diff options
Diffstat (limited to 'channels/sip')
-rw-r--r-- | channels/sip/include/sdp_crypto.h | 85 | ||||
-rw-r--r-- | channels/sip/include/sip.h | 6 | ||||
-rw-r--r-- | channels/sip/include/srtp.h | 59 | ||||
-rw-r--r-- | channels/sip/sdp_crypto.c | 318 | ||||
-rw-r--r-- | channels/sip/srtp.c | 55 |
5 files changed, 3 insertions, 520 deletions
diff --git a/channels/sip/include/sdp_crypto.h b/channels/sip/include/sdp_crypto.h deleted file mode 100644 index da1035e87..000000000 --- a/channels/sip/include/sdp_crypto.h +++ /dev/null @@ -1,85 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (C) 2006 - 2007, 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. - */ - -/*! \file sdp_crypto.h - * - * \brief SDP Security descriptions - * - * Specified in RFC 4568 - * - * \author Mikael Magnusson <mikma@users.sourceforge.net> - */ - -#ifndef _SDP_CRYPTO_H -#define _SDP_CRYPTO_H - -#include <asterisk/rtp_engine.h> - -struct sdp_crypto; -struct sip_srtp; - -/*! \brief Initialize an return an sdp_crypto struct - * - * \details - * This function allocates a new sdp_crypto struct and initializes its values - * - * \retval NULL on failure - * \retval a pointer to a new sdp_crypto structure - */ -struct sdp_crypto *sdp_crypto_setup(void); - -/*! \brief Destroy a previously allocated sdp_crypto struct */ -void sdp_crypto_destroy(struct sdp_crypto *crypto); - -/*! \brief Parse the a=crypto line from SDP and set appropriate values on the - * sdp_crypto struct. - * - * \param p A valid sdp_crypto struct - * \param attr the a:crypto line from SDP - * \param rtp The rtp instance associated with the SDP being parsed - * \param srtp SRTP structure - * - * \retval 0 success - * \retval nonzero failure - */ -int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp); - - -/*! \brief Generate an SRTP a=crypto offer - * - * \details - * The offer is stored on the sdp_crypto struct in a_crypto - * - * \param p A valid sdp_crypto struct - * \param taglen Length - * - * \retval 0 success - * \retval nonzero failure - */ -int sdp_crypto_offer(struct sdp_crypto *p, int taglen); - - -/*! \brief Return the a_crypto value of the sdp_crypto struct - * - * \param p An sdp_crypto struct that has had sdp_crypto_offer called - * - * \retval The value of the a_crypto for p - */ -const char *sdp_crypto_attrib(struct sdp_crypto *p); - -#endif /* _SDP_CRYPTO_H */ diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h index 0adde37f2..8b4672b25 100644 --- a/channels/sip/include/sip.h +++ b/channels/sip/include/sip.h @@ -1165,9 +1165,9 @@ struct sip_pvt { AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */ struct sip_invite_param *options; /*!< Options for INVITE */ struct sip_st_dlg *stimer; /*!< SIP Session-Timers */ - struct sip_srtp *srtp; /*!< Structure to hold Secure RTP session data for audio */ - struct sip_srtp *vsrtp; /*!< Structure to hold Secure RTP session data for video */ - struct sip_srtp *tsrtp; /*!< Structure to hold Secure RTP session data for text */ + struct ast_sdp_srtp *srtp; /*!< Structure to hold Secure RTP session data for audio */ + struct ast_sdp_srtp *vsrtp; /*!< Structure to hold Secure RTP session data for video */ + struct ast_sdp_srtp *tsrtp; /*!< Structure to hold Secure RTP session data for text */ int red; /*!< T.140 RTP Redundancy */ int hangupcause; /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */ diff --git a/channels/sip/include/srtp.h b/channels/sip/include/srtp.h deleted file mode 100644 index a4ded62ca..000000000 --- a/channels/sip/include/srtp.h +++ /dev/null @@ -1,59 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (C) 2006 - 2007, 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. - */ - -/*! \file srtp.h - * - * \brief SIP Secure RTP (SRTP) - * - * Specified in RFC 3711 - * - * \author Mikael Magnusson <mikma@users.sourceforge.net> - */ - -#ifndef _SIP_SRTP_H -#define _SIP_SRTP_H - -#include "sdp_crypto.h" - -/* SRTP flags */ -#define SRTP_ENCR_OPTIONAL (1 << 1) /* SRTP encryption optional */ -#define SRTP_CRYPTO_ENABLE (1 << 2) -#define SRTP_CRYPTO_OFFER_OK (1 << 3) -#define SRTP_CRYPTO_TAG_32 (1 << 4) -#define SRTP_CRYPTO_TAG_80 (1 << 5) - -/*! \brief structure for secure RTP audio */ -struct sip_srtp { - unsigned int flags; - struct sdp_crypto *crypto; -}; - -/*! - * \brief allocate a sip_srtp structure - * \retval a new malloc'd sip_srtp structure on success - * \retval NULL on failure -*/ -struct sip_srtp *sip_srtp_alloc(void); - -/*! - * \brief free a sip_srtp structure - * \param srtp a sip_srtp structure -*/ -void sip_srtp_destroy(struct sip_srtp *srtp); - -#endif /* _SIP_SRTP_H */ diff --git a/channels/sip/sdp_crypto.c b/channels/sip/sdp_crypto.c deleted file mode 100644 index c27e882c2..000000000 --- a/channels/sip/sdp_crypto.c +++ /dev/null @@ -1,318 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (C) 2006 - 2007, 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. - */ - -/*! \file sdp_crypto.c - * - * \brief SDP Security descriptions - * - * Specified in RFC 4568 - * - * \author Mikael Magnusson <mikma@users.sourceforge.net> - */ - -/*** MODULEINFO - <support_level>core</support_level> - ***/ - -#include "asterisk.h" - -ASTERISK_FILE_VERSION(__FILE__, "$Revision$") - -#include "asterisk/options.h" -#include "asterisk/utils.h" -#include "include/sdp_crypto.h" -#include "include/srtp.h" - -#define SRTP_MASTER_LEN 30 -#define SRTP_MASTERKEY_LEN 16 -#define SRTP_MASTERSALT_LEN ((SRTP_MASTER_LEN) - (SRTP_MASTERKEY_LEN)) -#define SRTP_MASTER_LEN64 (((SRTP_MASTER_LEN) * 8 + 5) / 6 + 1) - -extern struct ast_srtp_res *res_srtp; -extern struct ast_srtp_policy_res *res_srtp_policy; - -struct sdp_crypto { - char *a_crypto; - unsigned char local_key[SRTP_MASTER_LEN]; - char *tag; - char local_key64[SRTP_MASTER_LEN64]; - unsigned char remote_key[SRTP_MASTER_LEN]; -}; - -static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, unsigned long ssrc, int inbound); - -static struct sdp_crypto *sdp_crypto_alloc(void) -{ - return ast_calloc(1, sizeof(struct sdp_crypto)); -} - -void sdp_crypto_destroy(struct sdp_crypto *crypto) -{ - ast_free(crypto->a_crypto); - crypto->a_crypto = NULL; - ast_free(crypto->tag); - crypto->tag = NULL; - ast_free(crypto); -} - -struct sdp_crypto *sdp_crypto_setup(void) -{ - struct sdp_crypto *p; - int key_len; - unsigned char remote_key[SRTP_MASTER_LEN]; - - if (!ast_rtp_engine_srtp_is_registered()) { - return NULL; - } - - if (!(p = sdp_crypto_alloc())) { - return NULL; - } - - if (res_srtp->get_random(p->local_key, sizeof(p->local_key)) < 0) { - sdp_crypto_destroy(p); - return NULL; - } - - ast_base64encode(p->local_key64, p->local_key, SRTP_MASTER_LEN, sizeof(p->local_key64)); - - key_len = ast_base64decode(remote_key, p->local_key64, sizeof(remote_key)); - - if (key_len != SRTP_MASTER_LEN) { - ast_log(LOG_ERROR, "base64 encode/decode bad len %d != %d\n", key_len, SRTP_MASTER_LEN); - ast_free(p); - return NULL; - } - - if (memcmp(remote_key, p->local_key, SRTP_MASTER_LEN)) { - ast_log(LOG_ERROR, "base64 encode/decode bad key\n"); - ast_free(p); - return NULL; - } - - ast_debug(1 , "local_key64 %s len %zu\n", p->local_key64, strlen(p->local_key64)); - - return p; -} - -static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, unsigned long ssrc, int inbound) -{ - const unsigned char *master_salt = NULL; - - if (!ast_rtp_engine_srtp_is_registered()) { - return -1; - } - - master_salt = master_key + SRTP_MASTERKEY_LEN; - if (res_srtp_policy->set_master_key(policy, master_key, SRTP_MASTERKEY_LEN, master_salt, SRTP_MASTERSALT_LEN) < 0) { - return -1; - } - - if (res_srtp_policy->set_suite(policy, suite_val)) { - ast_log(LOG_WARNING, "Could not set remote SRTP suite\n"); - return -1; - } - - res_srtp_policy->set_ssrc(policy, ssrc, inbound); - - return 0; -} - -static int sdp_crypto_activate(struct sdp_crypto *p, int suite_val, unsigned char *remote_key, 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 (!ast_rtp_engine_srtp_is_registered()) { - return -1; - } - - if (!p) { - return -1; - } - - if (!(local_policy = res_srtp_policy->alloc())) { - return -1; - } - - if (!(remote_policy = res_srtp_policy->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, stats.local_ssrc, 0) < 0) { - goto err; - } - - if (set_crypto_policy(remote_policy, suite_val, remote_key, 0, 1) < 0) { - goto err; - } - - /* Add the SRTP policies */ - if (ast_rtp_instance_add_srtp_policy(rtp, remote_policy, local_policy)) { - 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) { - res_srtp_policy->destroy(local_policy); - } - - if (remote_policy) { - res_srtp_policy->destroy(remote_policy); - } - - return res; -} - -int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp) -{ - 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; - char *lifetime = NULL; - int found = 0; - int key_len = 0; - int suite_val = 0; - unsigned char remote_key[SRTP_MASTER_LEN]; - int taglen = 0; - - if (!ast_rtp_engine_srtp_is_registered()) { - return -1; - } - - str = ast_strdupa(attr); - - strsep(&str, ":"); - tag = strsep(&str, " "); - suite = strsep(&str, " "); - key_params = strsep(&str, " "); - session_params = strsep(&str, " "); - - if (!tag || !suite) { - ast_log(LOG_WARNING, "Unrecognized a=%s", attr); - return -1; - } - - if (!ast_strlen_zero(session_params)) { - ast_log(LOG_WARNING, "Unsupported crypto parameters: %s", session_params); - return -1; - } - - if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_80")) { - suite_val = AST_AES_CM_128_HMAC_SHA1_80; - ast_set_flag(srtp, SRTP_CRYPTO_TAG_80); - taglen = 80; - } else if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_32")) { - suite_val = AST_AES_CM_128_HMAC_SHA1_32; - ast_set_flag(srtp, SRTP_CRYPTO_TAG_32); - taglen = 32; - } else { - ast_log(LOG_WARNING, "Unsupported crypto suite: %s\n", suite); - return -1; - } - - while ((key_param = strsep(&key_params, ";"))) { - char *method = NULL; - char *info = NULL; - - method = strsep(&key_param, ":"); - info = strsep(&key_param, ";"); - - if (!strcmp(method, "inline")) { - key_salt = strsep(&info, "|"); - lifetime = strsep(&info, "|"); - - if (lifetime) { - ast_log(LOG_NOTICE, "Crypto life time unsupported: %s\n", attr); - continue; - } - - found = 1; - break; - } - } - - if (!found) { - ast_log(LOG_NOTICE, "SRTP crypto offer not acceptable\n"); - return -1; - } - - if ((key_len = ast_base64decode(remote_key, key_salt, sizeof(remote_key))) != SRTP_MASTER_LEN) { - ast_log(LOG_WARNING, "SRTP descriptions key %d != %d\n", key_len, SRTP_MASTER_LEN); - return -1; - } - - if (!memcmp(p->remote_key, remote_key, sizeof(p->remote_key))) { - ast_debug(1, "SRTP remote key unchanged; maintaining current policy\n"); - return 0; - } - memcpy(p->remote_key, remote_key, sizeof(p->remote_key)); - - if (sdp_crypto_activate(p, suite_val, remote_key, rtp) < 0) { - return -1; - } - - if (!p->tag) { - ast_log(LOG_DEBUG, "Accepting crypto tag %s\n", tag); - p->tag = ast_strdup(tag); - if (!p->tag) { - ast_log(LOG_ERROR, "Could not allocate memory for tag\n"); - return -1; - } - } - - /* Finally, rebuild the crypto line */ - return sdp_crypto_offer(p, taglen); -} - -int sdp_crypto_offer(struct sdp_crypto *p, int taglen) -{ - /* Rebuild the crypto line */ - if (p->a_crypto) { - ast_free(p->a_crypto); - } - - if (ast_asprintf(&p->a_crypto, "a=crypto:%s AES_CM_128_HMAC_SHA1_%i inline:%s\r\n", - p->tag ? p->tag : "1", taglen, p->local_key64) == -1) { - ast_log(LOG_ERROR, "Could not allocate memory for crypto line\n"); - return -1; - } - - ast_log(LOG_DEBUG, "Crypto line: %s", p->a_crypto); - - return 0; -} - -const char *sdp_crypto_attrib(struct sdp_crypto *p) -{ - return p->a_crypto; -} diff --git a/channels/sip/srtp.c b/channels/sip/srtp.c deleted file mode 100644 index 8b2718fc3..000000000 --- a/channels/sip/srtp.c +++ /dev/null @@ -1,55 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (C) 2006 - 2007, 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. - */ - -/*! \file srtp.c - * - * \brief SIP Secure RTP (SRTP) - * - * Specified in RFC 3711 - * - * \author Mikael Magnusson <mikma@users.sourceforge.net> - */ - -/*** MODULEINFO - <support_level>core</support_level> - ***/ - -#include "asterisk.h" - -ASTERISK_FILE_VERSION(__FILE__, "$Revision$") - -#include "asterisk/utils.h" -#include "include/srtp.h" - -struct sip_srtp *sip_srtp_alloc(void) -{ - struct sip_srtp *srtp; - - srtp = ast_calloc(1, sizeof(*srtp)); - - return srtp; -} - -void sip_srtp_destroy(struct sip_srtp *srtp) -{ - if (srtp->crypto) { - sdp_crypto_destroy(srtp->crypto); - } - srtp->crypto = NULL; - ast_free(srtp); -} |