857814f435
After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
404 lines
9.7 KiB
C
404 lines
9.7 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2005, Mikael Magnusson
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*
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* Mikael Magnusson <mikma@users.sourceforge.net>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*
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* Builds on libSRTP http://srtp.sourceforge.net
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*/
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/*! \file res_srtp.c
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*
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* \brief Secure RTP (SRTP)
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*
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* Secure RTP (SRTP)
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* Specified in RFC 3711.
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*
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* \author Mikael Magnusson <mikma@users.sourceforge.net>
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*/
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/*** MODULEINFO
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<depend>srtp</depend>
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***/
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/* The SIP channel will automatically use sdescriptions if received in a SDP offer,
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and res_srtp is loaded. SRTP with sdescriptions key exchange can be activated
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in outgoing offers by setting _SIPSRTP_CRYPTO=enable in extension.conf before executing Dial
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The dial fails if the callee doesn't support SRTP and sdescriptions.
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exten => 2345,1,Set(_SIPSRTP_CRYPTO=enable)
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exten => 2345,2,Dial(SIP/1001)
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <srtp/srtp.h>
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#include "asterisk/lock.h"
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#include "asterisk/sched.h"
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#include "asterisk/module.h"
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#include "asterisk/options.h"
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#include "asterisk/rtp_engine.h"
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struct ast_srtp {
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struct ast_rtp_instance *rtp;
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srtp_t session;
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const struct ast_srtp_cb *cb;
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void *data;
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unsigned char buf[8192 + AST_FRIENDLY_OFFSET];
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unsigned int has_stream:1;
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};
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struct ast_srtp_policy {
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srtp_policy_t sp;
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};
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static int g_initialized = 0;
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/* SRTP functions */
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static int ast_srtp_create(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy);
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static void ast_srtp_destroy(struct ast_srtp *srtp);
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static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy);
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static int ast_srtp_unprotect(struct ast_srtp *srtp, void *buf, int *len, int rtcp);
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static int ast_srtp_protect(struct ast_srtp *srtp, void **buf, int *len, int rtcp);
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static void ast_srtp_set_cb(struct ast_srtp *srtp, const struct ast_srtp_cb *cb, void *data);
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static int ast_srtp_get_random(unsigned char *key, size_t len);
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/* Policy functions */
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static struct ast_srtp_policy *ast_srtp_policy_alloc(void);
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static void ast_srtp_policy_destroy(struct ast_srtp_policy *policy);
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static int ast_srtp_policy_set_suite(struct ast_srtp_policy *policy, enum ast_srtp_suite suite);
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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);
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static void ast_srtp_policy_set_ssrc(struct ast_srtp_policy *policy, unsigned long ssrc, int inbound);
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static struct ast_srtp_res srtp_res = {
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.create = ast_srtp_create,
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.destroy = ast_srtp_destroy,
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.add_stream = ast_srtp_add_stream,
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.set_cb = ast_srtp_set_cb,
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.unprotect = ast_srtp_unprotect,
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.protect = ast_srtp_protect,
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.get_random = ast_srtp_get_random
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};
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static struct ast_srtp_policy_res policy_res = {
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.alloc = ast_srtp_policy_alloc,
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.destroy = ast_srtp_policy_destroy,
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.set_suite = ast_srtp_policy_set_suite,
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.set_master_key = ast_srtp_policy_set_master_key,
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.set_ssrc = ast_srtp_policy_set_ssrc
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};
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static const char *srtp_errstr(int err)
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{
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switch(err) {
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case err_status_ok:
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return "nothing to report";
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case err_status_fail:
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return "unspecified failure";
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case err_status_bad_param:
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return "unsupported parameter";
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case err_status_alloc_fail:
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return "couldn't allocate memory";
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case err_status_dealloc_fail:
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return "couldn't deallocate properly";
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case err_status_init_fail:
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return "couldn't initialize";
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case err_status_terminus:
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return "can't process as much data as requested";
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case err_status_auth_fail:
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return "authentication failure";
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case err_status_cipher_fail:
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return "cipher failure";
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case err_status_replay_fail:
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return "replay check failed (bad index)";
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case err_status_replay_old:
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return "replay check failed (index too old)";
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case err_status_algo_fail:
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return "algorithm failed test routine";
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case err_status_no_such_op:
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return "unsupported operation";
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case err_status_no_ctx:
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return "no appropriate context found";
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case err_status_cant_check:
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return "unable to perform desired validation";
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case err_status_key_expired:
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return "can't use key any more";
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default:
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return "unknown";
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}
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}
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static struct ast_srtp *res_srtp_new(void)
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{
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struct ast_srtp *srtp;
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if (!(srtp = ast_calloc(1, sizeof(*srtp)))) {
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ast_log(LOG_ERROR, "Unable to allocate memory for srtp\n");
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return NULL;
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}
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return srtp;
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}
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/*
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struct ast_srtp_policy
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*/
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static void srtp_event_cb(srtp_event_data_t *data)
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{
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switch (data->event) {
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case event_ssrc_collision:
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ast_debug(1, "SSRC collision\n");
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break;
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case event_key_soft_limit:
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ast_debug(1, "event_key_soft_limit\n");
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break;
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case event_key_hard_limit:
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ast_debug(1, "event_key_hard_limit\n");
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break;
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case event_packet_index_limit:
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ast_debug(1, "event_packet_index_limit\n");
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break;
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}
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}
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static void ast_srtp_policy_set_ssrc(struct ast_srtp_policy *policy,
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unsigned long ssrc, int inbound)
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{
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if (ssrc) {
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policy->sp.ssrc.type = ssrc_specific;
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policy->sp.ssrc.value = ssrc;
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} else {
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policy->sp.ssrc.type = inbound ? ssrc_any_inbound : ssrc_any_outbound;
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}
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}
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static struct ast_srtp_policy *ast_srtp_policy_alloc()
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{
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struct ast_srtp_policy *tmp;
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if (!(tmp = ast_calloc(1, sizeof(*tmp)))) {
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ast_log(LOG_ERROR, "Unable to allocate memory for srtp_policy\n");
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}
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return tmp;
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}
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static void ast_srtp_policy_destroy(struct ast_srtp_policy *policy)
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{
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if (policy->sp.key) {
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ast_free(policy->sp.key);
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policy->sp.key = NULL;
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}
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ast_free(policy);
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}
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static int policy_set_suite(crypto_policy_t *p, enum ast_srtp_suite suite)
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{
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switch (suite) {
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case AST_AES_CM_128_HMAC_SHA1_80:
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p->cipher_type = AES_128_ICM;
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p->cipher_key_len = 30;
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p->auth_type = HMAC_SHA1;
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p->auth_key_len = 20;
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p->auth_tag_len = 10;
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p->sec_serv = sec_serv_conf_and_auth;
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return 0;
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case AST_AES_CM_128_HMAC_SHA1_32:
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p->cipher_type = AES_128_ICM;
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p->cipher_key_len = 30;
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p->auth_type = HMAC_SHA1;
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p->auth_key_len = 20;
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p->auth_tag_len = 4;
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p->sec_serv = sec_serv_conf_and_auth;
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return 0;
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default:
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ast_log(LOG_ERROR, "Invalid crypto suite: %d\n", suite);
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return -1;
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}
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}
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static int ast_srtp_policy_set_suite(struct ast_srtp_policy *policy, enum ast_srtp_suite suite)
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{
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return policy_set_suite(&policy->sp.rtp, suite) | policy_set_suite(&policy->sp.rtcp, suite);
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}
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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)
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{
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size_t size = key_len + salt_len;
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unsigned char *master_key;
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if (policy->sp.key) {
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ast_free(policy->sp.key);
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policy->sp.key = NULL;
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}
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if (!(master_key = ast_calloc(1, size))) {
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return -1;
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}
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memcpy(master_key, key, key_len);
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memcpy(master_key + key_len, salt, salt_len);
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policy->sp.key = master_key;
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return 0;
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}
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static int ast_srtp_get_random(unsigned char *key, size_t len)
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{
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return crypto_get_random(key, len) != err_status_ok ? -1: 0;
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}
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static void ast_srtp_set_cb(struct ast_srtp *srtp, const struct ast_srtp_cb *cb, void *data)
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{
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if (!srtp) {
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return;
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}
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srtp->cb = cb;
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srtp->data = data;
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}
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/* Vtable functions */
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static int ast_srtp_unprotect(struct ast_srtp *srtp, void *buf, int *len, int rtcp)
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{
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int res = 0;
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int i;
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struct ast_rtp_instance_stats stats = {0,};
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for (i = 0; i < 2; i++) {
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res = rtcp ? srtp_unprotect_rtcp(srtp->session, buf, len) : srtp_unprotect(srtp->session, buf, len);
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if (res != err_status_no_ctx) {
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break;
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}
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if (srtp->cb && srtp->cb->no_ctx) {
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if (ast_rtp_instance_get_stats(srtp->rtp, &stats, AST_RTP_INSTANCE_STAT_REMOTE_SSRC)) {
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break;
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}
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if (srtp->cb->no_ctx(srtp->rtp, stats.remote_ssrc, srtp->data) < 0) {
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break;
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}
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} else {
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break;
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}
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}
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if (res != err_status_ok && res != err_status_replay_fail ) {
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ast_debug(1, "SRTP unprotect: %s\n", srtp_errstr(res));
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return -1;
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}
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return *len;
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}
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static int ast_srtp_protect(struct ast_srtp *srtp, void **buf, int *len, int rtcp)
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{
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int res;
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if ((*len + SRTP_MAX_TRAILER_LEN) > sizeof(srtp->buf)) {
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return -1;
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}
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memcpy(srtp->buf, *buf, *len);
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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) {
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ast_debug(1, "SRTP protect: %s\n", srtp_errstr(res));
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return -1;
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}
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*buf = srtp->buf;
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return *len;
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}
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static int ast_srtp_create(struct ast_srtp **srtp, struct ast_rtp_instance *rtp, struct ast_srtp_policy *policy)
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{
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struct ast_srtp *temp;
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if (!(temp = res_srtp_new())) {
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return -1;
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}
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if (srtp_create(&temp->session, &policy->sp) != err_status_ok) {
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return -1;
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}
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temp->rtp = rtp;
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*srtp = temp;
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return 0;
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}
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static void ast_srtp_destroy(struct ast_srtp *srtp)
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{
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if (srtp->session) {
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srtp_dealloc(srtp->session);
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}
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ast_free(srtp);
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}
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static int ast_srtp_add_stream(struct ast_srtp *srtp, struct ast_srtp_policy *policy)
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{
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if (!srtp->has_stream && srtp_add_stream(srtp->session, &policy->sp) != err_status_ok) {
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return -1;
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}
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srtp->has_stream = 1;
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return 0;
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}
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static int res_srtp_init(void)
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{
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if (g_initialized) {
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return 0;
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}
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if (srtp_init() != err_status_ok) {
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return -1;
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}
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srtp_install_event_handler(srtp_event_cb);
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return ast_rtp_engine_register_srtp(&srtp_res, &policy_res);
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}
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/*
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* Exported functions
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*/
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static int load_module(void)
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{
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return res_srtp_init();
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}
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static int unload_module(void)
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{
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ast_rtp_engine_unregister_srtp();
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return 0;
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS, "Secure RTP (SRTP)",
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.load = load_module,
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.unload = unload_module,
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);
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