| /* |
| * srtp.c |
| * |
| * the secure real-time transport protocol |
| * |
| * David A. McGrew |
| * Cisco Systems, Inc. |
| */ |
| /* |
| * |
| * Copyright (c) 2001-2006, Cisco Systems, Inc. |
| * All rights reserved. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions |
| * are met: |
| * |
| * Redistributions of source code must retain the above copyright |
| * notice, this list of conditions and the following disclaimer. |
| * |
| * Redistributions in binary form must reproduce the above |
| * copyright notice, this list of conditions and the following |
| * disclaimer in the documentation and/or other materials provided |
| * with the distribution. |
| * |
| * Neither the name of the Cisco Systems, Inc. nor the names of its |
| * contributors may be used to endorse or promote products derived |
| * from this software without specific prior written permission. |
| * |
| * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS |
| * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT |
| * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS |
| * FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE |
| * COPYRIGHT HOLDERS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, |
| * INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES |
| * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR |
| * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) |
| * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, |
| * STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) |
| * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED |
| * OF THE POSSIBILITY OF SUCH DAMAGE. |
| * |
| */ |
| |
| #include "srtp.h" |
| #include "srtp_priv.h" |
| #include "crypto_types.h" |
| #include "err.h" |
| #include "ekt.h" /* for SRTP Encrypted Key Transport */ |
| #include "alloc.h" /* for srtp_crypto_alloc() */ |
| #ifdef OPENSSL |
| #include "aes_gcm_ossl.h" /* for AES GCM mode */ |
| #endif |
| |
| #include <limits.h> |
| #ifdef HAVE_NETINET_IN_H |
| # include <netinet/in.h> |
| #elif defined(HAVE_WINSOCK2_H) |
| # include <winsock2.h> |
| #endif |
| |
| |
| /* the debug module for srtp */ |
| |
| srtp_debug_module_t mod_srtp = { |
| 0, /* debugging is off by default */ |
| "srtp" /* printable name for module */ |
| }; |
| |
| #define octets_in_rtp_header 12 |
| #define uint32s_in_rtp_header 3 |
| #define octets_in_rtcp_header 8 |
| #define uint32s_in_rtcp_header 2 |
| |
| const char *srtp_get_version_string () |
| { |
| /* |
| * Simply return the autotools generated string |
| */ |
| return SRTP_VER_STRING; |
| } |
| |
| unsigned int srtp_get_version () |
| { |
| unsigned int major = 0, minor = 0, micro = 0; |
| unsigned int rv = 0; |
| int parse_rv; |
| |
| /* |
| * Parse the autotools generated version |
| */ |
| parse_rv = sscanf(SRTP_VERSION, "%u.%u.%u", &major, &minor, µ); |
| if (parse_rv != 3) { |
| /* |
| * We're expected to parse all 3 version levels. |
| * If not, then this must not be an official release. |
| * Return all zeros on the version |
| */ |
| return (0); |
| } |
| |
| /* |
| * We allow 8 bits for the major and minor, while |
| * allowing 16 bits for the micro. 16 bits for the micro |
| * may be beneficial for a continuous delivery model |
| * in the future. |
| */ |
| rv |= (major & 0xFF) << 24; |
| rv |= (minor & 0xFF) << 16; |
| rv |= micro & 0xFF; |
| return rv; |
| } |
| |
| srtp_err_status_t |
| srtp_stream_alloc(srtp_stream_ctx_t **str_ptr, |
| const srtp_policy_t *p) { |
| srtp_stream_ctx_t *str; |
| srtp_err_status_t stat; |
| |
| /* |
| * This function allocates the stream context, rtp and rtcp ciphers |
| * and auth functions, and key limit structure. If there is a |
| * failure during allocation, we free all previously allocated |
| * memory and return a failure code. The code could probably |
| * be improved, but it works and should be clear. |
| */ |
| |
| /* allocate srtp stream and set str_ptr */ |
| str = (srtp_stream_ctx_t *) srtp_crypto_alloc(sizeof(srtp_stream_ctx_t)); |
| if (str == NULL) |
| return srtp_err_status_alloc_fail; |
| *str_ptr = str; |
| |
| /* allocate cipher */ |
| stat = srtp_crypto_kernel_alloc_cipher(p->rtp.cipher_type, |
| &str->rtp_cipher, |
| p->rtp.cipher_key_len, |
| p->rtp.auth_tag_len); |
| if (stat) { |
| srtp_crypto_free(str); |
| return stat; |
| } |
| |
| /* allocate auth function */ |
| stat = srtp_crypto_kernel_alloc_auth(p->rtp.auth_type, |
| &str->rtp_auth, |
| p->rtp.auth_key_len, |
| p->rtp.auth_tag_len); |
| if (stat) { |
| cipher_dealloc(str->rtp_cipher); |
| srtp_crypto_free(str); |
| return stat; |
| } |
| |
| /* allocate key limit structure */ |
| str->limit = (srtp_key_limit_ctx_t*) srtp_crypto_alloc(sizeof(srtp_key_limit_ctx_t)); |
| if (str->limit == NULL) { |
| auth_dealloc(str->rtp_auth); |
| cipher_dealloc(str->rtp_cipher); |
| srtp_crypto_free(str); |
| return srtp_err_status_alloc_fail; |
| } |
| |
| /* |
| * ...and now the RTCP-specific initialization - first, allocate |
| * the cipher |
| */ |
| stat = srtp_crypto_kernel_alloc_cipher(p->rtcp.cipher_type, |
| &str->rtcp_cipher, |
| p->rtcp.cipher_key_len, |
| p->rtcp.auth_tag_len); |
| if (stat) { |
| auth_dealloc(str->rtp_auth); |
| cipher_dealloc(str->rtp_cipher); |
| srtp_crypto_free(str->limit); |
| srtp_crypto_free(str); |
| return stat; |
| } |
| |
| /* allocate auth function */ |
| stat = srtp_crypto_kernel_alloc_auth(p->rtcp.auth_type, |
| &str->rtcp_auth, |
| p->rtcp.auth_key_len, |
| p->rtcp.auth_tag_len); |
| if (stat) { |
| cipher_dealloc(str->rtcp_cipher); |
| auth_dealloc(str->rtp_auth); |
| cipher_dealloc(str->rtp_cipher); |
| srtp_crypto_free(str->limit); |
| srtp_crypto_free(str); |
| return stat; |
| } |
| |
| /* allocate ekt data associated with stream */ |
| stat = srtp_ekt_alloc(&str->ekt, p->ekt); |
| if (stat) { |
| auth_dealloc(str->rtcp_auth); |
| cipher_dealloc(str->rtcp_cipher); |
| auth_dealloc(str->rtp_auth); |
| cipher_dealloc(str->rtp_cipher); |
| srtp_crypto_free(str->limit); |
| srtp_crypto_free(str); |
| return stat; |
| } |
| |
| return srtp_err_status_ok; |
| } |
| |
| srtp_err_status_t |
| srtp_stream_dealloc(srtp_t session, srtp_stream_ctx_t *stream) { |
| srtp_err_status_t status; |
| |
| /* |
| * we use a conservative deallocation strategy - if any deallocation |
| * fails, then we report that fact without trying to deallocate |
| * anything else |
| */ |
| |
| /* deallocate cipher, if it is not the same as that in template */ |
| if (session->stream_template |
| && stream->rtp_cipher == session->stream_template->rtp_cipher) { |
| /* do nothing */ |
| } else { |
| status = cipher_dealloc(stream->rtp_cipher); |
| if (status) |
| return status; |
| } |
| |
| /* deallocate auth function, if it is not the same as that in template */ |
| if (session->stream_template |
| && stream->rtp_auth == session->stream_template->rtp_auth) { |
| /* do nothing */ |
| } else { |
| status = auth_dealloc(stream->rtp_auth); |
| if (status) |
| return status; |
| } |
| |
| /* deallocate key usage limit, if it is not the same as that in template */ |
| if (session->stream_template |
| && stream->limit == session->stream_template->limit) { |
| /* do nothing */ |
| } else { |
| srtp_crypto_free(stream->limit); |
| } |
| |
| /* |
| * deallocate rtcp cipher, if it is not the same as that in |
| * template |
| */ |
| if (session->stream_template |
| && stream->rtcp_cipher == session->stream_template->rtcp_cipher) { |
| /* do nothing */ |
| } else { |
| status = cipher_dealloc(stream->rtcp_cipher); |
| if (status) |
| return status; |
| } |
| |
| /* |
| * deallocate rtcp auth function, if it is not the same as that in |
| * template |
| */ |
| if (session->stream_template |
| && stream->rtcp_auth == session->stream_template->rtcp_auth) { |
| /* do nothing */ |
| } else { |
| status = auth_dealloc(stream->rtcp_auth); |
| if (status) |
| return status; |
| } |
| |
| status = srtp_rdbx_dealloc(&stream->rtp_rdbx); |
| if (status) |
| return status; |
| |
| /* DAM - need to deallocate EKT here */ |
| |
| /* |
| * zeroize the salt value |
| */ |
| memset(stream->salt, 0, SRTP_AEAD_SALT_LEN); |
| memset(stream->c_salt, 0, SRTP_AEAD_SALT_LEN); |
| |
| |
| /* deallocate srtp stream context */ |
| srtp_crypto_free(stream); |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| /* |
| * srtp_stream_clone(stream_template, new) allocates a new stream and |
| * initializes it using the cipher and auth of the stream_template |
| * |
| * the only unique data in a cloned stream is the replay database and |
| * the SSRC |
| */ |
| |
| srtp_err_status_t |
| srtp_stream_clone(const srtp_stream_ctx_t *stream_template, |
| uint32_t ssrc, |
| srtp_stream_ctx_t **str_ptr) { |
| srtp_err_status_t status; |
| srtp_stream_ctx_t *str; |
| |
| debug_print(mod_srtp, "cloning stream (SSRC: 0x%08x)", ssrc); |
| |
| /* allocate srtp stream and set str_ptr */ |
| str = (srtp_stream_ctx_t *) srtp_crypto_alloc(sizeof(srtp_stream_ctx_t)); |
| if (str == NULL) |
| return srtp_err_status_alloc_fail; |
| *str_ptr = str; |
| |
| /* set cipher and auth pointers to those of the template */ |
| str->rtp_cipher = stream_template->rtp_cipher; |
| str->rtp_auth = stream_template->rtp_auth; |
| str->rtcp_cipher = stream_template->rtcp_cipher; |
| str->rtcp_auth = stream_template->rtcp_auth; |
| |
| /* set key limit to point to that of the template */ |
| status = srtp_key_limit_clone(stream_template->limit, &str->limit); |
| if (status) { |
| srtp_crypto_free(*str_ptr); |
| *str_ptr = NULL; |
| return status; |
| } |
| |
| /* initialize replay databases */ |
| status = srtp_rdbx_init(&str->rtp_rdbx, |
| srtp_rdbx_get_window_size(&stream_template->rtp_rdbx)); |
| if (status) { |
| srtp_crypto_free(*str_ptr); |
| *str_ptr = NULL; |
| return status; |
| } |
| srtp_rdb_init(&str->rtcp_rdb); |
| str->allow_repeat_tx = stream_template->allow_repeat_tx; |
| |
| /* set ssrc to that provided */ |
| str->ssrc = ssrc; |
| |
| /* set direction and security services */ |
| str->direction = stream_template->direction; |
| str->rtp_services = stream_template->rtp_services; |
| str->rtcp_services = stream_template->rtcp_services; |
| |
| /* set pointer to EKT data associated with stream */ |
| str->ekt = stream_template->ekt; |
| |
| /* Copy the salt values */ |
| memcpy(str->salt, stream_template->salt, SRTP_AEAD_SALT_LEN); |
| memcpy(str->c_salt, stream_template->c_salt, SRTP_AEAD_SALT_LEN); |
| |
| /* defensive coding */ |
| str->next = NULL; |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| /* |
| * key derivation functions, internal to libSRTP |
| * |
| * srtp_kdf_t is a key derivation context |
| * |
| * srtp_kdf_init(&kdf, cipher_id, k, keylen) initializes kdf to use cipher |
| * described by cipher_id, with the master key k with length in octets keylen. |
| * |
| * srtp_kdf_generate(&kdf, l, kl, keylen) derives the key |
| * corresponding to label l and puts it into kl; the length |
| * of the key in octets is provided as keylen. this function |
| * should be called once for each subkey that is derived. |
| * |
| * srtp_kdf_clear(&kdf) zeroizes and deallocates the kdf state |
| */ |
| |
| typedef enum { |
| label_rtp_encryption = 0x00, |
| label_rtp_msg_auth = 0x01, |
| label_rtp_salt = 0x02, |
| label_rtcp_encryption = 0x03, |
| label_rtcp_msg_auth = 0x04, |
| label_rtcp_salt = 0x05 |
| } srtp_prf_label; |
| |
| |
| /* |
| * srtp_kdf_t represents a key derivation function. The SRTP |
| * default KDF is the only one implemented at present. |
| */ |
| |
| typedef struct { |
| srtp_cipher_t *cipher; /* cipher used for key derivation */ |
| } srtp_kdf_t; |
| |
| srtp_err_status_t |
| srtp_kdf_init(srtp_kdf_t *kdf, srtp_cipher_type_id_t cipher_id, const uint8_t *key, int length) { |
| |
| srtp_err_status_t stat; |
| stat = srtp_crypto_kernel_alloc_cipher(cipher_id, &kdf->cipher, length, 0); |
| if (stat) |
| return stat; |
| |
| stat = cipher_init(kdf->cipher, key); |
| if (stat) { |
| cipher_dealloc(kdf->cipher); |
| return stat; |
| } |
| |
| return srtp_err_status_ok; |
| } |
| |
| srtp_err_status_t |
| srtp_kdf_generate(srtp_kdf_t *kdf, srtp_prf_label label, |
| uint8_t *key, unsigned int length) { |
| |
| v128_t nonce; |
| srtp_err_status_t status; |
| |
| /* set eigth octet of nonce to <label>, set the rest of it to zero */ |
| v128_set_to_zero(&nonce); |
| nonce.v8[7] = label; |
| |
| status = cipher_set_iv(kdf->cipher, &nonce, direction_encrypt); |
| if (status) |
| return status; |
| |
| /* generate keystream output */ |
| octet_string_set_to_zero(key, length); |
| status = srtp_cipher_encrypt(kdf->cipher, key, &length); |
| if (status) |
| return status; |
| |
| return srtp_err_status_ok; |
| } |
| |
| srtp_err_status_t |
| srtp_kdf_clear(srtp_kdf_t *kdf) { |
| srtp_err_status_t status; |
| status = cipher_dealloc(kdf->cipher); |
| if (status) |
| return status; |
| kdf->cipher = NULL; |
| |
| return srtp_err_status_ok; |
| } |
| |
| /* |
| * end of key derivation functions |
| */ |
| |
| #define MAX_SRTP_KEY_LEN 256 |
| |
| |
| /* Get the base key length corresponding to a given combined key+salt |
| * length for the given cipher. |
| * Assumption is that for AES-ICM a key length < 30 is Ismacryp using |
| * AES-128 and short salts; everything else uses a salt length of 14. |
| * TODO: key and salt lengths should be separate fields in the policy. */ |
| static inline int base_key_length(const srtp_cipher_type_t *cipher, int key_length) |
| { |
| switch (cipher->id) { |
| case SRTP_AES_128_ICM: |
| case SRTP_AES_192_ICM: |
| case SRTP_AES_256_ICM: |
| /* The legacy modes are derived from |
| * the configured key length on the policy */ |
| return key_length - 14; |
| break; |
| case SRTP_AES_128_GCM: |
| return 16; |
| break; |
| case SRTP_AES_256_GCM: |
| return 32; |
| break; |
| default: |
| return key_length; |
| break; |
| } |
| } |
| |
| srtp_err_status_t |
| srtp_stream_init_keys(srtp_stream_ctx_t *srtp, const void *key) { |
| srtp_err_status_t stat; |
| srtp_kdf_t kdf; |
| uint8_t tmp_key[MAX_SRTP_KEY_LEN]; |
| int kdf_keylen = 30, rtp_keylen, rtcp_keylen; |
| int rtp_base_key_len, rtp_salt_len; |
| int rtcp_base_key_len, rtcp_salt_len; |
| |
| /* If RTP or RTCP have a key length > AES-128, assume matching kdf. */ |
| /* TODO: kdf algorithm, master key length, and master salt length should |
| * be part of srtp_policy_t. */ |
| rtp_keylen = srtp_cipher_get_key_length(srtp->rtp_cipher); |
| rtcp_keylen = srtp_cipher_get_key_length(srtp->rtcp_cipher); |
| rtp_base_key_len = base_key_length(srtp->rtp_cipher->type, rtp_keylen); |
| rtp_salt_len = rtp_keylen - rtp_base_key_len; |
| |
| if (rtp_keylen > kdf_keylen) { |
| kdf_keylen = 46; /* AES-CTR mode is always used for KDF */ |
| } |
| |
| if (rtcp_keylen > kdf_keylen) { |
| kdf_keylen = 46; /* AES-CTR mode is always used for KDF */ |
| } |
| |
| debug_print(mod_srtp, "srtp key len: %d", rtp_keylen); |
| debug_print(mod_srtp, "srtcp key len: %d", rtcp_keylen); |
| debug_print(mod_srtp, "base key len: %d", rtp_base_key_len); |
| debug_print(mod_srtp, "kdf key len: %d", kdf_keylen); |
| debug_print(mod_srtp, "rtp salt len: %d", rtp_salt_len); |
| |
| /* |
| * Make sure the key given to us is 'zero' appended. GCM |
| * mode uses a shorter master SALT (96 bits), but still relies on |
| * the legacy CTR mode KDF, which uses a 112 bit master SALT. |
| */ |
| memset(tmp_key, 0x0, MAX_SRTP_KEY_LEN); |
| memcpy(tmp_key, key, (rtp_base_key_len + rtp_salt_len)); |
| |
| /* initialize KDF state */ |
| stat = srtp_kdf_init(&kdf, SRTP_AES_ICM, (const uint8_t *)tmp_key, kdf_keylen); |
| if (stat) { |
| return srtp_err_status_init_fail; |
| } |
| |
| /* generate encryption key */ |
| stat = srtp_kdf_generate(&kdf, label_rtp_encryption, |
| tmp_key, rtp_base_key_len); |
| if (stat) { |
| /* zeroize temp buffer */ |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| return srtp_err_status_init_fail; |
| } |
| debug_print(mod_srtp, "cipher key: %s", |
| srtp_octet_string_hex_string(tmp_key, rtp_base_key_len)); |
| |
| /* |
| * if the cipher in the srtp context uses a salt, then we need |
| * to generate the salt value |
| */ |
| if (rtp_salt_len > 0) { |
| debug_print(mod_srtp, "found rtp_salt_len > 0, generating salt", NULL); |
| |
| /* generate encryption salt, put after encryption key */ |
| stat = srtp_kdf_generate(&kdf, label_rtp_salt, |
| tmp_key + rtp_base_key_len, rtp_salt_len); |
| if (stat) { |
| /* zeroize temp buffer */ |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| return srtp_err_status_init_fail; |
| } |
| memcpy(srtp->salt, tmp_key + rtp_base_key_len, SRTP_AEAD_SALT_LEN); |
| } |
| if (rtp_salt_len > 0) { |
| debug_print(mod_srtp, "cipher salt: %s", |
| srtp_octet_string_hex_string(tmp_key + rtp_base_key_len, rtp_salt_len)); |
| } |
| |
| /* initialize cipher */ |
| stat = cipher_init(srtp->rtp_cipher, tmp_key); |
| if (stat) { |
| /* zeroize temp buffer */ |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| return srtp_err_status_init_fail; |
| } |
| |
| /* generate authentication key */ |
| stat = srtp_kdf_generate(&kdf, label_rtp_msg_auth, |
| tmp_key, srtp_auth_get_key_length(srtp->rtp_auth)); |
| if (stat) { |
| /* zeroize temp buffer */ |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| return srtp_err_status_init_fail; |
| } |
| debug_print(mod_srtp, "auth key: %s", |
| srtp_octet_string_hex_string(tmp_key, |
| srtp_auth_get_key_length(srtp->rtp_auth))); |
| |
| /* initialize auth function */ |
| stat = auth_init(srtp->rtp_auth, tmp_key); |
| if (stat) { |
| /* zeroize temp buffer */ |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| return srtp_err_status_init_fail; |
| } |
| |
| /* |
| * ...now initialize SRTCP keys |
| */ |
| |
| rtcp_base_key_len = base_key_length(srtp->rtcp_cipher->type, rtcp_keylen); |
| rtcp_salt_len = rtcp_keylen - rtcp_base_key_len; |
| debug_print(mod_srtp, "rtcp salt len: %d", rtcp_salt_len); |
| |
| /* generate encryption key */ |
| stat = srtp_kdf_generate(&kdf, label_rtcp_encryption, |
| tmp_key, rtcp_base_key_len); |
| if (stat) { |
| /* zeroize temp buffer */ |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| return srtp_err_status_init_fail; |
| } |
| |
| /* |
| * if the cipher in the srtp context uses a salt, then we need |
| * to generate the salt value |
| */ |
| if (rtcp_salt_len > 0) { |
| debug_print(mod_srtp, "found rtcp_salt_len > 0, generating rtcp salt", |
| NULL); |
| |
| /* generate encryption salt, put after encryption key */ |
| stat = srtp_kdf_generate(&kdf, label_rtcp_salt, |
| tmp_key + rtcp_base_key_len, rtcp_salt_len); |
| if (stat) { |
| /* zeroize temp buffer */ |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| return srtp_err_status_init_fail; |
| } |
| memcpy(srtp->c_salt, tmp_key + rtcp_base_key_len, SRTP_AEAD_SALT_LEN); |
| } |
| debug_print(mod_srtp, "rtcp cipher key: %s", |
| srtp_octet_string_hex_string(tmp_key, rtcp_base_key_len)); |
| if (rtcp_salt_len > 0) { |
| debug_print(mod_srtp, "rtcp cipher salt: %s", |
| srtp_octet_string_hex_string(tmp_key + rtcp_base_key_len, rtcp_salt_len)); |
| } |
| |
| /* initialize cipher */ |
| stat = cipher_init(srtp->rtcp_cipher, tmp_key); |
| if (stat) { |
| /* zeroize temp buffer */ |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| return srtp_err_status_init_fail; |
| } |
| |
| /* generate authentication key */ |
| stat = srtp_kdf_generate(&kdf, label_rtcp_msg_auth, |
| tmp_key, srtp_auth_get_key_length(srtp->rtcp_auth)); |
| if (stat) { |
| /* zeroize temp buffer */ |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| return srtp_err_status_init_fail; |
| } |
| |
| debug_print(mod_srtp, "rtcp auth key: %s", |
| srtp_octet_string_hex_string(tmp_key, |
| srtp_auth_get_key_length(srtp->rtcp_auth))); |
| |
| /* initialize auth function */ |
| stat = auth_init(srtp->rtcp_auth, tmp_key); |
| if (stat) { |
| /* zeroize temp buffer */ |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| return srtp_err_status_init_fail; |
| } |
| |
| /* clear memory then return */ |
| stat = srtp_kdf_clear(&kdf); |
| octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN); |
| if (stat) |
| return srtp_err_status_init_fail; |
| |
| return srtp_err_status_ok; |
| } |
| |
| srtp_err_status_t |
| srtp_stream_init(srtp_stream_ctx_t *srtp, |
| const srtp_policy_t *p) { |
| srtp_err_status_t err; |
| |
| debug_print(mod_srtp, "initializing stream (SSRC: 0x%08x)", |
| p->ssrc.value); |
| |
| /* initialize replay database */ |
| /* window size MUST be at least 64. MAY be larger. Values more than |
| * 2^15 aren't meaningful due to how extended sequence numbers are |
| * calculated. Let a window size of 0 imply the default value. */ |
| |
| if (p->window_size != 0 && (p->window_size < 64 || p->window_size >= 0x8000)) |
| return srtp_err_status_bad_param; |
| |
| if (p->window_size != 0) |
| err = srtp_rdbx_init(&srtp->rtp_rdbx, p->window_size); |
| else |
| err = srtp_rdbx_init(&srtp->rtp_rdbx, 128); |
| if (err) return err; |
| |
| /* initialize key limit to maximum value */ |
| #ifdef NO_64BIT_MATH |
| { |
| uint64_t temp; |
| temp = make64(UINT_MAX,UINT_MAX); |
| srtp_key_limit_set(srtp->limit, temp); |
| } |
| #else |
| srtp_key_limit_set(srtp->limit, 0xffffffffffffLL); |
| #endif |
| |
| /* set the SSRC value */ |
| srtp->ssrc = htonl(p->ssrc.value); |
| |
| /* set the security service flags */ |
| srtp->rtp_services = p->rtp.sec_serv; |
| srtp->rtcp_services = p->rtcp.sec_serv; |
| |
| /* |
| * set direction to unknown - this flag gets checked in srtp_protect(), |
| * srtp_unprotect(), srtp_protect_rtcp(), and srtp_unprotect_rtcp(), and |
| * gets set appropriately if it is set to unknown. |
| */ |
| srtp->direction = dir_unknown; |
| |
| /* initialize SRTCP replay database */ |
| srtp_rdb_init(&srtp->rtcp_rdb); |
| |
| /* initialize allow_repeat_tx */ |
| /* guard against uninitialized memory: allow only 0 or 1 here */ |
| if (p->allow_repeat_tx != 0 && p->allow_repeat_tx != 1) { |
| srtp_rdbx_dealloc(&srtp->rtp_rdbx); |
| return srtp_err_status_bad_param; |
| } |
| srtp->allow_repeat_tx = p->allow_repeat_tx; |
| |
| /* DAM - no RTCP key limit at present */ |
| |
| /* initialize keys */ |
| err = srtp_stream_init_keys(srtp, p->key); |
| if (err) { |
| srtp_rdbx_dealloc(&srtp->rtp_rdbx); |
| return err; |
| } |
| |
| /* |
| * if EKT is in use, then initialize the EKT data associated with |
| * the stream |
| */ |
| err = srtp_ekt_stream_init_from_policy(srtp->ekt, p->ekt); |
| if (err) { |
| srtp_rdbx_dealloc(&srtp->rtp_rdbx); |
| return err; |
| } |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| /* |
| * srtp_event_reporter is an event handler function that merely |
| * reports the events that are reported by the callbacks |
| */ |
| |
| void |
| srtp_event_reporter(srtp_event_data_t *data) { |
| |
| srtp_err_report(srtp_err_level_warning, "srtp: in stream 0x%x: ", |
| data->stream->ssrc); |
| |
| switch(data->event) { |
| case event_ssrc_collision: |
| srtp_err_report(srtp_err_level_warning, "\tSSRC collision\n"); |
| break; |
| case event_key_soft_limit: |
| srtp_err_report(srtp_err_level_warning, "\tkey usage soft limit reached\n"); |
| break; |
| case event_key_hard_limit: |
| srtp_err_report(srtp_err_level_warning, "\tkey usage hard limit reached\n"); |
| break; |
| case event_packet_index_limit: |
| srtp_err_report(srtp_err_level_warning, "\tpacket index limit reached\n"); |
| break; |
| default: |
| srtp_err_report(srtp_err_level_warning, "\tunknown event reported to handler\n"); |
| } |
| } |
| |
| /* |
| * srtp_event_handler is a global variable holding a pointer to the |
| * event handler function; this function is called for any unexpected |
| * event that needs to be handled out of the SRTP data path. see |
| * srtp_event_t in srtp.h for more info |
| * |
| * it is okay to set srtp_event_handler to NULL, but we set |
| * it to the srtp_event_reporter. |
| */ |
| |
| static srtp_event_handler_func_t *srtp_event_handler = srtp_event_reporter; |
| |
| srtp_err_status_t |
| srtp_install_event_handler(srtp_event_handler_func_t func) { |
| |
| /* |
| * note that we accept NULL arguments intentionally - calling this |
| * function with a NULL arguments removes an event handler that's |
| * been previously installed |
| */ |
| |
| /* set global event handling function */ |
| srtp_event_handler = func; |
| return srtp_err_status_ok; |
| } |
| |
| /* |
| * AEAD uses a new IV formation method. This function implements |
| * section 9.1 from draft-ietf-avtcore-srtp-aes-gcm-07.txt. The |
| * calculation is defined as, where (+) is the xor operation: |
| * |
| * |
| * 0 0 0 0 0 0 0 0 0 0 1 1 |
| * 0 1 2 3 4 5 6 7 8 9 0 1 |
| * +--+--+--+--+--+--+--+--+--+--+--+--+ |
| * |00|00| SSRC | ROC | SEQ |---+ |
| * +--+--+--+--+--+--+--+--+--+--+--+--+ | |
| * | |
| * +--+--+--+--+--+--+--+--+--+--+--+--+ | |
| * | Encryption Salt |->(+) |
| * +--+--+--+--+--+--+--+--+--+--+--+--+ | |
| * | |
| * +--+--+--+--+--+--+--+--+--+--+--+--+ | |
| * | Initialization Vector |<--+ |
| * +--+--+--+--+--+--+--+--+--+--+--+--+* |
| * |
| * Input: *stream - pointer to SRTP stream context, used to retrieve |
| * the SALT |
| * *iv - Pointer to receive the calculated IV |
| * *seq - The ROC and SEQ value to use for the |
| * IV calculation. |
| * *hdr - The RTP header, used to get the SSRC value |
| * |
| */ |
| static void srtp_calc_aead_iv(srtp_stream_ctx_t *stream, v128_t *iv, |
| srtp_xtd_seq_num_t *seq, srtp_hdr_t *hdr) |
| { |
| v128_t in; |
| v128_t salt; |
| |
| #ifdef NO_64BIT_MATH |
| uint32_t local_roc = ((high32(*seq) << 16) | |
| (low32(*seq) >> 16)); |
| uint16_t local_seq = (uint16_t) (low32(*seq)); |
| #else |
| uint32_t local_roc = (uint32_t)(*seq >> 16); |
| uint16_t local_seq = (uint16_t) *seq; |
| #endif |
| |
| memset(&in, 0, sizeof(v128_t)); |
| memset(&salt, 0, sizeof(v128_t)); |
| |
| in.v16[5] = htons(local_seq); |
| local_roc = htonl(local_roc); |
| memcpy(&in.v16[3], &local_roc, sizeof(local_roc)); |
| |
| /* |
| * Copy in the RTP SSRC value |
| */ |
| memcpy(&in.v8[2], &hdr->ssrc, 4); |
| debug_print(mod_srtp, "Pre-salted RTP IV = %s\n", v128_hex_string(&in)); |
| |
| /* |
| * Get the SALT value from the context |
| */ |
| memcpy(salt.v8, stream->salt, SRTP_AEAD_SALT_LEN); |
| debug_print(mod_srtp, "RTP SALT = %s\n", v128_hex_string(&salt)); |
| |
| /* |
| * Finally, apply tyhe SALT to the input |
| */ |
| v128_xor(iv, &in, &salt); |
| } |
| |
| |
| /* |
| * This function handles outgoing SRTP packets while in AEAD mode, |
| * which currently supports AES-GCM encryption. All packets are |
| * encrypted and authenticated. |
| */ |
| static srtp_err_status_t |
| srtp_protect_aead (srtp_ctx_t *ctx, srtp_stream_ctx_t *stream, |
| void *rtp_hdr, unsigned int *pkt_octet_len) |
| { |
| srtp_hdr_t *hdr = (srtp_hdr_t*)rtp_hdr; |
| uint32_t *enc_start; /* pointer to start of encrypted portion */ |
| unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */ |
| srtp_xtd_seq_num_t est; /* estimated xtd_seq_num_t of *hdr */ |
| int delta; /* delta of local pkt idx and that in hdr */ |
| srtp_err_status_t status; |
| int tag_len; |
| v128_t iv; |
| unsigned int aad_len; |
| |
| debug_print(mod_srtp, "function srtp_protect_aead", NULL); |
| |
| /* |
| * update the key usage limit, and check it to make sure that we |
| * didn't just hit either the soft limit or the hard limit, and call |
| * the event handler if we hit either. |
| */ |
| switch (srtp_key_limit_update(stream->limit)) { |
| case srtp_key_event_normal: |
| break; |
| case srtp_key_event_hard_limit: |
| srtp_handle_event(ctx, stream, event_key_hard_limit); |
| return srtp_err_status_key_expired; |
| case srtp_key_event_soft_limit: |
| default: |
| srtp_handle_event(ctx, stream, event_key_soft_limit); |
| break; |
| } |
| |
| /* get tag length from stream */ |
| tag_len = srtp_auth_get_tag_length(stream->rtp_auth); |
| |
| /* |
| * find starting point for encryption and length of data to be |
| * encrypted - the encrypted portion starts after the rtp header |
| * extension, if present; otherwise, it starts after the last csrc, |
| * if any are present |
| */ |
| enc_start = (uint32_t*)hdr + uint32s_in_rtp_header + hdr->cc; |
| if (hdr->x == 1) { |
| srtp_hdr_xtnd_t *xtn_hdr = (srtp_hdr_xtnd_t*)enc_start; |
| enc_start += (ntohs(xtn_hdr->length) + 1); |
| } |
| if (!((uint8_t*)enc_start < (uint8_t*)hdr + *pkt_octet_len)) |
| return srtp_err_status_parse_err; |
| enc_octet_len = (unsigned int)(*pkt_octet_len - |
| ((uint8_t*)enc_start - (uint8_t*)hdr)); |
| |
| /* |
| * estimate the packet index using the start of the replay window |
| * and the sequence number from the header |
| */ |
| delta = srtp_rdbx_estimate_index(&stream->rtp_rdbx, &est, ntohs(hdr->seq)); |
| status = srtp_rdbx_check(&stream->rtp_rdbx, delta); |
| if (status) { |
| if (status != srtp_err_status_replay_fail || !stream->allow_repeat_tx) { |
| return status; /* we've been asked to reuse an index */ |
| } |
| } else { |
| srtp_rdbx_add_index(&stream->rtp_rdbx, delta); |
| } |
| |
| #ifdef NO_64BIT_MATH |
| debug_print2(mod_srtp, "estimated packet index: %08x%08x", |
| high32(est), low32(est)); |
| #else |
| debug_print(mod_srtp, "estimated packet index: %016llx", est); |
| #endif |
| |
| /* |
| * AEAD uses a new IV formation method |
| */ |
| srtp_calc_aead_iv(stream, &iv, &est, hdr); |
| status = cipher_set_iv(stream->rtp_cipher, &iv, direction_encrypt); |
| if (status) { |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* shift est, put into network byte order */ |
| #ifdef NO_64BIT_MATH |
| est = be64_to_cpu(make64((high32(est) << 16) | |
| (low32(est) >> 16), |
| low32(est) << 16)); |
| #else |
| est = be64_to_cpu(est << 16); |
| #endif |
| |
| /* |
| * Set the AAD over the RTP header |
| */ |
| aad_len = (uint8_t *)enc_start - (uint8_t *)hdr; |
| status = cipher_set_aad(stream->rtp_cipher, (uint8_t*)hdr, aad_len); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| |
| /* Encrypt the payload */ |
| status = srtp_cipher_encrypt(stream->rtp_cipher, |
| (uint8_t*)enc_start, &enc_octet_len); |
| if (status) { |
| return srtp_err_status_cipher_fail; |
| } |
| /* |
| * If we're doing GCM, we need to get the tag |
| * and append that to the output |
| */ |
| status = cipher_get_tag(stream->rtp_cipher, |
| (uint8_t*)enc_start+enc_octet_len, &tag_len); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| enc_octet_len += tag_len; |
| |
| /* increase the packet length by the length of the auth tag */ |
| *pkt_octet_len += tag_len; |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| /* |
| * This function handles incoming SRTP packets while in AEAD mode, |
| * which currently supports AES-GCM encryption. All packets are |
| * encrypted and authenticated. Note, the auth tag is at the end |
| * of the packet stream and is automatically checked by GCM |
| * when decrypting the payload. |
| */ |
| static srtp_err_status_t |
| srtp_unprotect_aead (srtp_ctx_t *ctx, srtp_stream_ctx_t *stream, int delta, |
| srtp_xtd_seq_num_t est, void *srtp_hdr, unsigned int *pkt_octet_len) |
| { |
| srtp_hdr_t *hdr = (srtp_hdr_t*)srtp_hdr; |
| uint32_t *enc_start; /* pointer to start of encrypted portion */ |
| unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */ |
| v128_t iv; |
| srtp_err_status_t status; |
| int tag_len; |
| unsigned int aad_len; |
| |
| debug_print(mod_srtp, "function srtp_unprotect_aead", NULL); |
| |
| #ifdef NO_64BIT_MATH |
| debug_print2(mod_srtp, "estimated u_packet index: %08x%08x", high32(est), low32(est)); |
| #else |
| debug_print(mod_srtp, "estimated u_packet index: %016llx", est); |
| #endif |
| |
| /* get tag length from stream */ |
| tag_len = srtp_auth_get_tag_length(stream->rtp_auth); |
| |
| /* |
| * AEAD uses a new IV formation method |
| */ |
| srtp_calc_aead_iv(stream, &iv, &est, hdr); |
| status = cipher_set_iv(stream->rtp_cipher, &iv, direction_decrypt); |
| if (status) { |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* |
| * find starting point for decryption and length of data to be |
| * decrypted - the encrypted portion starts after the rtp header |
| * extension, if present; otherwise, it starts after the last csrc, |
| * if any are present |
| */ |
| enc_start = (uint32_t*)hdr + uint32s_in_rtp_header + hdr->cc; |
| if (hdr->x == 1) { |
| srtp_hdr_xtnd_t *xtn_hdr = (srtp_hdr_xtnd_t*)enc_start; |
| enc_start += (ntohs(xtn_hdr->length) + 1); |
| } |
| if (!((uint8_t*)enc_start < (uint8_t*)hdr + *pkt_octet_len)) |
| return srtp_err_status_parse_err; |
| /* |
| * We pass the tag down to the cipher when doing GCM mode |
| */ |
| enc_octet_len = (unsigned int)(*pkt_octet_len - |
| ((uint8_t*)enc_start - (uint8_t*)hdr)); |
| |
| /* |
| * Sanity check the encrypted payload length against |
| * the tag size. It must always be at least as large |
| * as the tag length. |
| */ |
| if (enc_octet_len < tag_len) { |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* |
| * update the key usage limit, and check it to make sure that we |
| * didn't just hit either the soft limit or the hard limit, and call |
| * the event handler if we hit either. |
| */ |
| switch (srtp_key_limit_update(stream->limit)) { |
| case srtp_key_event_normal: |
| break; |
| case srtp_key_event_soft_limit: |
| srtp_handle_event(ctx, stream, event_key_soft_limit); |
| break; |
| case srtp_key_event_hard_limit: |
| srtp_handle_event(ctx, stream, event_key_hard_limit); |
| return srtp_err_status_key_expired; |
| default: |
| break; |
| } |
| |
| /* |
| * Set the AAD for AES-GCM, which is the RTP header |
| */ |
| aad_len = (uint8_t *)enc_start - (uint8_t *)hdr; |
| status = cipher_set_aad(stream->rtp_cipher, (uint8_t*)hdr, aad_len); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| |
| /* Decrypt the ciphertext. This also checks the auth tag based |
| * on the AAD we just specified above */ |
| status = srtp_cipher_decrypt(stream->rtp_cipher, (uint8_t*)enc_start, &enc_octet_len); |
| if (status) { |
| return status; |
| } |
| |
| /* |
| * verify that stream is for received traffic - this check will |
| * detect SSRC collisions, since a stream that appears in both |
| * srtp_protect() and srtp_unprotect() will fail this test in one of |
| * those functions. |
| * |
| * we do this check *after* the authentication check, so that the |
| * latter check will catch any attempts to fool us into thinking |
| * that we've got a collision |
| */ |
| if (stream->direction != dir_srtp_receiver) { |
| if (stream->direction == dir_unknown) { |
| stream->direction = dir_srtp_receiver; |
| } else { |
| srtp_handle_event(ctx, stream, event_ssrc_collision); |
| } |
| } |
| |
| /* |
| * if the stream is a 'provisional' one, in which the template context |
| * is used, then we need to allocate a new stream at this point, since |
| * the authentication passed |
| */ |
| if (stream == ctx->stream_template) { |
| srtp_stream_ctx_t *new_stream; |
| |
| /* |
| * allocate and initialize a new stream |
| * |
| * note that we indicate failure if we can't allocate the new |
| * stream, and some implementations will want to not return |
| * failure here |
| */ |
| status = srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream); |
| if (status) { |
| return status; |
| } |
| |
| /* add new stream to the head of the stream_list */ |
| new_stream->next = ctx->stream_list; |
| ctx->stream_list = new_stream; |
| |
| /* set stream (the pointer used in this function) */ |
| stream = new_stream; |
| } |
| |
| /* |
| * the message authentication function passed, so add the packet |
| * index into the replay database |
| */ |
| srtp_rdbx_add_index(&stream->rtp_rdbx, delta); |
| |
| /* decrease the packet length by the length of the auth tag */ |
| *pkt_octet_len -= tag_len; |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| |
| |
| srtp_err_status_t |
| srtp_protect(srtp_ctx_t *ctx, void *rtp_hdr, int *pkt_octet_len) { |
| srtp_hdr_t *hdr = (srtp_hdr_t *)rtp_hdr; |
| uint32_t *enc_start; /* pointer to start of encrypted portion */ |
| uint32_t *auth_start; /* pointer to start of auth. portion */ |
| unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */ |
| srtp_xtd_seq_num_t est; /* estimated xtd_seq_num_t of *hdr */ |
| int delta; /* delta of local pkt idx and that in hdr */ |
| uint8_t *auth_tag = NULL; /* location of auth_tag within packet */ |
| srtp_err_status_t status; |
| int tag_len; |
| srtp_stream_ctx_t *stream; |
| uint32_t prefix_len; |
| |
| debug_print(mod_srtp, "function srtp_protect", NULL); |
| |
| /* we assume the hdr is 32-bit aligned to start */ |
| |
| /* check the packet length - it must at least contain a full header */ |
| if (*pkt_octet_len < octets_in_rtp_header) |
| return srtp_err_status_bad_param; |
| |
| /* |
| * look up ssrc in srtp_stream list, and process the packet with |
| * the appropriate stream. if we haven't seen this stream before, |
| * there's a template key for this srtp_session, and the cipher |
| * supports key-sharing, then we assume that a new stream using |
| * that key has just started up |
| */ |
| stream = srtp_get_stream(ctx, hdr->ssrc); |
| if (stream == NULL) { |
| if (ctx->stream_template != NULL) { |
| srtp_stream_ctx_t *new_stream; |
| |
| /* allocate and initialize a new stream */ |
| status = srtp_stream_clone(ctx->stream_template, |
| hdr->ssrc, &new_stream); |
| if (status) |
| return status; |
| |
| /* add new stream to the head of the stream_list */ |
| new_stream->next = ctx->stream_list; |
| ctx->stream_list = new_stream; |
| |
| /* set direction to outbound */ |
| new_stream->direction = dir_srtp_sender; |
| |
| /* set stream (the pointer used in this function) */ |
| stream = new_stream; |
| } else { |
| /* no template stream, so we return an error */ |
| return srtp_err_status_no_ctx; |
| } |
| } |
| |
| /* |
| * verify that stream is for sending traffic - this check will |
| * detect SSRC collisions, since a stream that appears in both |
| * srtp_protect() and srtp_unprotect() will fail this test in one of |
| * those functions. |
| */ |
| if (stream->direction != dir_srtp_sender) { |
| if (stream->direction == dir_unknown) { |
| stream->direction = dir_srtp_sender; |
| } else { |
| srtp_handle_event(ctx, stream, event_ssrc_collision); |
| } |
| } |
| |
| /* |
| * Check if this is an AEAD stream (GCM mode). If so, then dispatch |
| * the request to our AEAD handler. |
| */ |
| if (stream->rtp_cipher->algorithm == SRTP_AES_128_GCM || |
| stream->rtp_cipher->algorithm == SRTP_AES_256_GCM) { |
| return srtp_protect_aead(ctx, stream, rtp_hdr, (unsigned int*)pkt_octet_len); |
| } |
| |
| /* |
| * update the key usage limit, and check it to make sure that we |
| * didn't just hit either the soft limit or the hard limit, and call |
| * the event handler if we hit either. |
| */ |
| switch(srtp_key_limit_update(stream->limit)) { |
| case srtp_key_event_normal: |
| break; |
| case srtp_key_event_soft_limit: |
| srtp_handle_event(ctx, stream, event_key_soft_limit); |
| break; |
| case srtp_key_event_hard_limit: |
| srtp_handle_event(ctx, stream, event_key_hard_limit); |
| return srtp_err_status_key_expired; |
| default: |
| break; |
| } |
| |
| /* get tag length from stream */ |
| tag_len = srtp_auth_get_tag_length(stream->rtp_auth); |
| |
| /* |
| * find starting point for encryption and length of data to be |
| * encrypted - the encrypted portion starts after the rtp header |
| * extension, if present; otherwise, it starts after the last csrc, |
| * if any are present |
| * |
| * if we're not providing confidentiality, set enc_start to NULL |
| */ |
| if (stream->rtp_services & sec_serv_conf) { |
| enc_start = (uint32_t *)hdr + uint32s_in_rtp_header + hdr->cc; |
| if (hdr->x == 1) { |
| srtp_hdr_xtnd_t *xtn_hdr = (srtp_hdr_xtnd_t *)enc_start; |
| enc_start += (ntohs(xtn_hdr->length) + 1); |
| if (!((uint8_t*)enc_start < (uint8_t*)hdr + *pkt_octet_len)) |
| return srtp_err_status_parse_err; |
| } |
| enc_octet_len = (unsigned int)(*pkt_octet_len - |
| ((uint8_t*)enc_start - (uint8_t*)hdr)); |
| } else { |
| enc_start = NULL; |
| } |
| |
| /* |
| * if we're providing authentication, set the auth_start and auth_tag |
| * pointers to the proper locations; otherwise, set auth_start to NULL |
| * to indicate that no authentication is needed |
| */ |
| if (stream->rtp_services & sec_serv_auth) { |
| auth_start = (uint32_t *)hdr; |
| auth_tag = (uint8_t *)hdr + *pkt_octet_len; |
| } else { |
| auth_start = NULL; |
| auth_tag = NULL; |
| } |
| |
| /* |
| * estimate the packet index using the start of the replay window |
| * and the sequence number from the header |
| */ |
| delta = srtp_rdbx_estimate_index(&stream->rtp_rdbx, &est, ntohs(hdr->seq)); |
| status = srtp_rdbx_check(&stream->rtp_rdbx, delta); |
| if (status) { |
| if (status != srtp_err_status_replay_fail || !stream->allow_repeat_tx) |
| return status; /* we've been asked to reuse an index */ |
| } |
| else |
| srtp_rdbx_add_index(&stream->rtp_rdbx, delta); |
| |
| #ifdef NO_64BIT_MATH |
| debug_print2(mod_srtp, "estimated packet index: %08x%08x", |
| high32(est),low32(est)); |
| #else |
| debug_print(mod_srtp, "estimated packet index: %016llx", est); |
| #endif |
| |
| /* |
| * if we're using rindael counter mode, set nonce and seq |
| */ |
| if (stream->rtp_cipher->type->id == SRTP_AES_ICM || |
| stream->rtp_cipher->type->id == SRTP_AES_256_ICM) { |
| v128_t iv; |
| |
| iv.v32[0] = 0; |
| iv.v32[1] = hdr->ssrc; |
| #ifdef NO_64BIT_MATH |
| iv.v64[1] = be64_to_cpu(make64((high32(est) << 16) | (low32(est) >> 16), |
| low32(est) << 16)); |
| #else |
| iv.v64[1] = be64_to_cpu(est << 16); |
| #endif |
| status = cipher_set_iv(stream->rtp_cipher, &iv, direction_encrypt); |
| |
| } else { |
| v128_t iv; |
| |
| /* otherwise, set the index to est */ |
| #ifdef NO_64BIT_MATH |
| iv.v32[0] = 0; |
| iv.v32[1] = 0; |
| #else |
| iv.v64[0] = 0; |
| #endif |
| iv.v64[1] = be64_to_cpu(est); |
| status = cipher_set_iv(stream->rtp_cipher, &iv, direction_encrypt); |
| } |
| if (status) |
| return srtp_err_status_cipher_fail; |
| |
| /* shift est, put into network byte order */ |
| #ifdef NO_64BIT_MATH |
| est = be64_to_cpu(make64((high32(est) << 16) | |
| (low32(est) >> 16), |
| low32(est) << 16)); |
| #else |
| est = be64_to_cpu(est << 16); |
| #endif |
| |
| /* |
| * if we're authenticating using a universal hash, put the keystream |
| * prefix into the authentication tag |
| */ |
| if (auth_start) { |
| |
| prefix_len = srtp_auth_get_prefix_length(stream->rtp_auth); |
| if (prefix_len) { |
| status = srtp_cipher_output(stream->rtp_cipher, auth_tag, &prefix_len); |
| if (status) |
| return srtp_err_status_cipher_fail; |
| debug_print(mod_srtp, "keystream prefix: %s", |
| srtp_octet_string_hex_string(auth_tag, prefix_len)); |
| } |
| } |
| |
| /* if we're encrypting, exor keystream into the message */ |
| if (enc_start) { |
| status = srtp_cipher_encrypt(stream->rtp_cipher, |
| (uint8_t *)enc_start, &enc_octet_len); |
| if (status) |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* |
| * if we're authenticating, run authentication function and put result |
| * into the auth_tag |
| */ |
| if (auth_start) { |
| |
| /* initialize auth func context */ |
| status = auth_start(stream->rtp_auth); |
| if (status) return status; |
| |
| /* run auth func over packet */ |
| status = auth_update(stream->rtp_auth, |
| (uint8_t *)auth_start, *pkt_octet_len); |
| if (status) return status; |
| |
| /* run auth func over ROC, put result into auth_tag */ |
| debug_print(mod_srtp, "estimated packet index: %016llx", est); |
| status = auth_compute(stream->rtp_auth, (uint8_t *)&est, 4, auth_tag); |
| debug_print(mod_srtp, "srtp auth tag: %s", |
| srtp_octet_string_hex_string(auth_tag, tag_len)); |
| if (status) |
| return srtp_err_status_auth_fail; |
| |
| } |
| |
| if (auth_tag) { |
| |
| /* increase the packet length by the length of the auth tag */ |
| *pkt_octet_len += tag_len; |
| } |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| srtp_err_status_t |
| srtp_unprotect(srtp_ctx_t *ctx, void *srtp_hdr, int *pkt_octet_len) { |
| srtp_hdr_t *hdr = (srtp_hdr_t *)srtp_hdr; |
| uint32_t *enc_start; /* pointer to start of encrypted portion */ |
| uint32_t *auth_start; /* pointer to start of auth. portion */ |
| unsigned int enc_octet_len = 0;/* number of octets in encrypted portion */ |
| uint8_t *auth_tag = NULL; /* location of auth_tag within packet */ |
| srtp_xtd_seq_num_t est; /* estimated xtd_seq_num_t of *hdr */ |
| int delta; /* delta of local pkt idx and that in hdr */ |
| v128_t iv; |
| srtp_err_status_t status; |
| srtp_stream_ctx_t *stream; |
| uint8_t tmp_tag[SRTP_MAX_TAG_LEN]; |
| uint32_t tag_len, prefix_len; |
| |
| debug_print(mod_srtp, "function srtp_unprotect", NULL); |
| |
| /* we assume the hdr is 32-bit aligned to start */ |
| |
| /* check the packet length - it must at least contain a full header */ |
| if (*pkt_octet_len < octets_in_rtp_header) |
| return srtp_err_status_bad_param; |
| |
| /* |
| * look up ssrc in srtp_stream list, and process the packet with |
| * the appropriate stream. if we haven't seen this stream before, |
| * there's only one key for this srtp_session, and the cipher |
| * supports key-sharing, then we assume that a new stream using |
| * that key has just started up |
| */ |
| stream = srtp_get_stream(ctx, hdr->ssrc); |
| if (stream == NULL) { |
| if (ctx->stream_template != NULL) { |
| stream = ctx->stream_template; |
| debug_print(mod_srtp, "using provisional stream (SSRC: 0x%08x)", |
| hdr->ssrc); |
| |
| /* |
| * set estimated packet index to sequence number from header, |
| * and set delta equal to the same value |
| */ |
| #ifdef NO_64BIT_MATH |
| est = (srtp_xtd_seq_num_t) make64(0,ntohs(hdr->seq)); |
| delta = low32(est); |
| #else |
| est = (srtp_xtd_seq_num_t) ntohs(hdr->seq); |
| delta = (int)est; |
| #endif |
| } else { |
| |
| /* |
| * no stream corresponding to SSRC found, and we don't do |
| * key-sharing, so return an error |
| */ |
| return srtp_err_status_no_ctx; |
| } |
| } else { |
| |
| /* estimate packet index from seq. num. in header */ |
| delta = srtp_rdbx_estimate_index(&stream->rtp_rdbx, &est, ntohs(hdr->seq)); |
| |
| /* check replay database */ |
| status = srtp_rdbx_check(&stream->rtp_rdbx, delta); |
| if (status) |
| return status; |
| } |
| |
| #ifdef NO_64BIT_MATH |
| debug_print2(mod_srtp, "estimated u_packet index: %08x%08x", high32(est),low32(est)); |
| #else |
| debug_print(mod_srtp, "estimated u_packet index: %016llx", est); |
| #endif |
| |
| /* |
| * Check if this is an AEAD stream (GCM mode). If so, then dispatch |
| * the request to our AEAD handler. |
| */ |
| if (stream->rtp_cipher->algorithm == SRTP_AES_128_GCM || |
| stream->rtp_cipher->algorithm == SRTP_AES_256_GCM) { |
| return srtp_unprotect_aead(ctx, stream, delta, est, srtp_hdr, (unsigned int*)pkt_octet_len); |
| } |
| |
| /* get tag length from stream */ |
| tag_len = srtp_auth_get_tag_length(stream->rtp_auth); |
| |
| /* |
| * set the cipher's IV properly, depending on whatever cipher we |
| * happen to be using |
| */ |
| if (stream->rtp_cipher->type->id == SRTP_AES_ICM || |
| stream->rtp_cipher->type->id == SRTP_AES_256_ICM) { |
| |
| /* aes counter mode */ |
| iv.v32[0] = 0; |
| iv.v32[1] = hdr->ssrc; /* still in network order */ |
| #ifdef NO_64BIT_MATH |
| iv.v64[1] = be64_to_cpu(make64((high32(est) << 16) | (low32(est) >> 16), |
| low32(est) << 16)); |
| #else |
| iv.v64[1] = be64_to_cpu(est << 16); |
| #endif |
| status = cipher_set_iv(stream->rtp_cipher, &iv, direction_decrypt); |
| } else { |
| |
| /* no particular format - set the iv to the pakcet index */ |
| #ifdef NO_64BIT_MATH |
| iv.v32[0] = 0; |
| iv.v32[1] = 0; |
| #else |
| iv.v64[0] = 0; |
| #endif |
| iv.v64[1] = be64_to_cpu(est); |
| status = cipher_set_iv(stream->rtp_cipher, &iv, direction_decrypt); |
| } |
| if (status) |
| return srtp_err_status_cipher_fail; |
| |
| /* shift est, put into network byte order */ |
| #ifdef NO_64BIT_MATH |
| est = be64_to_cpu(make64((high32(est) << 16) | |
| (low32(est) >> 16), |
| low32(est) << 16)); |
| #else |
| est = be64_to_cpu(est << 16); |
| #endif |
| |
| /* |
| * find starting point for decryption and length of data to be |
| * decrypted - the encrypted portion starts after the rtp header |
| * extension, if present; otherwise, it starts after the last csrc, |
| * if any are present |
| * |
| * if we're not providing confidentiality, set enc_start to NULL |
| */ |
| if (stream->rtp_services & sec_serv_conf) { |
| enc_start = (uint32_t *)hdr + uint32s_in_rtp_header + hdr->cc; |
| if (hdr->x == 1) { |
| srtp_hdr_xtnd_t *xtn_hdr = (srtp_hdr_xtnd_t *)enc_start; |
| enc_start += (ntohs(xtn_hdr->length) + 1); |
| } |
| if (!((uint8_t*)enc_start < (uint8_t*)hdr + *pkt_octet_len)) |
| return srtp_err_status_parse_err; |
| enc_octet_len = (uint32_t)(*pkt_octet_len - tag_len - |
| ((uint8_t*)enc_start - (uint8_t*)hdr)); |
| } else { |
| enc_start = NULL; |
| } |
| |
| /* |
| * if we're providing authentication, set the auth_start and auth_tag |
| * pointers to the proper locations; otherwise, set auth_start to NULL |
| * to indicate that no authentication is needed |
| */ |
| if (stream->rtp_services & sec_serv_auth) { |
| auth_start = (uint32_t *)hdr; |
| auth_tag = (uint8_t *)hdr + *pkt_octet_len - tag_len; |
| } else { |
| auth_start = NULL; |
| auth_tag = NULL; |
| } |
| |
| /* |
| * if we expect message authentication, run the authentication |
| * function and compare the result with the value of the auth_tag |
| */ |
| if (auth_start) { |
| |
| /* |
| * if we're using a universal hash, then we need to compute the |
| * keystream prefix for encrypting the universal hash output |
| * |
| * if the keystream prefix length is zero, then we know that |
| * the authenticator isn't using a universal hash function |
| */ |
| if (stream->rtp_auth->prefix_len != 0) { |
| |
| prefix_len = srtp_auth_get_prefix_length(stream->rtp_auth); |
| status = srtp_cipher_output(stream->rtp_cipher, tmp_tag, &prefix_len); |
| debug_print(mod_srtp, "keystream prefix: %s", |
| srtp_octet_string_hex_string(tmp_tag, prefix_len)); |
| if (status) |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* initialize auth func context */ |
| status = auth_start(stream->rtp_auth); |
| if (status) return status; |
| |
| /* now compute auth function over packet */ |
| status = auth_update(stream->rtp_auth, (uint8_t *)auth_start, |
| *pkt_octet_len - tag_len); |
| |
| /* run auth func over ROC, then write tmp tag */ |
| status = auth_compute(stream->rtp_auth, (uint8_t *)&est, 4, tmp_tag); |
| |
| debug_print(mod_srtp, "computed auth tag: %s", |
| srtp_octet_string_hex_string(tmp_tag, tag_len)); |
| debug_print(mod_srtp, "packet auth tag: %s", |
| srtp_octet_string_hex_string(auth_tag, tag_len)); |
| if (status) |
| return srtp_err_status_auth_fail; |
| |
| if (octet_string_is_eq(tmp_tag, auth_tag, tag_len)) |
| return srtp_err_status_auth_fail; |
| } |
| |
| /* |
| * update the key usage limit, and check it to make sure that we |
| * didn't just hit either the soft limit or the hard limit, and call |
| * the event handler if we hit either. |
| */ |
| switch(srtp_key_limit_update(stream->limit)) { |
| case srtp_key_event_normal: |
| break; |
| case srtp_key_event_soft_limit: |
| srtp_handle_event(ctx, stream, event_key_soft_limit); |
| break; |
| case srtp_key_event_hard_limit: |
| srtp_handle_event(ctx, stream, event_key_hard_limit); |
| return srtp_err_status_key_expired; |
| default: |
| break; |
| } |
| |
| /* if we're decrypting, add keystream into ciphertext */ |
| if (enc_start) { |
| status = srtp_cipher_decrypt(stream->rtp_cipher, (uint8_t *)enc_start, &enc_octet_len); |
| if (status) |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* |
| * verify that stream is for received traffic - this check will |
| * detect SSRC collisions, since a stream that appears in both |
| * srtp_protect() and srtp_unprotect() will fail this test in one of |
| * those functions. |
| * |
| * we do this check *after* the authentication check, so that the |
| * latter check will catch any attempts to fool us into thinking |
| * that we've got a collision |
| */ |
| if (stream->direction != dir_srtp_receiver) { |
| if (stream->direction == dir_unknown) { |
| stream->direction = dir_srtp_receiver; |
| } else { |
| srtp_handle_event(ctx, stream, event_ssrc_collision); |
| } |
| } |
| |
| /* |
| * if the stream is a 'provisional' one, in which the template context |
| * is used, then we need to allocate a new stream at this point, since |
| * the authentication passed |
| */ |
| if (stream == ctx->stream_template) { |
| srtp_stream_ctx_t *new_stream; |
| |
| /* |
| * allocate and initialize a new stream |
| * |
| * note that we indicate failure if we can't allocate the new |
| * stream, and some implementations will want to not return |
| * failure here |
| */ |
| status = srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream); |
| if (status) |
| return status; |
| |
| /* add new stream to the head of the stream_list */ |
| new_stream->next = ctx->stream_list; |
| ctx->stream_list = new_stream; |
| |
| /* set stream (the pointer used in this function) */ |
| stream = new_stream; |
| } |
| |
| /* |
| * the message authentication function passed, so add the packet |
| * index into the replay database |
| */ |
| srtp_rdbx_add_index(&stream->rtp_rdbx, delta); |
| |
| /* decrease the packet length by the length of the auth tag */ |
| *pkt_octet_len -= tag_len; |
| |
| return srtp_err_status_ok; |
| } |
| |
| srtp_err_status_t |
| srtp_init() { |
| srtp_err_status_t status; |
| |
| /* initialize crypto kernel */ |
| status = srtp_crypto_kernel_init(); |
| if (status) |
| return status; |
| |
| /* load srtp debug module into the kernel */ |
| status = srtp_crypto_kernel_load_debug_module(&mod_srtp); |
| if (status) |
| return status; |
| |
| return srtp_err_status_ok; |
| } |
| |
| srtp_err_status_t |
| srtp_shutdown() { |
| srtp_err_status_t status; |
| |
| /* shut down crypto kernel */ |
| status = srtp_crypto_kernel_shutdown(); |
| if (status) |
| return status; |
| |
| /* shutting down crypto kernel frees the srtp debug module as well */ |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| /* |
| * The following code is under consideration for removal. See |
| * SRTP_MAX_TRAILER_LEN |
| */ |
| #if 0 |
| |
| /* |
| * srtp_get_trailer_length(&a) returns the number of octets that will |
| * be added to an RTP packet by the SRTP processing. This value |
| * is constant for a given srtp_stream_t (i.e. between initializations). |
| */ |
| |
| int |
| srtp_get_trailer_length(const srtp_stream_t s) { |
| return srtp_auth_get_tag_length(s->rtp_auth); |
| } |
| |
| #endif |
| |
| /* |
| * srtp_get_stream(ssrc) returns a pointer to the stream corresponding |
| * to ssrc, or NULL if no stream exists for that ssrc |
| * |
| * this is an internal function |
| */ |
| |
| srtp_stream_ctx_t * |
| srtp_get_stream(srtp_t srtp, uint32_t ssrc) { |
| srtp_stream_ctx_t *stream; |
| |
| /* walk down list until ssrc is found */ |
| stream = srtp->stream_list; |
| while (stream != NULL) { |
| if (stream->ssrc == ssrc) |
| return stream; |
| stream = stream->next; |
| } |
| |
| /* we haven't found our ssrc, so return a null */ |
| return NULL; |
| } |
| |
| srtp_err_status_t |
| srtp_dealloc(srtp_t session) { |
| srtp_stream_ctx_t *stream; |
| srtp_err_status_t status; |
| |
| /* |
| * we take a conservative deallocation strategy - if we encounter an |
| * error deallocating a stream, then we stop trying to deallocate |
| * memory and just return an error |
| */ |
| |
| /* walk list of streams, deallocating as we go */ |
| stream = session->stream_list; |
| while (stream != NULL) { |
| srtp_stream_t next = stream->next; |
| status = srtp_stream_dealloc(session, stream); |
| if (status) |
| return status; |
| stream = next; |
| } |
| |
| /* deallocate stream template, if there is one */ |
| if (session->stream_template != NULL) { |
| status = auth_dealloc(session->stream_template->rtcp_auth); |
| if (status) |
| return status; |
| status = cipher_dealloc(session->stream_template->rtcp_cipher); |
| if (status) |
| return status; |
| srtp_crypto_free(session->stream_template->limit); |
| status = cipher_dealloc(session->stream_template->rtp_cipher); |
| if (status) |
| return status; |
| status = auth_dealloc(session->stream_template->rtp_auth); |
| if (status) |
| return status; |
| status = srtp_rdbx_dealloc(&session->stream_template->rtp_rdbx); |
| if (status) |
| return status; |
| srtp_crypto_free(session->stream_template); |
| } |
| |
| /* deallocate session context */ |
| srtp_crypto_free(session); |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| srtp_err_status_t |
| srtp_add_stream(srtp_t session, |
| const srtp_policy_t *policy) { |
| srtp_err_status_t status; |
| srtp_stream_t tmp; |
| |
| /* sanity check arguments */ |
| if ((session == NULL) || (policy == NULL) || (policy->key == NULL)) |
| return srtp_err_status_bad_param; |
| |
| /* allocate stream */ |
| status = srtp_stream_alloc(&tmp, policy); |
| if (status) { |
| return status; |
| } |
| |
| /* initialize stream */ |
| status = srtp_stream_init(tmp, policy); |
| if (status) { |
| srtp_crypto_free(tmp); |
| return status; |
| } |
| |
| /* |
| * set the head of the stream list or the template to point to the |
| * stream that we've just alloced and init'ed, depending on whether |
| * or not it has a wildcard SSRC value or not |
| * |
| * if the template stream has already been set, then the policy is |
| * inconsistent, so we return a bad_param error code |
| */ |
| switch (policy->ssrc.type) { |
| case (ssrc_any_outbound): |
| if (session->stream_template) { |
| return srtp_err_status_bad_param; |
| } |
| session->stream_template = tmp; |
| session->stream_template->direction = dir_srtp_sender; |
| break; |
| case (ssrc_any_inbound): |
| if (session->stream_template) { |
| return srtp_err_status_bad_param; |
| } |
| session->stream_template = tmp; |
| session->stream_template->direction = dir_srtp_receiver; |
| break; |
| case (ssrc_specific): |
| tmp->next = session->stream_list; |
| session->stream_list = tmp; |
| break; |
| case (ssrc_undefined): |
| default: |
| srtp_crypto_free(tmp); |
| return srtp_err_status_bad_param; |
| } |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| srtp_err_status_t |
| srtp_create(srtp_t *session, /* handle for session */ |
| const srtp_policy_t *policy) { /* SRTP policy (list) */ |
| srtp_err_status_t stat; |
| srtp_ctx_t *ctx; |
| |
| /* sanity check arguments */ |
| if (session == NULL) |
| return srtp_err_status_bad_param; |
| |
| /* allocate srtp context and set ctx_ptr */ |
| ctx = (srtp_ctx_t *) srtp_crypto_alloc(sizeof(srtp_ctx_t)); |
| if (ctx == NULL) |
| return srtp_err_status_alloc_fail; |
| *session = ctx; |
| |
| /* |
| * loop over elements in the policy list, allocating and |
| * initializing a stream for each element |
| */ |
| ctx->stream_template = NULL; |
| ctx->stream_list = NULL; |
| ctx->user_data = NULL; |
| while (policy != NULL) { |
| |
| stat = srtp_add_stream(ctx, policy); |
| if (stat) { |
| /* clean up everything */ |
| srtp_dealloc(*session); |
| return stat; |
| } |
| |
| /* set policy to next item in list */ |
| policy = policy->next; |
| } |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| srtp_err_status_t |
| srtp_remove_stream(srtp_t session, uint32_t ssrc) { |
| srtp_stream_ctx_t *stream, *last_stream; |
| srtp_err_status_t status; |
| |
| /* sanity check arguments */ |
| if (session == NULL) |
| return srtp_err_status_bad_param; |
| |
| /* find stream in list; complain if not found */ |
| last_stream = stream = session->stream_list; |
| while ((stream != NULL) && (ssrc != stream->ssrc)) { |
| last_stream = stream; |
| stream = stream->next; |
| } |
| if (stream == NULL) |
| return srtp_err_status_no_ctx; |
| |
| /* remove stream from the list */ |
| if (last_stream == stream) |
| /* stream was first in list */ |
| session->stream_list = stream->next; |
| else |
| last_stream->next = stream->next; |
| |
| /* deallocate the stream */ |
| status = srtp_stream_dealloc(session, stream); |
| if (status) |
| return status; |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| /* |
| * the default policy - provides a convenient way for callers to use |
| * the default security policy |
| * |
| * this policy is that defined in the current SRTP internet draft. |
| * |
| */ |
| |
| /* |
| * NOTE: cipher_key_len is really key len (128 bits) plus salt len |
| * (112 bits) |
| */ |
| /* There are hard-coded 16's for base_key_len in the key generation code */ |
| |
| void |
| srtp_crypto_policy_set_rtp_default(srtp_crypto_policy_t *p) { |
| |
| p->cipher_type = SRTP_AES_ICM; |
| p->cipher_key_len = 30; /* default 128 bits per RFC 3711 */ |
| p->auth_type = SRTP_HMAC_SHA1; |
| p->auth_key_len = 20; /* default 160 bits per RFC 3711 */ |
| p->auth_tag_len = 10; /* default 80 bits per RFC 3711 */ |
| p->sec_serv = sec_serv_conf_and_auth; |
| |
| } |
| |
| void |
| srtp_crypto_policy_set_rtcp_default(srtp_crypto_policy_t *p) { |
| |
| p->cipher_type = SRTP_AES_ICM; |
| p->cipher_key_len = 30; /* default 128 bits per RFC 3711 */ |
| p->auth_type = SRTP_HMAC_SHA1; |
| p->auth_key_len = 20; /* default 160 bits per RFC 3711 */ |
| p->auth_tag_len = 10; /* default 80 bits per RFC 3711 */ |
| p->sec_serv = sec_serv_conf_and_auth; |
| |
| } |
| |
| void |
| srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(srtp_crypto_policy_t *p) { |
| |
| /* |
| * corresponds to RFC 4568 |
| * |
| * note that this crypto policy is intended for SRTP, but not SRTCP |
| */ |
| |
| p->cipher_type = SRTP_AES_ICM; |
| p->cipher_key_len = 30; /* 128 bit key, 112 bit salt */ |
| p->auth_type = SRTP_HMAC_SHA1; |
| p->auth_key_len = 20; /* 160 bit key */ |
| p->auth_tag_len = 4; /* 32 bit tag */ |
| p->sec_serv = sec_serv_conf_and_auth; |
| |
| } |
| |
| |
| void |
| srtp_crypto_policy_set_aes_cm_128_null_auth(srtp_crypto_policy_t *p) { |
| |
| /* |
| * corresponds to RFC 4568 |
| * |
| * note that this crypto policy is intended for SRTP, but not SRTCP |
| */ |
| |
| p->cipher_type = SRTP_AES_ICM; |
| p->cipher_key_len = 30; /* 128 bit key, 112 bit salt */ |
| p->auth_type = SRTP_NULL_AUTH; |
| p->auth_key_len = 0; |
| p->auth_tag_len = 0; |
| p->sec_serv = sec_serv_conf; |
| |
| } |
| |
| |
| void |
| srtp_crypto_policy_set_null_cipher_hmac_sha1_80(srtp_crypto_policy_t *p) { |
| |
| /* |
| * corresponds to RFC 4568 |
| */ |
| |
| p->cipher_type = SRTP_NULL_CIPHER; |
| p->cipher_key_len = 0; |
| p->auth_type = SRTP_HMAC_SHA1; |
| p->auth_key_len = 20; |
| p->auth_tag_len = 10; |
| p->sec_serv = sec_serv_auth; |
| |
| } |
| |
| void |
| srtp_crypto_policy_set_null_cipher_hmac_null(srtp_crypto_policy_t *p) { |
| |
| /* |
| * Should only be used for testing |
| */ |
| |
| p->cipher_type = SRTP_NULL_CIPHER; |
| p->cipher_key_len = 0; |
| p->auth_type = SRTP_NULL_AUTH; |
| p->auth_key_len = 0; |
| p->auth_tag_len = 0; |
| p->sec_serv = sec_serv_none; |
| |
| } |
| |
| |
| void |
| srtp_crypto_policy_set_aes_cm_256_hmac_sha1_80(srtp_crypto_policy_t *p) { |
| |
| /* |
| * corresponds to draft-ietf-avt-big-aes-03.txt |
| */ |
| |
| p->cipher_type = SRTP_AES_ICM; |
| p->cipher_key_len = 46; |
| p->auth_type = SRTP_HMAC_SHA1; |
| p->auth_key_len = 20; /* default 160 bits per RFC 3711 */ |
| p->auth_tag_len = 10; /* default 80 bits per RFC 3711 */ |
| p->sec_serv = sec_serv_conf_and_auth; |
| } |
| |
| |
| void |
| srtp_crypto_policy_set_aes_cm_256_hmac_sha1_32(srtp_crypto_policy_t *p) { |
| |
| /* |
| * corresponds to draft-ietf-avt-big-aes-03.txt |
| * |
| * note that this crypto policy is intended for SRTP, but not SRTCP |
| */ |
| |
| p->cipher_type = SRTP_AES_ICM; |
| p->cipher_key_len = 46; |
| p->auth_type = SRTP_HMAC_SHA1; |
| p->auth_key_len = 20; /* default 160 bits per RFC 3711 */ |
| p->auth_tag_len = 4; /* default 80 bits per RFC 3711 */ |
| p->sec_serv = sec_serv_conf_and_auth; |
| } |
| |
| /* |
| * AES-256 with no authentication. |
| */ |
| void |
| srtp_crypto_policy_set_aes_cm_256_null_auth (srtp_crypto_policy_t *p) |
| { |
| p->cipher_type = SRTP_AES_ICM; |
| p->cipher_key_len = 46; |
| p->auth_type = SRTP_NULL_AUTH; |
| p->auth_key_len = 0; |
| p->auth_tag_len = 0; |
| p->sec_serv = sec_serv_conf; |
| } |
| |
| #ifdef OPENSSL |
| /* |
| * AES-128 GCM mode with 8 octet auth tag. |
| */ |
| void |
| srtp_crypto_policy_set_aes_gcm_128_8_auth(srtp_crypto_policy_t *p) { |
| p->cipher_type = SRTP_AES_128_GCM; |
| p->cipher_key_len = SRTP_AES_128_GCM_KEYSIZE_WSALT; |
| p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */ |
| p->auth_key_len = 0; |
| p->auth_tag_len = 8; /* 8 octet tag length */ |
| p->sec_serv = sec_serv_conf_and_auth; |
| } |
| |
| /* |
| * AES-256 GCM mode with 8 octet auth tag. |
| */ |
| void |
| srtp_crypto_policy_set_aes_gcm_256_8_auth(srtp_crypto_policy_t *p) { |
| p->cipher_type = SRTP_AES_256_GCM; |
| p->cipher_key_len = SRTP_AES_256_GCM_KEYSIZE_WSALT; |
| p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */ |
| p->auth_key_len = 0; |
| p->auth_tag_len = 8; /* 8 octet tag length */ |
| p->sec_serv = sec_serv_conf_and_auth; |
| } |
| |
| /* |
| * AES-128 GCM mode with 8 octet auth tag, no RTCP encryption. |
| */ |
| void |
| srtp_crypto_policy_set_aes_gcm_128_8_only_auth(srtp_crypto_policy_t *p) { |
| p->cipher_type = SRTP_AES_128_GCM; |
| p->cipher_key_len = SRTP_AES_128_GCM_KEYSIZE_WSALT; |
| p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */ |
| p->auth_key_len = 0; |
| p->auth_tag_len = 8; /* 8 octet tag length */ |
| p->sec_serv = sec_serv_auth; /* This only applies to RTCP */ |
| } |
| |
| /* |
| * AES-256 GCM mode with 8 octet auth tag, no RTCP encryption. |
| */ |
| void |
| srtp_crypto_policy_set_aes_gcm_256_8_only_auth(srtp_crypto_policy_t *p) { |
| p->cipher_type = SRTP_AES_256_GCM; |
| p->cipher_key_len = SRTP_AES_256_GCM_KEYSIZE_WSALT; |
| p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */ |
| p->auth_key_len = 0; |
| p->auth_tag_len = 8; /* 8 octet tag length */ |
| p->sec_serv = sec_serv_auth; /* This only applies to RTCP */ |
| } |
| |
| /* |
| * AES-128 GCM mode with 16 octet auth tag. |
| */ |
| void |
| srtp_crypto_policy_set_aes_gcm_128_16_auth(srtp_crypto_policy_t *p) { |
| p->cipher_type = SRTP_AES_128_GCM; |
| p->cipher_key_len = SRTP_AES_128_GCM_KEYSIZE_WSALT; |
| p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */ |
| p->auth_key_len = 0; |
| p->auth_tag_len = 16; /* 16 octet tag length */ |
| p->sec_serv = sec_serv_conf_and_auth; |
| } |
| |
| /* |
| * AES-256 GCM mode with 16 octet auth tag. |
| */ |
| void |
| srtp_crypto_policy_set_aes_gcm_256_16_auth(srtp_crypto_policy_t *p) { |
| p->cipher_type = SRTP_AES_256_GCM; |
| p->cipher_key_len = SRTP_AES_256_GCM_KEYSIZE_WSALT; |
| p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */ |
| p->auth_key_len = 0; |
| p->auth_tag_len = 16; /* 16 octet tag length */ |
| p->sec_serv = sec_serv_conf_and_auth; |
| } |
| |
| #endif |
| |
| /* |
| * secure rtcp functions |
| */ |
| |
| /* |
| * AEAD uses a new IV formation method. This function implements |
| * section 10.1 from draft-ietf-avtcore-srtp-aes-gcm-07.txt. The |
| * calculation is defined as, where (+) is the xor operation: |
| * |
| * 0 1 2 3 4 5 6 7 8 9 10 11 |
| * +--+--+--+--+--+--+--+--+--+--+--+--+ |
| * |00|00| SSRC |00|00|0+SRTCP Idx|---+ |
| * +--+--+--+--+--+--+--+--+--+--+--+--+ | |
| * | |
| * +--+--+--+--+--+--+--+--+--+--+--+--+ | |
| * | Encryption Salt |->(+) |
| * +--+--+--+--+--+--+--+--+--+--+--+--+ | |
| * | |
| * +--+--+--+--+--+--+--+--+--+--+--+--+ | |
| * | Initialization Vector |<--+ |
| * +--+--+--+--+--+--+--+--+--+--+--+--+* |
| * |
| * Input: *stream - pointer to SRTP stream context, used to retrieve |
| * the SALT |
| * *iv - Pointer to recieve the calculated IV |
| * seq_num - The SEQ value to use for the IV calculation. |
| * *hdr - The RTP header, used to get the SSRC value |
| * |
| */ |
| static void srtp_calc_aead_iv_srtcp(srtp_stream_ctx_t *stream, v128_t *iv, |
| uint32_t seq_num, srtcp_hdr_t *hdr) |
| { |
| v128_t in; |
| v128_t salt; |
| |
| memset(&in, 0, sizeof(v128_t)); |
| memset(&salt, 0, sizeof(v128_t)); |
| |
| in.v16[0] = 0; |
| memcpy(&in.v16[1], &hdr->ssrc, 4); /* still in network order! */ |
| in.v16[3] = 0; |
| in.v32[2] = 0x7FFFFFFF & htonl(seq_num); /* bit 32 is suppose to be zero */ |
| |
| debug_print(mod_srtp, "Pre-salted RTCP IV = %s\n", v128_hex_string(&in)); |
| |
| /* |
| * Get the SALT value from the context |
| */ |
| memcpy(salt.v8, stream->c_salt, 12); |
| debug_print(mod_srtp, "RTCP SALT = %s\n", v128_hex_string(&salt)); |
| |
| /* |
| * Finally, apply the SALT to the input |
| */ |
| v128_xor(iv, &in, &salt); |
| } |
| |
| /* |
| * This code handles AEAD ciphers for outgoing RTCP. We currently support |
| * AES-GCM mode with 128 or 256 bit keys. |
| */ |
| static srtp_err_status_t |
| srtp_protect_rtcp_aead (srtp_t ctx, srtp_stream_ctx_t *stream, |
| void *rtcp_hdr, unsigned int *pkt_octet_len) |
| { |
| srtcp_hdr_t *hdr = (srtcp_hdr_t*)rtcp_hdr; |
| uint32_t *enc_start; /* pointer to start of encrypted portion */ |
| uint32_t *trailer; /* pointer to start of trailer */ |
| unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */ |
| uint8_t *auth_tag = NULL; /* location of auth_tag within packet */ |
| srtp_err_status_t status; |
| int tag_len; |
| uint32_t seq_num; |
| v128_t iv; |
| uint32_t tseq; |
| |
| /* get tag length from stream context */ |
| tag_len = srtp_auth_get_tag_length(stream->rtcp_auth); |
| |
| /* |
| * set encryption start and encryption length - if we're not |
| * providing confidentiality, set enc_start to NULL |
| */ |
| enc_start = (uint32_t*)hdr + uint32s_in_rtcp_header; |
| enc_octet_len = *pkt_octet_len - octets_in_rtcp_header; |
| |
| /* NOTE: hdr->length is not usable - it refers to only the first |
| RTCP report in the compound packet! */ |
| /* NOTE: trailer is 32-bit aligned because RTCP 'packets' are always |
| multiples of 32-bits (RFC 3550 6.1) */ |
| trailer = (uint32_t*)((char*)enc_start + enc_octet_len + tag_len); |
| |
| if (stream->rtcp_services & sec_serv_conf) { |
| *trailer = htonl(SRTCP_E_BIT); /* set encrypt bit */ |
| } else { |
| enc_start = NULL; |
| enc_octet_len = 0; |
| /* 0 is network-order independant */ |
| *trailer = 0x00000000; /* set encrypt bit */ |
| } |
| |
| /* |
| * set the auth_tag pointer to the proper location, which is after |
| * the payload, but before the trailer |
| * (note that srtpc *always* provides authentication, unlike srtp) |
| */ |
| /* Note: This would need to change for optional mikey data */ |
| auth_tag = (uint8_t*)hdr + *pkt_octet_len; |
| |
| /* |
| * check sequence number for overruns, and copy it into the packet |
| * if its value isn't too big |
| */ |
| status = srtp_rdb_increment(&stream->rtcp_rdb); |
| if (status) { |
| return status; |
| } |
| seq_num = srtp_rdb_get_value(&stream->rtcp_rdb); |
| *trailer |= htonl(seq_num); |
| debug_print(mod_srtp, "srtcp index: %x", seq_num); |
| |
| /* |
| * Calculating the IV and pass it down to the cipher |
| */ |
| srtp_calc_aead_iv_srtcp(stream, &iv, seq_num, hdr); |
| status = cipher_set_iv(stream->rtcp_cipher, &iv, direction_encrypt); |
| if (status) { |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* |
| * Set the AAD for GCM mode |
| */ |
| if (enc_start) { |
| /* |
| * If payload encryption is enabled, then the AAD consist of |
| * the RTCP header and the seq# at the end of the packet |
| */ |
| status = cipher_set_aad(stream->rtcp_cipher, (uint8_t*)hdr, |
| octets_in_rtcp_header); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| } else { |
| /* |
| * Since payload encryption is not enabled, we must authenticate |
| * the entire packet as described in section 10.3 in revision 07 |
| * of the draft. |
| */ |
| status = cipher_set_aad(stream->rtcp_cipher, (uint8_t*)hdr, |
| *pkt_octet_len); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| } |
| /* |
| * put the idx# into network byte order and process it as AAD |
| */ |
| tseq = htonl(*trailer); |
| status = cipher_set_aad(stream->rtcp_cipher, (uint8_t*)&tseq, |
| sizeof(srtcp_trailer_t)); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| |
| /* if we're encrypting, exor keystream into the message */ |
| if (enc_start) { |
| status = srtp_cipher_encrypt(stream->rtcp_cipher, |
| (uint8_t*)enc_start, &enc_octet_len); |
| if (status) { |
| return srtp_err_status_cipher_fail; |
| } |
| /* |
| * Get the tag and append that to the output |
| */ |
| status = cipher_get_tag(stream->rtcp_cipher, (uint8_t*)auth_tag, |
| &tag_len); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| enc_octet_len += tag_len; |
| } else { |
| /* |
| * Even though we're not encrypting the payload, we need |
| * to run the cipher to get the auth tag. |
| */ |
| unsigned int nolen = 0; |
| status = srtp_cipher_encrypt(stream->rtcp_cipher, NULL, &nolen); |
| if (status) { |
| return srtp_err_status_cipher_fail; |
| } |
| /* |
| * Get the tag and append that to the output |
| */ |
| status = cipher_get_tag(stream->rtcp_cipher, (uint8_t*)auth_tag, |
| &tag_len); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| enc_octet_len += tag_len; |
| } |
| |
| /* increase the packet length by the length of the auth tag and seq_num*/ |
| *pkt_octet_len += (tag_len + sizeof(srtcp_trailer_t)); |
| |
| return srtp_err_status_ok; |
| } |
| |
| /* |
| * This function handles incoming SRTCP packets while in AEAD mode, |
| * which currently supports AES-GCM encryption. Note, the auth tag is |
| * at the end of the packet stream and is automatically checked by GCM |
| * when decrypting the payload. |
| */ |
| static srtp_err_status_t |
| srtp_unprotect_rtcp_aead (srtp_t ctx, srtp_stream_ctx_t *stream, |
| void *srtcp_hdr, unsigned int *pkt_octet_len) |
| { |
| srtcp_hdr_t *hdr = (srtcp_hdr_t*)srtcp_hdr; |
| uint32_t *enc_start; /* pointer to start of encrypted portion */ |
| uint32_t *trailer; /* pointer to start of trailer */ |
| unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */ |
| uint8_t *auth_tag = NULL; /* location of auth_tag within packet */ |
| srtp_err_status_t status; |
| int tag_len; |
| unsigned int tmp_len; |
| uint32_t seq_num; |
| v128_t iv; |
| uint32_t tseq; |
| |
| /* get tag length from stream context */ |
| tag_len = srtp_auth_get_tag_length(stream->rtcp_auth); |
| |
| /* |
| * set encryption start, encryption length, and trailer |
| */ |
| /* index & E (encryption) bit follow normal data. hdr->len |
| is the number of words (32-bit) in the normal packet minus 1 */ |
| /* This should point trailer to the word past the end of the |
| normal data. */ |
| /* This would need to be modified for optional mikey data */ |
| /* |
| * NOTE: trailer is 32-bit aligned because RTCP 'packets' are always |
| * multiples of 32-bits (RFC 3550 6.1) |
| */ |
| trailer = (uint32_t*)((char*)hdr + *pkt_octet_len - sizeof(srtcp_trailer_t)); |
| /* |
| * We pass the tag down to the cipher when doing GCM mode |
| */ |
| enc_octet_len = *pkt_octet_len - (octets_in_rtcp_header + |
| sizeof(srtcp_trailer_t)); |
| auth_tag = (uint8_t*)hdr + *pkt_octet_len - tag_len - sizeof(srtcp_trailer_t); |
| |
| if (*((unsigned char*)trailer) & SRTCP_E_BYTE_BIT) { |
| enc_start = (uint32_t*)hdr + uint32s_in_rtcp_header; |
| } else { |
| enc_octet_len = 0; |
| enc_start = NULL; /* this indicates that there's no encryption */ |
| } |
| |
| /* |
| * check the sequence number for replays |
| */ |
| /* this is easier than dealing with bitfield access */ |
| seq_num = ntohl(*trailer) & SRTCP_INDEX_MASK; |
| debug_print(mod_srtp, "srtcp index: %x", seq_num); |
| status = srtp_rdb_check(&stream->rtcp_rdb, seq_num); |
| if (status) { |
| return status; |
| } |
| |
| /* |
| * Calculate and set the IV |
| */ |
| srtp_calc_aead_iv_srtcp(stream, &iv, seq_num, hdr); |
| status = cipher_set_iv(stream->rtcp_cipher, &iv, direction_decrypt); |
| if (status) { |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* |
| * Set the AAD for GCM mode |
| */ |
| if (enc_start) { |
| /* |
| * If payload encryption is enabled, then the AAD consist of |
| * the RTCP header and the seq# at the end of the packet |
| */ |
| status = cipher_set_aad(stream->rtcp_cipher, (uint8_t*)hdr, |
| octets_in_rtcp_header); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| } else { |
| /* |
| * Since payload encryption is not enabled, we must authenticate |
| * the entire packet as described in section 10.3 in revision 07 |
| * of the draft. |
| */ |
| status = cipher_set_aad(stream->rtcp_cipher, (uint8_t*)hdr, |
| (*pkt_octet_len - tag_len - sizeof(srtcp_trailer_t))); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| } |
| |
| /* |
| * put the idx# into network byte order, and process it as AAD |
| */ |
| tseq = htonl(*trailer); |
| status = cipher_set_aad(stream->rtcp_cipher, (uint8_t*)&tseq, |
| sizeof(srtcp_trailer_t)); |
| if (status) { |
| return ( srtp_err_status_cipher_fail); |
| } |
| |
| /* if we're decrypting, exor keystream into the message */ |
| if (enc_start) { |
| status = srtp_cipher_decrypt(stream->rtcp_cipher, (uint8_t*)enc_start, &enc_octet_len); |
| if (status) { |
| return status; |
| } |
| } else { |
| /* |
| * Still need to run the cipher to check the tag |
| */ |
| tmp_len = tag_len; |
| status = srtp_cipher_decrypt(stream->rtcp_cipher, (uint8_t*)auth_tag, &tmp_len); |
| if (status) { |
| return status; |
| } |
| } |
| |
| /* decrease the packet length by the length of the auth tag and seq_num*/ |
| *pkt_octet_len -= (tag_len + sizeof(srtcp_trailer_t)); |
| |
| /* |
| * verify that stream is for received traffic - this check will |
| * detect SSRC collisions, since a stream that appears in both |
| * srtp_protect() and srtp_unprotect() will fail this test in one of |
| * those functions. |
| * |
| * we do this check *after* the authentication check, so that the |
| * latter check will catch any attempts to fool us into thinking |
| * that we've got a collision |
| */ |
| if (stream->direction != dir_srtp_receiver) { |
| if (stream->direction == dir_unknown) { |
| stream->direction = dir_srtp_receiver; |
| } else { |
| srtp_handle_event(ctx, stream, event_ssrc_collision); |
| } |
| } |
| |
| /* |
| * if the stream is a 'provisional' one, in which the template context |
| * is used, then we need to allocate a new stream at this point, since |
| * the authentication passed |
| */ |
| if (stream == ctx->stream_template) { |
| srtp_stream_ctx_t *new_stream; |
| |
| /* |
| * allocate and initialize a new stream |
| * |
| * note that we indicate failure if we can't allocate the new |
| * stream, and some implementations will want to not return |
| * failure here |
| */ |
| status = srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream); |
| if (status) { |
| return status; |
| } |
| |
| /* add new stream to the head of the stream_list */ |
| new_stream->next = ctx->stream_list; |
| ctx->stream_list = new_stream; |
| |
| /* set stream (the pointer used in this function) */ |
| stream = new_stream; |
| } |
| |
| /* we've passed the authentication check, so add seq_num to the rdb */ |
| srtp_rdb_add_index(&stream->rtcp_rdb, seq_num); |
| |
| return srtp_err_status_ok; |
| } |
| |
| srtp_err_status_t |
| srtp_protect_rtcp(srtp_t ctx, void *rtcp_hdr, int *pkt_octet_len) { |
| srtcp_hdr_t *hdr = (srtcp_hdr_t *)rtcp_hdr; |
| uint32_t *enc_start; /* pointer to start of encrypted portion */ |
| uint32_t *auth_start; /* pointer to start of auth. portion */ |
| uint32_t *trailer; /* pointer to start of trailer */ |
| unsigned int enc_octet_len = 0;/* number of octets in encrypted portion */ |
| uint8_t *auth_tag = NULL; /* location of auth_tag within packet */ |
| srtp_err_status_t status; |
| int tag_len; |
| srtp_stream_ctx_t *stream; |
| uint32_t prefix_len; |
| uint32_t seq_num; |
| |
| /* we assume the hdr is 32-bit aligned to start */ |
| |
| /* check the packet length - it must at least contain a full header */ |
| if (*pkt_octet_len < octets_in_rtcp_header) |
| return srtp_err_status_bad_param; |
| |
| /* |
| * look up ssrc in srtp_stream list, and process the packet with |
| * the appropriate stream. if we haven't seen this stream before, |
| * there's only one key for this srtp_session, and the cipher |
| * supports key-sharing, then we assume that a new stream using |
| * that key has just started up |
| */ |
| stream = srtp_get_stream(ctx, hdr->ssrc); |
| if (stream == NULL) { |
| if (ctx->stream_template != NULL) { |
| srtp_stream_ctx_t *new_stream; |
| |
| /* allocate and initialize a new stream */ |
| status = srtp_stream_clone(ctx->stream_template, |
| hdr->ssrc, &new_stream); |
| if (status) |
| return status; |
| |
| /* add new stream to the head of the stream_list */ |
| new_stream->next = ctx->stream_list; |
| ctx->stream_list = new_stream; |
| |
| /* set stream (the pointer used in this function) */ |
| stream = new_stream; |
| } else { |
| /* no template stream, so we return an error */ |
| return srtp_err_status_no_ctx; |
| } |
| } |
| |
| /* |
| * verify that stream is for sending traffic - this check will |
| * detect SSRC collisions, since a stream that appears in both |
| * srtp_protect() and srtp_unprotect() will fail this test in one of |
| * those functions. |
| */ |
| if (stream->direction != dir_srtp_sender) { |
| if (stream->direction == dir_unknown) { |
| stream->direction = dir_srtp_sender; |
| } else { |
| srtp_handle_event(ctx, stream, event_ssrc_collision); |
| } |
| } |
| |
| /* |
| * Check if this is an AEAD stream (GCM mode). If so, then dispatch |
| * the request to our AEAD handler. |
| */ |
| if (stream->rtp_cipher->algorithm == SRTP_AES_128_GCM || |
| stream->rtp_cipher->algorithm == SRTP_AES_256_GCM) { |
| return srtp_protect_rtcp_aead(ctx, stream, rtcp_hdr, (unsigned int*)pkt_octet_len); |
| } |
| |
| /* get tag length from stream context */ |
| tag_len = srtp_auth_get_tag_length(stream->rtcp_auth); |
| |
| /* |
| * set encryption start and encryption length - if we're not |
| * providing confidentiality, set enc_start to NULL |
| */ |
| enc_start = (uint32_t *)hdr + uint32s_in_rtcp_header; |
| enc_octet_len = *pkt_octet_len - octets_in_rtcp_header; |
| |
| /* all of the packet, except the header, gets encrypted */ |
| /* NOTE: hdr->length is not usable - it refers to only the first |
| RTCP report in the compound packet! */ |
| /* NOTE: trailer is 32-bit aligned because RTCP 'packets' are always |
| multiples of 32-bits (RFC 3550 6.1) */ |
| trailer = (uint32_t *) ((char *)enc_start + enc_octet_len); |
| |
| if (stream->rtcp_services & sec_serv_conf) { |
| *trailer = htonl(SRTCP_E_BIT); /* set encrypt bit */ |
| } else { |
| enc_start = NULL; |
| enc_octet_len = 0; |
| /* 0 is network-order independant */ |
| *trailer = 0x00000000; /* set encrypt bit */ |
| } |
| |
| /* |
| * set the auth_start and auth_tag pointers to the proper locations |
| * (note that srtpc *always* provides authentication, unlike srtp) |
| */ |
| /* Note: This would need to change for optional mikey data */ |
| auth_start = (uint32_t *)hdr; |
| auth_tag = (uint8_t *)hdr + *pkt_octet_len + sizeof(srtcp_trailer_t); |
| |
| /* perform EKT processing if needed */ |
| srtp_ekt_write_data(stream->ekt, auth_tag, tag_len, pkt_octet_len, |
| srtp_rdbx_get_packet_index(&stream->rtp_rdbx)); |
| |
| /* |
| * check sequence number for overruns, and copy it into the packet |
| * if its value isn't too big |
| */ |
| status = srtp_rdb_increment(&stream->rtcp_rdb); |
| if (status) |
| return status; |
| seq_num = srtp_rdb_get_value(&stream->rtcp_rdb); |
| *trailer |= htonl(seq_num); |
| debug_print(mod_srtp, "srtcp index: %x", seq_num); |
| |
| /* |
| * if we're using rindael counter mode, set nonce and seq |
| */ |
| if (stream->rtcp_cipher->type->id == SRTP_AES_ICM) { |
| v128_t iv; |
| |
| iv.v32[0] = 0; |
| iv.v32[1] = hdr->ssrc; /* still in network order! */ |
| iv.v32[2] = htonl(seq_num >> 16); |
| iv.v32[3] = htonl(seq_num << 16); |
| status = cipher_set_iv(stream->rtcp_cipher, &iv, direction_encrypt); |
| |
| } else { |
| v128_t iv; |
| |
| /* otherwise, just set the index to seq_num */ |
| iv.v32[0] = 0; |
| iv.v32[1] = 0; |
| iv.v32[2] = 0; |
| iv.v32[3] = htonl(seq_num); |
| status = cipher_set_iv(stream->rtcp_cipher, &iv, direction_encrypt); |
| } |
| if (status) |
| return srtp_err_status_cipher_fail; |
| |
| /* |
| * if we're authenticating using a universal hash, put the keystream |
| * prefix into the authentication tag |
| */ |
| |
| /* if auth_start is non-null, then put keystream into tag */ |
| if (auth_start) { |
| |
| /* put keystream prefix into auth_tag */ |
| prefix_len = srtp_auth_get_prefix_length(stream->rtcp_auth); |
| status = srtp_cipher_output(stream->rtcp_cipher, auth_tag, &prefix_len); |
| |
| debug_print(mod_srtp, "keystream prefix: %s", |
| srtp_octet_string_hex_string(auth_tag, prefix_len)); |
| |
| if (status) |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* if we're encrypting, exor keystream into the message */ |
| if (enc_start) { |
| status = srtp_cipher_encrypt(stream->rtcp_cipher, |
| (uint8_t *)enc_start, &enc_octet_len); |
| if (status) |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* initialize auth func context */ |
| auth_start(stream->rtcp_auth); |
| |
| /* |
| * run auth func over packet (including trailer), and write the |
| * result at auth_tag |
| */ |
| status = auth_compute(stream->rtcp_auth, |
| (uint8_t *)auth_start, |
| (*pkt_octet_len) + sizeof(srtcp_trailer_t), |
| auth_tag); |
| debug_print(mod_srtp, "srtcp auth tag: %s", |
| srtp_octet_string_hex_string(auth_tag, tag_len)); |
| if (status) |
| return srtp_err_status_auth_fail; |
| |
| /* increase the packet length by the length of the auth tag and seq_num*/ |
| *pkt_octet_len += (tag_len + sizeof(srtcp_trailer_t)); |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| srtp_err_status_t |
| srtp_unprotect_rtcp(srtp_t ctx, void *srtcp_hdr, int *pkt_octet_len) { |
| srtcp_hdr_t *hdr = (srtcp_hdr_t *)srtcp_hdr; |
| uint32_t *enc_start; /* pointer to start of encrypted portion */ |
| uint32_t *auth_start; /* pointer to start of auth. portion */ |
| uint32_t *trailer; /* pointer to start of trailer */ |
| unsigned int enc_octet_len = 0;/* number of octets in encrypted portion */ |
| uint8_t *auth_tag = NULL; /* location of auth_tag within packet */ |
| uint8_t tmp_tag[SRTP_MAX_TAG_LEN]; |
| uint8_t tag_copy[SRTP_MAX_TAG_LEN]; |
| srtp_err_status_t status; |
| unsigned int auth_len; |
| int tag_len; |
| srtp_stream_ctx_t *stream; |
| uint32_t prefix_len; |
| uint32_t seq_num; |
| int e_bit_in_packet; /* whether the E-bit was found in the packet */ |
| int sec_serv_confidentiality; /* whether confidentiality was requested */ |
| |
| /* we assume the hdr is 32-bit aligned to start */ |
| |
| /* check that the length value is sane; we'll check again once we |
| know the tag length, but we at least want to know that it is |
| a positive value */ |
| if (*pkt_octet_len < octets_in_rtcp_header + sizeof(srtcp_trailer_t)) |
| return srtp_err_status_bad_param; |
| |
| /* |
| * look up ssrc in srtp_stream list, and process the packet with |
| * the appropriate stream. if we haven't seen this stream before, |
| * there's only one key for this srtp_session, and the cipher |
| * supports key-sharing, then we assume that a new stream using |
| * that key has just started up |
| */ |
| stream = srtp_get_stream(ctx, hdr->ssrc); |
| if (stream == NULL) { |
| if (ctx->stream_template != NULL) { |
| stream = ctx->stream_template; |
| |
| /* |
| * check to see if stream_template has an EKT data structure, in |
| * which case we initialize the template using the EKT policy |
| * referenced by that data (which consists of decrypting the |
| * master key from the EKT field) |
| * |
| * this function initializes a *provisional* stream, and this |
| * stream should not be accepted until and unless the packet |
| * passes its authentication check |
| */ |
| if (stream->ekt != NULL) { |
| status = srtp_stream_init_from_ekt(stream, srtcp_hdr, *pkt_octet_len); |
| if (status) |
| return status; |
| } |
| |
| debug_print(mod_srtp, "srtcp using provisional stream (SSRC: 0x%08x)", |
| hdr->ssrc); |
| } else { |
| /* no template stream, so we return an error */ |
| return srtp_err_status_no_ctx; |
| } |
| } |
| |
| /* get tag length from stream context */ |
| tag_len = srtp_auth_get_tag_length(stream->rtcp_auth); |
| |
| /* check the packet length - it must contain at least a full RTCP |
| header, an auth tag (if applicable), and the SRTCP encrypted flag |
| and 31-bit index value */ |
| if (*pkt_octet_len < (octets_in_rtcp_header + tag_len + sizeof(srtcp_trailer_t))) { |
| return srtp_err_status_bad_param; |
| } |
| |
| /* |
| * Check if this is an AEAD stream (GCM mode). If so, then dispatch |
| * the request to our AEAD handler. |
| */ |
| if (stream->rtp_cipher->algorithm == SRTP_AES_128_GCM || |
| stream->rtp_cipher->algorithm == SRTP_AES_256_GCM) { |
| return srtp_unprotect_rtcp_aead(ctx, stream, srtcp_hdr, (unsigned int*)pkt_octet_len); |
| } |
| |
| sec_serv_confidentiality = stream->rtcp_services == sec_serv_conf || |
| stream->rtcp_services == sec_serv_conf_and_auth; |
| |
| /* |
| * set encryption start, encryption length, and trailer |
| */ |
| enc_octet_len = *pkt_octet_len - |
| (octets_in_rtcp_header + tag_len + sizeof(srtcp_trailer_t)); |
| /* index & E (encryption) bit follow normal data. hdr->len |
| is the number of words (32-bit) in the normal packet minus 1 */ |
| /* This should point trailer to the word past the end of the |
| normal data. */ |
| /* This would need to be modified for optional mikey data */ |
| /* |
| * NOTE: trailer is 32-bit aligned because RTCP 'packets' are always |
| * multiples of 32-bits (RFC 3550 6.1) |
| */ |
| trailer = (uint32_t *) ((char *) hdr + |
| *pkt_octet_len -(tag_len + sizeof(srtcp_trailer_t))); |
| e_bit_in_packet = |
| (*((unsigned char *) trailer) & SRTCP_E_BYTE_BIT) == SRTCP_E_BYTE_BIT; |
| if (e_bit_in_packet != sec_serv_confidentiality) { |
| return srtp_err_status_cant_check; |
| } |
| if (sec_serv_confidentiality) { |
| enc_start = (uint32_t *)hdr + uint32s_in_rtcp_header; |
| } else { |
| enc_octet_len = 0; |
| enc_start = NULL; /* this indicates that there's no encryption */ |
| } |
| |
| /* |
| * set the auth_start and auth_tag pointers to the proper locations |
| * (note that srtcp *always* uses authentication, unlike srtp) |
| */ |
| auth_start = (uint32_t *)hdr; |
| auth_len = *pkt_octet_len - tag_len; |
| auth_tag = (uint8_t *)hdr + auth_len; |
| |
| /* |
| * if EKT is in use, then we make a copy of the tag from the packet, |
| * and then zeroize the location of the base tag |
| * |
| * we first re-position the auth_tag pointer so that it points to |
| * the base tag |
| */ |
| if (stream->ekt) { |
| auth_tag -= srtp_ekt_octets_after_base_tag(stream->ekt); |
| memcpy(tag_copy, auth_tag, tag_len); |
| octet_string_set_to_zero(auth_tag, tag_len); |
| auth_tag = tag_copy; |
| auth_len += tag_len; |
| } |
| |
| /* |
| * check the sequence number for replays |
| */ |
| /* this is easier than dealing with bitfield access */ |
| seq_num = ntohl(*trailer) & SRTCP_INDEX_MASK; |
| debug_print(mod_srtp, "srtcp index: %x", seq_num); |
| status = srtp_rdb_check(&stream->rtcp_rdb, seq_num); |
| if (status) |
| return status; |
| |
| /* |
| * if we're using aes counter mode, set nonce and seq |
| */ |
| if (stream->rtcp_cipher->type->id == SRTP_AES_ICM) { |
| v128_t iv; |
| |
| iv.v32[0] = 0; |
| iv.v32[1] = hdr->ssrc; /* still in network order! */ |
| iv.v32[2] = htonl(seq_num >> 16); |
| iv.v32[3] = htonl(seq_num << 16); |
| status = cipher_set_iv(stream->rtcp_cipher, &iv, direction_decrypt); |
| |
| } else { |
| v128_t iv; |
| |
| /* otherwise, just set the index to seq_num */ |
| iv.v32[0] = 0; |
| iv.v32[1] = 0; |
| iv.v32[2] = 0; |
| iv.v32[3] = htonl(seq_num); |
| status = cipher_set_iv(stream->rtcp_cipher, &iv, direction_decrypt); |
| |
| } |
| if (status) |
| return srtp_err_status_cipher_fail; |
| |
| /* initialize auth func context */ |
| auth_start(stream->rtcp_auth); |
| |
| /* run auth func over packet, put result into tmp_tag */ |
| status = auth_compute(stream->rtcp_auth, (uint8_t *)auth_start, |
| auth_len, tmp_tag); |
| debug_print(mod_srtp, "srtcp computed tag: %s", |
| srtp_octet_string_hex_string(tmp_tag, tag_len)); |
| if (status) |
| return srtp_err_status_auth_fail; |
| |
| /* compare the tag just computed with the one in the packet */ |
| debug_print(mod_srtp, "srtcp tag from packet: %s", |
| srtp_octet_string_hex_string(auth_tag, tag_len)); |
| if (octet_string_is_eq(tmp_tag, auth_tag, tag_len)) |
| return srtp_err_status_auth_fail; |
| |
| /* |
| * if we're authenticating using a universal hash, put the keystream |
| * prefix into the authentication tag |
| */ |
| prefix_len = srtp_auth_get_prefix_length(stream->rtcp_auth); |
| if (prefix_len) { |
| status = srtp_cipher_output(stream->rtcp_cipher, auth_tag, &prefix_len); |
| debug_print(mod_srtp, "keystream prefix: %s", |
| srtp_octet_string_hex_string(auth_tag, prefix_len)); |
| if (status) |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* if we're decrypting, exor keystream into the message */ |
| if (enc_start) { |
| status = srtp_cipher_decrypt(stream->rtcp_cipher, (uint8_t *)enc_start, &enc_octet_len); |
| if (status) |
| return srtp_err_status_cipher_fail; |
| } |
| |
| /* decrease the packet length by the length of the auth tag and seq_num */ |
| *pkt_octet_len -= (tag_len + sizeof(srtcp_trailer_t)); |
| |
| /* |
| * if EKT is in effect, subtract the EKT data out of the packet |
| * length |
| */ |
| *pkt_octet_len -= srtp_ekt_octets_after_base_tag(stream->ekt); |
| |
| /* |
| * verify that stream is for received traffic - this check will |
| * detect SSRC collisions, since a stream that appears in both |
| * srtp_protect() and srtp_unprotect() will fail this test in one of |
| * those functions. |
| * |
| * we do this check *after* the authentication check, so that the |
| * latter check will catch any attempts to fool us into thinking |
| * that we've got a collision |
| */ |
| if (stream->direction != dir_srtp_receiver) { |
| if (stream->direction == dir_unknown) { |
| stream->direction = dir_srtp_receiver; |
| } else { |
| srtp_handle_event(ctx, stream, event_ssrc_collision); |
| } |
| } |
| |
| /* |
| * if the stream is a 'provisional' one, in which the template context |
| * is used, then we need to allocate a new stream at this point, since |
| * the authentication passed |
| */ |
| if (stream == ctx->stream_template) { |
| srtp_stream_ctx_t *new_stream; |
| |
| /* |
| * allocate and initialize a new stream |
| * |
| * note that we indicate failure if we can't allocate the new |
| * stream, and some implementations will want to not return |
| * failure here |
| */ |
| status = srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream); |
| if (status) |
| return status; |
| |
| /* add new stream to the head of the stream_list */ |
| new_stream->next = ctx->stream_list; |
| ctx->stream_list = new_stream; |
| |
| /* set stream (the pointer used in this function) */ |
| stream = new_stream; |
| } |
| |
| /* we've passed the authentication check, so add seq_num to the rdb */ |
| srtp_rdb_add_index(&stream->rtcp_rdb, seq_num); |
| |
| |
| return srtp_err_status_ok; |
| } |
| |
| |
| /* |
| * user data within srtp_t context |
| */ |
| |
| void |
| srtp_set_user_data(srtp_t ctx, void *data) { |
| ctx->user_data = data; |
| } |
| |
| void* |
| srtp_get_user_data(srtp_t ctx) { |
| return ctx->user_data; |
| } |
| |
| |
| /* |
| * dtls keying for srtp |
| */ |
| |
| srtp_err_status_t |
| srtp_crypto_policy_set_from_profile_for_rtp(srtp_crypto_policy_t *policy, |
| srtp_profile_t profile) { |
| |
| /* set SRTP policy from the SRTP profile in the key set */ |
| switch(profile) { |
| case srtp_profile_aes128_cm_sha1_80: |
| srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(policy); |
| break; |
| case srtp_profile_aes128_cm_sha1_32: |
| srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(policy); |
| break; |
| case srtp_profile_null_sha1_80: |
| srtp_crypto_policy_set_null_cipher_hmac_sha1_80(policy); |
| break; |
| case srtp_profile_aes256_cm_sha1_80: |
| srtp_crypto_policy_set_aes_cm_256_hmac_sha1_80(policy); |
| break; |
| case srtp_profile_aes256_cm_sha1_32: |
| srtp_crypto_policy_set_aes_cm_256_hmac_sha1_32(policy); |
| break; |
| /* the following profiles are not (yet) supported */ |
| case srtp_profile_null_sha1_32: |
| default: |
| return srtp_err_status_bad_param; |
| } |
| |
| return srtp_err_status_ok; |
| } |
| |
| srtp_err_status_t |
| srtp_crypto_policy_set_from_profile_for_rtcp(srtp_crypto_policy_t *policy, |
| srtp_profile_t profile) { |
| |
| /* set SRTP policy from the SRTP profile in the key set */ |
| switch(profile) { |
| case srtp_profile_aes128_cm_sha1_80: |
| srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(policy); |
| break; |
| case srtp_profile_aes128_cm_sha1_32: |
| /* We do not honor the 32-bit auth tag request since |
| * this is not compliant with RFC 3711 */ |
| srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(policy); |
| break; |
| case srtp_profile_null_sha1_80: |
| srtp_crypto_policy_set_null_cipher_hmac_sha1_80(policy); |
| break; |
| case srtp_profile_aes256_cm_sha1_80: |
| srtp_crypto_policy_set_aes_cm_256_hmac_sha1_80(policy); |
| break; |
| case srtp_profile_aes256_cm_sha1_32: |
| /* We do not honor the 32-bit auth tag request since |
| * this is not compliant with RFC 3711 */ |
| srtp_crypto_policy_set_aes_cm_256_hmac_sha1_80(policy); |
| break; |
| /* the following profiles are not (yet) supported */ |
| case srtp_profile_null_sha1_32: |
| default: |
| return srtp_err_status_bad_param; |
| } |
| |
| return srtp_err_status_ok; |
| } |
| |
| void srtp_append_salt_to_key(uint8_t *key, unsigned int bytes_in_key, uint8_t *salt, unsigned int bytes_in_salt) { |
| memcpy(key + bytes_in_key, salt, bytes_in_salt); |
| } |
| |
| unsigned int |
| srtp_profile_get_master_key_length(srtp_profile_t profile) { |
| |
| switch(profile) { |
| case srtp_profile_aes128_cm_sha1_80: |
| return 16; |
| break; |
| case srtp_profile_aes128_cm_sha1_32: |
| return 16; |
| break; |
| case srtp_profile_null_sha1_80: |
| return 16; |
| break; |
| case srtp_profile_aes256_cm_sha1_80: |
| return 32; |
| break; |
| case srtp_profile_aes256_cm_sha1_32: |
| return 32; |
| break; |
| /* the following profiles are not (yet) supported */ |
| case srtp_profile_null_sha1_32: |
| default: |
| return 0; /* indicate error by returning a zero */ |
| } |
| } |
| |
| unsigned int |
| srtp_profile_get_master_salt_length(srtp_profile_t profile) { |
| |
| switch(profile) { |
| case srtp_profile_aes128_cm_sha1_80: |
| return 14; |
| break; |
| case srtp_profile_aes128_cm_sha1_32: |
| return 14; |
| break; |
| case srtp_profile_null_sha1_80: |
| return 14; |
| break; |
| case srtp_profile_aes256_cm_sha1_80: |
| return 14; |
| break; |
| case srtp_profile_aes256_cm_sha1_32: |
| return 14; |
| break; |
| /* the following profiles are not (yet) supported */ |
| case srtp_profile_null_sha1_32: |
| default: |
| return 0; /* indicate error by returning a zero */ |
| } |
| } |
| |
| /* |
| * SRTP debug interface |
| */ |
| srtp_err_status_t srtp_set_debug_module(char *mod_name, int v) |
| { |
| return srtp_crypto_kernel_set_debug_module(mod_name, v); |
| } |
| |
| srtp_err_status_t srtp_list_debug_modules(void) |
| { |
| return srtp_crypto_kernel_list_debug_modules(); |
| } |
| |