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-rw-r--r--third_party/libsrtp/src/srtp/srtp.c4723
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diff --git a/third_party/libsrtp/src/srtp/srtp.c b/third_party/libsrtp/src/srtp/srtp.c
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+++ b/third_party/libsrtp/src/srtp/srtp.c
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+/*
+ * srtp.c
+ *
+ * the secure real-time transport protocol
+ *
+ * David A. McGrew
+ * Cisco Systems, Inc.
+ */
+/*
+ *
+ * Copyright (c) 2001-2017, 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.
+ *
+ */
+
+// Leave this as the top level import. Ensures the existence of defines
+#include "config.h"
+
+#include "srtp_priv.h"
+#include "crypto_types.h"
+#include "err.h"
+#include "alloc.h" /* for srtp_crypto_alloc() */
+
+#ifdef GCM
+#include "aes_gcm.h" /* for AES GCM mode */
+#endif
+
+#ifdef OPENSSL_KDF
+#include <openssl/kdf.h>
+#include "aes_icm_ext.h"
+#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
+#define octets_in_rtp_extn_hdr 4
+
+static srtp_err_status_t srtp_validate_rtp_header(void *rtp_hdr,
+ int *pkt_octet_len)
+{
+ srtp_hdr_t *hdr = (srtp_hdr_t *)rtp_hdr;
+ int rtp_header_len;
+
+ if (*pkt_octet_len < octets_in_rtp_header)
+ return srtp_err_status_bad_param;
+
+ /* Check RTP header length */
+ rtp_header_len = octets_in_rtp_header + 4 * hdr->cc;
+ if (hdr->x == 1)
+ rtp_header_len += octets_in_rtp_extn_hdr;
+
+ if (*pkt_octet_len < rtp_header_len)
+ return srtp_err_status_bad_param;
+
+ /* Verifing profile length. */
+ if (hdr->x == 1) {
+ srtp_hdr_xtnd_t *xtn_hdr =
+ (srtp_hdr_xtnd_t *)((uint32_t *)hdr + uint32s_in_rtp_header +
+ hdr->cc);
+ int profile_len = ntohs(xtn_hdr->length);
+ rtp_header_len += profile_len * 4;
+ /* profile length counts the number of 32-bit words */
+ if (*pkt_octet_len < rtp_header_len)
+ return srtp_err_status_bad_param;
+ }
+ return srtp_err_status_ok;
+}
+
+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, &micro);
+ 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_dealloc(srtp_stream_ctx_t *stream,
+ const srtp_stream_ctx_t *stream_template)
+{
+ srtp_err_status_t status;
+ unsigned int i = 0;
+ srtp_session_keys_t *session_keys = NULL;
+ srtp_session_keys_t *template_session_keys = NULL;
+
+ /*
+ * we use a conservative deallocation strategy - if any deallocation
+ * fails, then we report that fact without trying to deallocate
+ * anything else
+ */
+ if (stream->session_keys) {
+ for (i = 0; i < stream->num_master_keys; i++) {
+ session_keys = &stream->session_keys[i];
+
+ if (stream_template &&
+ stream->num_master_keys == stream_template->num_master_keys) {
+ template_session_keys = &stream_template->session_keys[i];
+ } else {
+ template_session_keys = NULL;
+ }
+
+ /*
+ * deallocate cipher, if it is not the same as that in template
+ */
+ if (template_session_keys &&
+ session_keys->rtp_cipher == template_session_keys->rtp_cipher) {
+ /* do nothing */
+ } else if (session_keys->rtp_cipher) {
+ status = srtp_cipher_dealloc(session_keys->rtp_cipher);
+ if (status)
+ return status;
+ }
+
+ /*
+ * deallocate auth function, if it is not the same as that in
+ * template
+ */
+ if (template_session_keys &&
+ session_keys->rtp_auth == template_session_keys->rtp_auth) {
+ /* do nothing */
+ } else if (session_keys->rtp_auth) {
+ status = srtp_auth_dealloc(session_keys->rtp_auth);
+ if (status)
+ return status;
+ }
+
+ if (template_session_keys &&
+ session_keys->rtp_xtn_hdr_cipher ==
+ template_session_keys->rtp_xtn_hdr_cipher) {
+ /* do nothing */
+ } else if (session_keys->rtp_xtn_hdr_cipher) {
+ status = srtp_cipher_dealloc(session_keys->rtp_xtn_hdr_cipher);
+ if (status)
+ return status;
+ }
+
+ /*
+ * deallocate rtcp cipher, if it is not the same as that in
+ * template
+ */
+ if (template_session_keys &&
+ session_keys->rtcp_cipher ==
+ template_session_keys->rtcp_cipher) {
+ /* do nothing */
+ } else if (session_keys->rtcp_cipher) {
+ status = srtp_cipher_dealloc(session_keys->rtcp_cipher);
+ if (status)
+ return status;
+ }
+
+ /*
+ * deallocate rtcp auth function, if it is not the same as that in
+ * template
+ */
+ if (template_session_keys &&
+ session_keys->rtcp_auth == template_session_keys->rtcp_auth) {
+ /* do nothing */
+ } else if (session_keys->rtcp_auth) {
+ status = srtp_auth_dealloc(session_keys->rtcp_auth);
+ if (status)
+ return status;
+ }
+
+ /*
+ * zeroize the salt value
+ */
+ octet_string_set_to_zero(session_keys->salt, SRTP_AEAD_SALT_LEN);
+ octet_string_set_to_zero(session_keys->c_salt, SRTP_AEAD_SALT_LEN);
+
+ if (session_keys->mki_id) {
+ octet_string_set_to_zero(session_keys->mki_id,
+ session_keys->mki_size);
+ srtp_crypto_free(session_keys->mki_id);
+ session_keys->mki_id = NULL;
+ }
+
+ /*
+ * deallocate key usage limit, if it is not the same as that in
+ * template
+ */
+ if (template_session_keys &&
+ session_keys->limit == template_session_keys->limit) {
+ /* do nothing */
+ } else if (session_keys->limit) {
+ srtp_crypto_free(session_keys->limit);
+ }
+ }
+ srtp_crypto_free(stream->session_keys);
+ }
+
+ status = srtp_rdbx_dealloc(&stream->rtp_rdbx);
+ if (status)
+ return status;
+
+ if (stream_template &&
+ stream->enc_xtn_hdr == stream_template->enc_xtn_hdr) {
+ /* do nothing */
+ } else if (stream->enc_xtn_hdr) {
+ srtp_crypto_free(stream->enc_xtn_hdr);
+ }
+
+ /* deallocate srtp stream context */
+ srtp_crypto_free(stream);
+
+ return srtp_err_status_ok;
+}
+
+static srtp_err_status_t srtp_valid_policy(const srtp_policy_t *p)
+{
+ if (p != NULL && p->deprecated_ekt != NULL) {
+ return srtp_err_status_bad_param;
+ }
+
+ return srtp_err_status_ok;
+}
+
+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;
+ unsigned int i = 0;
+ srtp_session_keys_t *session_keys = NULL;
+
+ stat = srtp_valid_policy(p);
+ if (stat != srtp_err_status_ok) {
+ return 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;
+
+ /*
+ *To keep backwards API compatible if someone is using multiple master
+ * keys then key should be set to NULL
+ */
+ if (p->key != NULL) {
+ str->num_master_keys = 1;
+ } else {
+ str->num_master_keys = p->num_master_keys;
+ }
+
+ str->session_keys = (srtp_session_keys_t *)srtp_crypto_alloc(
+ sizeof(srtp_session_keys_t) * str->num_master_keys);
+
+ if (str->session_keys == NULL) {
+ srtp_stream_dealloc(str, NULL);
+ return srtp_err_status_alloc_fail;
+ }
+
+ for (i = 0; i < str->num_master_keys; i++) {
+ session_keys = &str->session_keys[i];
+
+ /* allocate cipher */
+ stat = srtp_crypto_kernel_alloc_cipher(
+ p->rtp.cipher_type, &session_keys->rtp_cipher,
+ p->rtp.cipher_key_len, p->rtp.auth_tag_len);
+ if (stat) {
+ srtp_stream_dealloc(str, NULL);
+ return stat;
+ }
+
+ /* allocate auth function */
+ stat = srtp_crypto_kernel_alloc_auth(
+ p->rtp.auth_type, &session_keys->rtp_auth, p->rtp.auth_key_len,
+ p->rtp.auth_tag_len);
+ if (stat) {
+ srtp_stream_dealloc(str, NULL);
+ return stat;
+ }
+
+ /*
+ * ...and now the RTCP-specific initialization - first, allocate
+ * the cipher
+ */
+ stat = srtp_crypto_kernel_alloc_cipher(
+ p->rtcp.cipher_type, &session_keys->rtcp_cipher,
+ p->rtcp.cipher_key_len, p->rtcp.auth_tag_len);
+ if (stat) {
+ srtp_stream_dealloc(str, NULL);
+ return stat;
+ }
+
+ /* allocate auth function */
+ stat = srtp_crypto_kernel_alloc_auth(
+ p->rtcp.auth_type, &session_keys->rtcp_auth, p->rtcp.auth_key_len,
+ p->rtcp.auth_tag_len);
+ if (stat) {
+ srtp_stream_dealloc(str, NULL);
+ return stat;
+ }
+
+ session_keys->mki_id = NULL;
+
+ /* allocate key limit structure */
+ session_keys->limit = (srtp_key_limit_ctx_t *)srtp_crypto_alloc(
+ sizeof(srtp_key_limit_ctx_t));
+ if (session_keys->limit == NULL) {
+ srtp_stream_dealloc(str, NULL);
+ return srtp_err_status_alloc_fail;
+ }
+ }
+
+ if (p->enc_xtn_hdr && p->enc_xtn_hdr_count > 0) {
+ srtp_cipher_type_id_t enc_xtn_hdr_cipher_type;
+ int enc_xtn_hdr_cipher_key_len;
+
+ str->enc_xtn_hdr = (int *)srtp_crypto_alloc(p->enc_xtn_hdr_count *
+ sizeof(p->enc_xtn_hdr[0]));
+ if (!str->enc_xtn_hdr) {
+ srtp_stream_dealloc(str, NULL);
+ return srtp_err_status_alloc_fail;
+ }
+ memcpy(str->enc_xtn_hdr, p->enc_xtn_hdr,
+ p->enc_xtn_hdr_count * sizeof(p->enc_xtn_hdr[0]));
+ str->enc_xtn_hdr_count = p->enc_xtn_hdr_count;
+
+ /*
+ * For GCM ciphers, the corresponding ICM cipher is used for header
+ * extensions encryption.
+ */
+ switch (p->rtp.cipher_type) {
+ case SRTP_AES_GCM_128:
+ enc_xtn_hdr_cipher_type = SRTP_AES_ICM_128;
+ enc_xtn_hdr_cipher_key_len = SRTP_AES_ICM_128_KEY_LEN_WSALT;
+ break;
+ case SRTP_AES_GCM_256:
+ enc_xtn_hdr_cipher_type = SRTP_AES_ICM_256;
+ enc_xtn_hdr_cipher_key_len = SRTP_AES_ICM_256_KEY_LEN_WSALT;
+ break;
+ default:
+ enc_xtn_hdr_cipher_type = p->rtp.cipher_type;
+ enc_xtn_hdr_cipher_key_len = p->rtp.cipher_key_len;
+ break;
+ }
+
+ for (i = 0; i < str->num_master_keys; i++) {
+ session_keys = &str->session_keys[i];
+
+ /* allocate cipher for extensions header encryption */
+ stat = srtp_crypto_kernel_alloc_cipher(
+ enc_xtn_hdr_cipher_type, &session_keys->rtp_xtn_hdr_cipher,
+ enc_xtn_hdr_cipher_key_len, 0);
+ if (stat) {
+ srtp_stream_dealloc(str, NULL);
+ return stat;
+ }
+ }
+ } else {
+ for (i = 0; i < str->num_master_keys; i++) {
+ session_keys = &str->session_keys[i];
+ session_keys->rtp_xtn_hdr_cipher = NULL;
+ }
+
+ str->enc_xtn_hdr = NULL;
+ str->enc_xtn_hdr_count = 0;
+ }
+
+ 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;
+ unsigned int i = 0;
+ srtp_session_keys_t *session_keys = NULL;
+ const srtp_session_keys_t *template_session_keys = NULL;
+
+ debug_print(mod_srtp, "cloning stream (SSRC: 0x%08x)", ntohl(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;
+
+ str->num_master_keys = stream_template->num_master_keys;
+ str->session_keys = (srtp_session_keys_t *)srtp_crypto_alloc(
+ sizeof(srtp_session_keys_t) * str->num_master_keys);
+
+ if (str->session_keys == NULL) {
+ srtp_stream_dealloc(*str_ptr, stream_template);
+ *str_ptr = NULL;
+ return srtp_err_status_alloc_fail;
+ }
+
+ for (i = 0; i < stream_template->num_master_keys; i++) {
+ session_keys = &str->session_keys[i];
+ template_session_keys = &stream_template->session_keys[i];
+
+ /* set cipher and auth pointers to those of the template */
+ session_keys->rtp_cipher = template_session_keys->rtp_cipher;
+ session_keys->rtp_auth = template_session_keys->rtp_auth;
+ session_keys->rtp_xtn_hdr_cipher =
+ template_session_keys->rtp_xtn_hdr_cipher;
+ session_keys->rtcp_cipher = template_session_keys->rtcp_cipher;
+ session_keys->rtcp_auth = template_session_keys->rtcp_auth;
+ session_keys->mki_size = template_session_keys->mki_size;
+
+ if (template_session_keys->mki_size == 0) {
+ session_keys->mki_id = NULL;
+ } else {
+ session_keys->mki_id =
+ srtp_crypto_alloc(template_session_keys->mki_size);
+
+ if (session_keys->mki_id == NULL) {
+ srtp_stream_dealloc(*str_ptr, stream_template);
+ *str_ptr = NULL;
+ return srtp_err_status_init_fail;
+ }
+ memcpy(session_keys->mki_id, template_session_keys->mki_id,
+ session_keys->mki_size);
+ }
+ /* Copy the salt values */
+ memcpy(session_keys->salt, template_session_keys->salt,
+ SRTP_AEAD_SALT_LEN);
+ memcpy(session_keys->c_salt, template_session_keys->c_salt,
+ SRTP_AEAD_SALT_LEN);
+
+ /* set key limit to point to that of the template */
+ status = srtp_key_limit_clone(template_session_keys->limit,
+ &session_keys->limit);
+ if (status) {
+ srtp_stream_dealloc(*str_ptr, stream_template);
+ *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_stream_dealloc(*str_ptr, stream_template);
+ *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;
+
+ /* reset pending ROC */
+ str->pending_roc = 0;
+
+ /* set direction and security services */
+ str->direction = stream_template->direction;
+ str->rtp_services = stream_template->rtp_services;
+ str->rtcp_services = stream_template->rtcp_services;
+
+ /* copy information about extensions header encryption */
+ str->enc_xtn_hdr = stream_template->enc_xtn_hdr;
+ str->enc_xtn_hdr_count = stream_template->enc_xtn_hdr_count;
+
+ /* 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,
+ label_rtp_header_encryption = 0x06,
+ label_rtp_header_salt = 0x07
+} srtp_prf_label;
+
+#define MAX_SRTP_KEY_LEN 256
+
+#if defined(OPENSSL) && defined(OPENSSL_KDF)
+#define MAX_SRTP_AESKEY_LEN 32
+#define MAX_SRTP_SALT_LEN 14
+
+/*
+ * srtp_kdf_t represents a key derivation function. The SRTP
+ * default KDF is the only one implemented at present.
+ */
+typedef struct {
+ uint8_t master_key[MAX_SRTP_AESKEY_LEN];
+ uint8_t master_salt[MAX_SRTP_SALT_LEN];
+ const EVP_CIPHER *evp;
+} srtp_kdf_t;
+
+static srtp_err_status_t srtp_kdf_init(srtp_kdf_t *kdf,
+ const uint8_t *key,
+ int key_len,
+ int salt_len)
+{
+ memset(kdf, 0x0, sizeof(srtp_kdf_t));
+
+ /* The NULL cipher has zero key length */
+ if (key_len == 0)
+ return srtp_err_status_ok;
+
+ if ((key_len > MAX_SRTP_AESKEY_LEN) || (salt_len > MAX_SRTP_SALT_LEN)) {
+ return srtp_err_status_bad_param;
+ }
+ switch (key_len) {
+ case SRTP_AES_256_KEYSIZE:
+ kdf->evp = EVP_aes_256_ctr();
+ break;
+ case SRTP_AES_192_KEYSIZE:
+ kdf->evp = EVP_aes_192_ctr();
+ break;
+ case SRTP_AES_128_KEYSIZE:
+ kdf->evp = EVP_aes_128_ctr();
+ break;
+ default:
+ return srtp_err_status_bad_param;
+ break;
+ }
+ memcpy(kdf->master_key, key, key_len);
+ memcpy(kdf->master_salt, key + key_len, salt_len);
+ return srtp_err_status_ok;
+}
+
+static srtp_err_status_t srtp_kdf_generate(srtp_kdf_t *kdf,
+ srtp_prf_label label,
+ uint8_t *key,
+ unsigned int length)
+{
+ int ret;
+
+ /* The NULL cipher will not have an EVP */
+ if (!kdf->evp)
+ return srtp_err_status_ok;
+ octet_string_set_to_zero(key, length);
+
+ /*
+ * Invoke the OpenSSL SRTP KDF function
+ * This is useful if OpenSSL is in FIPS mode and FIP
+ * compliance is required for SRTP.
+ */
+ ret = kdf_srtp(kdf->evp, (char *)&kdf->master_key,
+ (char *)&kdf->master_salt, NULL, NULL, label, (char *)key);
+ if (ret == -1) {
+ return (srtp_err_status_algo_fail);
+ }
+
+ return srtp_err_status_ok;
+}
+
+static srtp_err_status_t srtp_kdf_clear(srtp_kdf_t *kdf)
+{
+ octet_string_set_to_zero(kdf->master_key, MAX_SRTP_AESKEY_LEN);
+ octet_string_set_to_zero(kdf->master_salt, MAX_SRTP_SALT_LEN);
+ kdf->evp = NULL;
+
+ return srtp_err_status_ok;
+}
+
+#else /* if OPENSSL_KDF */
+
+/*
+ * 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;
+
+static srtp_err_status_t srtp_kdf_init(srtp_kdf_t *kdf,
+ const uint8_t *key,
+ int key_len)
+{
+ srtp_cipher_type_id_t cipher_id;
+ srtp_err_status_t stat;
+
+ switch (key_len) {
+ case SRTP_AES_ICM_256_KEY_LEN_WSALT:
+ cipher_id = SRTP_AES_ICM_256;
+ break;
+ case SRTP_AES_ICM_192_KEY_LEN_WSALT:
+ cipher_id = SRTP_AES_ICM_192;
+ break;
+ case SRTP_AES_ICM_128_KEY_LEN_WSALT:
+ cipher_id = SRTP_AES_ICM_128;
+ break;
+ default:
+ return srtp_err_status_bad_param;
+ break;
+ }
+
+ stat = srtp_crypto_kernel_alloc_cipher(cipher_id, &kdf->cipher, key_len, 0);
+ if (stat)
+ return stat;
+
+ stat = srtp_cipher_init(kdf->cipher, key);
+ if (stat) {
+ srtp_cipher_dealloc(kdf->cipher);
+ return stat;
+ }
+ return srtp_err_status_ok;
+}
+
+static srtp_err_status_t srtp_kdf_generate(srtp_kdf_t *kdf,
+ srtp_prf_label label,
+ uint8_t *key,
+ unsigned int length)
+{
+ srtp_err_status_t status;
+ v128_t nonce;
+
+ /* set eigth octet of nonce to <label>, set the rest of it to zero */
+ v128_set_to_zero(&nonce);
+ nonce.v8[7] = label;
+
+ status = srtp_cipher_set_iv(kdf->cipher, (uint8_t *)&nonce,
+ srtp_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;
+}
+
+static srtp_err_status_t srtp_kdf_clear(srtp_kdf_t *kdf)
+{
+ srtp_err_status_t status;
+ status = srtp_cipher_dealloc(kdf->cipher);
+ if (status)
+ return status;
+ kdf->cipher = NULL;
+ return srtp_err_status_ok;
+}
+#endif /* else OPENSSL_KDF */
+
+/*
+ * end of key derivation functions
+ */
+
+/* Get the base key length corresponding to a given combined key+salt
+ * length for the given cipher.
+ * 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_NULL_CIPHER:
+ return 0;
+ case SRTP_AES_ICM_128:
+ case SRTP_AES_ICM_192:
+ case SRTP_AES_ICM_256:
+ /* The legacy modes are derived from
+ * the configured key length on the policy */
+ return key_length - SRTP_SALT_LEN;
+ case SRTP_AES_GCM_128:
+ return key_length - SRTP_AEAD_SALT_LEN;
+ case SRTP_AES_GCM_256:
+ return key_length - SRTP_AEAD_SALT_LEN;
+ default:
+ return key_length;
+ }
+}
+
+/* Get the key length that the application should supply for the given cipher */
+static inline int full_key_length(const srtp_cipher_type_t *cipher)
+{
+ switch (cipher->id) {
+ case SRTP_NULL_CIPHER:
+ case SRTP_AES_ICM_128:
+ return SRTP_AES_ICM_128_KEY_LEN_WSALT;
+ case SRTP_AES_ICM_192:
+ return SRTP_AES_ICM_192_KEY_LEN_WSALT;
+ case SRTP_AES_ICM_256:
+ return SRTP_AES_ICM_256_KEY_LEN_WSALT;
+ case SRTP_AES_GCM_128:
+ return SRTP_AES_GCM_128_KEY_LEN_WSALT;
+ case SRTP_AES_GCM_256:
+ return SRTP_AES_GCM_256_KEY_LEN_WSALT;
+ default:
+ return 0;
+ }
+}
+
+unsigned int srtp_validate_policy_master_keys(const srtp_policy_t *policy)
+{
+ unsigned long i = 0;
+
+ if (policy->key == NULL) {
+ if (policy->num_master_keys <= 0)
+ return 0;
+
+ if (policy->num_master_keys > SRTP_MAX_NUM_MASTER_KEYS)
+ return 0;
+
+ for (i = 0; i < policy->num_master_keys; i++) {
+ if (policy->keys[i]->key == NULL)
+ return 0;
+ if (policy->keys[i]->mki_size > SRTP_MAX_MKI_LEN)
+ return 0;
+ }
+ }
+
+ return 1;
+}
+
+srtp_session_keys_t *srtp_get_session_keys_with_mki_index(
+ srtp_stream_ctx_t *stream,
+ unsigned int use_mki,
+ unsigned int mki_index)
+{
+ if (use_mki) {
+ if (mki_index >= stream->num_master_keys) {
+ return NULL;
+ }
+ return &stream->session_keys[mki_index];
+ }
+
+ return &stream->session_keys[0];
+}
+
+unsigned int srtp_inject_mki(uint8_t *mki_tag_location,
+ srtp_session_keys_t *session_keys,
+ unsigned int use_mki)
+{
+ unsigned int mki_size = 0;
+
+ if (use_mki) {
+ mki_size = session_keys->mki_size;
+
+ if (mki_size != 0) {
+ // Write MKI into memory
+ memcpy(mki_tag_location, session_keys->mki_id, mki_size);
+ }
+ }
+
+ return mki_size;
+}
+
+srtp_err_status_t srtp_stream_init_all_master_keys(
+ srtp_stream_ctx_t *srtp,
+ unsigned char *key,
+ srtp_master_key_t **keys,
+ const unsigned int max_master_keys)
+{
+ unsigned int i = 0;
+ srtp_err_status_t status = srtp_err_status_ok;
+ srtp_master_key_t single_master_key;
+
+ if (key != NULL) {
+ srtp->num_master_keys = 1;
+ single_master_key.key = key;
+ single_master_key.mki_id = NULL;
+ single_master_key.mki_size = 0;
+ status = srtp_stream_init_keys(srtp, &single_master_key, 0);
+ } else {
+ srtp->num_master_keys = max_master_keys;
+
+ for (i = 0; i < srtp->num_master_keys && i < SRTP_MAX_NUM_MASTER_KEYS;
+ i++) {
+ status = srtp_stream_init_keys(srtp, keys[i], i);
+
+ if (status) {
+ return status;
+ }
+ }
+ }
+
+ return status;
+}
+
+srtp_err_status_t srtp_stream_init_keys(srtp_stream_ctx_t *srtp,
+ srtp_master_key_t *master_key,
+ const unsigned int current_mki_index)
+{
+ srtp_err_status_t stat;
+ srtp_kdf_t kdf;
+ uint8_t tmp_key[MAX_SRTP_KEY_LEN];
+ int input_keylen, input_keylen_rtcp;
+ int kdf_keylen = 30, rtp_keylen, rtcp_keylen;
+ int rtp_base_key_len, rtp_salt_len;
+ int rtcp_base_key_len, rtcp_salt_len;
+ srtp_session_keys_t *session_keys = NULL;
+ unsigned char *key = master_key->key;
+
+ /* 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.
+ */
+ session_keys = &srtp->session_keys[current_mki_index];
+
+/* initialize key limit to maximum value */
+#ifdef NO_64BIT_MATH
+ {
+ uint64_t temp;
+ temp = make64(UINT_MAX, UINT_MAX);
+ srtp_key_limit_set(session_keys->limit, temp);
+ }
+#else
+ srtp_key_limit_set(session_keys->limit, 0xffffffffffffLL);
+#endif
+
+ if (master_key->mki_size != 0) {
+ session_keys->mki_id = srtp_crypto_alloc(master_key->mki_size);
+
+ if (session_keys->mki_id == NULL) {
+ return srtp_err_status_init_fail;
+ }
+ memcpy(session_keys->mki_id, master_key->mki_id, master_key->mki_size);
+ } else {
+ session_keys->mki_id = NULL;
+ }
+
+ session_keys->mki_size = master_key->mki_size;
+
+ input_keylen = full_key_length(session_keys->rtp_cipher->type);
+ input_keylen_rtcp = full_key_length(session_keys->rtcp_cipher->type);
+ if (input_keylen_rtcp > input_keylen) {
+ input_keylen = input_keylen_rtcp;
+ }
+
+ rtp_keylen = srtp_cipher_get_key_length(session_keys->rtp_cipher);
+ rtcp_keylen = srtp_cipher_get_key_length(session_keys->rtcp_cipher);
+ rtp_base_key_len =
+ base_key_length(session_keys->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 */
+ }
+
+ if (input_keylen > kdf_keylen) {
+ kdf_keylen = 46; /* AES-CTR mode is always used for KDF */
+ }
+
+ debug_print(mod_srtp, "input key len: %d", input_keylen);
+ 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, input_keylen);
+
+/* initialize KDF state */
+#if defined(OPENSSL) && defined(OPENSSL_KDF)
+ stat = srtp_kdf_init(&kdf, (const uint8_t *)tmp_key, rtp_base_key_len,
+ rtp_salt_len);
+#else
+ stat = srtp_kdf_init(&kdf, (const uint8_t *)tmp_key, kdf_keylen);
+#endif
+ if (stat) {
+ /* zeroize temp buffer */
+ octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+ 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_print0(mod_srtp, "found rtp_salt_len > 0, generating salt");
+
+ /* 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(session_keys->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 = srtp_cipher_init(session_keys->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;
+ }
+
+ if (session_keys->rtp_xtn_hdr_cipher) {
+ /* generate extensions header encryption key */
+ int rtp_xtn_hdr_keylen;
+ int rtp_xtn_hdr_base_key_len;
+ int rtp_xtn_hdr_salt_len;
+ srtp_kdf_t tmp_kdf;
+ srtp_kdf_t *xtn_hdr_kdf;
+
+ if (session_keys->rtp_xtn_hdr_cipher->type !=
+ session_keys->rtp_cipher->type) {
+ /*
+ * With GCM ciphers, the header extensions are still encrypted using
+ * the corresponding ICM cipher.
+ * See https://tools.ietf.org/html/rfc7714#section-8.3
+ */
+ uint8_t tmp_xtn_hdr_key[MAX_SRTP_KEY_LEN];
+ rtp_xtn_hdr_keylen =
+ srtp_cipher_get_key_length(session_keys->rtp_xtn_hdr_cipher);
+ rtp_xtn_hdr_base_key_len = base_key_length(
+ session_keys->rtp_xtn_hdr_cipher->type, rtp_xtn_hdr_keylen);
+ rtp_xtn_hdr_salt_len =
+ rtp_xtn_hdr_keylen - rtp_xtn_hdr_base_key_len;
+ if (rtp_xtn_hdr_salt_len > rtp_salt_len) {
+ switch (session_keys->rtp_cipher->type->id) {
+ case SRTP_AES_GCM_128:
+ case SRTP_AES_GCM_256:
+ /*
+ * The shorter GCM salt is padded to the required ICM salt
+ * length.
+ */
+ rtp_xtn_hdr_salt_len = rtp_salt_len;
+ break;
+ default:
+ /* zeroize temp buffer */
+ octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+ return srtp_err_status_bad_param;
+ }
+ }
+ memset(tmp_xtn_hdr_key, 0x0, MAX_SRTP_KEY_LEN);
+ memcpy(tmp_xtn_hdr_key, key,
+ (rtp_xtn_hdr_base_key_len + rtp_xtn_hdr_salt_len));
+ xtn_hdr_kdf = &tmp_kdf;
+
+/* initialize KDF state */
+#if defined(OPENSSL) && defined(OPENSSL_KDF)
+ stat =
+ srtp_kdf_init(xtn_hdr_kdf, (const uint8_t *)tmp_xtn_hdr_key,
+ rtp_xtn_hdr_base_key_len, rtp_xtn_hdr_salt_len);
+#else
+ stat = srtp_kdf_init(xtn_hdr_kdf, (const uint8_t *)tmp_xtn_hdr_key,
+ kdf_keylen);
+#endif
+ octet_string_set_to_zero(tmp_xtn_hdr_key, MAX_SRTP_KEY_LEN);
+ if (stat) {
+ /* zeroize temp buffer */
+ octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+ return srtp_err_status_init_fail;
+ }
+ } else {
+ /* Reuse main KDF. */
+ rtp_xtn_hdr_keylen = rtp_keylen;
+ rtp_xtn_hdr_base_key_len = rtp_base_key_len;
+ rtp_xtn_hdr_salt_len = rtp_salt_len;
+ xtn_hdr_kdf = &kdf;
+ }
+
+ stat = srtp_kdf_generate(xtn_hdr_kdf, label_rtp_header_encryption,
+ tmp_key, rtp_xtn_hdr_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, "extensions cipher key: %s",
+ srtp_octet_string_hex_string(tmp_key, rtp_xtn_hdr_base_key_len));
+
+ /*
+ * if the cipher in the srtp context uses a salt, then we need
+ * to generate the salt value
+ */
+ if (rtp_xtn_hdr_salt_len > 0) {
+ debug_print0(mod_srtp,
+ "found rtp_xtn_hdr_salt_len > 0, generating salt");
+
+ /* generate encryption salt, put after encryption key */
+ stat = srtp_kdf_generate(xtn_hdr_kdf, label_rtp_header_salt,
+ tmp_key + rtp_xtn_hdr_base_key_len,
+ rtp_xtn_hdr_salt_len);
+ if (stat) {
+ /* zeroize temp buffer */
+ octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
+ return srtp_err_status_init_fail;
+ }
+ }
+ if (rtp_xtn_hdr_salt_len > 0) {
+ debug_print(
+ mod_srtp, "extensions cipher salt: %s",
+ srtp_octet_string_hex_string(tmp_key + rtp_xtn_hdr_base_key_len,
+ rtp_xtn_hdr_salt_len));
+ }
+
+ /* initialize extensions header cipher */
+ stat = srtp_cipher_init(session_keys->rtp_xtn_hdr_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;
+ }
+
+ if (xtn_hdr_kdf != &kdf) {
+ /* release memory for custom header extension encryption kdf */
+ stat = srtp_kdf_clear(xtn_hdr_kdf);
+ 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(session_keys->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(session_keys->rtp_auth)));
+
+ /* initialize auth function */
+ stat = srtp_auth_init(session_keys->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(session_keys->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_print0(mod_srtp, "found rtcp_salt_len > 0, generating rtcp salt");
+
+ /* 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(session_keys->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 = srtp_cipher_init(session_keys->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(session_keys->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(session_keys->rtcp_auth)));
+
+ /* initialize auth function */
+ stat = srtp_auth_init(session_keys->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;
+
+ err = srtp_valid_policy(p);
+ if (err != srtp_err_status_ok) {
+ return 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;
+
+ /* set the SSRC value */
+ srtp->ssrc = htonl(p->ssrc.value);
+
+ /* reset pending ROC */
+ srtp->pending_roc = 0;
+
+ /* 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_all_master_keys(srtp, p->key, p->keys,
+ p->num_master_keys);
+ 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->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;
+}
+
+/*
+ * Check if the given extension header id is / should be encrypted.
+ * Returns 1 if yes, otherwise 0.
+ */
+static int srtp_protect_extension_header(srtp_stream_ctx_t *stream, int id)
+{
+ int *enc_xtn_hdr = stream->enc_xtn_hdr;
+ int count = stream->enc_xtn_hdr_count;
+
+ if (!enc_xtn_hdr || count <= 0) {
+ return 0;
+ }
+
+ while (count > 0) {
+ if (*enc_xtn_hdr == id) {
+ return 1;
+ }
+
+ enc_xtn_hdr++;
+ count--;
+ }
+ return 0;
+}
+
+/*
+ * extensions header encryption RFC 6904
+ */
+static srtp_err_status_t srtp_process_header_encryption(
+ srtp_stream_ctx_t *stream,
+ srtp_hdr_xtnd_t *xtn_hdr,
+ srtp_session_keys_t *session_keys)
+{
+ srtp_err_status_t status;
+ uint8_t keystream[257]; /* Maximum 2 bytes header + 255 bytes data. */
+ int keystream_pos;
+ uint8_t *xtn_hdr_data = ((uint8_t *)xtn_hdr) + octets_in_rtp_extn_hdr;
+ uint8_t *xtn_hdr_end =
+ xtn_hdr_data + (ntohs(xtn_hdr->length) * sizeof(uint32_t));
+
+ if (ntohs(xtn_hdr->profile_specific) == 0xbede) {
+ /* RFC 5285, section 4.2. One-Byte Header */
+ while (xtn_hdr_data < xtn_hdr_end) {
+ uint8_t xid = (*xtn_hdr_data & 0xf0) >> 4;
+ unsigned int xlen = (*xtn_hdr_data & 0x0f) + 1;
+ uint32_t xlen_with_header = 1 + xlen;
+ xtn_hdr_data++;
+
+ if (xtn_hdr_data + xlen > xtn_hdr_end)
+ return srtp_err_status_parse_err;
+
+ if (xid == 15) {
+ /* found header 15, stop further processing. */
+ break;
+ }
+
+ status = srtp_cipher_output(session_keys->rtp_xtn_hdr_cipher,
+ keystream, &xlen_with_header);
+ if (status)
+ return srtp_err_status_cipher_fail;
+
+ if (srtp_protect_extension_header(stream, xid)) {
+ keystream_pos = 1;
+ while (xlen > 0) {
+ *xtn_hdr_data ^= keystream[keystream_pos++];
+ xtn_hdr_data++;
+ xlen--;
+ }
+ } else {
+ xtn_hdr_data += xlen;
+ }
+
+ /* skip padding bytes. */
+ while (xtn_hdr_data < xtn_hdr_end && *xtn_hdr_data == 0) {
+ xtn_hdr_data++;
+ }
+ }
+ } else if ((ntohs(xtn_hdr->profile_specific) & 0xfff0) == 0x1000) {
+ /* RFC 5285, section 4.3. Two-Byte Header */
+ while (xtn_hdr_data + 1 < xtn_hdr_end) {
+ uint8_t xid = *xtn_hdr_data;
+ unsigned int xlen = *(xtn_hdr_data + 1);
+ uint32_t xlen_with_header = 2 + xlen;
+ xtn_hdr_data += 2;
+
+ if (xtn_hdr_data + xlen > xtn_hdr_end)
+ return srtp_err_status_parse_err;
+
+ status = srtp_cipher_output(session_keys->rtp_xtn_hdr_cipher,
+ keystream, &xlen_with_header);
+ if (status)
+ return srtp_err_status_cipher_fail;
+
+ if (xlen > 0 && srtp_protect_extension_header(stream, xid)) {
+ keystream_pos = 2;
+ while (xlen > 0) {
+ *xtn_hdr_data ^= keystream[keystream_pos++];
+ xtn_hdr_data++;
+ xlen--;
+ }
+ } else {
+ xtn_hdr_data += xlen;
+ }
+
+ /* skip padding bytes. */
+ while (xtn_hdr_data < xtn_hdr_end && *xtn_hdr_data == 0) {
+ xtn_hdr_data++;
+ }
+ }
+ } else {
+ /* unsupported extension header format. */
+ return srtp_err_status_parse_err;
+ }
+
+ return srtp_err_status_ok;
+}
+
+/*
+ * AEAD uses a new IV formation method. This function implements
+ * section 8.1. (SRTP IV Formation for AES-GCM) of RFC7714.
+ * 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: *session_keys - pointer to SRTP stream context session keys,
+ * 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_session_keys_t *session_keys,
+ 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, session_keys->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);
+}
+
+srtp_session_keys_t *srtp_get_session_keys(srtp_stream_ctx_t *stream,
+ uint8_t *hdr,
+ const unsigned int *pkt_octet_len,
+ unsigned int *mki_size)
+{
+ unsigned int base_mki_start_location = *pkt_octet_len;
+ unsigned int mki_start_location = 0;
+ unsigned int tag_len = 0;
+ unsigned int i = 0;
+
+ // Determine the authentication tag size
+ if (stream->session_keys[0].rtp_cipher->algorithm == SRTP_AES_GCM_128 ||
+ stream->session_keys[0].rtp_cipher->algorithm == SRTP_AES_GCM_256) {
+ tag_len = 0;
+ } else {
+ tag_len = srtp_auth_get_tag_length(stream->session_keys[0].rtp_auth);
+ }
+
+ if (tag_len > base_mki_start_location) {
+ *mki_size = 0;
+ return NULL;
+ }
+
+ base_mki_start_location -= tag_len;
+
+ for (i = 0; i < stream->num_master_keys; i++) {
+ if (stream->session_keys[i].mki_size != 0 &&
+ stream->session_keys[i].mki_size <= base_mki_start_location) {
+ *mki_size = stream->session_keys[i].mki_size;
+ mki_start_location = base_mki_start_location - *mki_size;
+
+ if (memcmp(hdr + mki_start_location, stream->session_keys[i].mki_id,
+ *mki_size) == 0) {
+ return &stream->session_keys[i];
+ }
+ }
+ }
+
+ *mki_size = 0;
+ return NULL;
+}
+
+static srtp_err_status_t srtp_estimate_index(srtp_rdbx_t *rdbx,
+ uint32_t roc,
+ srtp_xtd_seq_num_t *est,
+ srtp_sequence_number_t seq,
+ int *delta)
+{
+#ifdef NO_64BIT_MATH
+ uint32_t internal_pkt_idx_reduced;
+ uint32_t external_pkt_idx_reduced;
+ uint32_t internal_roc;
+ uint32_t roc_difference;
+#endif
+
+#ifdef NO_64BIT_MATH
+ *est = (srtp_xtd_seq_num_t)make64(roc >> 16, (roc << 16) | seq);
+ *delta = low32(est) - rdbx->index;
+#else
+ *est = (srtp_xtd_seq_num_t)(((uint64_t)roc) << 16) | seq;
+ *delta = (int)(*est - rdbx->index);
+#endif
+
+ if (*est > rdbx->index) {
+#ifdef NO_64BIT_MATH
+ internal_roc = (uint32_t)(rdbx->index >> 16);
+ roc_difference = roc - internal_roc;
+ if (roc_difference > 1) {
+ *delta = 0;
+ return srtp_err_status_pkt_idx_adv;
+ }
+
+ internal_pkt_idx_reduced = (uint32_t)(rdbx->index & 0xFFFF);
+ external_pkt_idx_reduced = (uint32_t)((roc_difference << 16) | seq);
+
+ if (external_pkt_idx_reduced - internal_pkt_idx_reduced >
+ seq_num_median) {
+ *delta = 0;
+ return srtp_err_status_pkt_idx_adv;
+ }
+#else
+ if (*est - rdbx->index > seq_num_median) {
+ *delta = 0;
+ return srtp_err_status_pkt_idx_adv;
+ }
+#endif
+ } else if (*est < rdbx->index) {
+#ifdef NO_64BIT_MATH
+
+ internal_roc = (uint32_t)(rdbx->index >> 16);
+ roc_difference = internal_roc - roc;
+ if (roc_difference > 1) {
+ *delta = 0;
+ return srtp_err_status_pkt_idx_adv;
+ }
+
+ internal_pkt_idx_reduced =
+ (uint32_t)((roc_difference << 16) | rdbx->index & 0xFFFF);
+ external_pkt_idx_reduced = (uint32_t)(seq);
+
+ if (internal_pkt_idx_reduced - external_pkt_idx_reduced >
+ seq_num_median) {
+ *delta = 0;
+ return srtp_err_status_pkt_idx_old;
+ }
+#else
+ if (rdbx->index - *est > seq_num_median) {
+ *delta = 0;
+ return srtp_err_status_pkt_idx_old;
+ }
+#endif
+ }
+
+ return srtp_err_status_ok;
+}
+
+static srtp_err_status_t srtp_get_est_pkt_index(srtp_hdr_t *hdr,
+ srtp_stream_ctx_t *stream,
+ srtp_xtd_seq_num_t *est,
+ int *delta)
+{
+ srtp_err_status_t result = srtp_err_status_ok;
+
+ if (stream->pending_roc) {
+ result = srtp_estimate_index(&stream->rtp_rdbx, stream->pending_roc,
+ est, ntohs(hdr->seq), delta);
+ } else {
+ /* estimate packet index from seq. num. in header */
+ *delta =
+ srtp_rdbx_estimate_index(&stream->rtp_rdbx, est, ntohs(hdr->seq));
+ }
+
+#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: %016" PRIx64, *est);
+#endif
+ return result;
+}
+
+/*
+ * 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_session_keys_t *session_keys,
+ unsigned int use_mki)
+{
+ srtp_hdr_t *hdr = (srtp_hdr_t *)rtp_hdr;
+ uint32_t *enc_start; /* pointer to start of encrypted portion */
+ 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;
+ uint32_t tag_len;
+ v128_t iv;
+ unsigned int aad_len;
+ srtp_hdr_xtnd_t *xtn_hdr = NULL;
+ unsigned int mki_size = 0;
+ uint8_t *mki_location = NULL;
+
+ debug_print0(mod_srtp, "function srtp_protect_aead");
+
+ /*
+ * 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(session_keys->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(session_keys->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) {
+ xtn_hdr = (srtp_hdr_xtnd_t *)enc_start;
+ enc_start += (ntohs(xtn_hdr->length) + 1);
+ }
+ /* note: the passed size is without the auth tag */
+ if (!((uint8_t *)enc_start <= (uint8_t *)hdr + *pkt_octet_len))
+ return srtp_err_status_parse_err;
+ enc_octet_len =
+ (int)(*pkt_octet_len - ((uint8_t *)enc_start - (uint8_t *)hdr));
+ if (enc_octet_len < 0)
+ return srtp_err_status_parse_err;
+
+ /*
+ * 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: %016" PRIx64, est);
+#endif
+
+ /*
+ * AEAD uses a new IV formation method
+ */
+ srtp_calc_aead_iv(session_keys, &iv, &est, hdr);
+/* 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
+
+ status = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
+ srtp_direction_encrypt);
+ if (!status && session_keys->rtp_xtn_hdr_cipher) {
+ iv.v32[0] = 0;
+ iv.v32[1] = hdr->ssrc;
+ iv.v64[1] = est;
+ status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
+ (uint8_t *)&iv, srtp_direction_encrypt);
+ }
+ if (status) {
+ return srtp_err_status_cipher_fail;
+ }
+
+ if (xtn_hdr && session_keys->rtp_xtn_hdr_cipher) {
+ /*
+ * extensions header encryption RFC 6904
+ */
+ status = srtp_process_header_encryption(stream, xtn_hdr, session_keys);
+ if (status) {
+ return status;
+ }
+ }
+
+ /*
+ * Set the AAD over the RTP header
+ */
+ aad_len = (uint8_t *)enc_start - (uint8_t *)hdr;
+ status =
+ srtp_cipher_set_aad(session_keys->rtp_cipher, (uint8_t *)hdr, aad_len);
+ if (status) {
+ return (srtp_err_status_cipher_fail);
+ }
+
+ /* Encrypt the payload */
+ status = srtp_cipher_encrypt(session_keys->rtp_cipher, (uint8_t *)enc_start,
+ (unsigned int *)&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 =
+ srtp_cipher_get_tag(session_keys->rtp_cipher,
+ (uint8_t *)enc_start + enc_octet_len, &tag_len);
+ if (status) {
+ return (srtp_err_status_cipher_fail);
+ }
+
+ mki_location = (uint8_t *)hdr + *pkt_octet_len + tag_len;
+ mki_size = srtp_inject_mki(mki_location, session_keys, use_mki);
+
+ /* increase the packet length by the length of the auth tag */
+ *pkt_octet_len += tag_len;
+
+ /* increase the packet length by the length of the mki_size */
+ *pkt_octet_len += mki_size;
+
+ 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_session_keys_t *session_keys,
+ unsigned int mki_size)
+{
+ 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;
+ srtp_hdr_xtnd_t *xtn_hdr = NULL;
+
+ debug_print0(mod_srtp, "function srtp_unprotect_aead");
+
+#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: %016" PRIx64, est);
+#endif
+
+ /* get tag length from stream */
+ tag_len = srtp_auth_get_tag_length(session_keys->rtp_auth);
+
+ /*
+ * AEAD uses a new IV formation method
+ */
+ srtp_calc_aead_iv(session_keys, &iv, &est, hdr);
+ status = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
+ srtp_direction_decrypt);
+ if (!status && session_keys->rtp_xtn_hdr_cipher) {
+ 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 = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
+ (uint8_t *)&iv, srtp_direction_encrypt);
+ }
+ 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) {
+ 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 - tag_len - mki_size)))
+ 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 - mki_size -
+ ((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 < (unsigned int)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(session_keys->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 =
+ srtp_cipher_set_aad(session_keys->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(session_keys->rtp_cipher, (uint8_t *)enc_start,
+ &enc_octet_len);
+ if (status) {
+ return status;
+ }
+
+ if (xtn_hdr && session_keys->rtp_xtn_hdr_cipher) {
+ /*
+ * extensions header encryption RFC 6904
+ */
+ status = srtp_process_header_encryption(stream, xtn_hdr, session_keys);
+ 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;
+
+ /* decrease the packet length by the length of the mki_size */
+ *pkt_octet_len -= mki_size;
+
+ return srtp_err_status_ok;
+}
+
+srtp_err_status_t srtp_protect(srtp_ctx_t *ctx,
+ void *rtp_hdr,
+ int *pkt_octet_len)
+{
+ return srtp_protect_mki(ctx, rtp_hdr, pkt_octet_len, 0, 0);
+}
+
+srtp_err_status_t srtp_protect_mki(srtp_ctx_t *ctx,
+ void *rtp_hdr,
+ int *pkt_octet_len,
+ unsigned int use_mki,
+ unsigned int mki_index)
+{
+ 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 */
+ 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;
+ srtp_hdr_xtnd_t *xtn_hdr = NULL;
+ unsigned int mki_size = 0;
+ srtp_session_keys_t *session_keys = NULL;
+ uint8_t *mki_location = NULL;
+ int advance_packet_index = 0;
+
+ debug_print0(mod_srtp, "function srtp_protect");
+
+ /* we assume the hdr is 32-bit aligned to start */
+
+ /* Verify RTP header */
+ status = srtp_validate_rtp_header(rtp_hdr, pkt_octet_len);
+ if (status)
+ return status;
+
+ /* 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);
+ }
+ }
+
+ session_keys =
+ srtp_get_session_keys_with_mki_index(stream, use_mki, mki_index);
+
+ if (session_keys == NULL)
+ return srtp_err_status_bad_mki;
+
+ /*
+ * Check if this is an AEAD stream (GCM mode). If so, then dispatch
+ * the request to our AEAD handler.
+ */
+ if (session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_128 ||
+ session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_256) {
+ return srtp_protect_aead(ctx, stream, rtp_hdr,
+ (unsigned int *)pkt_octet_len, session_keys,
+ use_mki);
+ }
+
+ /*
+ * 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(session_keys->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(session_keys->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) {
+ xtn_hdr = (srtp_hdr_xtnd_t *)enc_start;
+ enc_start += (ntohs(xtn_hdr->length) + 1);
+ }
+ /* note: the passed size is without the auth tag */
+ if (!((uint8_t *)enc_start <= (uint8_t *)hdr + *pkt_octet_len))
+ return srtp_err_status_parse_err;
+ enc_octet_len =
+ (int)(*pkt_octet_len - ((uint8_t *)enc_start - (uint8_t *)hdr));
+ if (enc_octet_len < 0)
+ return srtp_err_status_parse_err;
+ } else {
+ enc_start = NULL;
+ }
+
+ mki_location = (uint8_t *)hdr + *pkt_octet_len;
+ mki_size = srtp_inject_mki(mki_location, session_keys, use_mki);
+
+ /*
+ * 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 + mki_size;
+ } 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
+ */
+ status = srtp_get_est_pkt_index(hdr, stream, &est, &delta);
+
+ if (status && (status != srtp_err_status_pkt_idx_adv))
+ return status;
+
+ if (status == srtp_err_status_pkt_idx_adv)
+ advance_packet_index = 1;
+
+ if (advance_packet_index) {
+ srtp_rdbx_set_roc_seq(&stream->rtp_rdbx, (uint32_t)(est >> 16),
+ (uint16_t)(est & 0xFFFF));
+ stream->pending_roc = 0;
+ srtp_rdbx_add_index(&stream->rtp_rdbx, 0);
+ } else {
+ 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 */
+ }
+ 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: %016" PRIx64, est);
+#endif
+
+ /*
+ * if we're using rindael counter mode, set nonce and seq
+ */
+ if (session_keys->rtp_cipher->type->id == SRTP_AES_ICM_128 ||
+ session_keys->rtp_cipher->type->id == SRTP_AES_ICM_192 ||
+ session_keys->rtp_cipher->type->id == SRTP_AES_ICM_256) {
+ 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 = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
+ srtp_direction_encrypt);
+ if (!status && session_keys->rtp_xtn_hdr_cipher) {
+ status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
+ (uint8_t *)&iv, srtp_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 = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
+ srtp_direction_encrypt);
+ if (!status && session_keys->rtp_xtn_hdr_cipher) {
+ status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
+ (uint8_t *)&iv, srtp_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(session_keys->rtp_auth);
+ if (prefix_len) {
+ status = srtp_cipher_output(session_keys->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 (xtn_hdr && session_keys->rtp_xtn_hdr_cipher) {
+ /*
+ * extensions header encryption RFC 6904
+ */
+ status = srtp_process_header_encryption(stream, xtn_hdr, session_keys);
+ if (status) {
+ return status;
+ }
+ }
+
+ /* if we're encrypting, exor keystream into the message */
+ if (enc_start) {
+ status =
+ srtp_cipher_encrypt(session_keys->rtp_cipher, (uint8_t *)enc_start,
+ (unsigned int *)&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 = srtp_auth_start(session_keys->rtp_auth);
+ if (status)
+ return status;
+
+ /* run auth func over packet */
+ status = srtp_auth_update(session_keys->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: %016" PRIx64, est);
+ status = srtp_auth_compute(session_keys->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;
+ }
+
+ if (use_mki) {
+ /* increate the packet length by the mki size */
+ *pkt_octet_len += mki_size;
+ }
+
+ return srtp_err_status_ok;
+}
+
+srtp_err_status_t srtp_unprotect(srtp_ctx_t *ctx,
+ void *srtp_hdr,
+ int *pkt_octet_len)
+{
+ return srtp_unprotect_mki(ctx, srtp_hdr, pkt_octet_len, 0);
+}
+
+srtp_err_status_t srtp_unprotect_mki(srtp_ctx_t *ctx,
+ void *srtp_hdr,
+ int *pkt_octet_len,
+ unsigned int use_mki)
+{
+ 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;
+ srtp_hdr_xtnd_t *xtn_hdr = NULL;
+ unsigned int mki_size = 0;
+ srtp_session_keys_t *session_keys = NULL;
+ int advance_packet_index = 0;
+ uint32_t roc_to_set = 0;
+ uint16_t seq_to_set = 0;
+
+ debug_print0(mod_srtp, "function srtp_unprotect");
+
+ /* we assume the hdr is 32-bit aligned to start */
+
+ /* Verify RTP header */
+ status = srtp_validate_rtp_header(srtp_hdr, pkt_octet_len);
+ if (status)
+ return status;
+
+ /* 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)",
+ ntohl(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 {
+ status = srtp_get_est_pkt_index(hdr, stream, &est, &delta);
+
+ if (status && (status != srtp_err_status_pkt_idx_adv))
+ return status;
+
+ if (status == srtp_err_status_pkt_idx_adv) {
+ advance_packet_index = 1;
+ roc_to_set = (uint32_t)(est >> 16);
+ seq_to_set = (uint16_t)(est & 0xFFFF);
+ }
+
+ /* check replay database */
+ if (!advance_packet_index) {
+ 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: %016" PRIx64, est);
+#endif
+
+ /* Determine if MKI is being used and what session keys should be used */
+ if (use_mki) {
+ session_keys = srtp_get_session_keys(
+ stream, (uint8_t *)hdr, (const unsigned int *)pkt_octet_len,
+ &mki_size);
+
+ if (session_keys == NULL)
+ return srtp_err_status_bad_mki;
+ } else {
+ session_keys = &stream->session_keys[0];
+ }
+
+ /*
+ * Check if this is an AEAD stream (GCM mode). If so, then dispatch
+ * the request to our AEAD handler.
+ */
+ if (session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_128 ||
+ session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_256) {
+ return srtp_unprotect_aead(ctx, stream, delta, est, srtp_hdr,
+ (unsigned int *)pkt_octet_len, session_keys,
+ mki_size);
+ }
+
+ /* get tag length from stream */
+ tag_len = srtp_auth_get_tag_length(session_keys->rtp_auth);
+
+ /*
+ * set the cipher's IV properly, depending on whatever cipher we
+ * happen to be using
+ */
+ if (session_keys->rtp_cipher->type->id == SRTP_AES_ICM_128 ||
+ session_keys->rtp_cipher->type->id == SRTP_AES_ICM_192 ||
+ session_keys->rtp_cipher->type->id == SRTP_AES_ICM_256) {
+ /* 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 = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
+ srtp_direction_decrypt);
+ if (!status && session_keys->rtp_xtn_hdr_cipher) {
+ status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
+ (uint8_t *)&iv, srtp_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 = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
+ srtp_direction_decrypt);
+ if (!status && session_keys->rtp_xtn_hdr_cipher) {
+ status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
+ (uint8_t *)&iv, srtp_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) {
+ 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 - tag_len - mki_size)))
+ return srtp_err_status_parse_err;
+ enc_octet_len = (uint32_t)(*pkt_octet_len - tag_len - mki_size -
+ ((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 (session_keys->rtp_auth->prefix_len != 0) {
+ prefix_len = srtp_auth_get_prefix_length(session_keys->rtp_auth);
+ status = srtp_cipher_output(session_keys->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 = srtp_auth_start(session_keys->rtp_auth);
+ if (status)
+ return status;
+
+ /* now compute auth function over packet */
+ status = srtp_auth_update(session_keys->rtp_auth, (uint8_t *)auth_start,
+ *pkt_octet_len - tag_len - mki_size);
+ if (status)
+ return status;
+
+ /* run auth func over ROC, then write tmp tag */
+ status = srtp_auth_compute(session_keys->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 (srtp_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(session_keys->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 (xtn_hdr && session_keys->rtp_xtn_hdr_cipher) {
+ /* extensions header encryption RFC 6904 */
+ status = srtp_process_header_encryption(stream, xtn_hdr, session_keys);
+ if (status) {
+ return status;
+ }
+ }
+
+ /* if we're decrypting, add keystream into ciphertext */
+ if (enc_start) {
+ status = srtp_cipher_decrypt(session_keys->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
+ */
+ if (advance_packet_index) {
+ srtp_rdbx_set_roc_seq(&stream->rtp_rdbx, roc_to_set, seq_to_set);
+ stream->pending_roc = 0;
+ srtp_rdbx_add_index(&stream->rtp_rdbx, 0);
+ } else {
+ srtp_rdbx_add_index(&stream->rtp_rdbx, delta);
+ }
+
+ /* decrease the packet length by the length of the auth tag */
+ *pkt_octet_len -= tag_len;
+
+ /* decrease the packet length by the mki size */
+ *pkt_octet_len -= mki_size;
+
+ 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;
+}
+
+/*
+ * 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(stream, session->stream_template);
+ if (status)
+ return status;
+ stream = next;
+ }
+
+ /* deallocate stream template, if there is one */
+ if (session->stream_template != NULL) {
+ status = srtp_stream_dealloc(session->stream_template, NULL);
+ if (status)
+ return status;
+ }
+
+ /* 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;
+
+ status = srtp_valid_policy(policy);
+ if (status != srtp_err_status_ok) {
+ return status;
+ }
+
+ /* sanity check arguments */
+ if ((session == NULL) || (policy == NULL) ||
+ (!srtp_validate_policy_master_keys(policy)))
+ 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_stream_dealloc(tmp, NULL);
+ 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) {
+ srtp_stream_dealloc(tmp, NULL);
+ 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) {
+ srtp_stream_dealloc(tmp, NULL);
+ 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_stream_dealloc(tmp, NULL);
+ 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;
+
+ stat = srtp_valid_policy(policy);
+ if (stat != srtp_err_status_ok) {
+ return stat;
+ }
+
+ /* 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);
+ *session = NULL;
+ 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(stream, session->stream_template);
+ if (status)
+ return status;
+
+ return srtp_err_status_ok;
+}
+
+srtp_err_status_t srtp_update(srtp_t session, const srtp_policy_t *policy)
+{
+ srtp_err_status_t stat;
+
+ stat = srtp_valid_policy(policy);
+ if (stat != srtp_err_status_ok) {
+ return stat;
+ }
+
+ /* sanity check arguments */
+ if ((session == NULL) || (policy == NULL) ||
+ (!srtp_validate_policy_master_keys(policy))) {
+ return srtp_err_status_bad_param;
+ }
+
+ while (policy != NULL) {
+ stat = srtp_update_stream(session, policy);
+ if (stat) {
+ return stat;
+ }
+
+ /* set policy to next item in list */
+ policy = policy->next;
+ }
+ return srtp_err_status_ok;
+}
+
+static srtp_err_status_t update_template_streams(srtp_t session,
+ const srtp_policy_t *policy)
+{
+ srtp_err_status_t status;
+ srtp_stream_t new_stream_template;
+ srtp_stream_t new_stream_list = NULL;
+
+ status = srtp_valid_policy(policy);
+ if (status != srtp_err_status_ok) {
+ return status;
+ }
+
+ if (session->stream_template == NULL) {
+ return srtp_err_status_bad_param;
+ }
+
+ /* allocate new template stream */
+ status = srtp_stream_alloc(&new_stream_template, policy);
+ if (status) {
+ return status;
+ }
+
+ /* initialize new template stream */
+ status = srtp_stream_init(new_stream_template, policy);
+ if (status) {
+ srtp_crypto_free(new_stream_template);
+ return status;
+ }
+
+ /* for all old templated streams */
+ for (;;) {
+ srtp_stream_t stream;
+ uint32_t ssrc;
+ srtp_xtd_seq_num_t old_index;
+ srtp_rdb_t old_rtcp_rdb;
+
+ stream = session->stream_list;
+ while ((stream != NULL) &&
+ (stream->session_keys[0].rtp_auth !=
+ session->stream_template->session_keys[0].rtp_auth)) {
+ stream = stream->next;
+ }
+ if (stream == NULL) {
+ /* no more templated streams */
+ break;
+ }
+
+ /* save old extendard seq */
+ ssrc = stream->ssrc;
+ old_index = stream->rtp_rdbx.index;
+ old_rtcp_rdb = stream->rtcp_rdb;
+
+ /* remove stream */
+ status = srtp_remove_stream(session, ssrc);
+ if (status) {
+ /* free new allocations */
+ while (new_stream_list != NULL) {
+ srtp_stream_t next = new_stream_list->next;
+ srtp_stream_dealloc(new_stream_list, new_stream_template);
+ new_stream_list = next;
+ }
+ srtp_stream_dealloc(new_stream_template, NULL);
+ return status;
+ }
+
+ /* allocate and initialize a new stream */
+ status = srtp_stream_clone(new_stream_template, ssrc, &stream);
+ if (status) {
+ /* free new allocations */
+ while (new_stream_list != NULL) {
+ srtp_stream_t next = new_stream_list->next;
+ srtp_stream_dealloc(new_stream_list, new_stream_template);
+ new_stream_list = next;
+ }
+ srtp_stream_dealloc(new_stream_template, NULL);
+ return status;
+ }
+
+ /* add new stream to the head of the new_stream_list */
+ stream->next = new_stream_list;
+ new_stream_list = stream;
+
+ /* restore old extended seq */
+ stream->rtp_rdbx.index = old_index;
+ stream->rtcp_rdb = old_rtcp_rdb;
+ }
+ /* dealloc old template */
+ srtp_stream_dealloc(session->stream_template, NULL);
+ /* set new template */
+ session->stream_template = new_stream_template;
+ /* add new list */
+ if (new_stream_list) {
+ srtp_stream_t tail = new_stream_list;
+ while (tail->next) {
+ tail = tail->next;
+ }
+ tail->next = session->stream_list;
+ session->stream_list = new_stream_list;
+ }
+ return status;
+}
+
+static srtp_err_status_t update_stream(srtp_t session,
+ const srtp_policy_t *policy)
+{
+ srtp_err_status_t status;
+ srtp_xtd_seq_num_t old_index;
+ srtp_rdb_t old_rtcp_rdb;
+ srtp_stream_t stream;
+
+ status = srtp_valid_policy(policy);
+ if (status != srtp_err_status_ok) {
+ return status;
+ }
+
+ stream = srtp_get_stream(session, htonl(policy->ssrc.value));
+ if (stream == NULL) {
+ return srtp_err_status_bad_param;
+ }
+
+ /* save old extendard seq */
+ old_index = stream->rtp_rdbx.index;
+ old_rtcp_rdb = stream->rtcp_rdb;
+
+ status = srtp_remove_stream(session, htonl(policy->ssrc.value));
+ if (status) {
+ return status;
+ }
+
+ status = srtp_add_stream(session, policy);
+ if (status) {
+ return status;
+ }
+
+ stream = srtp_get_stream(session, htonl(policy->ssrc.value));
+ if (stream == NULL) {
+ return srtp_err_status_fail;
+ }
+
+ /* restore old extended seq */
+ stream->rtp_rdbx.index = old_index;
+ stream->rtcp_rdb = old_rtcp_rdb;
+
+ return srtp_err_status_ok;
+}
+
+srtp_err_status_t srtp_update_stream(srtp_t session,
+ const srtp_policy_t *policy)
+{
+ srtp_err_status_t status;
+
+ status = srtp_valid_policy(policy);
+ if (status != srtp_err_status_ok) {
+ return status;
+ }
+
+ /* sanity check arguments */
+ if ((session == NULL) || (policy == NULL) ||
+ (!srtp_validate_policy_master_keys(policy)))
+ return srtp_err_status_bad_param;
+
+ switch (policy->ssrc.type) {
+ case (ssrc_any_outbound):
+ case (ssrc_any_inbound):
+ status = update_template_streams(session, policy);
+ break;
+ case (ssrc_specific):
+ status = update_stream(session, policy);
+ break;
+ case (ssrc_undefined):
+ default:
+ return srtp_err_status_bad_param;
+ }
+
+ return status;
+}
+
+/*
+ * The default policy - provides a convenient way for callers to use
+ * the default security policy
+ *
+ * The default policy is defined in RFC 3711
+ * (Section 5. Default and mandatory-to-implement Transforms)
+ *
+ */
+
+/*
+ * 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_128;
+ p->cipher_key_len =
+ SRTP_AES_ICM_128_KEY_LEN_WSALT; /* 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_128;
+ p->cipher_key_len =
+ SRTP_AES_ICM_128_KEY_LEN_WSALT; /* 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_128;
+ p->cipher_key_len =
+ SRTP_AES_ICM_128_KEY_LEN_WSALT; /* 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_128;
+ p->cipher_key_len =
+ SRTP_AES_ICM_128_KEY_LEN_WSALT; /* 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 = 16;
+ 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 = 16;
+ 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 RFC 6188
+ */
+
+ p->cipher_type = SRTP_AES_ICM_256;
+ p->cipher_key_len = SRTP_AES_ICM_256_KEY_LEN_WSALT;
+ 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 RFC 6188
+ *
+ * note that this crypto policy is intended for SRTP, but not SRTCP
+ */
+
+ p->cipher_type = SRTP_AES_ICM_256;
+ p->cipher_key_len = SRTP_AES_ICM_256_KEY_LEN_WSALT;
+ 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_256;
+ p->cipher_key_len = SRTP_AES_ICM_256_KEY_LEN_WSALT;
+ 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_aes_cm_192_hmac_sha1_80(srtp_crypto_policy_t *p)
+{
+ /*
+ * corresponds to RFC 6188
+ */
+
+ p->cipher_type = SRTP_AES_ICM_192;
+ p->cipher_key_len = SRTP_AES_ICM_192_KEY_LEN_WSALT;
+ 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_192_hmac_sha1_32(srtp_crypto_policy_t *p)
+{
+ /*
+ * corresponds to RFC 6188
+ *
+ * note that this crypto policy is intended for SRTP, but not SRTCP
+ */
+
+ p->cipher_type = SRTP_AES_ICM_192;
+ p->cipher_key_len = SRTP_AES_ICM_192_KEY_LEN_WSALT;
+ 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-192 with no authentication.
+ */
+void srtp_crypto_policy_set_aes_cm_192_null_auth(srtp_crypto_policy_t *p)
+{
+ p->cipher_type = SRTP_AES_ICM_192;
+ p->cipher_key_len = SRTP_AES_ICM_192_KEY_LEN_WSALT;
+ p->auth_type = SRTP_NULL_AUTH;
+ p->auth_key_len = 0;
+ p->auth_tag_len = 0;
+ p->sec_serv = sec_serv_conf;
+}
+
+/*
+ * 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_GCM_128;
+ p->cipher_key_len = SRTP_AES_GCM_128_KEY_LEN_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_GCM_256;
+ p->cipher_key_len = SRTP_AES_GCM_256_KEY_LEN_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_GCM_128;
+ p->cipher_key_len = SRTP_AES_GCM_128_KEY_LEN_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_GCM_256;
+ p->cipher_key_len = SRTP_AES_GCM_256_KEY_LEN_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_GCM_128;
+ p->cipher_key_len = SRTP_AES_GCM_128_KEY_LEN_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_GCM_256;
+ p->cipher_key_len = SRTP_AES_GCM_256_KEY_LEN_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;
+}
+
+/*
+ * secure rtcp functions
+ */
+
+/*
+ * AEAD uses a new IV formation method. This function implements
+ * section 9.1 (SRTCP IV Formation for AES-GCM) from RFC7714.
+ * 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: *session_keys - pointer to SRTP stream context session keys,
+ * 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
+ *
+ * Returns: srtp_err_status_ok if no error or srtp_err_status_bad_param
+ * if seq_num is invalid
+ *
+ */
+static srtp_err_status_t srtp_calc_aead_iv_srtcp(
+ srtp_session_keys_t *session_keys,
+ 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;
+
+ /*
+ * The SRTCP index (seq_num) spans bits 0 through 30 inclusive.
+ * The most significant bit should be zero.
+ */
+ if (seq_num & 0x80000000UL) {
+ return srtp_err_status_bad_param;
+ }
+ in.v32[2] = htonl(seq_num);
+
+ debug_print(mod_srtp, "Pre-salted RTCP IV = %s\n", v128_hex_string(&in));
+
+ /*
+ * Get the SALT value from the context
+ */
+ memcpy(salt.v8, session_keys->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);
+
+ return srtp_err_status_ok;
+}
+
+/*
+ * 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,
+ srtp_session_keys_t *session_keys,
+ unsigned int use_mki)
+{
+ srtcp_hdr_t *hdr = (srtcp_hdr_t *)rtcp_hdr;
+ uint32_t *enc_start; /* pointer to start of encrypted portion */
+ uint32_t *trailer_p; /* pointer to start of trailer */
+ uint32_t trailer; /* trailer value */
+ 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;
+ uint32_t tag_len;
+ uint32_t seq_num;
+ v128_t iv;
+ uint32_t tseq;
+ unsigned int mki_size = 0;
+
+ /* get tag length from stream context */
+ tag_len = srtp_auth_get_tag_length(session_keys->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!
+ */
+ trailer_p = (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 */
+ }
+
+ mki_size = srtp_inject_mki((uint8_t *)hdr + *pkt_octet_len + tag_len +
+ sizeof(srtcp_trailer_t),
+ session_keys, use_mki);
+
+ /*
+ * 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);
+
+ memcpy(trailer_p, &trailer, sizeof(trailer));
+
+ /*
+ * Calculate and set the IV
+ */
+ status = srtp_calc_aead_iv_srtcp(session_keys, &iv, seq_num, hdr);
+ if (status) {
+ return srtp_err_status_cipher_fail;
+ }
+ status = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
+ srtp_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 = srtp_cipher_set_aad(session_keys->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 RFC 7714 (Section 9.3. Data
+ * Types in Unencrypted SRTCP Compound Packets)
+ */
+ status = srtp_cipher_set_aad(session_keys->rtcp_cipher, (uint8_t *)hdr,
+ *pkt_octet_len);
+ if (status) {
+ return (srtp_err_status_cipher_fail);
+ }
+ }
+ /*
+ * Process the sequence# as AAD
+ */
+ tseq = trailer;
+ status = srtp_cipher_set_aad(session_keys->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(session_keys->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 = srtp_cipher_get_tag(session_keys->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(session_keys->rtcp_cipher, NULL, &nolen);
+ if (status) {
+ return srtp_err_status_cipher_fail;
+ }
+ /*
+ * Get the tag and append that to the output
+ */
+ status = srtp_cipher_get_tag(session_keys->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));
+
+ /* increase the packet by the mki_size */
+ *pkt_octet_len += mki_size;
+
+ 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,
+ srtp_session_keys_t *session_keys,
+ unsigned int use_mki)
+{
+ srtcp_hdr_t *hdr = (srtcp_hdr_t *)srtcp_hdr;
+ uint32_t *enc_start; /* pointer to start of encrypted portion */
+ uint32_t *trailer_p; /* pointer to start of trailer */
+ uint32_t trailer; /* trailer value */
+ 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;
+ unsigned int mki_size = 0;
+
+ /* get tag length from stream context */
+ tag_len = srtp_auth_get_tag_length(session_keys->rtcp_auth);
+
+ if (use_mki) {
+ mki_size = session_keys->mki_size;
+ }
+
+ /*
+ * 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 */
+ trailer_p = (uint32_t *)((char *)hdr + *pkt_octet_len -
+ sizeof(srtcp_trailer_t) - mki_size);
+ memcpy(&trailer, trailer_p, sizeof(trailer));
+
+ /*
+ * 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) + mki_size);
+ auth_tag = (uint8_t *)hdr + *pkt_octet_len - tag_len - mki_size -
+ sizeof(srtcp_trailer_t);
+
+ if (*((unsigned char *)trailer_p) & 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
+ */
+ status = srtp_calc_aead_iv_srtcp(session_keys, &iv, seq_num, hdr);
+ if (status) {
+ return srtp_err_status_cipher_fail;
+ }
+ status = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
+ srtp_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 = srtp_cipher_set_aad(session_keys->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 RFC 7714 (Section 9.3. Data
+ * Types in Unencrypted SRTCP Compound Packets)
+ */
+ status = srtp_cipher_set_aad(
+ session_keys->rtcp_cipher, (uint8_t *)hdr,
+ (*pkt_octet_len - tag_len - sizeof(srtcp_trailer_t) - mki_size));
+ if (status) {
+ return (srtp_err_status_cipher_fail);
+ }
+ }
+
+ /*
+ * Process the sequence# as AAD
+ */
+ tseq = trailer;
+ status = srtp_cipher_set_aad(session_keys->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(session_keys->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(session_keys->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) + mki_size);
+
+ /*
+ * 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)
+{
+ return srtp_protect_rtcp_mki(ctx, rtcp_hdr, pkt_octet_len, 0, 0);
+}
+
+srtp_err_status_t srtp_protect_rtcp_mki(srtp_t ctx,
+ void *rtcp_hdr,
+ int *pkt_octet_len,
+ unsigned int use_mki,
+ unsigned int mki_index)
+{
+ 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_p; /* pointer to start of trailer */
+ uint32_t trailer; /* trailer value */
+ 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;
+ unsigned int mki_size = 0;
+ srtp_session_keys_t *session_keys = 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_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);
+ }
+ }
+
+ session_keys =
+ srtp_get_session_keys_with_mki_index(stream, use_mki, mki_index);
+
+ if (session_keys == NULL)
+ return srtp_err_status_bad_mki;
+
+ /*
+ * Check if this is an AEAD stream (GCM mode). If so, then dispatch
+ * the request to our AEAD handler.
+ */
+ if (session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_128 ||
+ session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_256) {
+ return srtp_protect_rtcp_aead(ctx, stream, rtcp_hdr,
+ (unsigned int *)pkt_octet_len,
+ session_keys, use_mki);
+ }
+
+ /* get tag length from stream context */
+ tag_len = srtp_auth_get_tag_length(session_keys->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!
+ */
+ trailer_p = (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 */
+ }
+
+ mki_size = srtp_inject_mki((uint8_t *)hdr + *pkt_octet_len +
+ sizeof(srtcp_trailer_t),
+ session_keys, use_mki);
+
+ /*
+ * 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) + mki_size;
+
+ /*
+ * 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);
+
+ memcpy(trailer_p, &trailer, sizeof(trailer));
+
+ /*
+ * if we're using rindael counter mode, set nonce and seq
+ */
+ if (session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_128 ||
+ session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_192 ||
+ session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_256) {
+ 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 = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
+ srtp_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 = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
+ srtp_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(session_keys->rtcp_auth);
+ status = srtp_cipher_output(session_keys->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(session_keys->rtcp_cipher,
+ (uint8_t *)enc_start, &enc_octet_len);
+ if (status)
+ return srtp_err_status_cipher_fail;
+ }
+
+ /* initialize auth func context */
+ status = srtp_auth_start(session_keys->rtcp_auth);
+ if (status)
+ return status;
+
+ /*
+ * run auth func over packet (including trailer), and write the
+ * result at auth_tag
+ */
+ status =
+ srtp_auth_compute(session_keys->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));
+
+ /* increase the packet by the mki_size */
+ *pkt_octet_len += mki_size;
+
+ return srtp_err_status_ok;
+}
+
+srtp_err_status_t srtp_unprotect_rtcp(srtp_t ctx,
+ void *srtcp_hdr,
+ int *pkt_octet_len)
+{
+ return srtp_unprotect_rtcp_mki(ctx, srtcp_hdr, pkt_octet_len, 0);
+}
+
+srtp_err_status_t srtp_unprotect_rtcp_mki(srtp_t ctx,
+ void *srtcp_hdr,
+ int *pkt_octet_len,
+ unsigned int use_mki)
+{
+ 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_p; /* pointer to start of trailer */
+ uint32_t trailer; /* trailer value */
+ 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];
+ 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 */
+ unsigned int mki_size = 0;
+ srtp_session_keys_t *session_keys = NULL;
+
+ /* we assume the hdr is 32-bit aligned to start */
+
+ if (*pkt_octet_len < 0)
+ return srtp_err_status_bad_param;
+
+ /*
+ * 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 ((unsigned int)(*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;
+
+ debug_print(mod_srtp,
+ "srtcp using provisional stream (SSRC: 0x%08x)",
+ ntohl(hdr->ssrc));
+ } else {
+ /* no template stream, so we return an error */
+ return srtp_err_status_no_ctx;
+ }
+ }
+
+ /*
+ * Determine if MKI is being used and what session keys should be used
+ */
+ if (use_mki) {
+ session_keys = srtp_get_session_keys(
+ stream, (uint8_t *)hdr, (const unsigned int *)pkt_octet_len,
+ &mki_size);
+
+ if (session_keys == NULL)
+ return srtp_err_status_bad_mki;
+ } else {
+ session_keys = &stream->session_keys[0];
+ }
+
+ /* get tag length from stream context */
+ tag_len = srtp_auth_get_tag_length(session_keys->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 < (int)(octets_in_rtcp_header + tag_len + mki_size +
+ 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 (session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_128 ||
+ session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_256) {
+ return srtp_unprotect_rtcp_aead(ctx, stream, srtcp_hdr,
+ (unsigned int *)pkt_octet_len,
+ session_keys, mki_size);
+ }
+
+ 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 +
+ mki_size + 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 */
+ trailer_p = (uint32_t *)((char *)hdr + *pkt_octet_len -
+ (tag_len + mki_size + sizeof(srtcp_trailer_t)));
+ memcpy(&trailer, trailer_p, sizeof(trailer));
+
+ e_bit_in_packet =
+ (*((unsigned char *)trailer_p) & 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;
+
+ /*
+ * The location of the auth tag in the packet needs to know MKI
+ * could be present. The data needed to calculate the Auth tag
+ * must not include the MKI
+ */
+ auth_len = *pkt_octet_len - tag_len - mki_size;
+ auth_tag = (uint8_t *)hdr + auth_len + mki_size;
+
+ /*
+ * 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 (session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_128 ||
+ session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_192 ||
+ session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_256) {
+ 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 = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
+ srtp_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 = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
+ srtp_direction_decrypt);
+ }
+ if (status)
+ return srtp_err_status_cipher_fail;
+
+ /* initialize auth func context */
+ status = srtp_auth_start(session_keys->rtcp_auth);
+ if (status)
+ return status;
+
+ /* run auth func over packet, put result into tmp_tag */
+ status = srtp_auth_compute(session_keys->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 (srtp_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(session_keys->rtcp_auth);
+ if (prefix_len) {
+ status = srtp_cipher_output(session_keys->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(session_keys->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));
+
+ /* decrease the packet length by the length of the mki_size */
+ *pkt_octet_len -= mki_size;
+
+ /*
+ * 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;
+}
+
+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;
+#ifdef GCM
+ case srtp_profile_aead_aes_128_gcm:
+ srtp_crypto_policy_set_aes_gcm_128_16_auth(policy);
+ break;
+ case srtp_profile_aead_aes_256_gcm:
+ srtp_crypto_policy_set_aes_gcm_256_16_auth(policy);
+ break;
+#endif
+ /* 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;
+#ifdef GCM
+ case srtp_profile_aead_aes_128_gcm:
+ srtp_crypto_policy_set_aes_gcm_128_16_auth(policy);
+ break;
+ case srtp_profile_aead_aes_256_gcm:
+ srtp_crypto_policy_set_aes_gcm_256_16_auth(policy);
+ break;
+#endif
+ /* 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 SRTP_AES_128_KEY_LEN;
+ break;
+ case srtp_profile_aes128_cm_sha1_32:
+ return SRTP_AES_128_KEY_LEN;
+ break;
+ case srtp_profile_null_sha1_80:
+ return SRTP_AES_128_KEY_LEN;
+ break;
+ case srtp_profile_aead_aes_128_gcm:
+ return SRTP_AES_128_KEY_LEN;
+ break;
+ case srtp_profile_aead_aes_256_gcm:
+ return SRTP_AES_256_KEY_LEN;
+ 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 SRTP_SALT_LEN;
+ break;
+ case srtp_profile_aes128_cm_sha1_32:
+ return SRTP_SALT_LEN;
+ break;
+ case srtp_profile_null_sha1_80:
+ return SRTP_SALT_LEN;
+ break;
+ case srtp_profile_aead_aes_128_gcm:
+ return SRTP_AEAD_SALT_LEN;
+ break;
+ case srtp_profile_aead_aes_256_gcm:
+ return SRTP_AEAD_SALT_LEN;
+ break;
+ /* the following profiles are not (yet) supported */
+ case srtp_profile_null_sha1_32:
+ default:
+ return 0; /* indicate error by returning a zero */
+ }
+}
+
+srtp_err_status_t stream_get_protect_trailer_length(srtp_stream_ctx_t *stream,
+ uint32_t is_rtp,
+ uint32_t use_mki,
+ uint32_t mki_index,
+ uint32_t *length)
+{
+ srtp_session_keys_t *session_key;
+
+ *length = 0;
+
+ if (use_mki) {
+ if (mki_index >= stream->num_master_keys) {
+ return srtp_err_status_bad_mki;
+ }
+ session_key = &stream->session_keys[mki_index];
+
+ *length += session_key->mki_size;
+
+ } else {
+ session_key = &stream->session_keys[0];
+ }
+ if (is_rtp) {
+ *length += srtp_auth_get_tag_length(session_key->rtp_auth);
+ } else {
+ *length += srtp_auth_get_tag_length(session_key->rtcp_auth);
+ *length += sizeof(srtcp_trailer_t);
+ }
+
+ return srtp_err_status_ok;
+}
+
+srtp_err_status_t get_protect_trailer_length(srtp_t session,
+ uint32_t is_rtp,
+ uint32_t use_mki,
+ uint32_t mki_index,
+ uint32_t *length)
+{
+ srtp_stream_ctx_t *stream;
+
+ if (session == NULL) {
+ return srtp_err_status_bad_param;
+ }
+
+ if (session->stream_template == NULL && session->stream_list == NULL) {
+ return srtp_err_status_bad_param;
+ }
+
+ *length = 0;
+
+ stream = session->stream_template;
+
+ if (stream != NULL) {
+ stream_get_protect_trailer_length(stream, is_rtp, use_mki, mki_index,
+ length);
+ }
+
+ stream = session->stream_list;
+
+ while (stream != NULL) {
+ uint32_t temp_length;
+ if (stream_get_protect_trailer_length(stream, is_rtp, use_mki,
+ mki_index, &temp_length) ==
+ srtp_err_status_ok) {
+ if (temp_length > *length) {
+ *length = temp_length;
+ }
+ }
+ stream = stream->next;
+ }
+
+ return srtp_err_status_ok;
+}
+
+srtp_err_status_t srtp_get_protect_trailer_length(srtp_t session,
+ uint32_t use_mki,
+ uint32_t mki_index,
+ uint32_t *length)
+{
+ return get_protect_trailer_length(session, 1, use_mki, mki_index, length);
+}
+
+srtp_err_status_t srtp_get_protect_rtcp_trailer_length(srtp_t session,
+ uint32_t use_mki,
+ uint32_t mki_index,
+ uint32_t *length)
+{
+ return get_protect_trailer_length(session, 0, use_mki, mki_index, length);
+}
+
+/*
+ * SRTP debug interface
+ */
+srtp_err_status_t srtp_set_debug_module(const 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();
+}
+
+/*
+ * srtp_log_handler is a global variable holding a pointer to the
+ * log handler function; this function is called for any log
+ * output.
+ */
+
+static srtp_log_handler_func_t *srtp_log_handler = NULL;
+static void *srtp_log_handler_data = NULL;
+
+void srtp_err_handler(srtp_err_reporting_level_t level, const char *msg)
+{
+ if (srtp_log_handler) {
+ srtp_log_level_t log_level = srtp_log_level_error;
+ switch (level) {
+ case srtp_err_level_error:
+ log_level = srtp_log_level_error;
+ break;
+ case srtp_err_level_warning:
+ log_level = srtp_log_level_warning;
+ break;
+ case srtp_err_level_info:
+ log_level = srtp_log_level_info;
+ break;
+ case srtp_err_level_debug:
+ log_level = srtp_log_level_debug;
+ break;
+ }
+
+ srtp_log_handler(log_level, msg, srtp_log_handler_data);
+ }
+}
+
+srtp_err_status_t srtp_install_log_handler(srtp_log_handler_func_t func,
+ void *data)
+{
+ /*
+ * note that we accept NULL arguments intentionally - calling this
+ * function with a NULL arguments removes a log handler that's
+ * been previously installed
+ */
+
+ if (srtp_log_handler) {
+ srtp_install_err_report_handler(NULL);
+ }
+ srtp_log_handler = func;
+ srtp_log_handler_data = data;
+ if (srtp_log_handler) {
+ srtp_install_err_report_handler(srtp_err_handler);
+ }
+ return srtp_err_status_ok;
+}
+
+srtp_err_status_t srtp_set_stream_roc(srtp_t session,
+ uint32_t ssrc,
+ uint32_t roc)
+{
+ srtp_stream_t stream;
+
+ stream = srtp_get_stream(session, htonl(ssrc));
+ if (stream == NULL)
+ return srtp_err_status_bad_param;
+
+ stream->pending_roc = roc;
+
+ return srtp_err_status_ok;
+}
+
+srtp_err_status_t srtp_get_stream_roc(srtp_t session,
+ uint32_t ssrc,
+ uint32_t *roc)
+{
+ srtp_stream_t stream;
+
+ stream = srtp_get_stream(session, htonl(ssrc));
+ if (stream == NULL)
+ return srtp_err_status_bad_param;
+
+ *roc = srtp_rdbx_get_roc(&stream->rtp_rdbx);
+
+ return srtp_err_status_ok;
+}