5035 lines
160 KiB
C
5035 lines
160 KiB
C
/*
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* srtp.c
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*
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* the secure real-time transport protocol
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*
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* David A. McGrew
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* Cisco Systems, Inc.
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*/
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/*
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*
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* Copyright (c) 2001-2017, Cisco Systems, Inc.
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* All rights reserved.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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*
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* Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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*
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* Redistributions in binary form must reproduce the above
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* copyright notice, this list of conditions and the following
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* disclaimer in the documentation and/or other materials provided
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* with the distribution.
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*
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* Neither the name of the Cisco Systems, Inc. nor the names of its
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* contributors may be used to endorse or promote products derived
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* from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
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* "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
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* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
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* FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
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* COPYRIGHT HOLDERS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
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* INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
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* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
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* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
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* STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED
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* OF THE POSSIBILITY OF SUCH DAMAGE.
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*
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*/
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// Leave this as the top level import. Ensures the existence of defines
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#include "config.h"
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#include "srtp_priv.h"
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#include "stream_list_priv.h"
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#include "crypto_types.h"
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#include "err.h"
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#include "alloc.h" /* for srtp_crypto_alloc() */
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#ifdef GCM
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#include "aes_gcm.h" /* for AES GCM mode */
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#endif
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#ifdef OPENSSL_KDF
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#include <openssl/kdf.h>
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#include "aes_icm_ext.h"
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#endif
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#ifdef WOLFSSL
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#ifndef WOLFSSL_USER_SETTINGS
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#include <wolfssl/options.h>
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#endif
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#include <wolfssl/wolfcrypt/settings.h>
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#ifdef WOLFSSL_KDF
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#include <wolfssl/wolfcrypt/kdf.h>
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#endif
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#endif
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#include <limits.h>
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#ifdef HAVE_NETINET_IN_H
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#include <netinet/in.h>
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#elif defined(HAVE_WINSOCK2_H)
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#include <winsock2.h>
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#endif
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/* the debug module for srtp */
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srtp_debug_module_t mod_srtp = {
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false, /* debugging is off by default */
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"srtp" /* printable name for module */
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};
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static const size_t octets_in_rtp_header = 12;
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static const size_t octets_in_rtcp_header = 8;
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static const size_t octets_in_rtp_xtn_hdr = 4;
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static size_t srtp_get_rtp_hdr_len(const srtp_hdr_t *hdr)
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{
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return octets_in_rtp_header + 4 * hdr->cc;
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}
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/*
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* Returns the location of the header extention cast to a srtp_hdr_xtnd_t
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* struct. Will always return a value and assumes that the caller has already
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* verified that a header extension is present by checking the x bit of
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* srtp_hdr_t.
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*/
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static srtp_hdr_xtnd_t *srtp_get_rtp_xtn_hdr(const srtp_hdr_t *hdr,
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uint8_t *rtp)
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{
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return (srtp_hdr_xtnd_t *)(rtp + srtp_get_rtp_hdr_len(hdr));
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}
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/*
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* Returns the length of the extension header including the extension header
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* header so will return a minium of 4. Assumes the srtp_hdr_xtnd_t is a valid
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* pointer and that the caller has already verified that a header extension is
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* valid by checking the x bit of the RTP header.
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*/
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static size_t srtp_get_rtp_xtn_hdr_len(const srtp_hdr_t *hdr,
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const uint8_t *rtp)
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{
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const srtp_hdr_xtnd_t *xtn_hdr =
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(const srtp_hdr_xtnd_t *)(rtp + srtp_get_rtp_hdr_len(hdr));
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return (ntohs(xtn_hdr->length) + 1u) * 4u;
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}
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static srtp_err_status_t srtp_validate_rtp_header(const uint8_t *rtp,
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size_t pkt_octet_len)
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{
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const srtp_hdr_t *hdr = (const srtp_hdr_t *)rtp;
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size_t rtp_header_len;
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if (pkt_octet_len < octets_in_rtp_header) {
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return srtp_err_status_bad_param;
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}
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/* Check RTP header length */
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rtp_header_len = srtp_get_rtp_hdr_len(hdr);
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if (pkt_octet_len < rtp_header_len) {
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return srtp_err_status_bad_param;
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}
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/* Verifying profile length. */
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if (hdr->x == 1) {
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if (pkt_octet_len < rtp_header_len + octets_in_rtp_xtn_hdr) {
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return srtp_err_status_bad_param;
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}
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rtp_header_len += srtp_get_rtp_xtn_hdr_len(hdr, rtp);
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if (pkt_octet_len < rtp_header_len) {
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return srtp_err_status_bad_param;
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}
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}
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return srtp_err_status_ok;
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}
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const char *srtp_get_version_string(void)
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{
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/*
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* Simply return the autotools generated string
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*/
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return SRTP_VER_STRING;
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}
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unsigned int srtp_get_version(void)
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{
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unsigned int major = 0, minor = 0, micro = 0;
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unsigned int rv = 0;
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int parse_rv;
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/*
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* Parse the autotools generated version
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*/
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parse_rv = sscanf(SRTP_VERSION, "%u.%u.%u", &major, &minor, µ);
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if (parse_rv != 3) {
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/*
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* We're expected to parse all 3 version levels.
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* If not, then this must not be an official release.
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* Return all zeros on the version
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*/
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return (0);
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}
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/*
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* We allow 8 bits for the major and minor, while
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* allowing 16 bits for the micro. 16 bits for the micro
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* may be beneficial for a continuous delivery model
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* in the future.
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*/
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rv |= (major & 0xFF) << 24;
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rv |= (minor & 0xFF) << 16;
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rv |= micro & 0xFF;
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return rv;
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}
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static srtp_err_status_t srtp_stream_dealloc(
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srtp_stream_ctx_t *stream,
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const srtp_stream_ctx_t *stream_template)
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{
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srtp_err_status_t status;
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srtp_session_keys_t *session_keys = NULL;
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srtp_session_keys_t *template_session_keys = NULL;
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/*
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* we use a conservative deallocation strategy - if any deallocation
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* fails, then we report that fact without trying to deallocate
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* anything else
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*/
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if (stream->session_keys) {
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for (size_t i = 0; i < stream->num_master_keys; i++) {
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session_keys = &stream->session_keys[i];
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if (stream_template &&
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stream->num_master_keys == stream_template->num_master_keys) {
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template_session_keys = &stream_template->session_keys[i];
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} else {
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template_session_keys = NULL;
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}
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/*
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* deallocate cipher, if it is not the same as that in template
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*/
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if (template_session_keys &&
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session_keys->rtp_cipher == template_session_keys->rtp_cipher) {
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/* do nothing */
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} else if (session_keys->rtp_cipher) {
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status = srtp_cipher_dealloc(session_keys->rtp_cipher);
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if (status) {
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return status;
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}
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}
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/*
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* deallocate auth function, if it is not the same as that in
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* template
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*/
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if (template_session_keys &&
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session_keys->rtp_auth == template_session_keys->rtp_auth) {
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/* do nothing */
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} else if (session_keys->rtp_auth) {
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status = srtp_auth_dealloc(session_keys->rtp_auth);
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if (status) {
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return status;
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}
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}
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if (template_session_keys &&
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session_keys->rtp_xtn_hdr_cipher ==
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template_session_keys->rtp_xtn_hdr_cipher) {
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/* do nothing */
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} else if (session_keys->rtp_xtn_hdr_cipher) {
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status = srtp_cipher_dealloc(session_keys->rtp_xtn_hdr_cipher);
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if (status) {
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return status;
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}
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}
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/*
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* deallocate rtcp cipher, if it is not the same as that in
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* template
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*/
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if (template_session_keys &&
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session_keys->rtcp_cipher ==
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template_session_keys->rtcp_cipher) {
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/* do nothing */
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} else if (session_keys->rtcp_cipher) {
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status = srtp_cipher_dealloc(session_keys->rtcp_cipher);
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if (status) {
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return status;
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}
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}
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/*
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* deallocate rtcp auth function, if it is not the same as that in
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* template
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*/
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if (template_session_keys &&
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session_keys->rtcp_auth == template_session_keys->rtcp_auth) {
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/* do nothing */
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} else if (session_keys->rtcp_auth) {
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status = srtp_auth_dealloc(session_keys->rtcp_auth);
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if (status) {
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return status;
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}
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}
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/*
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* zeroize the salt value
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*/
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octet_string_set_to_zero(session_keys->salt, SRTP_AEAD_SALT_LEN);
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octet_string_set_to_zero(session_keys->c_salt, SRTP_AEAD_SALT_LEN);
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if (session_keys->mki_id) {
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octet_string_set_to_zero(session_keys->mki_id,
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stream->mki_size);
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srtp_crypto_free(session_keys->mki_id);
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session_keys->mki_id = NULL;
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}
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/*
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* deallocate key usage limit, if it is not the same as that in
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* template
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*/
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if (template_session_keys &&
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session_keys->limit == template_session_keys->limit) {
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/* do nothing */
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} else if (session_keys->limit) {
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srtp_crypto_free(session_keys->limit);
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}
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}
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srtp_crypto_free(stream->session_keys);
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}
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status = srtp_rdbx_dealloc(&stream->rtp_rdbx);
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if (status) {
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return status;
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}
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if (stream_template &&
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stream->enc_xtn_hdr == stream_template->enc_xtn_hdr) {
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/* do nothing */
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} else if (stream->enc_xtn_hdr) {
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srtp_crypto_free(stream->enc_xtn_hdr);
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}
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/* deallocate srtp stream context */
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srtp_crypto_free(stream);
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return srtp_err_status_ok;
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}
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/* try to insert stream in list or deallocate it */
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static srtp_err_status_t srtp_insert_or_dealloc_stream(srtp_stream_list_t list,
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srtp_stream_t stream,
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srtp_stream_t template)
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{
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srtp_err_status_t status = srtp_stream_list_insert(list, stream);
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/* on failure, ownership wasn't transferred and we need to deallocate */
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if (status) {
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srtp_stream_dealloc(stream, template);
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}
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return status;
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}
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struct remove_and_dealloc_streams_data {
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srtp_err_status_t status;
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srtp_stream_list_t list;
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srtp_stream_t template;
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};
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static bool remove_and_dealloc_streams_cb(srtp_stream_t stream, void *data)
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{
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struct remove_and_dealloc_streams_data *d =
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(struct remove_and_dealloc_streams_data *)data;
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srtp_stream_list_remove(d->list, stream);
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d->status = srtp_stream_dealloc(stream, d->template);
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if (d->status) {
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return false;
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}
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return true;
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}
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static srtp_err_status_t srtp_remove_and_dealloc_streams(
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srtp_stream_list_t list,
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srtp_stream_t template)
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{
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struct remove_and_dealloc_streams_data data = { srtp_err_status_ok, list,
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template };
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srtp_stream_list_for_each(list, remove_and_dealloc_streams_cb, &data);
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return data.status;
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}
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static srtp_err_status_t srtp_valid_policy(const srtp_policy_t *policy)
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{
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if (policy == NULL) {
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return srtp_err_status_bad_param;
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}
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if (policy->key == NULL) {
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if (policy->num_master_keys <= 0) {
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return srtp_err_status_bad_param;
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}
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if (policy->num_master_keys > SRTP_MAX_NUM_MASTER_KEYS) {
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return srtp_err_status_bad_param;
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}
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if (policy->use_mki) {
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if (policy->mki_size == 0 || policy->mki_size > SRTP_MAX_MKI_LEN) {
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return srtp_err_status_bad_param;
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}
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} else if (policy->mki_size != 0) {
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return srtp_err_status_bad_param;
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}
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for (size_t i = 0; i < policy->num_master_keys; i++) {
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if (policy->keys[i]->key == NULL) {
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return srtp_err_status_bad_param;
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}
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if (policy->use_mki && policy->keys[i]->mki_id == NULL) {
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return srtp_err_status_bad_param;
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}
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}
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} else {
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if (policy->use_mki || policy->mki_size != 0) {
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return srtp_err_status_bad_param;
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}
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}
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return srtp_err_status_ok;
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}
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static srtp_err_status_t srtp_stream_alloc(srtp_stream_ctx_t **str_ptr,
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const srtp_policy_t *p)
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{
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srtp_stream_ctx_t *str;
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srtp_err_status_t stat;
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size_t i = 0;
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srtp_session_keys_t *session_keys = NULL;
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stat = srtp_valid_policy(p);
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if (stat != srtp_err_status_ok) {
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return stat;
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}
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/*
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* This function allocates the stream context, rtp and rtcp ciphers
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* and auth functions, and key limit structure. If there is a
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* failure during allocation, we free all previously allocated
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* memory and return a failure code. The code could probably
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* be improved, but it works and should be clear.
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*/
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/* allocate srtp stream and set str_ptr */
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str = (srtp_stream_ctx_t *)srtp_crypto_alloc(sizeof(srtp_stream_ctx_t));
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if (str == NULL) {
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return srtp_err_status_alloc_fail;
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}
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*str_ptr = str;
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/*
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*To keep backwards API compatible if someone is using multiple master
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* keys then key should be set to NULL
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*/
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if (p->key != NULL) {
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str->num_master_keys = 1;
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} else {
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str->num_master_keys = p->num_master_keys;
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}
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str->session_keys = (srtp_session_keys_t *)srtp_crypto_alloc(
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sizeof(srtp_session_keys_t) * str->num_master_keys);
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if (str->session_keys == NULL) {
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srtp_stream_dealloc(str, NULL);
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return srtp_err_status_alloc_fail;
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}
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for (i = 0; i < str->num_master_keys; i++) {
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session_keys = &str->session_keys[i];
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/* allocate cipher */
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stat = srtp_crypto_kernel_alloc_cipher(
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p->rtp.cipher_type, &session_keys->rtp_cipher,
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p->rtp.cipher_key_len, p->rtp.auth_tag_len);
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if (stat) {
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srtp_stream_dealloc(str, NULL);
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return stat;
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}
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/* allocate auth function */
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stat = srtp_crypto_kernel_alloc_auth(
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p->rtp.auth_type, &session_keys->rtp_auth, p->rtp.auth_key_len,
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p->rtp.auth_tag_len);
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if (stat) {
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srtp_stream_dealloc(str, NULL);
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return stat;
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}
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|
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/*
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* ...and now the RTCP-specific initialization - first, allocate
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* the cipher
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*/
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stat = srtp_crypto_kernel_alloc_cipher(
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p->rtcp.cipher_type, &session_keys->rtcp_cipher,
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p->rtcp.cipher_key_len, p->rtcp.auth_tag_len);
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if (stat) {
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srtp_stream_dealloc(str, NULL);
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return stat;
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}
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/* allocate auth function */
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stat = srtp_crypto_kernel_alloc_auth(
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p->rtcp.auth_type, &session_keys->rtcp_auth, p->rtcp.auth_key_len,
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p->rtcp.auth_tag_len);
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if (stat) {
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srtp_stream_dealloc(str, NULL);
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return stat;
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}
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session_keys->mki_id = NULL;
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|
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/* allocate key limit structure */
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session_keys->limit = (srtp_key_limit_ctx_t *)srtp_crypto_alloc(
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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;
|
|
size_t enc_xtn_hdr_cipher_key_len;
|
|
|
|
str->enc_xtn_hdr = (uint8_t *)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
|
|
*/
|
|
|
|
static 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;
|
|
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)",
|
|
(unsigned int)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 (size_t 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;
|
|
|
|
if (stream_template->mki_size == 0) {
|
|
session_keys->mki_id = NULL;
|
|
} else {
|
|
session_keys->mki_id = srtp_crypto_alloc(stream_template->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,
|
|
stream_template->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;
|
|
}
|
|
}
|
|
|
|
str->use_mki = stream_template->use_mki;
|
|
str->mki_size = stream_template->mki_size;
|
|
|
|
/* 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;
|
|
|
|
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,
|
|
size_t key_len,
|
|
size_t 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,
|
|
size_t 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, &kdf->master_salt, NULL,
|
|
NULL, label, 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;
|
|
}
|
|
|
|
#elif defined(WOLFSSL) && defined(WOLFSSL_KDF)
|
|
#define MAX_SRTP_AESKEY_LEN AES_256_KEY_SIZE
|
|
#define MAX_SRTP_SALT_LEN WC_SRTP_MAX_SALT
|
|
|
|
/*
|
|
* 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];
|
|
int master_key_len;
|
|
uint8_t master_salt[MAX_SRTP_SALT_LEN];
|
|
} srtp_kdf_t;
|
|
|
|
static srtp_err_status_t srtp_kdf_init(srtp_kdf_t *kdf,
|
|
const uint8_t *key,
|
|
size_t key_len)
|
|
{
|
|
size_t salt_len;
|
|
|
|
memset(kdf, 0x0, sizeof(srtp_kdf_t));
|
|
|
|
switch (key_len) {
|
|
case SRTP_AES_ICM_256_KEY_LEN_WSALT:
|
|
kdf->master_key_len = AES_256_KEY_SIZE;
|
|
break;
|
|
case SRTP_AES_ICM_192_KEY_LEN_WSALT:
|
|
kdf->master_key_len = AES_192_KEY_SIZE;
|
|
break;
|
|
case SRTP_AES_ICM_128_KEY_LEN_WSALT:
|
|
kdf->master_key_len = AES_128_KEY_SIZE;
|
|
break;
|
|
default:
|
|
return srtp_err_status_bad_param;
|
|
break;
|
|
}
|
|
|
|
memcpy(kdf->master_key, key, kdf->master_key_len);
|
|
salt_len = key_len - kdf->master_key_len;
|
|
memcpy(kdf->master_salt, key + kdf->master_key_len, salt_len);
|
|
memset(kdf->master_salt + salt_len, 0, MAX_SRTP_SALT_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,
|
|
size_t length)
|
|
{
|
|
int err;
|
|
|
|
if (length == 0) {
|
|
return srtp_err_status_ok;
|
|
}
|
|
if (kdf->master_key_len == 0) {
|
|
return srtp_err_status_ok;
|
|
}
|
|
octet_string_set_to_zero(key, length);
|
|
|
|
PRIVATE_KEY_UNLOCK();
|
|
err = wc_SRTP_KDF_label(kdf->master_key, kdf->master_key_len,
|
|
kdf->master_salt, MAX_SRTP_SALT_LEN, -1, NULL,
|
|
label, key, length);
|
|
PRIVATE_KEY_LOCK();
|
|
if (err < 0) {
|
|
debug_print(mod_srtp, "wolfSSL SRTP KDF error: %d", err);
|
|
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);
|
|
kdf->master_key_len = 0;
|
|
octet_string_set_to_zero(kdf->master_salt, MAX_SRTP_SALT_LEN);
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
#else /* if OPENSSL_KDF || WOLFSSL_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,
|
|
size_t 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,
|
|
size_t 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, 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 || WOLFSSL_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 size_t base_key_length(const srtp_cipher_type_t *cipher,
|
|
size_t 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 size_t 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;
|
|
}
|
|
}
|
|
|
|
srtp_err_status_t srtp_get_session_keys(srtp_stream_ctx_t *stream,
|
|
size_t mki_index,
|
|
srtp_session_keys_t **session_keys)
|
|
{
|
|
if (stream->use_mki) {
|
|
if (mki_index >= stream->num_master_keys) {
|
|
return srtp_err_status_bad_mki;
|
|
}
|
|
*session_keys = &stream->session_keys[mki_index];
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
*session_keys = &stream->session_keys[0];
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
void srtp_inject_mki(uint8_t *mki_tag_location,
|
|
const srtp_session_keys_t *session_keys,
|
|
size_t mki_size)
|
|
{
|
|
if (mki_size > 0) {
|
|
// Write MKI into memory
|
|
memcpy(mki_tag_location, session_keys->mki_id, mki_size);
|
|
}
|
|
}
|
|
|
|
srtp_err_status_t srtp_stream_init_keys(srtp_session_keys_t *session_keys,
|
|
const srtp_master_key_t *master_key,
|
|
size_t mki_size)
|
|
{
|
|
srtp_err_status_t stat;
|
|
srtp_kdf_t kdf;
|
|
uint8_t tmp_key[MAX_SRTP_KEY_LEN];
|
|
size_t input_keylen, input_keylen_rtcp;
|
|
size_t kdf_keylen = 30, rtp_keylen, rtcp_keylen;
|
|
size_t rtp_base_key_len, rtp_salt_len;
|
|
size_t rtcp_base_key_len, rtcp_salt_len;
|
|
|
|
/* If RTP or RTCP have a key length > AES-128, assume matching kdf. */
|
|
/* TODO: kdf algorithm, master key length, and master salt length should
|
|
* be part of srtp_policy_t.
|
|
*/
|
|
|
|
/* initialize key limit to maximum value */
|
|
srtp_key_limit_set(session_keys->limit, 0xffffffffffffLL);
|
|
|
|
if (mki_size != 0) {
|
|
if (master_key->mki_id == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
session_keys->mki_id = srtp_crypto_alloc(mki_size);
|
|
|
|
if (session_keys->mki_id == NULL) {
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
memcpy(session_keys->mki_id, master_key->mki_id, mki_size);
|
|
} else {
|
|
session_keys->mki_id = NULL;
|
|
}
|
|
|
|
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;
|
|
|
|
/*
|
|
* We assume that the `key` buffer provided by the caller has a length
|
|
* equal to the greater of `rtp_keylen` and `rtcp_keylen`. Since we are
|
|
* about to read `input_keylen` bytes from it, we need to check that we will
|
|
* not overrun.
|
|
*/
|
|
if ((rtp_keylen < input_keylen) && (rtcp_keylen < input_keylen)) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
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: %zu", input_keylen);
|
|
debug_print(mod_srtp, "srtp key len: %zu", rtp_keylen);
|
|
debug_print(mod_srtp, "srtcp key len: %zu", rtcp_keylen);
|
|
debug_print(mod_srtp, "base key len: %zu", rtp_base_key_len);
|
|
debug_print(mod_srtp, "kdf key len: %zu", kdf_keylen);
|
|
debug_print(mod_srtp, "rtp salt len: %zu", 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, master_key->key, input_keylen);
|
|
|
|
/* initialize KDF state */
|
|
#if defined(OPENSSL) && defined(OPENSSL_KDF)
|
|
stat = srtp_kdf_init(&kdf, tmp_key, rtp_base_key_len, rtp_salt_len);
|
|
#else
|
|
stat = srtp_kdf_init(&kdf, 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 */
|
|
size_t rtp_xtn_hdr_keylen;
|
|
size_t rtp_xtn_hdr_base_key_len;
|
|
size_t 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, master_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, tmp_xtn_hdr_key,
|
|
rtp_xtn_hdr_base_key_len, rtp_xtn_hdr_salt_len);
|
|
#else
|
|
stat = srtp_kdf_init(xtn_hdr_kdf, 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: %zu", 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_all_master_keys(srtp_stream_ctx_t *srtp,
|
|
const srtp_policy_t *p)
|
|
{
|
|
srtp_err_status_t status = srtp_err_status_ok;
|
|
if (p->key != NULL) {
|
|
if (p->use_mki) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
srtp_master_key_t single_master_key;
|
|
srtp->num_master_keys = 1;
|
|
srtp->use_mki = false;
|
|
srtp->mki_size = 0;
|
|
single_master_key.key = p->key;
|
|
single_master_key.mki_id = NULL;
|
|
status = srtp_stream_init_keys(&srtp->session_keys[0],
|
|
&single_master_key, 0);
|
|
} else {
|
|
if (p->num_master_keys > SRTP_MAX_NUM_MASTER_KEYS) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
if (p->use_mki && p->mki_size == 0) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
srtp->num_master_keys = p->num_master_keys;
|
|
srtp->use_mki = p->use_mki;
|
|
srtp->mki_size = p->mki_size;
|
|
|
|
for (size_t i = 0; i < srtp->num_master_keys; i++) {
|
|
status = srtp_stream_init_keys(&srtp->session_keys[i], p->keys[i],
|
|
srtp->mki_size);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
}
|
|
|
|
return status;
|
|
}
|
|
|
|
static 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)",
|
|
(unsigned int)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 */
|
|
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);
|
|
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: ", (unsigned int)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 true if yes, otherwise false.
|
|
*/
|
|
static bool srtp_protect_extension_header(srtp_stream_ctx_t *stream, uint8_t id)
|
|
{
|
|
uint8_t *enc_xtn_hdr = stream->enc_xtn_hdr;
|
|
size_t count = stream->enc_xtn_hdr_count;
|
|
|
|
if (!enc_xtn_hdr || count <= 0) {
|
|
return false;
|
|
}
|
|
|
|
while (count > 0) {
|
|
if (*enc_xtn_hdr == id) {
|
|
return true;
|
|
}
|
|
|
|
enc_xtn_hdr++;
|
|
count--;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
/*
|
|
* 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. */
|
|
size_t keystream_pos;
|
|
uint8_t *xtn_hdr_data = ((uint8_t *)xtn_hdr) + octets_in_rtp_xtn_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;
|
|
size_t xlen = (*xtn_hdr_data & 0x0f) + 1;
|
|
size_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;
|
|
size_t xlen = *(xtn_hdr_data + 1);
|
|
size_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,
|
|
const srtp_hdr_t *hdr)
|
|
{
|
|
v128_t in;
|
|
v128_t salt;
|
|
|
|
uint32_t local_roc = (uint32_t)(*seq >> 16);
|
|
uint16_t local_seq = (uint16_t)*seq;
|
|
|
|
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);
|
|
}
|
|
|
|
static srtp_err_status_t srtp_get_session_keys_for_packet(
|
|
srtp_stream_ctx_t *stream,
|
|
const uint8_t *hdr,
|
|
size_t pkt_octet_len,
|
|
size_t tag_len,
|
|
srtp_session_keys_t **session_keys)
|
|
{
|
|
if (!stream->use_mki) {
|
|
*session_keys = &stream->session_keys[0];
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
size_t mki_start_location = pkt_octet_len;
|
|
|
|
if (tag_len > mki_start_location) {
|
|
return srtp_err_status_bad_mki;
|
|
}
|
|
|
|
mki_start_location -= tag_len;
|
|
|
|
if (stream->mki_size > mki_start_location) {
|
|
return srtp_err_status_bad_mki;
|
|
}
|
|
|
|
mki_start_location -= stream->mki_size;
|
|
|
|
for (size_t i = 0; i < stream->num_master_keys; i++) {
|
|
if (memcmp(hdr + mki_start_location, stream->session_keys[i].mki_id,
|
|
stream->mki_size) == 0) {
|
|
*session_keys = &stream->session_keys[i];
|
|
return srtp_err_status_ok;
|
|
}
|
|
}
|
|
|
|
return srtp_err_status_bad_mki;
|
|
}
|
|
|
|
static srtp_err_status_t srtp_get_session_keys_for_rtp_packet(
|
|
srtp_stream_ctx_t *stream,
|
|
const uint8_t *hdr,
|
|
size_t pkt_octet_len,
|
|
srtp_session_keys_t **session_keys)
|
|
{
|
|
size_t tag_len = 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);
|
|
}
|
|
|
|
return srtp_get_session_keys_for_packet(stream, hdr, pkt_octet_len, tag_len,
|
|
session_keys);
|
|
}
|
|
|
|
static srtp_err_status_t srtp_get_session_keys_for_rtcp_packet(
|
|
srtp_stream_ctx_t *stream,
|
|
const uint8_t *hdr,
|
|
size_t pkt_octet_len,
|
|
srtp_session_keys_t **session_keys)
|
|
{
|
|
size_t tag_len = 0;
|
|
|
|
// Determine the authentication tag size
|
|
if (stream->session_keys[0].rtcp_cipher->algorithm == SRTP_AES_GCM_128 ||
|
|
stream->session_keys[0].rtcp_cipher->algorithm == SRTP_AES_GCM_256) {
|
|
tag_len = 0;
|
|
} else {
|
|
tag_len = srtp_auth_get_tag_length(stream->session_keys[0].rtcp_auth);
|
|
}
|
|
|
|
return srtp_get_session_keys_for_packet(stream, hdr, pkt_octet_len, tag_len,
|
|
session_keys);
|
|
}
|
|
|
|
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,
|
|
ssize_t *delta)
|
|
{
|
|
*est = (srtp_xtd_seq_num_t)(((uint64_t)roc) << 16) | seq;
|
|
*delta = *est - rdbx->index;
|
|
|
|
if (*est > rdbx->index) {
|
|
if (*est - rdbx->index > seq_num_median) {
|
|
*delta = 0;
|
|
return srtp_err_status_pkt_idx_adv;
|
|
}
|
|
} else if (*est < rdbx->index) {
|
|
if (rdbx->index - *est > seq_num_median) {
|
|
*delta = 0;
|
|
return srtp_err_status_pkt_idx_old;
|
|
}
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
static srtp_err_status_t srtp_get_est_pkt_index(const srtp_hdr_t *hdr,
|
|
srtp_stream_ctx_t *stream,
|
|
srtp_xtd_seq_num_t *est,
|
|
ssize_t *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));
|
|
}
|
|
|
|
debug_print(mod_srtp, "estimated u_packet index: %016" PRIx64, *est);
|
|
|
|
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,
|
|
const uint8_t *rtp,
|
|
size_t rtp_len,
|
|
uint8_t *srtp,
|
|
size_t *srtp_len,
|
|
srtp_session_keys_t *session_keys)
|
|
{
|
|
const srtp_hdr_t *hdr = (const srtp_hdr_t *)rtp;
|
|
size_t enc_start; /* offset to start of encrypted portion */
|
|
size_t enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
srtp_xtd_seq_num_t est; /* estimated xtd_seq_num_t of *hdr */
|
|
ssize_t delta; /* delta of local pkt idx and that in hdr */
|
|
srtp_err_status_t status;
|
|
size_t tag_len;
|
|
v128_t iv;
|
|
size_t aad_len;
|
|
|
|
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 = srtp_get_rtp_hdr_len(hdr);
|
|
if (hdr->x == 1) {
|
|
enc_start += srtp_get_rtp_xtn_hdr_len(hdr, rtp);
|
|
}
|
|
|
|
/* note: the passed size is without the auth tag */
|
|
if (enc_start > rtp_len) {
|
|
return srtp_err_status_parse_err;
|
|
}
|
|
enc_octet_len = rtp_len - enc_start;
|
|
|
|
/* check output length */
|
|
if (*srtp_len < rtp_len + tag_len + stream->mki_size) {
|
|
return srtp_err_status_buffer_small;
|
|
}
|
|
|
|
/* if not-inplace then need to copy full rtp header */
|
|
if (rtp != srtp) {
|
|
memcpy(srtp, rtp, enc_start);
|
|
}
|
|
|
|
/*
|
|
* 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) {
|
|
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);
|
|
}
|
|
|
|
debug_print(mod_srtp, "estimated packet index: %016" PRIx64, est);
|
|
|
|
/*
|
|
* AEAD uses a new IV formation method
|
|
*/
|
|
srtp_calc_aead_iv(session_keys, &iv, &est, hdr);
|
|
/* shift est, put into network byte order */
|
|
est = be64_to_cpu(est << 16);
|
|
|
|
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 (hdr->x == 1 && session_keys->rtp_xtn_hdr_cipher) {
|
|
/*
|
|
* extensions header encryption RFC 6904
|
|
*/
|
|
status = srtp_process_header_encryption(
|
|
stream, srtp_get_rtp_xtn_hdr(hdr, srtp), session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Set the AAD over the RTP header
|
|
*/
|
|
aad_len = enc_start;
|
|
status = srtp_cipher_set_aad(session_keys->rtp_cipher, srtp, aad_len);
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
|
|
/* Encrypt the payload */
|
|
size_t outlen = *srtp_len - enc_start;
|
|
status = srtp_cipher_encrypt(session_keys->rtp_cipher, rtp + enc_start,
|
|
enc_octet_len, srtp + enc_start, &outlen);
|
|
enc_octet_len = outlen;
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
if (stream->use_mki) {
|
|
srtp_inject_mki(srtp + enc_start + enc_octet_len, session_keys,
|
|
stream->mki_size);
|
|
}
|
|
|
|
*srtp_len = enc_start + enc_octet_len;
|
|
|
|
/* increase the packet length by the length of the mki_size */
|
|
*srtp_len += stream->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,
|
|
ssize_t delta,
|
|
srtp_xtd_seq_num_t est,
|
|
const uint8_t *srtp,
|
|
size_t srtp_len,
|
|
uint8_t *rtp,
|
|
size_t *rtp_len,
|
|
srtp_session_keys_t *session_keys,
|
|
bool advance_packet_index)
|
|
{
|
|
const srtp_hdr_t *hdr = (const srtp_hdr_t *)srtp;
|
|
size_t enc_start; /* offset to start of encrypted portion */
|
|
size_t enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
v128_t iv;
|
|
srtp_err_status_t status;
|
|
size_t tag_len;
|
|
size_t aad_len;
|
|
|
|
debug_print0(mod_srtp, "function srtp_unprotect_aead");
|
|
|
|
debug_print(mod_srtp, "estimated u_packet index: %016" PRIx64, est);
|
|
|
|
/* 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;
|
|
iv.v64[1] = be64_to_cpu(est << 16);
|
|
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;
|
|
}
|
|
|
|
enc_start = srtp_get_rtp_hdr_len(hdr);
|
|
if (hdr->x == 1) {
|
|
enc_start += srtp_get_rtp_xtn_hdr_len(hdr, srtp);
|
|
}
|
|
|
|
if (enc_start > srtp_len - tag_len - stream->mki_size) {
|
|
return srtp_err_status_parse_err;
|
|
}
|
|
|
|
/*
|
|
* We pass the tag down to the cipher when doing GCM mode
|
|
*/
|
|
enc_octet_len = srtp_len - enc_start - stream->mki_size;
|
|
|
|
/*
|
|
* Sanity check the encrypted payload length against
|
|
* the tag size. It must always be at least as large
|
|
* as the tag length.
|
|
*/
|
|
if (enc_octet_len < tag_len) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/* check output length */
|
|
if (*rtp_len < srtp_len - stream->mki_size - tag_len) {
|
|
return srtp_err_status_buffer_small;
|
|
}
|
|
|
|
/* if not-inplace then need to copy full rtp header */
|
|
if (srtp != rtp) {
|
|
memcpy(rtp, srtp, enc_start);
|
|
}
|
|
|
|
/*
|
|
* 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 = enc_start;
|
|
status = srtp_cipher_set_aad(session_keys->rtp_cipher, srtp, 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, srtp + enc_start,
|
|
enc_octet_len, rtp + enc_start, &enc_octet_len);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
if (hdr->x == 1 && session_keys->rtp_xtn_hdr_cipher) {
|
|
/*
|
|
* extensions header encryption RFC 6904
|
|
*/
|
|
status = srtp_process_header_encryption(
|
|
stream, srtp_get_rtp_xtn_hdr(hdr, rtp), 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 list */
|
|
status = srtp_insert_or_dealloc_stream(ctx->stream_list, new_stream,
|
|
ctx->stream_template);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* 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) {
|
|
uint32_t roc_to_set = (uint32_t)(est >> 16);
|
|
uint16_t seq_to_set = (uint16_t)(est & 0xFFFF);
|
|
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);
|
|
}
|
|
|
|
*rtp_len = enc_start + enc_octet_len;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_protect(srtp_t ctx,
|
|
const uint8_t *rtp,
|
|
size_t rtp_len,
|
|
uint8_t *srtp,
|
|
size_t *srtp_len,
|
|
size_t mki_index)
|
|
{
|
|
const srtp_hdr_t *hdr = (const srtp_hdr_t *)rtp;
|
|
size_t enc_start; /* offset to start of encrypted portion */
|
|
uint8_t *auth_start; /* pointer to start of auth. portion */
|
|
size_t enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
srtp_xtd_seq_num_t est; /* estimated xtd_seq_num_t of *hdr */
|
|
ssize_t 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;
|
|
size_t tag_len;
|
|
srtp_stream_ctx_t *stream;
|
|
size_t prefix_len;
|
|
srtp_session_keys_t *session_keys = NULL;
|
|
|
|
debug_print0(mod_srtp, "function srtp_protect");
|
|
|
|
/* Verify RTP header */
|
|
status = srtp_validate_rtp_header(rtp, rtp_len);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* check the packet length - it must at least contain a full header */
|
|
if (rtp_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 list */
|
|
status = srtp_insert_or_dealloc_stream(ctx->stream_list, new_stream,
|
|
ctx->stream_template);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* 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);
|
|
}
|
|
}
|
|
|
|
status = srtp_get_session_keys(stream, mki_index, &session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/*
|
|
* 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, rtp_len, srtp, srtp_len,
|
|
session_keys);
|
|
}
|
|
|
|
/*
|
|
* 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
|
|
*/
|
|
enc_start = srtp_get_rtp_hdr_len(hdr);
|
|
if (hdr->x == 1) {
|
|
enc_start += srtp_get_rtp_xtn_hdr_len(hdr, rtp);
|
|
}
|
|
|
|
if (enc_start > rtp_len) {
|
|
return srtp_err_status_parse_err;
|
|
}
|
|
enc_octet_len = rtp_len - enc_start;
|
|
|
|
/* check output length */
|
|
if (*srtp_len < rtp_len + stream->mki_size + tag_len) {
|
|
return srtp_err_status_buffer_small;
|
|
}
|
|
|
|
/* if not-inplace then need to copy full rtp header */
|
|
if (rtp != srtp) {
|
|
memcpy(srtp, rtp, enc_start);
|
|
}
|
|
|
|
if (stream->use_mki) {
|
|
srtp_inject_mki(srtp + rtp_len, session_keys, stream->mki_size);
|
|
}
|
|
|
|
/*
|
|
* 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 = srtp;
|
|
auth_tag = srtp + rtp_len + stream->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) {
|
|
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);
|
|
}
|
|
|
|
debug_print(mod_srtp, "estimated packet index: %016" PRIx64, est);
|
|
|
|
/*
|
|
* 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;
|
|
iv.v64[1] = be64_to_cpu(est << 16);
|
|
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 */
|
|
iv.v64[0] = 0;
|
|
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 */
|
|
est = be64_to_cpu(est << 16);
|
|
|
|
/*
|
|
* 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 (hdr->x == 1 && session_keys->rtp_xtn_hdr_cipher) {
|
|
/*
|
|
* extensions header encryption RFC 6904
|
|
*/
|
|
status = srtp_process_header_encryption(
|
|
stream, srtp_get_rtp_xtn_hdr(hdr, srtp), session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
|
|
/* if we're encrypting, exor keystream into the message */
|
|
if (stream->rtp_services & sec_serv_conf) {
|
|
status = srtp_cipher_encrypt(session_keys->rtp_cipher, rtp + enc_start,
|
|
enc_octet_len, srtp + enc_start,
|
|
&enc_octet_len);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
} else if (rtp != srtp) {
|
|
/* if no encryption and not-inplace then need to copy rest of packet */
|
|
memcpy(srtp + enc_start, rtp + enc_start, enc_octet_len);
|
|
}
|
|
|
|
/*
|
|
* 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, auth_start, rtp_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 status;
|
|
}
|
|
}
|
|
|
|
*srtp_len = enc_start + enc_octet_len;
|
|
|
|
/* increase the packet length by the length of the auth tag */
|
|
*srtp_len += tag_len;
|
|
|
|
/* increate the packet length by the mki size if used */
|
|
*srtp_len += stream->mki_size;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_unprotect(srtp_t ctx,
|
|
const uint8_t *srtp,
|
|
size_t srtp_len,
|
|
uint8_t *rtp,
|
|
size_t *rtp_len)
|
|
{
|
|
const srtp_hdr_t *hdr = (const srtp_hdr_t *)srtp;
|
|
size_t enc_start; /* pointer to start of encrypted portion */
|
|
const uint8_t *auth_start; /* pointer to start of auth. portion */
|
|
size_t enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
const uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
srtp_xtd_seq_num_t est; /* estimated xtd_seq_num_t of *hdr */
|
|
ssize_t 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];
|
|
size_t tag_len, prefix_len;
|
|
srtp_session_keys_t *session_keys = NULL;
|
|
bool advance_packet_index = false;
|
|
uint32_t roc_to_set = 0;
|
|
uint16_t seq_to_set = 0;
|
|
|
|
debug_print0(mod_srtp, "function srtp_unprotect");
|
|
|
|
/* Verify RTP header */
|
|
status = srtp_validate_rtp_header(srtp, srtp_len);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* check the packet length - it must at least contain a full header */
|
|
if (srtp_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)",
|
|
(unsigned int)ntohl(hdr->ssrc));
|
|
|
|
/*
|
|
* set estimated packet index to sequence number from header,
|
|
* and set delta equal to the same value
|
|
*/
|
|
est = (srtp_xtd_seq_num_t)ntohs(hdr->seq);
|
|
delta = (int)est;
|
|
} 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 = true;
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
|
|
debug_print(mod_srtp, "estimated u_packet index: %016" PRIx64, est);
|
|
|
|
/* Determine if MKI is being used and what session keys should be used */
|
|
status = srtp_get_session_keys_for_rtp_packet(stream, srtp, srtp_len,
|
|
&session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/*
|
|
* 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, srtp_len, rtp,
|
|
rtp_len, session_keys, advance_packet_index);
|
|
}
|
|
|
|
/* 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 */
|
|
iv.v64[1] = be64_to_cpu(est << 16);
|
|
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 packet index */
|
|
iv.v64[0] = 0;
|
|
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 */
|
|
est = be64_to_cpu(est << 16);
|
|
|
|
enc_start = srtp_get_rtp_hdr_len(hdr);
|
|
if (hdr->x == 1) {
|
|
enc_start += srtp_get_rtp_xtn_hdr_len(hdr, srtp);
|
|
}
|
|
|
|
if (enc_start > srtp_len - tag_len - stream->mki_size) {
|
|
return srtp_err_status_parse_err;
|
|
}
|
|
enc_octet_len = srtp_len - enc_start - stream->mki_size - tag_len;
|
|
|
|
/* check output length */
|
|
if (*rtp_len < srtp_len - stream->mki_size - tag_len) {
|
|
return srtp_err_status_buffer_small;
|
|
}
|
|
|
|
/* if not-inplace then need to copy full rtp header */
|
|
if (srtp != rtp) {
|
|
memcpy(rtp, srtp, enc_start);
|
|
}
|
|
|
|
/*
|
|
* 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 = srtp;
|
|
auth_tag = srtp + srtp_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, auth_start,
|
|
srtp_len - tag_len - stream->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_equal(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 (hdr->x == 1 && session_keys->rtp_xtn_hdr_cipher) {
|
|
/* extensions header encryption RFC 6904 */
|
|
status = srtp_process_header_encryption(
|
|
stream, srtp_get_rtp_xtn_hdr(hdr, rtp), session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
|
|
/* if we're decrypting, add keystream into ciphertext */
|
|
if (stream->rtp_services & sec_serv_conf) {
|
|
status =
|
|
srtp_cipher_decrypt(session_keys->rtp_cipher, srtp + enc_start,
|
|
enc_octet_len, rtp + enc_start, &enc_octet_len);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
} else if (rtp != srtp) {
|
|
/* if no encryption and not-inplace then need to copy rest of packet */
|
|
memcpy(rtp + enc_start, srtp + enc_start, enc_octet_len);
|
|
}
|
|
|
|
/*
|
|
* 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 list */
|
|
status = srtp_insert_or_dealloc_stream(ctx->stream_list, new_stream,
|
|
ctx->stream_template);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* 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);
|
|
}
|
|
|
|
*rtp_len = enc_start + enc_octet_len;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_init(void)
|
|
{
|
|
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(void)
|
|
{
|
|
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_stream_ctx_t *srtp_get_stream(srtp_t srtp, uint32_t ssrc)
|
|
{
|
|
return srtp_stream_list_get(srtp->stream_list, ssrc);
|
|
}
|
|
|
|
srtp_err_status_t srtp_dealloc(srtp_t session)
|
|
{
|
|
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
|
|
*/
|
|
|
|
/* deallocate streams */
|
|
status = srtp_remove_and_dealloc_streams(session->stream_list,
|
|
session->stream_template);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* 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 stream list */
|
|
status = srtp_stream_list_dealloc(session->stream_list);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* deallocate session context */
|
|
srtp_crypto_free(session);
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_stream_add(srtp_t session, const srtp_policy_t *policy)
|
|
{
|
|
srtp_err_status_t status;
|
|
srtp_stream_t tmp;
|
|
|
|
/* sanity check arguments */
|
|
if (session == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
status = srtp_valid_policy(policy);
|
|
if (status != srtp_err_status_ok) {
|
|
return status;
|
|
}
|
|
|
|
/* 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):
|
|
status = srtp_insert_or_dealloc_stream(session->stream_list, tmp,
|
|
session->stream_template);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
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;
|
|
|
|
/* sanity check arguments */
|
|
if (session == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
if (policy) {
|
|
stat = srtp_valid_policy(policy);
|
|
if (stat != srtp_err_status_ok) {
|
|
return stat;
|
|
}
|
|
}
|
|
|
|
/* 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;
|
|
|
|
ctx->stream_template = NULL;
|
|
ctx->stream_list = NULL;
|
|
ctx->user_data = NULL;
|
|
|
|
/* allocate stream list */
|
|
stat = srtp_stream_list_alloc(&ctx->stream_list);
|
|
if (stat) {
|
|
/* clean up everything */
|
|
srtp_dealloc(*session);
|
|
*session = NULL;
|
|
return stat;
|
|
}
|
|
|
|
/*
|
|
* loop over elements in the policy list, allocating and
|
|
* initializing a stream for each element
|
|
*/
|
|
while (policy != NULL) {
|
|
stat = srtp_stream_add(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_stream_remove(srtp_t session, uint32_t ssrc)
|
|
{
|
|
srtp_stream_ctx_t *stream;
|
|
srtp_err_status_t status;
|
|
|
|
/* sanity check arguments */
|
|
if (session == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
/* find and remove stream from the list */
|
|
stream = srtp_stream_list_get(session->stream_list, htonl(ssrc));
|
|
if (stream == NULL) {
|
|
return srtp_err_status_no_ctx;
|
|
}
|
|
|
|
srtp_stream_list_remove(session->stream_list, stream);
|
|
|
|
/* 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;
|
|
|
|
/* sanity check arguments */
|
|
if (session == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
stat = srtp_valid_policy(policy);
|
|
if (stat != srtp_err_status_ok) {
|
|
return stat;
|
|
}
|
|
|
|
while (policy != NULL) {
|
|
stat = srtp_stream_update(session, policy);
|
|
if (stat) {
|
|
return stat;
|
|
}
|
|
|
|
/* set policy to next item in list */
|
|
policy = policy->next;
|
|
}
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
struct update_template_stream_data {
|
|
srtp_err_status_t status;
|
|
srtp_t session;
|
|
srtp_stream_t new_stream_template;
|
|
srtp_stream_list_t new_stream_list;
|
|
};
|
|
|
|
static bool update_template_stream_cb(srtp_stream_t stream, void *raw_data)
|
|
{
|
|
struct update_template_stream_data *data =
|
|
(struct update_template_stream_data *)raw_data;
|
|
srtp_t session = data->session;
|
|
uint32_t ssrc = stream->ssrc;
|
|
srtp_xtd_seq_num_t old_index;
|
|
srtp_rdb_t old_rtcp_rdb;
|
|
|
|
/* old / non-template streams are copied unchanged */
|
|
if (stream->session_keys[0].rtp_auth !=
|
|
session->stream_template->session_keys[0].rtp_auth) {
|
|
srtp_stream_list_remove(session->stream_list, stream);
|
|
data->status = srtp_insert_or_dealloc_stream(
|
|
data->new_stream_list, stream, session->stream_template);
|
|
if (data->status) {
|
|
return false;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
/* save old extended seq */
|
|
old_index = stream->rtp_rdbx.index;
|
|
old_rtcp_rdb = stream->rtcp_rdb;
|
|
|
|
/* remove stream */
|
|
data->status = srtp_stream_remove(session, ntohl(ssrc));
|
|
if (data->status) {
|
|
return false;
|
|
}
|
|
|
|
/* allocate and initialize a new stream */
|
|
data->status = srtp_stream_clone(data->new_stream_template, ssrc, &stream);
|
|
if (data->status) {
|
|
return false;
|
|
}
|
|
|
|
/* add new stream to the head of the new_stream_list */
|
|
data->status = srtp_insert_or_dealloc_stream(data->new_stream_list, stream,
|
|
data->new_stream_template);
|
|
if (data->status) {
|
|
return false;
|
|
}
|
|
|
|
/* restore old extended seq */
|
|
stream->rtp_rdbx.index = old_index;
|
|
stream->rtcp_rdb = old_rtcp_rdb;
|
|
|
|
return true;
|
|
}
|
|
|
|
static srtp_err_status_t is_update_policy_compatable(
|
|
srtp_stream_t stream,
|
|
const srtp_policy_t *policy)
|
|
{
|
|
if (stream->use_mki != policy->use_mki) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
if (stream->use_mki && stream->mki_size != policy->mki_size) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
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_list_t new_stream_list;
|
|
|
|
status = srtp_valid_policy(policy);
|
|
if (status != srtp_err_status_ok) {
|
|
return status;
|
|
}
|
|
|
|
if (session->stream_template == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
status = is_update_policy_compatable(session->stream_template, policy);
|
|
if (status != srtp_err_status_ok) {
|
|
return status;
|
|
}
|
|
|
|
/* 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;
|
|
}
|
|
|
|
/* allocate new stream list */
|
|
status = srtp_stream_list_alloc(&new_stream_list);
|
|
if (status) {
|
|
srtp_crypto_free(new_stream_template);
|
|
return status;
|
|
}
|
|
|
|
/* process streams */
|
|
struct update_template_stream_data data = { srtp_err_status_ok, session,
|
|
new_stream_template,
|
|
new_stream_list };
|
|
srtp_stream_list_for_each(session->stream_list, update_template_stream_cb,
|
|
&data);
|
|
if (data.status) {
|
|
/* free new allocations */
|
|
srtp_remove_and_dealloc_streams(new_stream_list, new_stream_template);
|
|
srtp_stream_list_dealloc(new_stream_list);
|
|
srtp_stream_dealloc(new_stream_template, NULL);
|
|
return data.status;
|
|
}
|
|
|
|
/* dealloc old list / template */
|
|
srtp_remove_and_dealloc_streams(session->stream_list,
|
|
session->stream_template);
|
|
srtp_stream_list_dealloc(session->stream_list);
|
|
srtp_stream_dealloc(session->stream_template, NULL);
|
|
|
|
/* set new list / template */
|
|
session->stream_template = new_stream_template;
|
|
session->stream_list = new_stream_list;
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
static srtp_err_status_t stream_update(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;
|
|
}
|
|
|
|
status = is_update_policy_compatable(stream, policy);
|
|
if (status != srtp_err_status_ok) {
|
|
return status;
|
|
}
|
|
|
|
/* save old extendard seq */
|
|
old_index = stream->rtp_rdbx.index;
|
|
old_rtcp_rdb = stream->rtcp_rdb;
|
|
|
|
status = srtp_stream_remove(session, policy->ssrc.value);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
status = srtp_stream_add(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_stream_update(srtp_t session,
|
|
const srtp_policy_t *policy)
|
|
{
|
|
srtp_err_status_t status;
|
|
|
|
/* sanity check arguments */
|
|
if (session == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
status = srtp_valid_policy(policy);
|
|
if (status != srtp_err_status_ok) {
|
|
return status;
|
|
}
|
|
|
|
switch (policy->ssrc.type) {
|
|
case (ssrc_any_outbound):
|
|
case (ssrc_any_inbound):
|
|
status = update_template_streams(session, policy);
|
|
break;
|
|
case (ssrc_specific):
|
|
status = stream_update(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 =
|
|
SRTP_AES_ICM_128_KEY_LEN_WSALT; /* 128 bit key, 112 bit salt */
|
|
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 =
|
|
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_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 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,
|
|
const 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_stream_ctx_t *stream,
|
|
const uint8_t *rtcp,
|
|
size_t rtcp_len,
|
|
uint8_t *srtcp,
|
|
size_t *srtcp_len,
|
|
srtp_session_keys_t *session_keys)
|
|
{
|
|
const srtcp_hdr_t *hdr = (const srtcp_hdr_t *)rtcp;
|
|
size_t enc_start; /* pointer to start of encrypted portion */
|
|
uint8_t *trailer_p; /* pointer to start of trailer */
|
|
uint32_t trailer; /* trailer value */
|
|
size_t enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
srtp_err_status_t status;
|
|
size_t tag_len;
|
|
uint32_t seq_num;
|
|
v128_t iv;
|
|
|
|
/* 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 = octets_in_rtcp_header;
|
|
enc_octet_len = rtcp_len - enc_start;
|
|
|
|
/* check output length */
|
|
if (*srtcp_len <
|
|
rtcp_len + sizeof(srtcp_trailer_t) + stream->mki_size + tag_len) {
|
|
return srtp_err_status_buffer_small;
|
|
}
|
|
|
|
/* if not-inplace then need to copy full rtcp header */
|
|
if (rtcp != srtcp) {
|
|
memcpy(srtcp, rtcp, enc_start);
|
|
}
|
|
|
|
/* NOTE: hdr->length is not usable - it refers to only the first
|
|
* RTCP report in the compound packet!
|
|
*/
|
|
trailer_p = srtcp + enc_start + enc_octet_len + tag_len;
|
|
|
|
if (stream->rtcp_services & sec_serv_conf) {
|
|
trailer = htonl(SRTCP_E_BIT); /* set encrypt bit */
|
|
} else {
|
|
/* 0 is network-order independent */
|
|
trailer = 0x00000000; /* set encrypt bit */
|
|
}
|
|
|
|
if (stream->use_mki) {
|
|
srtp_inject_mki(srtcp + rtcp_len + tag_len + sizeof(srtcp_trailer_t),
|
|
session_keys, stream->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", (unsigned int)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 (stream->rtcp_services & sec_serv_conf) {
|
|
/*
|
|
* 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, rtcp,
|
|
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, rtcp, rtcp_len);
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
}
|
|
/*
|
|
* Process the sequence# as AAD
|
|
*/
|
|
status = srtp_cipher_set_aad(session_keys->rtcp_cipher, (uint8_t *)&trailer,
|
|
sizeof(trailer));
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
|
|
/* if we're encrypting, exor keystream into the message */
|
|
if (stream->rtcp_services & sec_serv_conf) {
|
|
size_t out_len = *srtcp_len - enc_start;
|
|
status =
|
|
srtp_cipher_encrypt(session_keys->rtcp_cipher, rtcp + enc_start,
|
|
enc_octet_len, srtcp + enc_start, &out_len);
|
|
enc_octet_len = out_len;
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
} else {
|
|
/* if no encryption and not-inplace then need to copy rest of packet */
|
|
if (rtcp != srtcp) {
|
|
memcpy(srtcp + enc_start, rtcp + enc_start, enc_octet_len);
|
|
}
|
|
|
|
/*
|
|
* Even though we're not encrypting the payload, we need
|
|
* to run the cipher to get the auth tag.
|
|
*/
|
|
uint8_t *auth_tag = srtcp + enc_start + enc_octet_len;
|
|
size_t out_len = *srtcp_len - enc_start - enc_octet_len;
|
|
status = srtp_cipher_encrypt(session_keys->rtcp_cipher, NULL, 0,
|
|
auth_tag, &out_len);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
enc_octet_len += out_len;
|
|
}
|
|
|
|
*srtcp_len = octets_in_rtcp_header + enc_octet_len;
|
|
|
|
/* increase the packet length by the length of the seq_num*/
|
|
*srtcp_len += sizeof(srtcp_trailer_t);
|
|
|
|
/* increase the packet by the mki_size */
|
|
*srtcp_len += stream->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,
|
|
const uint8_t *srtcp,
|
|
size_t srtcp_len,
|
|
uint8_t *rtcp,
|
|
size_t *rtcp_len,
|
|
srtp_session_keys_t *session_keys)
|
|
{
|
|
const srtcp_hdr_t *hdr = (const srtcp_hdr_t *)srtcp;
|
|
size_t enc_start; /* pointer to start of encrypted portion */
|
|
const uint8_t *trailer_p; /* pointer to start of trailer */
|
|
uint32_t trailer; /* trailer value */
|
|
size_t enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
const uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
srtp_err_status_t status;
|
|
size_t tag_len;
|
|
size_t tmp_len;
|
|
uint32_t seq_num;
|
|
v128_t iv;
|
|
|
|
/* get tag length from stream context */
|
|
tag_len = srtp_auth_get_tag_length(session_keys->rtcp_auth);
|
|
|
|
enc_start = octets_in_rtcp_header;
|
|
|
|
/*
|
|
* 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 = srtcp + srtcp_len - sizeof(srtcp_trailer_t) - stream->mki_size;
|
|
memcpy(&trailer, trailer_p, sizeof(trailer));
|
|
|
|
/*
|
|
* We pass the tag down to the cipher when doing GCM mode
|
|
*/
|
|
enc_octet_len = srtcp_len - (octets_in_rtcp_header +
|
|
sizeof(srtcp_trailer_t) + stream->mki_size);
|
|
auth_tag = srtcp + (srtcp_len - tag_len - stream->mki_size -
|
|
sizeof(srtcp_trailer_t));
|
|
|
|
/*
|
|
* 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", (unsigned int)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;
|
|
}
|
|
|
|
/* check output length */
|
|
if (*rtcp_len <
|
|
srtcp_len - sizeof(srtcp_trailer_t) - stream->mki_size - tag_len) {
|
|
return srtp_err_status_buffer_small;
|
|
}
|
|
|
|
/* if not inplace need to copy rtcp header */
|
|
if (srtcp != rtcp) {
|
|
memcpy(rtcp, srtcp, enc_start);
|
|
}
|
|
|
|
/*
|
|
* Set the AAD for GCM mode
|
|
*/
|
|
if (*trailer_p & SRTCP_E_BYTE_BIT) {
|
|
/*
|
|
* 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, srtcp,
|
|
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, srtcp,
|
|
(srtcp_len - tag_len - sizeof(srtcp_trailer_t) - stream->mki_size));
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Process the sequence# as AAD
|
|
*/
|
|
status = srtp_cipher_set_aad(session_keys->rtcp_cipher, (uint8_t *)&trailer,
|
|
sizeof(trailer));
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
|
|
/* if we're decrypting, exor keystream into the message */
|
|
if (*trailer_p & SRTCP_E_BYTE_BIT) {
|
|
status = srtp_cipher_decrypt(session_keys->rtcp_cipher,
|
|
srtcp + enc_start, enc_octet_len,
|
|
rtcp + enc_start, &enc_octet_len);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
} else {
|
|
/* if no encryption and not-inplace then need to copy rest of packet */
|
|
if (rtcp != srtcp) {
|
|
memcpy(rtcp + enc_start, srtcp + enc_start, enc_octet_len);
|
|
}
|
|
|
|
/*
|
|
* Still need to run the cipher to check the tag
|
|
*/
|
|
tmp_len = 0;
|
|
status = srtp_cipher_decrypt(session_keys->rtcp_cipher, auth_tag,
|
|
tag_len, NULL, &tmp_len);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
|
|
*rtcp_len = srtcp_len;
|
|
|
|
/* decrease the packet length by the length of the auth tag and seq_num*/
|
|
*rtcp_len -= (tag_len + sizeof(srtcp_trailer_t) + stream->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 list */
|
|
status = srtp_insert_or_dealloc_stream(ctx->stream_list, new_stream,
|
|
ctx->stream_template);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* 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,
|
|
const uint8_t *rtcp,
|
|
size_t rtcp_len,
|
|
uint8_t *srtcp,
|
|
size_t *srtcp_len,
|
|
size_t mki_index)
|
|
{
|
|
const srtcp_hdr_t *hdr = (const srtcp_hdr_t *)rtcp;
|
|
size_t enc_start; /* pointer to start of encrypted portion */
|
|
uint8_t *auth_start; /* pointer to start of auth. portion */
|
|
uint8_t *trailer_p; /* pointer to start of trailer */
|
|
uint32_t trailer; /* trailer value */
|
|
size_t 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;
|
|
size_t tag_len;
|
|
srtp_stream_ctx_t *stream;
|
|
size_t prefix_len;
|
|
uint32_t seq_num;
|
|
srtp_session_keys_t *session_keys = NULL;
|
|
|
|
/* check the packet length - it must at least contain a full header */
|
|
if (rtcp_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 list */
|
|
status = srtp_insert_or_dealloc_stream(ctx->stream_list, new_stream,
|
|
ctx->stream_template);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* 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);
|
|
}
|
|
}
|
|
|
|
status = srtp_get_session_keys(stream, mki_index, &session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/*
|
|
* 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(stream, rtcp, rtcp_len, srtcp, srtcp_len,
|
|
session_keys);
|
|
}
|
|
|
|
/* get tag length from stream context */
|
|
tag_len = srtp_auth_get_tag_length(session_keys->rtcp_auth);
|
|
|
|
/*
|
|
* set encryption start and encryption length
|
|
*/
|
|
enc_start = octets_in_rtcp_header;
|
|
enc_octet_len = rtcp_len - enc_start;
|
|
|
|
/* check output length */
|
|
if (*srtcp_len <
|
|
rtcp_len + sizeof(srtcp_trailer_t) + stream->mki_size + tag_len) {
|
|
return srtp_err_status_buffer_small;
|
|
}
|
|
|
|
/* if not in place then need to copy rtcp header */
|
|
if (rtcp != srtcp) {
|
|
memcpy(srtcp, rtcp, enc_start);
|
|
}
|
|
|
|
/* 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 = srtcp + enc_start + enc_octet_len;
|
|
|
|
if (stream->rtcp_services & sec_serv_conf) {
|
|
trailer = htonl(SRTCP_E_BIT); /* set encrypt bit */
|
|
} else {
|
|
/* 0 is network-order independant */
|
|
trailer = 0x00000000; /* set encrypt bit */
|
|
}
|
|
|
|
if (stream->use_mki) {
|
|
srtp_inject_mki(srtcp + rtcp_len + sizeof(srtcp_trailer_t),
|
|
session_keys, stream->mki_size);
|
|
}
|
|
|
|
/*
|
|
* 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 = srtcp;
|
|
auth_tag = srtcp + rtcp_len + sizeof(srtcp_trailer_t) + stream->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", (unsigned int)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 (stream->rtcp_services & sec_serv_conf) {
|
|
status = srtp_cipher_encrypt(session_keys->rtcp_cipher,
|
|
rtcp + enc_start, enc_octet_len,
|
|
srtcp + enc_start, &enc_octet_len);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
} else if (rtcp != srtcp) {
|
|
/* if no encryption and not-inplace then need to copy rest of packet */
|
|
memcpy(srtcp + enc_start, rtcp + enc_start, enc_octet_len);
|
|
}
|
|
|
|
/* 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, auth_start,
|
|
rtcp_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;
|
|
}
|
|
|
|
*srtcp_len = enc_start + enc_octet_len;
|
|
|
|
/* increase the packet length by the length of the auth tag and seq_num*/
|
|
*srtcp_len += (tag_len + sizeof(srtcp_trailer_t));
|
|
|
|
/* increase the packet by the mki_size */
|
|
*srtcp_len += stream->mki_size;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_unprotect_rtcp(srtp_t ctx,
|
|
const uint8_t *srtcp,
|
|
size_t srtcp_len,
|
|
uint8_t *rtcp,
|
|
size_t *rtcp_len)
|
|
{
|
|
const srtcp_hdr_t *hdr = (const srtcp_hdr_t *)srtcp;
|
|
size_t enc_start; /* pointer to start of encrypted portion */
|
|
const uint8_t *auth_start; /* pointer to start of auth. portion */
|
|
const uint8_t *trailer_p; /* pointer to start of trailer */
|
|
uint32_t trailer; /* trailer value */
|
|
size_t enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
const uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
uint8_t tmp_tag[SRTP_MAX_TAG_LEN];
|
|
srtp_err_status_t status;
|
|
size_t auth_len;
|
|
size_t tag_len;
|
|
srtp_stream_ctx_t *stream;
|
|
size_t prefix_len;
|
|
uint32_t seq_num;
|
|
bool e_bit_in_packet; /* E-bit was found in the packet */
|
|
bool sec_serv_confidentiality; /* whether confidentiality was requested */
|
|
srtp_session_keys_t *session_keys = NULL;
|
|
|
|
/*
|
|
* 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 (srtcp_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)",
|
|
(unsigned int)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
|
|
*/
|
|
status = srtp_get_session_keys_for_rtcp_packet(stream, srtcp, srtcp_len,
|
|
&session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* 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 (srtcp_len < octets_in_rtcp_header + sizeof(srtcp_trailer_t) +
|
|
stream->mki_size + tag_len) {
|
|
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, srtcp_len, rtcp,
|
|
rtcp_len, session_keys);
|
|
}
|
|
|
|
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_start = octets_in_rtcp_header;
|
|
enc_octet_len = srtcp_len - (octets_in_rtcp_header + tag_len +
|
|
stream->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 = srtcp + srtcp_len -
|
|
(tag_len + stream->mki_size + sizeof(srtcp_trailer_t));
|
|
memcpy(&trailer, trailer_p, sizeof(trailer));
|
|
|
|
e_bit_in_packet = (*trailer_p & SRTCP_E_BYTE_BIT) == SRTCP_E_BYTE_BIT;
|
|
if (e_bit_in_packet != sec_serv_confidentiality) {
|
|
return srtp_err_status_cant_check;
|
|
}
|
|
|
|
/*
|
|
* set the auth_start and auth_tag pointers to the proper locations
|
|
* (note that srtcp *always* uses authentication, unlike srtp)
|
|
*/
|
|
auth_start = srtcp;
|
|
|
|
/*
|
|
* 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 = srtcp_len - tag_len - stream->mki_size;
|
|
auth_tag = srtcp + auth_len + stream->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", (unsigned int)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;
|
|
}
|
|
|
|
/*
|
|
* 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, 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->rtcp_auth);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* run auth func over packet, put result into tmp_tag */
|
|
status = srtp_auth_compute(session_keys->rtcp_auth, 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_equal(tmp_tag, auth_tag, tag_len)) {
|
|
return srtp_err_status_auth_fail;
|
|
}
|
|
|
|
/* check output length */
|
|
if (*rtcp_len <
|
|
srtcp_len - sizeof(srtcp_trailer_t) - stream->mki_size - tag_len) {
|
|
return srtp_err_status_buffer_small;
|
|
}
|
|
|
|
/* if not inplace need to copy rtcp header */
|
|
if (srtcp != rtcp) {
|
|
memcpy(rtcp, srtcp, enc_start);
|
|
}
|
|
|
|
/* if we're decrypting, exor keystream into the message */
|
|
if (sec_serv_confidentiality) {
|
|
status = srtp_cipher_decrypt(session_keys->rtcp_cipher,
|
|
srtcp + enc_start, enc_octet_len,
|
|
rtcp + enc_start, &enc_octet_len);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
} else if (srtcp != rtcp) {
|
|
/* if no encryption and not-inplace then need to copy rest of packet */
|
|
memcpy(rtcp + enc_start, srtcp + enc_start, enc_octet_len);
|
|
}
|
|
|
|
*rtcp_len = srtcp_len;
|
|
|
|
/* decrease the packet length by the length of the auth tag and seq_num */
|
|
*rtcp_len -= (tag_len + sizeof(srtcp_trailer_t));
|
|
|
|
/* decrease the packet length by the length of the mki_size */
|
|
*rtcp_len -= stream->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 list */
|
|
status = srtp_insert_or_dealloc_stream(ctx->stream_list, new_stream,
|
|
ctx->stream_template);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* 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,
|
|
size_t bytes_in_key,
|
|
uint8_t *salt,
|
|
size_t bytes_in_salt)
|
|
{
|
|
memcpy(key + bytes_in_key, salt, bytes_in_salt);
|
|
}
|
|
|
|
size_t 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 */
|
|
}
|
|
}
|
|
|
|
size_t 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,
|
|
bool is_rtp,
|
|
size_t mki_index,
|
|
size_t *length)
|
|
{
|
|
srtp_session_keys_t *session_key;
|
|
|
|
*length = 0;
|
|
|
|
if (stream->use_mki) {
|
|
if (mki_index >= stream->num_master_keys) {
|
|
return srtp_err_status_bad_mki;
|
|
}
|
|
session_key = &stream->session_keys[mki_index];
|
|
|
|
*length += stream->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;
|
|
}
|
|
|
|
struct get_protect_trailer_length_data {
|
|
bool found_stream; /* whether at least one matching stream was found */
|
|
size_t length; /* maximum trailer length found so far */
|
|
bool is_rtp;
|
|
size_t mki_index;
|
|
};
|
|
|
|
static bool get_protect_trailer_length_cb(srtp_stream_t stream, void *raw_data)
|
|
{
|
|
struct get_protect_trailer_length_data *data =
|
|
(struct get_protect_trailer_length_data *)raw_data;
|
|
size_t temp_length;
|
|
|
|
if (stream_get_protect_trailer_length(stream, data->is_rtp, data->mki_index,
|
|
&temp_length) == srtp_err_status_ok) {
|
|
data->found_stream = true;
|
|
if (temp_length > data->length) {
|
|
data->length = temp_length;
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
srtp_err_status_t get_protect_trailer_length(srtp_t session,
|
|
bool is_rtp,
|
|
size_t mki_index,
|
|
size_t *length)
|
|
{
|
|
srtp_stream_ctx_t *stream;
|
|
struct get_protect_trailer_length_data data = { false, 0, is_rtp,
|
|
mki_index };
|
|
|
|
if (session == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
stream = session->stream_template;
|
|
|
|
if (stream != NULL) {
|
|
data.found_stream = true;
|
|
stream_get_protect_trailer_length(stream, is_rtp, mki_index,
|
|
&data.length);
|
|
}
|
|
|
|
srtp_stream_list_for_each(session->stream_list,
|
|
get_protect_trailer_length_cb, &data);
|
|
|
|
if (!data.found_stream) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
*length = data.length;
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_get_protect_trailer_length(srtp_t session,
|
|
size_t mki_index,
|
|
size_t *length)
|
|
{
|
|
return get_protect_trailer_length(session, true, mki_index, length);
|
|
}
|
|
|
|
srtp_err_status_t srtp_get_protect_rtcp_trailer_length(srtp_t session,
|
|
size_t mki_index,
|
|
size_t *length)
|
|
{
|
|
return get_protect_trailer_length(session, false, mki_index, length);
|
|
}
|
|
|
|
/*
|
|
* SRTP debug interface
|
|
*/
|
|
srtp_err_status_t srtp_set_debug_module(const char *mod_name, bool 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;
|
|
|
|
static 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_stream_set_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_stream_get_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;
|
|
}
|
|
|
|
#ifndef SRTP_NO_STREAM_LIST
|
|
|
|
#define INITIAL_STREAM_INDEX_SIZE 2
|
|
|
|
typedef struct list_entry {
|
|
uint32_t ssrc;
|
|
srtp_stream_t stream;
|
|
} list_entry;
|
|
|
|
typedef struct srtp_stream_list_ctx_t_ {
|
|
list_entry *entries;
|
|
size_t capacity;
|
|
size_t size;
|
|
} srtp_stream_list_ctx_t_;
|
|
|
|
srtp_err_status_t srtp_stream_list_alloc(srtp_stream_list_t *list_ptr)
|
|
{
|
|
srtp_stream_list_t list =
|
|
srtp_crypto_alloc(sizeof(srtp_stream_list_ctx_t_));
|
|
if (list == NULL) {
|
|
return srtp_err_status_alloc_fail;
|
|
}
|
|
|
|
list->entries =
|
|
srtp_crypto_alloc(sizeof(list_entry) * INITIAL_STREAM_INDEX_SIZE);
|
|
if (list->entries == NULL) {
|
|
srtp_crypto_free(list);
|
|
return srtp_err_status_alloc_fail;
|
|
}
|
|
|
|
list->capacity = INITIAL_STREAM_INDEX_SIZE;
|
|
list->size = 0;
|
|
|
|
*list_ptr = list;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_stream_list_dealloc(srtp_stream_list_t list)
|
|
{
|
|
/* list must be empty */
|
|
if (list->size != 0) {
|
|
return srtp_err_status_fail;
|
|
}
|
|
|
|
srtp_crypto_free(list->entries);
|
|
srtp_crypto_free(list);
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* inserting a new entry in the list may require reallocating memory in order
|
|
* to keep all the items in a contiguous memory block.
|
|
*/
|
|
srtp_err_status_t srtp_stream_list_insert(srtp_stream_list_t list,
|
|
srtp_stream_t stream)
|
|
{
|
|
/*
|
|
* there is no space to hold the new entry in the entries buffer,
|
|
* double the size of the buffer.
|
|
*/
|
|
if (list->size == list->capacity) {
|
|
size_t new_capacity = list->capacity * 2;
|
|
|
|
// Check for capacity overflow.
|
|
if (new_capacity < list->capacity ||
|
|
new_capacity > SIZE_MAX / sizeof(list_entry)) {
|
|
return srtp_err_status_alloc_fail;
|
|
}
|
|
|
|
list_entry *new_entries =
|
|
srtp_crypto_alloc(sizeof(list_entry) * new_capacity);
|
|
if (new_entries == NULL) {
|
|
return srtp_err_status_alloc_fail;
|
|
}
|
|
|
|
// Copy previous entries into the new buffer.
|
|
memcpy(new_entries, list->entries, sizeof(list_entry) * list->capacity);
|
|
|
|
// Release previous entries.
|
|
srtp_crypto_free(list->entries);
|
|
|
|
// Assign new entries to the list.
|
|
list->entries = new_entries;
|
|
|
|
// Update list capacity.
|
|
list->capacity = new_capacity;
|
|
}
|
|
|
|
// fill the first available entry
|
|
size_t next_index = list->size;
|
|
list->entries[next_index].ssrc = stream->ssrc;
|
|
list->entries[next_index].stream = stream;
|
|
|
|
// update size value
|
|
list->size++;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* removing an entry from the list performs a memory move of the following
|
|
* entries one position back in order to keep all the entries in the buffer
|
|
* contiguous.
|
|
*/
|
|
void srtp_stream_list_remove(srtp_stream_list_t list,
|
|
srtp_stream_t stream_to_remove)
|
|
{
|
|
size_t end = list->size;
|
|
|
|
for (size_t i = 0; i < end; i++) {
|
|
if (list->entries[i].ssrc == stream_to_remove->ssrc) {
|
|
size_t entries_to_move = list->size - i - 1;
|
|
memmove(&list->entries[i], &list->entries[i + 1],
|
|
sizeof(list_entry) * entries_to_move);
|
|
list->size--;
|
|
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
srtp_stream_t srtp_stream_list_get(srtp_stream_list_t list, uint32_t ssrc)
|
|
{
|
|
size_t end = list->size;
|
|
|
|
list_entry *entries = list->entries;
|
|
|
|
for (size_t i = 0; i < end; i++) {
|
|
if (entries[i].ssrc == ssrc) {
|
|
return entries[i].stream;
|
|
}
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
void srtp_stream_list_for_each(srtp_stream_list_t list,
|
|
bool (*callback)(srtp_stream_t, void *),
|
|
void *data)
|
|
{
|
|
list_entry *entries = list->entries;
|
|
|
|
size_t size = list->size;
|
|
|
|
/*
|
|
* the second statement of the expression needs to be recalculated on each
|
|
* iteration as the available number of entries may change within the given
|
|
* callback.
|
|
* Ie: in case the callback calls srtp_stream_list_remove().
|
|
*/
|
|
for (size_t i = 0; i < list->size;) {
|
|
if (!callback(entries[i].stream, data)) {
|
|
break;
|
|
}
|
|
|
|
// the entry was not removed, increase the counter.
|
|
if (size == list->size) {
|
|
++i;
|
|
}
|
|
|
|
size = list->size;
|
|
}
|
|
}
|
|
|
|
#endif
|