY_OFFSET];
};
struct ast_srtp_policy {
srtp_policy_t sp;
};
/*! Tracks whether or not we've initialized the libsrtp library */
static int g_initialized = 0;
/* SRTP functions */
static int ast_srtp_create(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy);
static int ast_srtp_replace(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy);
static void ast_srtp_destroy(struct ast_srtp *srtp);
static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy);
static int ast_srtp_change_source(struct ast_srtp *srtp, unsigned int from_ssrc, unsigned int to_ssrc);
static int ast_srtp_unprotect(struct ast_srtp *srtp, void *buf, int *len, int rtcp);
static int ast_srtp_protect(struct ast_srtp *srtp, void **buf, int *len, int rtcp);
static void ast_srtp_set_cb(struct ast_srtp *srtp, const struct ast_srtp_cb *cb, void *data);
static int ast_srtp_get_random(unsigned char *key, size_t len);
/* Policy functions */
static struct ast_srtp_policy *ast_srtp_policy_alloc(void);
static void ast_srtp_policy_destroy(struct ast_srtp_policy *policy);
static int ast_srtp_policy_set_suite(struct ast_srtp_policy *policy, enum ast_srtp_suite suite);
static int ast_srtp_policy_set_master_key(struct ast_srtp_policy *policy, const unsigned char *key, size_t key_len, const unsigned char *salt, size_t salt_len);
static void ast_srtp_policy_set_ssrc(struct ast_srtp_policy *policy, unsigned long ssrc, int inbound);
static struct ast_srtp_res srtp_res = {
.create = ast_srtp_create,
.replace = ast_srtp_replace,
.destroy = ast_srtp_destroy,
.add_stream = ast_srtp_add_stream,
.change_source = ast_srtp_change_source,
.set_cb = ast_srtp_set_cb,
.unprotect = ast_srtp_unprotect,
.protect = ast_srtp_protect,
.get_random = ast_srtp_get_random
};
static struct ast_srtp_policy_res policy_res = {
.alloc = ast_srtp_policy_alloc,
.destroy = ast_srtp_policy_destroy,
.set_suite = ast_srtp_policy_set_suite,
.set_master_key = ast_srtp_policy_set_master_key,
.set_ssrc = ast_srtp_policy_set_ssrc
};
static const char *srtp_errstr(int err)
{
switch(err) {
case err_status_ok:
return "nothing to report";
case err_status_fail:
return "unspecified failure";
case err_status_bad_param:
return "unsupported parameter";
case err_status_alloc_fail:
return "couldn't allocate memory";
case err_status_dealloc_fail:
return "couldn't deallocate properly";
case err_status_init_fail:
return "couldn't initialize";
case err_status_terminus:
return "can't process as much data as requested";
case err_status_auth_fail:
return "authentication failure";
case err_status_cipher_fail:
return "cipher failure";
case err_status_replay_fail:
return "replay check failed (bad index)";
case err_status_replay_old:
return "replay check failed (index too old)";
case err_status_algo_fail:
return "algorithm failed test routine";
case err_status_no_such_op:
return "unsupported operation";
case err_status_no_ctx:
return "no appropriate context found";
case err_status_cant_check:
return "unable to perform desired validation";
case err_status_key_expired:
return "can't use key any more";
default:
return "unknown";
}
}
static int policy_hash_fn(const void *obj, const int flags)
{
const struct ast_srtp_policy *policy = obj;
return policy->sp.ssrc.type == ssrc_specific ? policy->sp.ssrc.value : policy->sp.ssrc.type;
}
static int policy_cmp_fn(void *obj, void *arg, int flags)
{
const struct ast_srtp_policy *one = obj, *two = arg;
return one->sp.ssrc.type == two->sp.ssrc.type && one->sp.ssrc.value == two->sp.ssrc.value;
}
static struct ast_srtp_policy *find_policy(struct ast_srtp *srtp, const srtp_policy_t *policy, int flags)
{
struct ast_srtp_policy tmp = {
.sp = {
.ssrc.type = policy->ssrc.type,
.ssrc.value = policy->ssrc.value,
},
};
return ao2_t_find(srtp->policies, &tmp, flags, "Looking for policy");
}
static struct ast_srtp *res_srtp_new(void)
{
struct ast_srtp *srtp;
if (!(srtp = ast_calloc(1, sizeof(*srtp)))) {
ast_log(LOG_ERROR, "Unable to allocate memory for srtp\n");
return NULL;
}
if (!(srtp->policies = ao2_t_container_alloc(5, policy_hash_fn, policy_cmp_fn, "SRTP policy container"))) {
ast_free(srtp);
return NULL;
}
srtp->warned = 1;
return srtp;
}
/*
struct ast_srtp_policy
*/
static void srtp_event_cb(srtp_event_data_t *data)
{
switch (data->event) {
case event_ssrc_collision:
ast_debug(1, "SSRC collision\n");
break;
case event_key_soft_limit:
ast_debug(1, "event_key_soft_limit\n");
break;
case event_key_hard_limit:
ast_debug(1, "event_key_hard_limit\n");
break;
case event_packet_index_limit:
ast_debug(1, "event_packet_index_limit\n");
break;
}
}
static void ast_srtp_policy_set_ssrc(struct ast_srtp_policy *policy,
unsigned long ssrc, int inbound)
{
if (ssrc) {
policy->sp.ssrc.type = ssrc_specific;
policy->sp.ssrc.value = ssrc;
} else {
policy->sp.ssrc.type = inbound ? ssrc_any_inbound : ssrc_any_outbound;
}
}
static void policy_destructor(void *obj)
{
struct ast_srtp_policy *policy = obj;
if (policy->sp.key) {
ast_free(policy->sp.key);
policy->sp.key = NULL;
}
}
static struct ast_srtp_policy *ast_srtp_policy_alloc()
{
struct ast_srtp_policy *tmp;
if (!(tmp = ao2_t_alloc(sizeof(*tmp), policy_destructor, "Allocating policy"))) {
ast_log(LOG_ERROR, "Unable to allocate memory for srtp_policy\n");
}
return tmp;
}
static void ast_srtp_policy_destroy(struct ast_srtp_policy *policy)
{
ao2_t_ref(policy, -1, "Destroying policy");
}
static int policy_set_suite(crypto_policy_t *p, enum ast_srtp_suite suite)
{
switch (suite) {
case AST_AES_CM_128_HMAC_SHA1_80:
p->cipher_type = AES_128_ICM;
p->cipher_key_len = 30;
p->auth_type = HMAC_SHA1;
p->auth_key_len = 20;
p->auth_tag_len = 10;
p->sec_serv = sec_serv_conf_and_auth;
return 0;
case AST_AES_CM_128_HMAC_SHA1_32:
p->cipher_type = AES_128_ICM;
p->cipher_key_len = 30;
p->auth_type = HMAC_SHA1;
p->auth_key_len = 20;
p->auth_tag_len = 4;
p->sec_serv = sec_serv_conf_and_auth;
return 0;
default:
ast_log(LOG_ERROR, "Invalid crypto suite: %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;
int retry = 0;
struct ast_rtp_instance_stats stats = {0,};
tryagain:
for (i = 0; i < 2; i++) {
res = rtcp ? srtp_unprotect_rtcp(srtp->session, buf, len) : srtp_unprotect(srtp->session, buf, len);
if (res != err_status_no_ctx) {
break;
}
if (srtp->cb && srtp->cb->no_ctx) {
if (ast_rtp_instance_get_stats(srtp->rtp, &stats, AST_RTP_INSTANCE_STAT_REMOTE_SSRC)) {
break;
}
if (srtp->cb->no_ctx(srtp->rtp, stats.remote_ssrc, srtp->data) < 0) {
break;
}
} else {
break;
}
}
if (retry == 0 && res == err_status_replay_old) {
ast_log(AST_LOG_NOTICE, "SRTP unprotect failed with %s, retrying\n", srtp_errstr(res));
if (srtp->session) {
struct ast_srtp_policy *policy;
struct ao2_iterator it;
int policies_count;
/* dealloc first */
ast_debug(5, "SRTP destroy before re-create\n");
srtp_dealloc(srtp->session);
/* get the count */
policies_count = ao2_container_count(srtp->policies);
/* get the first to build up */
it = ao2_iterator_init(srtp->policies, 0);
policy = ao2_iterator_next(&it);
ast_debug(5, "SRTP try to re-create\n");
if (policy) {
if (srtp_create(&srtp->session, &policy->sp) == 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;
Merged revisions 377260,377263 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r377260 | file | 2012-12-05 10:51:58 -0600 (Wed, 05 Dec 2012) | 25 lines Fix a SIP request memory leak with TLS connections. During the TLS re-work in chan_sip some TLS specific code was moved into a separate function. This function operates on a copy of the incoming SIP request. This copy was never deinitialized causing a memory leak for each request processed. This function is now given a SIP request structure which it can use to copy the incoming request into. This reduces the amount of memory allocations done since the internal allocated components are reused between packets and also ensures the SIP request structure is deinitialized when the TLS connection is torn down. (closes issue ASTERISK-20763) Reported by: deti ........ Merged revisions 377257 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377258 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377259 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r377263 | jrose | 2012-12-05 11:17:06 -0600 (Wed, 05 Dec 2012) | 21 lines res_srtp: Fix a crash caused by srtp_dealloc on an already dealloced session When srtp_create fails, the session may be dealloced or just not alloced. At the same time though, the session pointer might not be set to NULL in this process and attempting to srtp_dealloc it again will cause a segfault. This patch checks for failure of srtp_create and sets the session pointer to NULL if it fails. (closes issue ASTERISK-20499) Reported by: tootai Review: https://reviewboard.asterisk.org/r/2228/ ........ Merged revisions 377256 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377261 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377262 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-05 17:20:37 +00:00
} else {
srtp->session = NULL;
}
ao2_t_ref(policy, -1, "Unreffing first policy after srtp_create failed");
}
ao2_iterator_destroy(&it);
}
}
if (res != err_status_ok && res != err_status_replay_fail ) {
if ((srtp->warned >= 10) && !((srtp->warned - 10) % 100)) {
ast_log(AST_LOG_WARNING, "SRTP unprotect failed with: %s %d\n", srtp_errstr(res), srtp->warned);
srtp->warned = 11;
} else {
srtp->warned++;
}
errno = EAGAIN;
return -1;
}
return *len;
}
static int ast_srtp_protect(struct ast_srtp *srtp, void **buf, int *len, int rtcp)
{
int res;
unsigned char *localbuf;
if ((*len + SRTP_MAX_TRAILER_LEN) > sizeof(srtp->buf)) {
return -1;
}
localbuf = rtcp ? srtp->rtcpbuf : srtp->buf;
memcpy(localbuf, *buf, *len);
if ((res = rtcp ? srtp_protect_rtcp(srtp->session, localbuf, len) : srtp_protect(srtp->session, localbuf, len)) != err_status_ok && res != err_status_replay_fail) {
ast_log(LOG_WARNING, "SRTP protect: %s\n", srtp_errstr(res));
return -1;
}
*buf = localbuf;
return *len;
}
static int ast_srtp_create(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy)
{
struct ast_srtp *temp;
if (!(temp = res_srtp_new())) {
return -1;
}
Fix a variety of memory leaks This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-18 14:43:44 +00:00
ast_module_ref(ast_module_info->self);
Fix a variety of memory leaks This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-18 14:43:44 +00:00
/* Any failures after this point can use ast_srtp_destroy to destroy the instance */
if (srtp_create(&temp->session, &policy->sp) != err_status_ok) {
Merged revisions 377260,377263 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r377260 | file | 2012-12-05 10:51:58 -0600 (Wed, 05 Dec 2012) | 25 lines Fix a SIP request memory leak with TLS connections. During the TLS re-work in chan_sip some TLS specific code was moved into a separate function. This function operates on a copy of the incoming SIP request. This copy was never deinitialized causing a memory leak for each request processed. This function is now given a SIP request structure which it can use to copy the incoming request into. This reduces the amount of memory allocations done since the internal allocated components are reused between packets and also ensures the SIP request structure is deinitialized when the TLS connection is torn down. (closes issue ASTERISK-20763) Reported by: deti ........ Merged revisions 377257 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377258 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377259 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r377263 | jrose | 2012-12-05 11:17:06 -0600 (Wed, 05 Dec 2012) | 21 lines res_srtp: Fix a crash caused by srtp_dealloc on an already dealloced session When srtp_create fails, the session may be dealloced or just not alloced. At the same time though, the session pointer might not be set to NULL in this process and attempting to srtp_dealloc it again will cause a segfault. This patch checks for failure of srtp_create and sets the session pointer to NULL if it fails. (closes issue ASTERISK-20499) Reported by: tootai Review: https://reviewboard.asterisk.org/r/2228/ ........ Merged revisions 377256 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377261 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377262 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-05 17:20:37 +00:00
/* Session either wasn't created or was created and dealloced. */
temp->session = NULL;
Fix a variety of memory leaks This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-18 14:43:44 +00:00
ast_srtp_destroy(temp);
return -1;
}
temp->rtp = rtp;
*srtp = temp;
ao2_t_link((*srtp)->policies, policy, "Created initial policy");
return 0;
}
static int ast_srtp_replace(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy)
{
if ((*srtp) != NULL) {
ast_srtp_destroy(*srtp);
}
return ast_srtp_create(srtp, rtp, policy);
}
static void ast_srtp_destroy(struct ast_srtp *srtp)
{
if (srtp->session) {
srtp_dealloc(srtp->session);
}
ao2_t_callback(srtp->policies, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, NULL, NULL, "Unallocate policy");
ao2_t_ref(srtp->policies, -1, "Destroying container");
ast_free(srtp);
ast_module_unref(ast_module_info->self);
}
static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy)
{
struct ast_srtp_policy *match;
/* For existing streams, replace if its an SSRC stream, or bail if its a wildcard */
if ((match = find_policy(srtp, &policy->sp, OBJ_POINTER))) {
if (policy->sp.ssrc.type != ssrc_specific) {
ast_log(AST_LOG_WARNING, "Cannot replace an existing wildcard policy\n");
ao2_t_ref(match, -1, "Unreffing already existing policy");
return -1;
} else {
if (srtp_remove_stream(srtp->session, match->sp.ssrc.value) != err_status_ok) {
ast_log(AST_LOG_WARNING, "Failed to remove SRTP stream for SSRC %d\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 %d\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 %d\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 (%d)\n", status);
}
ao2_t_ref(match, -1, "Unreffing found policy in change_source");
}
return 0;
}
static void res_srtp_shutdown(void)
{
srtp_install_event_handler(NULL);
ast_rtp_engine_unregister_srtp();
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;
}
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)",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
);