4733 lines
154 KiB
C
4733 lines
154 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 "crypto_types.h"
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#include "err.h"
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#include "ekt.h" /* for SRTP Encrypted Key Transport */
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#include "alloc.h" /* for srtp_crypto_alloc() */
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#ifdef OPENSSL
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#include "aes_gcm_ossl.h" /* for AES GCM mode */
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#ifdef OPENSSL_KDF
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#include <openssl/kdf.h>
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#include "aes_icm_ossl.h" /* for AES GCM mode */
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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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0, /* debugging is off by default */
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"srtp" /* printable name for module */
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};
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#define octets_in_rtp_header 12
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#define uint32s_in_rtp_header 3
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#define octets_in_rtcp_header 8
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#define uint32s_in_rtcp_header 2
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#define octets_in_rtp_extn_hdr 4
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static srtp_err_status_t srtp_validate_rtp_header(void *rtp_hdr,
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int *pkt_octet_len)
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{
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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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srtp_hdr_t *hdr = (srtp_hdr_t *)rtp_hdr;
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/* Check RTP header length */
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int rtp_header_len = octets_in_rtp_header + 4 * hdr->cc;
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if (hdr->x == 1)
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rtp_header_len += octets_in_rtp_extn_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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/* Verifing profile length. */
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if (hdr->x == 1) {
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srtp_hdr_xtnd_t *xtn_hdr =
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(srtp_hdr_xtnd_t *)((uint32_t *)hdr + uint32s_in_rtp_header +
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hdr->cc);
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int profile_len = ntohs(xtn_hdr->length);
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rtp_header_len += profile_len * 4;
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/* profile length counts the number of 32-bit words */
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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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return srtp_err_status_ok;
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}
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const char *srtp_get_version_string()
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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()
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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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srtp_err_status_t srtp_stream_dealloc(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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unsigned int i = 0;
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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 (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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* 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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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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* 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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* 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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* 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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session_keys->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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/* DAM - need to deallocate EKT here */
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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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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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unsigned int i = 0;
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srtp_session_keys_t *session_keys = NULL;
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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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*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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* ...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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/* 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));
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if (session_keys->limit == 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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}
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/* allocate ekt data associated with stream */
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stat = srtp_ekt_alloc(&str->ekt, p->ekt);
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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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if (p->enc_xtn_hdr && p->enc_xtn_hdr_count > 0) {
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srtp_cipher_type_id_t enc_xtn_hdr_cipher_type;
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int enc_xtn_hdr_cipher_key_len;
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str->enc_xtn_hdr = (int *)srtp_crypto_alloc(p->enc_xtn_hdr_count *
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sizeof(p->enc_xtn_hdr[0]));
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if (!str->enc_xtn_hdr) {
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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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memcpy(str->enc_xtn_hdr, p->enc_xtn_hdr,
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p->enc_xtn_hdr_count * sizeof(p->enc_xtn_hdr[0]));
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str->enc_xtn_hdr_count = p->enc_xtn_hdr_count;
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/*
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* For GCM ciphers, the corresponding ICM cipher is used for header
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* extensions encryption.
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*/
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switch (p->rtp.cipher_type) {
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case SRTP_AES_GCM_128:
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enc_xtn_hdr_cipher_type = SRTP_AES_ICM_128;
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enc_xtn_hdr_cipher_key_len = SRTP_AES_ICM_128_KEY_LEN_WSALT;
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break;
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case SRTP_AES_GCM_256:
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enc_xtn_hdr_cipher_type = SRTP_AES_ICM_256;
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enc_xtn_hdr_cipher_key_len = SRTP_AES_ICM_256_KEY_LEN_WSALT;
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break;
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default:
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enc_xtn_hdr_cipher_type = p->rtp.cipher_type;
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enc_xtn_hdr_cipher_key_len = p->rtp.cipher_key_len;
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break;
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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 for extensions header encryption */
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stat = srtp_crypto_kernel_alloc_cipher(
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enc_xtn_hdr_cipher_type, &session_keys->rtp_xtn_hdr_cipher,
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enc_xtn_hdr_cipher_key_len, 0);
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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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} else {
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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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session_keys->rtp_xtn_hdr_cipher = NULL;
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}
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str->enc_xtn_hdr = NULL;
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str->enc_xtn_hdr_count = 0;
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}
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return srtp_err_status_ok;
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}
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/*
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* srtp_stream_clone(stream_template, new) allocates a new stream and
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* initializes it using the cipher and auth of the stream_template
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*
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* the only unique data in a cloned stream is the replay database and
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* the SSRC
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*/
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srtp_err_status_t srtp_stream_clone(const srtp_stream_ctx_t *stream_template,
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uint32_t ssrc,
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srtp_stream_ctx_t **str_ptr)
|
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{
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srtp_err_status_t status;
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srtp_stream_ctx_t *str;
|
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unsigned int i = 0;
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srtp_session_keys_t *session_keys = NULL;
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const srtp_session_keys_t *template_session_keys = NULL;
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debug_print(mod_srtp, "cloning stream (SSRC: 0x%08x)", ntohl(ssrc));
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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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*str_ptr = str;
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str->num_master_keys = stream_template->num_master_keys;
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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_ptr, stream_template);
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*str_ptr = NULL;
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return srtp_err_status_alloc_fail;
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}
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for (i = 0; i < stream_template->num_master_keys; i++) {
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session_keys = &str->session_keys[i];
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template_session_keys = &stream_template->session_keys[i];
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/* set cipher and auth pointers to those of the template */
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session_keys->rtp_cipher = template_session_keys->rtp_cipher;
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session_keys->rtp_auth = template_session_keys->rtp_auth;
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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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session_keys->rtcp_cipher = template_session_keys->rtcp_cipher;
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session_keys->rtcp_auth = template_session_keys->rtcp_auth;
|
|
session_keys->mki_size = template_session_keys->mki_size;
|
|
|
|
if (template_session_keys->mki_size == 0) {
|
|
session_keys->mki_id = NULL;
|
|
} else {
|
|
session_keys->mki_id =
|
|
srtp_crypto_alloc(template_session_keys->mki_size);
|
|
|
|
if (session_keys->mki_id == NULL) {
|
|
srtp_stream_dealloc(*str_ptr, stream_template);
|
|
*str_ptr = NULL;
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
memcpy(session_keys->mki_id, template_session_keys->mki_id,
|
|
session_keys->mki_size);
|
|
}
|
|
/* Copy the salt values */
|
|
memcpy(session_keys->salt, template_session_keys->salt,
|
|
SRTP_AEAD_SALT_LEN);
|
|
memcpy(session_keys->c_salt, template_session_keys->c_salt,
|
|
SRTP_AEAD_SALT_LEN);
|
|
|
|
/* set key limit to point to that of the template */
|
|
status = srtp_key_limit_clone(template_session_keys->limit,
|
|
&session_keys->limit);
|
|
if (status) {
|
|
srtp_stream_dealloc(*str_ptr, stream_template);
|
|
*str_ptr = NULL;
|
|
return status;
|
|
}
|
|
}
|
|
|
|
/* initialize replay databases */
|
|
status = srtp_rdbx_init(
|
|
&str->rtp_rdbx, srtp_rdbx_get_window_size(&stream_template->rtp_rdbx));
|
|
if (status) {
|
|
srtp_stream_dealloc(*str_ptr, stream_template);
|
|
*str_ptr = NULL;
|
|
return status;
|
|
}
|
|
srtp_rdb_init(&str->rtcp_rdb);
|
|
str->allow_repeat_tx = stream_template->allow_repeat_tx;
|
|
|
|
/* set ssrc to that provided */
|
|
str->ssrc = ssrc;
|
|
|
|
/* reset pending ROC */
|
|
str->pending_roc = 0;
|
|
|
|
/* set direction and security services */
|
|
str->direction = stream_template->direction;
|
|
str->rtp_services = stream_template->rtp_services;
|
|
str->rtcp_services = stream_template->rtcp_services;
|
|
|
|
/* set pointer to EKT data associated with stream */
|
|
str->ekt = stream_template->ekt;
|
|
|
|
/* copy information about extensions header encryption */
|
|
str->enc_xtn_hdr = stream_template->enc_xtn_hdr;
|
|
str->enc_xtn_hdr_count = stream_template->enc_xtn_hdr_count;
|
|
|
|
/* defensive coding */
|
|
str->next = NULL;
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* key derivation functions, internal to libSRTP
|
|
*
|
|
* srtp_kdf_t is a key derivation context
|
|
*
|
|
* srtp_kdf_init(&kdf, cipher_id, k, keylen) initializes kdf to use cipher
|
|
* described by cipher_id, with the master key k with length in octets keylen.
|
|
*
|
|
* srtp_kdf_generate(&kdf, l, kl, keylen) derives the key
|
|
* corresponding to label l and puts it into kl; the length
|
|
* of the key in octets is provided as keylen. this function
|
|
* should be called once for each subkey that is derived.
|
|
*
|
|
* srtp_kdf_clear(&kdf) zeroizes and deallocates the kdf state
|
|
*/
|
|
|
|
typedef enum {
|
|
label_rtp_encryption = 0x00,
|
|
label_rtp_msg_auth = 0x01,
|
|
label_rtp_salt = 0x02,
|
|
label_rtcp_encryption = 0x03,
|
|
label_rtcp_msg_auth = 0x04,
|
|
label_rtcp_salt = 0x05,
|
|
label_rtp_header_encryption = 0x06,
|
|
label_rtp_header_salt = 0x07
|
|
} srtp_prf_label;
|
|
|
|
#define MAX_SRTP_KEY_LEN 256
|
|
|
|
#if defined(OPENSSL) && defined(OPENSSL_KDF)
|
|
#define MAX_SRTP_AESKEY_LEN 32
|
|
#define MAX_SRTP_SALT_LEN 14
|
|
|
|
/*
|
|
* srtp_kdf_t represents a key derivation function. The SRTP
|
|
* default KDF is the only one implemented at present.
|
|
*/
|
|
typedef struct {
|
|
uint8_t master_key[MAX_SRTP_AESKEY_LEN];
|
|
uint8_t master_salt[MAX_SRTP_SALT_LEN];
|
|
const EVP_CIPHER *evp;
|
|
} srtp_kdf_t;
|
|
|
|
static srtp_err_status_t srtp_kdf_init(srtp_kdf_t *kdf,
|
|
const uint8_t *key,
|
|
int key_len,
|
|
int salt_len)
|
|
{
|
|
memset(kdf, 0x0, sizeof(srtp_kdf_t));
|
|
|
|
/* The NULL cipher has zero key length */
|
|
if (key_len == 0)
|
|
return srtp_err_status_ok;
|
|
|
|
if ((key_len > MAX_SRTP_AESKEY_LEN) || (salt_len > MAX_SRTP_SALT_LEN)) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
switch (key_len) {
|
|
case SRTP_AES_256_KEYSIZE:
|
|
kdf->evp = EVP_aes_256_ctr();
|
|
break;
|
|
case SRTP_AES_192_KEYSIZE:
|
|
kdf->evp = EVP_aes_192_ctr();
|
|
break;
|
|
case SRTP_AES_128_KEYSIZE:
|
|
kdf->evp = EVP_aes_128_ctr();
|
|
break;
|
|
default:
|
|
return srtp_err_status_bad_param;
|
|
break;
|
|
}
|
|
memcpy(kdf->master_key, key, key_len);
|
|
memcpy(kdf->master_salt, key + key_len, salt_len);
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
static srtp_err_status_t srtp_kdf_generate(srtp_kdf_t *kdf,
|
|
srtp_prf_label label,
|
|
uint8_t *key,
|
|
unsigned int length)
|
|
{
|
|
int ret;
|
|
|
|
/* The NULL cipher will not have an EVP */
|
|
if (!kdf->evp)
|
|
return srtp_err_status_ok;
|
|
octet_string_set_to_zero(key, length);
|
|
|
|
/*
|
|
* Invoke the OpenSSL SRTP KDF function
|
|
* This is useful if OpenSSL is in FIPS mode and FIP
|
|
* compliance is required for SRTP.
|
|
*/
|
|
ret = kdf_srtp(kdf->evp, (char *)&kdf->master_key,
|
|
(char *)&kdf->master_salt, NULL, NULL, label, (char *)key);
|
|
if (ret == -1) {
|
|
return (srtp_err_status_algo_fail);
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
static srtp_err_status_t srtp_kdf_clear(srtp_kdf_t *kdf)
|
|
{
|
|
octet_string_set_to_zero(kdf->master_key, MAX_SRTP_AESKEY_LEN);
|
|
octet_string_set_to_zero(kdf->master_salt, MAX_SRTP_SALT_LEN);
|
|
kdf->evp = NULL;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
#else /* if OPENSSL_KDF */
|
|
|
|
/*
|
|
* srtp_kdf_t represents a key derivation function. The SRTP
|
|
* default KDF is the only one implemented at present.
|
|
*/
|
|
typedef struct {
|
|
srtp_cipher_t *cipher; /* cipher used for key derivation */
|
|
} srtp_kdf_t;
|
|
|
|
static srtp_err_status_t srtp_kdf_init(srtp_kdf_t *kdf,
|
|
const uint8_t *key,
|
|
int key_len)
|
|
{
|
|
srtp_cipher_type_id_t cipher_id;
|
|
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;
|
|
}
|
|
|
|
srtp_err_status_t stat;
|
|
stat = srtp_crypto_kernel_alloc_cipher(cipher_id, &kdf->cipher, key_len, 0);
|
|
if (stat)
|
|
return stat;
|
|
|
|
stat = srtp_cipher_init(kdf->cipher, key);
|
|
if (stat) {
|
|
srtp_cipher_dealloc(kdf->cipher);
|
|
return stat;
|
|
}
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
static srtp_err_status_t srtp_kdf_generate(srtp_kdf_t *kdf,
|
|
srtp_prf_label label,
|
|
uint8_t *key,
|
|
unsigned int length)
|
|
{
|
|
srtp_err_status_t status;
|
|
v128_t nonce;
|
|
|
|
/* set eigth octet of nonce to <label>, set the rest of it to zero */
|
|
v128_set_to_zero(&nonce);
|
|
nonce.v8[7] = label;
|
|
|
|
status = srtp_cipher_set_iv(kdf->cipher, (uint8_t *)&nonce,
|
|
srtp_direction_encrypt);
|
|
if (status)
|
|
return status;
|
|
|
|
/* generate keystream output */
|
|
octet_string_set_to_zero(key, length);
|
|
status = srtp_cipher_encrypt(kdf->cipher, key, &length);
|
|
if (status)
|
|
return status;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
static srtp_err_status_t srtp_kdf_clear(srtp_kdf_t *kdf)
|
|
{
|
|
srtp_err_status_t status;
|
|
status = srtp_cipher_dealloc(kdf->cipher);
|
|
if (status)
|
|
return status;
|
|
kdf->cipher = NULL;
|
|
return srtp_err_status_ok;
|
|
}
|
|
#endif /* else OPENSSL_KDF */
|
|
|
|
/*
|
|
* end of key derivation functions
|
|
*/
|
|
|
|
/* Get the base key length corresponding to a given combined key+salt
|
|
* length for the given cipher.
|
|
* TODO: key and salt lengths should be separate fields in the policy. */
|
|
static inline int base_key_length(const srtp_cipher_type_t *cipher,
|
|
int key_length)
|
|
{
|
|
switch (cipher->id) {
|
|
case SRTP_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;
|
|
break;
|
|
case SRTP_AES_GCM_128:
|
|
return key_length - SRTP_AEAD_SALT_LEN;
|
|
break;
|
|
case SRTP_AES_GCM_256:
|
|
return key_length - SRTP_AEAD_SALT_LEN;
|
|
break;
|
|
default:
|
|
return key_length;
|
|
break;
|
|
}
|
|
}
|
|
|
|
unsigned int srtp_validate_policy_master_keys(const srtp_policy_t *policy)
|
|
{
|
|
unsigned long i = 0;
|
|
|
|
if (policy->key == NULL) {
|
|
if (policy->num_master_keys <= 0)
|
|
return 0;
|
|
|
|
if (policy->num_master_keys > SRTP_MAX_NUM_MASTER_KEYS)
|
|
return 0;
|
|
|
|
for (i = 0; i < policy->num_master_keys; i++) {
|
|
if (policy->keys[i]->key == NULL)
|
|
return 0;
|
|
if (policy->keys[i]->mki_size > SRTP_MAX_MKI_LEN)
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
srtp_session_keys_t *srtp_get_session_keys_with_mki_index(
|
|
srtp_stream_ctx_t *stream,
|
|
unsigned int use_mki,
|
|
unsigned int mki_index)
|
|
{
|
|
if (use_mki) {
|
|
if (mki_index >= stream->num_master_keys) {
|
|
return NULL;
|
|
}
|
|
return &stream->session_keys[mki_index];
|
|
}
|
|
|
|
return &stream->session_keys[0];
|
|
}
|
|
|
|
unsigned int srtp_inject_mki(uint8_t *mki_tag_location,
|
|
srtp_session_keys_t *session_keys,
|
|
unsigned int use_mki)
|
|
{
|
|
unsigned int mki_size = 0;
|
|
|
|
if (use_mki) {
|
|
mki_size = session_keys->mki_size;
|
|
|
|
if (mki_size != 0) {
|
|
// Write MKI into memory
|
|
memcpy(mki_tag_location, session_keys->mki_id, mki_size);
|
|
}
|
|
}
|
|
|
|
return mki_size;
|
|
}
|
|
|
|
srtp_err_status_t srtp_stream_init_all_master_keys(
|
|
srtp_stream_ctx_t *srtp,
|
|
unsigned char *key,
|
|
srtp_master_key_t **keys,
|
|
const unsigned int max_master_keys)
|
|
{
|
|
unsigned int i = 0;
|
|
srtp_err_status_t status = srtp_err_status_ok;
|
|
srtp_master_key_t single_master_key;
|
|
|
|
if (key != NULL) {
|
|
srtp->num_master_keys = 1;
|
|
single_master_key.key = key;
|
|
single_master_key.mki_id = NULL;
|
|
single_master_key.mki_size = 0;
|
|
status = srtp_stream_init_keys(srtp, &single_master_key, 0);
|
|
} else {
|
|
srtp->num_master_keys = max_master_keys;
|
|
|
|
for (i = 0; i < srtp->num_master_keys && i < SRTP_MAX_NUM_MASTER_KEYS;
|
|
i++) {
|
|
status = srtp_stream_init_keys(srtp, keys[i], i);
|
|
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
}
|
|
|
|
return status;
|
|
}
|
|
|
|
srtp_err_status_t srtp_stream_init_keys(srtp_stream_ctx_t *srtp,
|
|
srtp_master_key_t *master_key,
|
|
const unsigned int current_mki_index)
|
|
{
|
|
srtp_err_status_t stat;
|
|
srtp_kdf_t kdf;
|
|
uint8_t tmp_key[MAX_SRTP_KEY_LEN];
|
|
int kdf_keylen = 30, rtp_keylen, rtcp_keylen;
|
|
int rtp_base_key_len, rtp_salt_len;
|
|
int rtcp_base_key_len, rtcp_salt_len;
|
|
srtp_session_keys_t *session_keys = NULL;
|
|
unsigned char *key = master_key->key;
|
|
|
|
/* If RTP or RTCP have a key length > AES-128, assume matching kdf. */
|
|
/* TODO: kdf algorithm, master key length, and master salt length should
|
|
* be part of srtp_policy_t.
|
|
*/
|
|
session_keys = &srtp->session_keys[current_mki_index];
|
|
|
|
/* initialize key limit to maximum value */
|
|
#ifdef NO_64BIT_MATH
|
|
{
|
|
uint64_t temp;
|
|
temp = make64(UINT_MAX, UINT_MAX);
|
|
srtp_key_limit_set(session_keys->limit, temp);
|
|
}
|
|
#else
|
|
srtp_key_limit_set(session_keys->limit, 0xffffffffffffLL);
|
|
#endif
|
|
|
|
if (master_key->mki_size != 0) {
|
|
session_keys->mki_id = srtp_crypto_alloc(master_key->mki_size);
|
|
|
|
if (session_keys->mki_id == NULL) {
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
memcpy(session_keys->mki_id, master_key->mki_id, master_key->mki_size);
|
|
} else {
|
|
session_keys->mki_id = NULL;
|
|
}
|
|
|
|
session_keys->mki_size = master_key->mki_size;
|
|
|
|
rtp_keylen = srtp_cipher_get_key_length(session_keys->rtp_cipher);
|
|
rtcp_keylen = srtp_cipher_get_key_length(session_keys->rtcp_cipher);
|
|
rtp_base_key_len =
|
|
base_key_length(session_keys->rtp_cipher->type, rtp_keylen);
|
|
rtp_salt_len = rtp_keylen - rtp_base_key_len;
|
|
|
|
if (rtp_keylen > kdf_keylen) {
|
|
kdf_keylen = 46; /* AES-CTR mode is always used for KDF */
|
|
}
|
|
|
|
if (rtcp_keylen > kdf_keylen) {
|
|
kdf_keylen = 46; /* AES-CTR mode is always used for KDF */
|
|
}
|
|
|
|
debug_print(mod_srtp, "srtp key len: %d", rtp_keylen);
|
|
debug_print(mod_srtp, "srtcp key len: %d", rtcp_keylen);
|
|
debug_print(mod_srtp, "base key len: %d", rtp_base_key_len);
|
|
debug_print(mod_srtp, "kdf key len: %d", kdf_keylen);
|
|
debug_print(mod_srtp, "rtp salt len: %d", rtp_salt_len);
|
|
|
|
/*
|
|
* Make sure the key given to us is 'zero' appended. GCM
|
|
* mode uses a shorter master SALT (96 bits), but still relies on
|
|
* the legacy CTR mode KDF, which uses a 112 bit master SALT.
|
|
*/
|
|
memset(tmp_key, 0x0, MAX_SRTP_KEY_LEN);
|
|
memcpy(tmp_key, key, (rtp_base_key_len + rtp_salt_len));
|
|
|
|
/* initialize KDF state */
|
|
#if defined(OPENSSL) && defined(OPENSSL_KDF)
|
|
stat = srtp_kdf_init(&kdf, (const uint8_t *)tmp_key, rtp_base_key_len,
|
|
rtp_salt_len);
|
|
#else
|
|
stat = srtp_kdf_init(&kdf, (const uint8_t *)tmp_key, kdf_keylen);
|
|
#endif
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
|
|
/* generate encryption key */
|
|
stat = srtp_kdf_generate(&kdf, label_rtp_encryption, tmp_key,
|
|
rtp_base_key_len);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
debug_print(mod_srtp, "cipher key: %s",
|
|
srtp_octet_string_hex_string(tmp_key, rtp_base_key_len));
|
|
|
|
/*
|
|
* if the cipher in the srtp context uses a salt, then we need
|
|
* to generate the salt value
|
|
*/
|
|
if (rtp_salt_len > 0) {
|
|
debug_print(mod_srtp, "found rtp_salt_len > 0, generating salt", NULL);
|
|
|
|
/* generate encryption salt, put after encryption key */
|
|
stat = srtp_kdf_generate(&kdf, label_rtp_salt,
|
|
tmp_key + rtp_base_key_len, rtp_salt_len);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
memcpy(session_keys->salt, tmp_key + rtp_base_key_len,
|
|
SRTP_AEAD_SALT_LEN);
|
|
}
|
|
if (rtp_salt_len > 0) {
|
|
debug_print(mod_srtp, "cipher salt: %s",
|
|
srtp_octet_string_hex_string(tmp_key + rtp_base_key_len,
|
|
rtp_salt_len));
|
|
}
|
|
|
|
/* initialize cipher */
|
|
stat = srtp_cipher_init(session_keys->rtp_cipher, tmp_key);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
|
|
if (session_keys->rtp_xtn_hdr_cipher) {
|
|
/* generate extensions header encryption key */
|
|
int rtp_xtn_hdr_keylen;
|
|
int rtp_xtn_hdr_base_key_len;
|
|
int rtp_xtn_hdr_salt_len;
|
|
srtp_kdf_t tmp_kdf;
|
|
srtp_kdf_t *xtn_hdr_kdf;
|
|
|
|
if (session_keys->rtp_xtn_hdr_cipher->type !=
|
|
session_keys->rtp_cipher->type) {
|
|
/*
|
|
* With GCM ciphers, the header extensions are still encrypted using
|
|
* the corresponding ICM cipher.
|
|
* See https://tools.ietf.org/html/rfc7714#section-8.3
|
|
*/
|
|
uint8_t tmp_xtn_hdr_key[MAX_SRTP_KEY_LEN];
|
|
rtp_xtn_hdr_keylen =
|
|
srtp_cipher_get_key_length(session_keys->rtp_xtn_hdr_cipher);
|
|
rtp_xtn_hdr_base_key_len = base_key_length(
|
|
session_keys->rtp_xtn_hdr_cipher->type, rtp_xtn_hdr_keylen);
|
|
rtp_xtn_hdr_salt_len =
|
|
rtp_xtn_hdr_keylen - rtp_xtn_hdr_base_key_len;
|
|
if (rtp_xtn_hdr_salt_len > rtp_salt_len) {
|
|
switch (session_keys->rtp_cipher->type->id) {
|
|
case SRTP_AES_GCM_128:
|
|
case SRTP_AES_GCM_256:
|
|
/*
|
|
* The shorter GCM salt is padded to the required ICM salt
|
|
* length.
|
|
*/
|
|
rtp_xtn_hdr_salt_len = rtp_salt_len;
|
|
break;
|
|
default:
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
}
|
|
memset(tmp_xtn_hdr_key, 0x0, MAX_SRTP_KEY_LEN);
|
|
memcpy(tmp_xtn_hdr_key, key,
|
|
(rtp_xtn_hdr_base_key_len + rtp_xtn_hdr_salt_len));
|
|
xtn_hdr_kdf = &tmp_kdf;
|
|
|
|
/* initialize KDF state */
|
|
#if defined(OPENSSL) && defined(OPENSSL_KDF)
|
|
stat =
|
|
srtp_kdf_init(xtn_hdr_kdf, (const uint8_t *)tmp_xtn_hdr_key,
|
|
rtp_xtn_hdr_base_key_len, rtp_xtn_hdr_salt_len);
|
|
#else
|
|
stat = srtp_kdf_init(xtn_hdr_kdf, (const uint8_t *)tmp_xtn_hdr_key,
|
|
kdf_keylen);
|
|
#endif
|
|
octet_string_set_to_zero(tmp_xtn_hdr_key, MAX_SRTP_KEY_LEN);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
} else {
|
|
/* Reuse main KDF. */
|
|
rtp_xtn_hdr_keylen = rtp_keylen;
|
|
rtp_xtn_hdr_base_key_len = rtp_base_key_len;
|
|
rtp_xtn_hdr_salt_len = rtp_salt_len;
|
|
xtn_hdr_kdf = &kdf;
|
|
}
|
|
|
|
stat = srtp_kdf_generate(xtn_hdr_kdf, label_rtp_header_encryption,
|
|
tmp_key, rtp_xtn_hdr_base_key_len);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
debug_print(
|
|
mod_srtp, "extensions cipher key: %s",
|
|
srtp_octet_string_hex_string(tmp_key, rtp_xtn_hdr_base_key_len));
|
|
|
|
/*
|
|
* if the cipher in the srtp context uses a salt, then we need
|
|
* to generate the salt value
|
|
*/
|
|
if (rtp_xtn_hdr_salt_len > 0) {
|
|
debug_print(mod_srtp,
|
|
"found rtp_xtn_hdr_salt_len > 0, generating salt",
|
|
NULL);
|
|
|
|
/* generate encryption salt, put after encryption key */
|
|
stat = srtp_kdf_generate(xtn_hdr_kdf, label_rtp_header_salt,
|
|
tmp_key + rtp_xtn_hdr_base_key_len,
|
|
rtp_xtn_hdr_salt_len);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
}
|
|
if (rtp_xtn_hdr_salt_len > 0) {
|
|
debug_print(
|
|
mod_srtp, "extensions cipher salt: %s",
|
|
srtp_octet_string_hex_string(tmp_key + rtp_xtn_hdr_base_key_len,
|
|
rtp_xtn_hdr_salt_len));
|
|
}
|
|
|
|
/* initialize extensions header cipher */
|
|
stat = srtp_cipher_init(session_keys->rtp_xtn_hdr_cipher, tmp_key);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
|
|
if (xtn_hdr_kdf != &kdf) {
|
|
/* release memory for custom header extension encryption kdf */
|
|
stat = srtp_kdf_clear(xtn_hdr_kdf);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* generate authentication key */
|
|
stat = srtp_kdf_generate(&kdf, label_rtp_msg_auth, tmp_key,
|
|
srtp_auth_get_key_length(session_keys->rtp_auth));
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
debug_print(mod_srtp, "auth key: %s",
|
|
srtp_octet_string_hex_string(
|
|
tmp_key, srtp_auth_get_key_length(session_keys->rtp_auth)));
|
|
|
|
/* initialize auth function */
|
|
stat = srtp_auth_init(session_keys->rtp_auth, tmp_key);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
|
|
/*
|
|
* ...now initialize SRTCP keys
|
|
*/
|
|
|
|
rtcp_base_key_len =
|
|
base_key_length(session_keys->rtcp_cipher->type, rtcp_keylen);
|
|
rtcp_salt_len = rtcp_keylen - rtcp_base_key_len;
|
|
debug_print(mod_srtp, "rtcp salt len: %d", rtcp_salt_len);
|
|
|
|
/* generate encryption key */
|
|
stat = srtp_kdf_generate(&kdf, label_rtcp_encryption, tmp_key,
|
|
rtcp_base_key_len);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
|
|
/*
|
|
* if the cipher in the srtp context uses a salt, then we need
|
|
* to generate the salt value
|
|
*/
|
|
if (rtcp_salt_len > 0) {
|
|
debug_print(mod_srtp, "found rtcp_salt_len > 0, generating rtcp salt",
|
|
NULL);
|
|
|
|
/* generate encryption salt, put after encryption key */
|
|
stat = srtp_kdf_generate(&kdf, label_rtcp_salt,
|
|
tmp_key + rtcp_base_key_len, rtcp_salt_len);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
memcpy(session_keys->c_salt, tmp_key + rtcp_base_key_len,
|
|
SRTP_AEAD_SALT_LEN);
|
|
}
|
|
debug_print(mod_srtp, "rtcp cipher key: %s",
|
|
srtp_octet_string_hex_string(tmp_key, rtcp_base_key_len));
|
|
if (rtcp_salt_len > 0) {
|
|
debug_print(mod_srtp, "rtcp cipher salt: %s",
|
|
srtp_octet_string_hex_string(tmp_key + rtcp_base_key_len,
|
|
rtcp_salt_len));
|
|
}
|
|
|
|
/* initialize cipher */
|
|
stat = srtp_cipher_init(session_keys->rtcp_cipher, tmp_key);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
|
|
/* generate authentication key */
|
|
stat = srtp_kdf_generate(&kdf, label_rtcp_msg_auth, tmp_key,
|
|
srtp_auth_get_key_length(session_keys->rtcp_auth));
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
|
|
debug_print(
|
|
mod_srtp, "rtcp auth key: %s",
|
|
srtp_octet_string_hex_string(
|
|
tmp_key, srtp_auth_get_key_length(session_keys->rtcp_auth)));
|
|
|
|
/* initialize auth function */
|
|
stat = srtp_auth_init(session_keys->rtcp_auth, tmp_key);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return srtp_err_status_init_fail;
|
|
}
|
|
|
|
/* clear memory then return */
|
|
stat = srtp_kdf_clear(&kdf);
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
if (stat)
|
|
return srtp_err_status_init_fail;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_stream_init(srtp_stream_ctx_t *srtp,
|
|
const srtp_policy_t *p)
|
|
{
|
|
srtp_err_status_t err;
|
|
|
|
debug_print(mod_srtp, "initializing stream (SSRC: 0x%08x)", p->ssrc.value);
|
|
|
|
/* initialize replay database */
|
|
/*
|
|
* window size MUST be at least 64. MAY be larger. Values more than
|
|
* 2^15 aren't meaningful due to how extended sequence numbers are
|
|
* calculated.
|
|
* Let a window size of 0 imply the default value.
|
|
*/
|
|
|
|
if (p->window_size != 0 &&
|
|
(p->window_size < 64 || p->window_size >= 0x8000))
|
|
return srtp_err_status_bad_param;
|
|
|
|
if (p->window_size != 0)
|
|
err = srtp_rdbx_init(&srtp->rtp_rdbx, p->window_size);
|
|
else
|
|
err = srtp_rdbx_init(&srtp->rtp_rdbx, 128);
|
|
if (err)
|
|
return err;
|
|
|
|
/* set the SSRC value */
|
|
srtp->ssrc = htonl(p->ssrc.value);
|
|
|
|
/* reset pending ROC */
|
|
srtp->pending_roc = 0;
|
|
|
|
/* set the security service flags */
|
|
srtp->rtp_services = p->rtp.sec_serv;
|
|
srtp->rtcp_services = p->rtcp.sec_serv;
|
|
|
|
/*
|
|
* set direction to unknown - this flag gets checked in srtp_protect(),
|
|
* srtp_unprotect(), srtp_protect_rtcp(), and srtp_unprotect_rtcp(), and
|
|
* gets set appropriately if it is set to unknown.
|
|
*/
|
|
srtp->direction = dir_unknown;
|
|
|
|
/* initialize SRTCP replay database */
|
|
srtp_rdb_init(&srtp->rtcp_rdb);
|
|
|
|
/* initialize allow_repeat_tx */
|
|
/* guard against uninitialized memory: allow only 0 or 1 here */
|
|
if (p->allow_repeat_tx != 0 && p->allow_repeat_tx != 1) {
|
|
srtp_rdbx_dealloc(&srtp->rtp_rdbx);
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
srtp->allow_repeat_tx = p->allow_repeat_tx;
|
|
|
|
/* DAM - no RTCP key limit at present */
|
|
|
|
/* initialize keys */
|
|
err = srtp_stream_init_all_master_keys(srtp, p->key, p->keys,
|
|
p->num_master_keys);
|
|
if (err) {
|
|
srtp_rdbx_dealloc(&srtp->rtp_rdbx);
|
|
return err;
|
|
}
|
|
|
|
/*
|
|
* if EKT is in use, then initialize the EKT data associated with
|
|
* the stream
|
|
*/
|
|
err = srtp_ekt_stream_init_from_policy(srtp->ekt, p->ekt);
|
|
if (err) {
|
|
srtp_rdbx_dealloc(&srtp->rtp_rdbx);
|
|
return err;
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* srtp_event_reporter is an event handler function that merely
|
|
* reports the events that are reported by the callbacks
|
|
*/
|
|
|
|
void srtp_event_reporter(srtp_event_data_t *data)
|
|
{
|
|
srtp_err_report(srtp_err_level_warning, "srtp: in stream 0x%x: ",
|
|
data->ssrc);
|
|
|
|
switch (data->event) {
|
|
case event_ssrc_collision:
|
|
srtp_err_report(srtp_err_level_warning, "\tSSRC collision\n");
|
|
break;
|
|
case event_key_soft_limit:
|
|
srtp_err_report(srtp_err_level_warning,
|
|
"\tkey usage soft limit reached\n");
|
|
break;
|
|
case event_key_hard_limit:
|
|
srtp_err_report(srtp_err_level_warning,
|
|
"\tkey usage hard limit reached\n");
|
|
break;
|
|
case event_packet_index_limit:
|
|
srtp_err_report(srtp_err_level_warning,
|
|
"\tpacket index limit reached\n");
|
|
break;
|
|
default:
|
|
srtp_err_report(srtp_err_level_warning,
|
|
"\tunknown event reported to handler\n");
|
|
}
|
|
}
|
|
|
|
/*
|
|
* srtp_event_handler is a global variable holding a pointer to the
|
|
* event handler function; this function is called for any unexpected
|
|
* event that needs to be handled out of the SRTP data path. see
|
|
* srtp_event_t in srtp.h for more info
|
|
*
|
|
* it is okay to set srtp_event_handler to NULL, but we set
|
|
* it to the srtp_event_reporter.
|
|
*/
|
|
|
|
static srtp_event_handler_func_t *srtp_event_handler = srtp_event_reporter;
|
|
|
|
srtp_err_status_t srtp_install_event_handler(srtp_event_handler_func_t func)
|
|
{
|
|
/*
|
|
* note that we accept NULL arguments intentionally - calling this
|
|
* function with a NULL arguments removes an event handler that's
|
|
* been previously installed
|
|
*/
|
|
|
|
/* set global event handling function */
|
|
srtp_event_handler = func;
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* Check if the given extension header id is / should be encrypted.
|
|
* Returns 1 if yes, otherwise 0.
|
|
*/
|
|
static int srtp_protect_extension_header(srtp_stream_ctx_t *stream, int id)
|
|
{
|
|
int *enc_xtn_hdr = stream->enc_xtn_hdr;
|
|
int count = stream->enc_xtn_hdr_count;
|
|
|
|
if (!enc_xtn_hdr || count <= 0) {
|
|
return 0;
|
|
}
|
|
|
|
while (count > 0) {
|
|
if (*enc_xtn_hdr == id) {
|
|
return 1;
|
|
}
|
|
|
|
enc_xtn_hdr++;
|
|
count--;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* extensions header encryption RFC 6904
|
|
*/
|
|
static srtp_err_status_t srtp_process_header_encryption(
|
|
srtp_stream_ctx_t *stream,
|
|
srtp_hdr_xtnd_t *xtn_hdr,
|
|
srtp_session_keys_t *session_keys)
|
|
{
|
|
srtp_err_status_t status;
|
|
uint8_t keystream[257]; /* Maximum 2 bytes header + 255 bytes data. */
|
|
int keystream_pos;
|
|
uint8_t *xtn_hdr_data = ((uint8_t *)xtn_hdr) + octets_in_rtp_extn_hdr;
|
|
uint8_t *xtn_hdr_end =
|
|
xtn_hdr_data + (ntohs(xtn_hdr->length) * sizeof(uint32_t));
|
|
|
|
if (ntohs(xtn_hdr->profile_specific) == 0xbede) {
|
|
/* RFC 5285, section 4.2. One-Byte Header */
|
|
while (xtn_hdr_data < xtn_hdr_end) {
|
|
uint8_t xid = (*xtn_hdr_data & 0xf0) >> 4;
|
|
unsigned int xlen = (*xtn_hdr_data & 0x0f) + 1;
|
|
uint32_t xlen_with_header = 1 + xlen;
|
|
xtn_hdr_data++;
|
|
|
|
if (xtn_hdr_data + xlen > xtn_hdr_end)
|
|
return srtp_err_status_parse_err;
|
|
|
|
if (xid == 15) {
|
|
/* found header 15, stop further processing. */
|
|
break;
|
|
}
|
|
|
|
status = srtp_cipher_output(session_keys->rtp_xtn_hdr_cipher,
|
|
keystream, &xlen_with_header);
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
|
|
if (srtp_protect_extension_header(stream, xid)) {
|
|
keystream_pos = 1;
|
|
while (xlen > 0) {
|
|
*xtn_hdr_data ^= keystream[keystream_pos++];
|
|
xtn_hdr_data++;
|
|
xlen--;
|
|
}
|
|
} else {
|
|
xtn_hdr_data += xlen;
|
|
}
|
|
|
|
/* skip padding bytes. */
|
|
while (xtn_hdr_data < xtn_hdr_end && *xtn_hdr_data == 0) {
|
|
xtn_hdr_data++;
|
|
}
|
|
}
|
|
} else if ((ntohs(xtn_hdr->profile_specific) & 0x1fff) == 0x100) {
|
|
/* RFC 5285, section 4.3. Two-Byte Header */
|
|
while (xtn_hdr_data + 1 < xtn_hdr_end) {
|
|
uint8_t xid = *xtn_hdr_data;
|
|
unsigned int xlen = *(xtn_hdr_data + 1);
|
|
uint32_t xlen_with_header = 2 + xlen;
|
|
xtn_hdr_data += 2;
|
|
|
|
if (xtn_hdr_data + xlen > xtn_hdr_end)
|
|
return srtp_err_status_parse_err;
|
|
|
|
status = srtp_cipher_output(session_keys->rtp_xtn_hdr_cipher,
|
|
keystream, &xlen_with_header);
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
|
|
if (xlen > 0 && srtp_protect_extension_header(stream, xid)) {
|
|
keystream_pos = 2;
|
|
while (xlen > 0) {
|
|
*xtn_hdr_data ^= keystream[keystream_pos++];
|
|
xtn_hdr_data++;
|
|
xlen--;
|
|
}
|
|
} else {
|
|
xtn_hdr_data += xlen;
|
|
}
|
|
|
|
/* skip padding bytes. */
|
|
while (xtn_hdr_data < xtn_hdr_end && *xtn_hdr_data == 0) {
|
|
xtn_hdr_data++;
|
|
}
|
|
}
|
|
} else {
|
|
/* unsupported extension header format. */
|
|
return srtp_err_status_parse_err;
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* AEAD uses a new IV formation method. This function implements
|
|
* section 8.1. (SRTP IV Formation for AES-GCM) of RFC7714.
|
|
* The calculation is defined as, where (+) is the xor operation:
|
|
*
|
|
*
|
|
* 0 0 0 0 0 0 0 0 0 0 1 1
|
|
* 0 1 2 3 4 5 6 7 8 9 0 1
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+
|
|
* |00|00| SSRC | ROC | SEQ |---+
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+ |
|
|
* |
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+ |
|
|
* | Encryption Salt |->(+)
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+ |
|
|
* |
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+ |
|
|
* | Initialization Vector |<--+
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+*
|
|
*
|
|
* Input: *session_keys - pointer to SRTP stream context session keys,
|
|
* used to retrieve the SALT
|
|
* *iv - Pointer to receive the calculated IV
|
|
* *seq - The ROC and SEQ value to use for the
|
|
* IV calculation.
|
|
* *hdr - The RTP header, used to get the SSRC value
|
|
*
|
|
*/
|
|
|
|
static void srtp_calc_aead_iv(srtp_session_keys_t *session_keys,
|
|
v128_t *iv,
|
|
srtp_xtd_seq_num_t *seq,
|
|
srtp_hdr_t *hdr)
|
|
{
|
|
v128_t in;
|
|
v128_t salt;
|
|
|
|
#ifdef NO_64BIT_MATH
|
|
uint32_t local_roc = ((high32(*seq) << 16) | (low32(*seq) >> 16));
|
|
uint16_t local_seq = (uint16_t)(low32(*seq));
|
|
#else
|
|
uint32_t local_roc = (uint32_t)(*seq >> 16);
|
|
uint16_t local_seq = (uint16_t)*seq;
|
|
#endif
|
|
|
|
memset(&in, 0, sizeof(v128_t));
|
|
memset(&salt, 0, sizeof(v128_t));
|
|
|
|
in.v16[5] = htons(local_seq);
|
|
local_roc = htonl(local_roc);
|
|
memcpy(&in.v16[3], &local_roc, sizeof(local_roc));
|
|
|
|
/*
|
|
* Copy in the RTP SSRC value
|
|
*/
|
|
memcpy(&in.v8[2], &hdr->ssrc, 4);
|
|
debug_print(mod_srtp, "Pre-salted RTP IV = %s\n", v128_hex_string(&in));
|
|
|
|
/*
|
|
* Get the SALT value from the context
|
|
*/
|
|
memcpy(salt.v8, session_keys->salt, SRTP_AEAD_SALT_LEN);
|
|
debug_print(mod_srtp, "RTP SALT = %s\n", v128_hex_string(&salt));
|
|
|
|
/*
|
|
* Finally, apply tyhe SALT to the input
|
|
*/
|
|
v128_xor(iv, &in, &salt);
|
|
}
|
|
|
|
srtp_session_keys_t *srtp_get_session_keys(srtp_stream_ctx_t *stream,
|
|
uint8_t *hdr,
|
|
const unsigned int *pkt_octet_len,
|
|
unsigned int *mki_size)
|
|
{
|
|
unsigned int base_mki_start_location = *pkt_octet_len;
|
|
unsigned int mki_start_location = 0;
|
|
unsigned int tag_len = 0;
|
|
unsigned int i = 0;
|
|
|
|
// Determine the authentication tag size
|
|
if (stream->session_keys[0].rtp_cipher->algorithm == SRTP_AES_GCM_128 ||
|
|
stream->session_keys[0].rtp_cipher->algorithm == SRTP_AES_GCM_256) {
|
|
tag_len = 0;
|
|
} else {
|
|
tag_len = srtp_auth_get_tag_length(stream->session_keys[0].rtp_auth);
|
|
}
|
|
|
|
if (tag_len > base_mki_start_location) {
|
|
*mki_size = 0;
|
|
return NULL;
|
|
}
|
|
|
|
base_mki_start_location -= tag_len;
|
|
|
|
for (i = 0; i < stream->num_master_keys; i++) {
|
|
if (stream->session_keys[i].mki_size != 0 &&
|
|
stream->session_keys[i].mki_size <= base_mki_start_location) {
|
|
*mki_size = stream->session_keys[i].mki_size;
|
|
mki_start_location = base_mki_start_location - *mki_size;
|
|
|
|
if (memcmp(hdr + mki_start_location, stream->session_keys[i].mki_id,
|
|
*mki_size) == 0) {
|
|
return &stream->session_keys[i];
|
|
}
|
|
}
|
|
}
|
|
|
|
*mki_size = 0;
|
|
return NULL;
|
|
}
|
|
|
|
static srtp_err_status_t srtp_estimate_index(srtp_rdbx_t *rdbx,
|
|
uint32_t roc,
|
|
srtp_xtd_seq_num_t *est,
|
|
srtp_sequence_number_t seq,
|
|
int *delta)
|
|
{
|
|
#ifdef NO_64BIT_MATH
|
|
uint32_t internal_pkt_idx_reduced;
|
|
uint32_t external_pkt_idx_reduced;
|
|
uint32_t internal_roc;
|
|
uint32_t roc_difference;
|
|
#endif
|
|
|
|
#ifdef NO_64BIT_MATH
|
|
*est = (srtp_xtd_seq_num_t)make64(roc >> 16, (roc << 16) | seq);
|
|
*delta = low32(est) - rdbx->index;
|
|
#else
|
|
*est = (srtp_xtd_seq_num_t)(((uint64_t)roc) << 16) | seq;
|
|
*delta = (int)(*est - rdbx->index);
|
|
#endif
|
|
|
|
if (*est > rdbx->index) {
|
|
#ifdef NO_64BIT_MATH
|
|
internal_roc = (uint32_t)(rdbx->index >> 16);
|
|
roc_difference = roc - internal_roc;
|
|
if (roc_difference > 1) {
|
|
*delta = 0;
|
|
return srtp_err_status_pkt_idx_adv;
|
|
}
|
|
|
|
internal_pkt_idx_reduced = (uint32_t)(rdbx->index & 0xFFFF);
|
|
external_pkt_idx_reduced = (uint32_t)((roc_difference << 16) | seq);
|
|
|
|
if (external_pkt_idx_reduced - internal_pkt_idx_reduced >
|
|
seq_num_median) {
|
|
*delta = 0;
|
|
return srtp_err_status_pkt_idx_adv;
|
|
}
|
|
#else
|
|
if (*est - rdbx->index > seq_num_median) {
|
|
*delta = 0;
|
|
return srtp_err_status_pkt_idx_adv;
|
|
}
|
|
#endif
|
|
} else if (*est < rdbx->index) {
|
|
#ifdef NO_64BIT_MATH
|
|
|
|
internal_roc = (uint32_t)(rdbx->index >> 16);
|
|
roc_difference = internal_roc - roc;
|
|
if (roc_difference > 1) {
|
|
*delta = 0;
|
|
return srtp_err_status_pkt_idx_adv;
|
|
}
|
|
|
|
internal_pkt_idx_reduced =
|
|
(uint32_t)((roc_difference << 16) | rdbx->index & 0xFFFF);
|
|
external_pkt_idx_reduced = (uint32_t)(seq);
|
|
|
|
if (internal_pkt_idx_reduced - external_pkt_idx_reduced >
|
|
seq_num_median) {
|
|
*delta = 0;
|
|
return srtp_err_status_pkt_idx_old;
|
|
}
|
|
#else
|
|
if (rdbx->index - *est > seq_num_median) {
|
|
*delta = 0;
|
|
return srtp_err_status_pkt_idx_old;
|
|
}
|
|
#endif
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
static srtp_err_status_t srtp_get_est_pkt_index(srtp_hdr_t *hdr,
|
|
srtp_stream_ctx_t *stream,
|
|
srtp_xtd_seq_num_t *est,
|
|
int *delta)
|
|
{
|
|
srtp_err_status_t result = srtp_err_status_ok;
|
|
|
|
if (stream->pending_roc) {
|
|
result = srtp_estimate_index(&stream->rtp_rdbx, stream->pending_roc,
|
|
est, ntohs(hdr->seq), delta);
|
|
} else {
|
|
/* estimate packet index from seq. num. in header */
|
|
*delta =
|
|
srtp_rdbx_estimate_index(&stream->rtp_rdbx, est, ntohs(hdr->seq));
|
|
}
|
|
|
|
#ifdef NO_64BIT_MATH
|
|
debug_print2(mod_srtp, "estimated u_packet index: %08x%08x", high32(*est),
|
|
low32(*est));
|
|
#else
|
|
debug_print(mod_srtp, "estimated u_packet index: %016llx", *est);
|
|
#endif
|
|
return result;
|
|
}
|
|
|
|
/*
|
|
* This function handles outgoing SRTP packets while in AEAD mode,
|
|
* which currently supports AES-GCM encryption. All packets are
|
|
* encrypted and authenticated.
|
|
*/
|
|
static srtp_err_status_t srtp_protect_aead(srtp_ctx_t *ctx,
|
|
srtp_stream_ctx_t *stream,
|
|
void *rtp_hdr,
|
|
unsigned int *pkt_octet_len,
|
|
srtp_session_keys_t *session_keys,
|
|
unsigned int use_mki)
|
|
{
|
|
srtp_hdr_t *hdr = (srtp_hdr_t *)rtp_hdr;
|
|
uint32_t *enc_start; /* pointer to start of encrypted portion */
|
|
int enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
srtp_xtd_seq_num_t est; /* estimated xtd_seq_num_t of *hdr */
|
|
int delta; /* delta of local pkt idx and that in hdr */
|
|
srtp_err_status_t status;
|
|
uint32_t tag_len;
|
|
v128_t iv;
|
|
unsigned int aad_len;
|
|
srtp_hdr_xtnd_t *xtn_hdr = NULL;
|
|
unsigned int mki_size = 0;
|
|
uint8_t *mki_location = NULL;
|
|
|
|
debug_print(mod_srtp, "function srtp_protect_aead", NULL);
|
|
|
|
/*
|
|
* update the key usage limit, and check it to make sure that we
|
|
* didn't just hit either the soft limit or the hard limit, and call
|
|
* the event handler if we hit either.
|
|
*/
|
|
switch (srtp_key_limit_update(session_keys->limit)) {
|
|
case srtp_key_event_normal:
|
|
break;
|
|
case srtp_key_event_hard_limit:
|
|
srtp_handle_event(ctx, stream, event_key_hard_limit);
|
|
return srtp_err_status_key_expired;
|
|
case srtp_key_event_soft_limit:
|
|
default:
|
|
srtp_handle_event(ctx, stream, event_key_soft_limit);
|
|
break;
|
|
}
|
|
|
|
/* get tag length from stream */
|
|
tag_len = srtp_auth_get_tag_length(session_keys->rtp_auth);
|
|
|
|
/*
|
|
* find starting point for encryption and length of data to be
|
|
* encrypted - the encrypted portion starts after the rtp header
|
|
* extension, if present; otherwise, it starts after the last csrc,
|
|
* if any are present
|
|
*/
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtp_header + hdr->cc;
|
|
if (hdr->x == 1) {
|
|
xtn_hdr = (srtp_hdr_xtnd_t *)enc_start;
|
|
enc_start += (ntohs(xtn_hdr->length) + 1);
|
|
}
|
|
/* note: the passed size is without the auth tag */
|
|
if (!((uint8_t *)enc_start <= (uint8_t *)hdr + *pkt_octet_len))
|
|
return srtp_err_status_parse_err;
|
|
enc_octet_len =
|
|
(int)(*pkt_octet_len - ((uint8_t *)enc_start - (uint8_t *)hdr));
|
|
if (enc_octet_len < 0)
|
|
return srtp_err_status_parse_err;
|
|
|
|
/*
|
|
* estimate the packet index using the start of the replay window
|
|
* and the sequence number from the header
|
|
*/
|
|
delta = srtp_rdbx_estimate_index(&stream->rtp_rdbx, &est, ntohs(hdr->seq));
|
|
status = srtp_rdbx_check(&stream->rtp_rdbx, delta);
|
|
if (status) {
|
|
if (status != srtp_err_status_replay_fail || !stream->allow_repeat_tx) {
|
|
return status; /* we've been asked to reuse an index */
|
|
}
|
|
} else {
|
|
srtp_rdbx_add_index(&stream->rtp_rdbx, delta);
|
|
}
|
|
|
|
#ifdef NO_64BIT_MATH
|
|
debug_print2(mod_srtp, "estimated packet index: %08x%08x", high32(est),
|
|
low32(est));
|
|
#else
|
|
debug_print(mod_srtp, "estimated packet index: %016llx", est);
|
|
#endif
|
|
|
|
/*
|
|
* AEAD uses a new IV formation method
|
|
*/
|
|
srtp_calc_aead_iv(session_keys, &iv, &est, hdr);
|
|
/* shift est, put into network byte order */
|
|
#ifdef NO_64BIT_MATH
|
|
est = be64_to_cpu(
|
|
make64((high32(est) << 16) | (low32(est) >> 16), low32(est) << 16));
|
|
#else
|
|
est = be64_to_cpu(est << 16);
|
|
#endif
|
|
|
|
status = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_encrypt);
|
|
if (!status && session_keys->rtp_xtn_hdr_cipher) {
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = hdr->ssrc;
|
|
iv.v64[1] = est;
|
|
status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
|
|
(uint8_t *)&iv, srtp_direction_encrypt);
|
|
}
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
if (xtn_hdr && session_keys->rtp_xtn_hdr_cipher) {
|
|
/*
|
|
* extensions header encryption RFC 6904
|
|
*/
|
|
status = srtp_process_header_encryption(stream, xtn_hdr, session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Set the AAD over the RTP header
|
|
*/
|
|
aad_len = (uint8_t *)enc_start - (uint8_t *)hdr;
|
|
status =
|
|
srtp_cipher_set_aad(session_keys->rtp_cipher, (uint8_t *)hdr, aad_len);
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
|
|
/* Encrypt the payload */
|
|
status = srtp_cipher_encrypt(session_keys->rtp_cipher, (uint8_t *)enc_start,
|
|
(unsigned int *)&enc_octet_len);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
/*
|
|
* If we're doing GCM, we need to get the tag
|
|
* and append that to the output
|
|
*/
|
|
status =
|
|
srtp_cipher_get_tag(session_keys->rtp_cipher,
|
|
(uint8_t *)enc_start + enc_octet_len, &tag_len);
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
|
|
mki_location = (uint8_t *)hdr + *pkt_octet_len + tag_len;
|
|
mki_size = srtp_inject_mki(mki_location, session_keys, use_mki);
|
|
|
|
/* increase the packet length by the length of the auth tag */
|
|
*pkt_octet_len += tag_len;
|
|
|
|
/* increase the packet length by the length of the mki_size */
|
|
*pkt_octet_len += mki_size;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* This function handles incoming SRTP packets while in AEAD mode,
|
|
* which currently supports AES-GCM encryption. All packets are
|
|
* encrypted and authenticated. Note, the auth tag is at the end
|
|
* of the packet stream and is automatically checked by GCM
|
|
* when decrypting the payload.
|
|
*/
|
|
static srtp_err_status_t srtp_unprotect_aead(srtp_ctx_t *ctx,
|
|
srtp_stream_ctx_t *stream,
|
|
int delta,
|
|
srtp_xtd_seq_num_t est,
|
|
void *srtp_hdr,
|
|
unsigned int *pkt_octet_len,
|
|
srtp_session_keys_t *session_keys,
|
|
unsigned int mki_size)
|
|
{
|
|
srtp_hdr_t *hdr = (srtp_hdr_t *)srtp_hdr;
|
|
uint32_t *enc_start; /* pointer to start of encrypted portion */
|
|
unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
v128_t iv;
|
|
srtp_err_status_t status;
|
|
int tag_len;
|
|
unsigned int aad_len;
|
|
srtp_hdr_xtnd_t *xtn_hdr = NULL;
|
|
|
|
debug_print(mod_srtp, "function srtp_unprotect_aead", NULL);
|
|
|
|
#ifdef NO_64BIT_MATH
|
|
debug_print2(mod_srtp, "estimated u_packet index: %08x%08x", high32(est),
|
|
low32(est));
|
|
#else
|
|
debug_print(mod_srtp, "estimated u_packet index: %016llx", est);
|
|
#endif
|
|
|
|
/* get tag length from stream */
|
|
tag_len = srtp_auth_get_tag_length(session_keys->rtp_auth);
|
|
|
|
/*
|
|
* AEAD uses a new IV formation method
|
|
*/
|
|
srtp_calc_aead_iv(session_keys, &iv, &est, hdr);
|
|
status = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_decrypt);
|
|
if (!status && session_keys->rtp_xtn_hdr_cipher) {
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = hdr->ssrc;
|
|
#ifdef NO_64BIT_MATH
|
|
iv.v64[1] = be64_to_cpu(
|
|
make64((high32(est) << 16) | (low32(est) >> 16), low32(est) << 16));
|
|
#else
|
|
iv.v64[1] = be64_to_cpu(est << 16);
|
|
#endif
|
|
status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
|
|
(uint8_t *)&iv, srtp_direction_encrypt);
|
|
}
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/*
|
|
* find starting point for decryption and length of data to be
|
|
* decrypted - the encrypted portion starts after the rtp header
|
|
* extension, if present; otherwise, it starts after the last csrc,
|
|
* if any are present
|
|
*/
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtp_header + hdr->cc;
|
|
if (hdr->x == 1) {
|
|
xtn_hdr = (srtp_hdr_xtnd_t *)enc_start;
|
|
enc_start += (ntohs(xtn_hdr->length) + 1);
|
|
}
|
|
if (!((uint8_t *)enc_start <=
|
|
(uint8_t *)hdr + (*pkt_octet_len - tag_len - mki_size)))
|
|
return srtp_err_status_parse_err;
|
|
/*
|
|
* We pass the tag down to the cipher when doing GCM mode
|
|
*/
|
|
enc_octet_len = (unsigned int)(*pkt_octet_len - mki_size -
|
|
((uint8_t *)enc_start - (uint8_t *)hdr));
|
|
|
|
/*
|
|
* Sanity check the encrypted payload length against
|
|
* the tag size. It must always be at least as large
|
|
* as the tag length.
|
|
*/
|
|
if (enc_octet_len < (unsigned int)tag_len) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/*
|
|
* update the key usage limit, and check it to make sure that we
|
|
* didn't just hit either the soft limit or the hard limit, and call
|
|
* the event handler if we hit either.
|
|
*/
|
|
switch (srtp_key_limit_update(session_keys->limit)) {
|
|
case srtp_key_event_normal:
|
|
break;
|
|
case srtp_key_event_soft_limit:
|
|
srtp_handle_event(ctx, stream, event_key_soft_limit);
|
|
break;
|
|
case srtp_key_event_hard_limit:
|
|
srtp_handle_event(ctx, stream, event_key_hard_limit);
|
|
return srtp_err_status_key_expired;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/*
|
|
* Set the AAD for AES-GCM, which is the RTP header
|
|
*/
|
|
aad_len = (uint8_t *)enc_start - (uint8_t *)hdr;
|
|
status =
|
|
srtp_cipher_set_aad(session_keys->rtp_cipher, (uint8_t *)hdr, aad_len);
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
|
|
/* Decrypt the ciphertext. This also checks the auth tag based
|
|
* on the AAD we just specified above */
|
|
status = srtp_cipher_decrypt(session_keys->rtp_cipher, (uint8_t *)enc_start,
|
|
&enc_octet_len);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
if (xtn_hdr && session_keys->rtp_xtn_hdr_cipher) {
|
|
/*
|
|
* extensions header encryption RFC 6904
|
|
*/
|
|
status = srtp_process_header_encryption(stream, xtn_hdr, session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
|
|
/*
|
|
* verify that stream is for received traffic - this check will
|
|
* detect SSRC collisions, since a stream that appears in both
|
|
* srtp_protect() and srtp_unprotect() will fail this test in one of
|
|
* those functions.
|
|
*
|
|
* we do this check *after* the authentication check, so that the
|
|
* latter check will catch any attempts to fool us into thinking
|
|
* that we've got a collision
|
|
*/
|
|
if (stream->direction != dir_srtp_receiver) {
|
|
if (stream->direction == dir_unknown) {
|
|
stream->direction = dir_srtp_receiver;
|
|
} else {
|
|
srtp_handle_event(ctx, stream, event_ssrc_collision);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* if the stream is a 'provisional' one, in which the template context
|
|
* is used, then we need to allocate a new stream at this point, since
|
|
* the authentication passed
|
|
*/
|
|
if (stream == ctx->stream_template) {
|
|
srtp_stream_ctx_t *new_stream;
|
|
|
|
/*
|
|
* allocate and initialize a new stream
|
|
*
|
|
* note that we indicate failure if we can't allocate the new
|
|
* stream, and some implementations will want to not return
|
|
* failure here
|
|
*/
|
|
status =
|
|
srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* add new stream to the head of the stream_list */
|
|
new_stream->next = ctx->stream_list;
|
|
ctx->stream_list = new_stream;
|
|
|
|
/* set stream (the pointer used in this function) */
|
|
stream = new_stream;
|
|
}
|
|
|
|
/*
|
|
* the message authentication function passed, so add the packet
|
|
* index into the replay database
|
|
*/
|
|
srtp_rdbx_add_index(&stream->rtp_rdbx, delta);
|
|
|
|
/* decrease the packet length by the length of the auth tag */
|
|
*pkt_octet_len -= tag_len;
|
|
|
|
/* decrease the packet length by the length of the mki_size */
|
|
*pkt_octet_len -= mki_size;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_protect(srtp_ctx_t *ctx,
|
|
void *rtp_hdr,
|
|
int *pkt_octet_len)
|
|
{
|
|
return srtp_protect_mki(ctx, rtp_hdr, pkt_octet_len, 0, 0);
|
|
}
|
|
|
|
srtp_err_status_t srtp_protect_mki(srtp_ctx_t *ctx,
|
|
void *rtp_hdr,
|
|
int *pkt_octet_len,
|
|
unsigned int use_mki,
|
|
unsigned int mki_index)
|
|
{
|
|
srtp_hdr_t *hdr = (srtp_hdr_t *)rtp_hdr;
|
|
uint32_t *enc_start; /* pointer to start of encrypted portion */
|
|
uint32_t *auth_start; /* pointer to start of auth. portion */
|
|
int enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
srtp_xtd_seq_num_t est; /* estimated xtd_seq_num_t of *hdr */
|
|
int delta; /* delta of local pkt idx and that in hdr */
|
|
uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
srtp_err_status_t status;
|
|
int tag_len;
|
|
srtp_stream_ctx_t *stream;
|
|
uint32_t prefix_len;
|
|
srtp_hdr_xtnd_t *xtn_hdr = NULL;
|
|
unsigned int mki_size = 0;
|
|
srtp_session_keys_t *session_keys = NULL;
|
|
uint8_t *mki_location = NULL;
|
|
int advance_packet_index = 0;
|
|
|
|
debug_print(mod_srtp, "function srtp_protect", NULL);
|
|
|
|
/* we assume the hdr is 32-bit aligned to start */
|
|
|
|
/* Verify RTP header */
|
|
status = srtp_validate_rtp_header(rtp_hdr, pkt_octet_len);
|
|
if (status)
|
|
return status;
|
|
|
|
/* check the packet length - it must at least contain a full header */
|
|
if (*pkt_octet_len < octets_in_rtp_header)
|
|
return srtp_err_status_bad_param;
|
|
|
|
/*
|
|
* look up ssrc in srtp_stream list, and process the packet with
|
|
* the appropriate stream. if we haven't seen this stream before,
|
|
* there's a template key for this srtp_session, and the cipher
|
|
* supports key-sharing, then we assume that a new stream using
|
|
* that key has just started up
|
|
*/
|
|
stream = srtp_get_stream(ctx, hdr->ssrc);
|
|
if (stream == NULL) {
|
|
if (ctx->stream_template != NULL) {
|
|
srtp_stream_ctx_t *new_stream;
|
|
|
|
/* allocate and initialize a new stream */
|
|
status =
|
|
srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream);
|
|
if (status)
|
|
return status;
|
|
|
|
/* add new stream to the head of the stream_list */
|
|
new_stream->next = ctx->stream_list;
|
|
ctx->stream_list = new_stream;
|
|
|
|
/* set direction to outbound */
|
|
new_stream->direction = dir_srtp_sender;
|
|
|
|
/* set stream (the pointer used in this function) */
|
|
stream = new_stream;
|
|
} else {
|
|
/* no template stream, so we return an error */
|
|
return srtp_err_status_no_ctx;
|
|
}
|
|
}
|
|
|
|
/*
|
|
* verify that stream is for sending traffic - this check will
|
|
* detect SSRC collisions, since a stream that appears in both
|
|
* srtp_protect() and srtp_unprotect() will fail this test in one of
|
|
* those functions.
|
|
*/
|
|
|
|
if (stream->direction != dir_srtp_sender) {
|
|
if (stream->direction == dir_unknown) {
|
|
stream->direction = dir_srtp_sender;
|
|
} else {
|
|
srtp_handle_event(ctx, stream, event_ssrc_collision);
|
|
}
|
|
}
|
|
|
|
session_keys =
|
|
srtp_get_session_keys_with_mki_index(stream, use_mki, mki_index);
|
|
|
|
if (session_keys == NULL)
|
|
return srtp_err_status_bad_mki;
|
|
|
|
/*
|
|
* Check if this is an AEAD stream (GCM mode). If so, then dispatch
|
|
* the request to our AEAD handler.
|
|
*/
|
|
if (session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_128 ||
|
|
session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_256) {
|
|
return srtp_protect_aead(ctx, stream, rtp_hdr,
|
|
(unsigned int *)pkt_octet_len, session_keys,
|
|
use_mki);
|
|
}
|
|
|
|
/*
|
|
* update the key usage limit, and check it to make sure that we
|
|
* didn't just hit either the soft limit or the hard limit, and call
|
|
* the event handler if we hit either.
|
|
*/
|
|
switch (srtp_key_limit_update(session_keys->limit)) {
|
|
case srtp_key_event_normal:
|
|
break;
|
|
case srtp_key_event_soft_limit:
|
|
srtp_handle_event(ctx, stream, event_key_soft_limit);
|
|
break;
|
|
case srtp_key_event_hard_limit:
|
|
srtp_handle_event(ctx, stream, event_key_hard_limit);
|
|
return srtp_err_status_key_expired;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* get tag length from stream */
|
|
tag_len = srtp_auth_get_tag_length(session_keys->rtp_auth);
|
|
|
|
/*
|
|
* find starting point for encryption and length of data to be
|
|
* encrypted - the encrypted portion starts after the rtp header
|
|
* extension, if present; otherwise, it starts after the last csrc,
|
|
* if any are present
|
|
*
|
|
* if we're not providing confidentiality, set enc_start to NULL
|
|
*/
|
|
if (stream->rtp_services & sec_serv_conf) {
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtp_header + hdr->cc;
|
|
if (hdr->x == 1) {
|
|
xtn_hdr = (srtp_hdr_xtnd_t *)enc_start;
|
|
enc_start += (ntohs(xtn_hdr->length) + 1);
|
|
}
|
|
/* note: the passed size is without the auth tag */
|
|
if (!((uint8_t *)enc_start <= (uint8_t *)hdr + *pkt_octet_len))
|
|
return srtp_err_status_parse_err;
|
|
enc_octet_len =
|
|
(int)(*pkt_octet_len - ((uint8_t *)enc_start - (uint8_t *)hdr));
|
|
if (enc_octet_len < 0)
|
|
return srtp_err_status_parse_err;
|
|
} else {
|
|
enc_start = NULL;
|
|
}
|
|
|
|
mki_location = (uint8_t *)hdr + *pkt_octet_len;
|
|
mki_size = srtp_inject_mki(mki_location, session_keys, use_mki);
|
|
|
|
/*
|
|
* if we're providing authentication, set the auth_start and auth_tag
|
|
* pointers to the proper locations; otherwise, set auth_start to NULL
|
|
* to indicate that no authentication is needed
|
|
*/
|
|
if (stream->rtp_services & sec_serv_auth) {
|
|
auth_start = (uint32_t *)hdr;
|
|
auth_tag = (uint8_t *)hdr + *pkt_octet_len + mki_size;
|
|
} else {
|
|
auth_start = NULL;
|
|
auth_tag = NULL;
|
|
}
|
|
|
|
/*
|
|
* estimate the packet index using the start of the replay window
|
|
* and the sequence number from the header
|
|
*/
|
|
status = srtp_get_est_pkt_index(hdr, stream, &est, &delta);
|
|
|
|
if (status && (status != srtp_err_status_pkt_idx_adv))
|
|
return status;
|
|
|
|
if (status == srtp_err_status_pkt_idx_adv)
|
|
advance_packet_index = 1;
|
|
|
|
if (advance_packet_index) {
|
|
srtp_rdbx_set_roc_seq(&stream->rtp_rdbx, (uint32_t)(est >> 16),
|
|
(uint16_t)(est & 0xFFFF));
|
|
stream->pending_roc = 0;
|
|
srtp_rdbx_add_index(&stream->rtp_rdbx, 0);
|
|
} else {
|
|
status = srtp_rdbx_check(&stream->rtp_rdbx, delta);
|
|
if (status) {
|
|
if (status != srtp_err_status_replay_fail ||
|
|
!stream->allow_repeat_tx)
|
|
return status; /* we've been asked to reuse an index */
|
|
}
|
|
srtp_rdbx_add_index(&stream->rtp_rdbx, delta);
|
|
}
|
|
|
|
#ifdef NO_64BIT_MATH
|
|
debug_print2(mod_srtp, "estimated packet index: %08x%08x", high32(est),
|
|
low32(est));
|
|
#else
|
|
debug_print(mod_srtp, "estimated packet index: %016llx", est);
|
|
#endif
|
|
|
|
/*
|
|
* if we're using rindael counter mode, set nonce and seq
|
|
*/
|
|
if (session_keys->rtp_cipher->type->id == SRTP_AES_ICM_128 ||
|
|
session_keys->rtp_cipher->type->id == SRTP_AES_ICM_192 ||
|
|
session_keys->rtp_cipher->type->id == SRTP_AES_ICM_256) {
|
|
v128_t iv;
|
|
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = hdr->ssrc;
|
|
#ifdef NO_64BIT_MATH
|
|
iv.v64[1] = be64_to_cpu(
|
|
make64((high32(est) << 16) | (low32(est) >> 16), low32(est) << 16));
|
|
#else
|
|
iv.v64[1] = be64_to_cpu(est << 16);
|
|
#endif
|
|
status = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_encrypt);
|
|
if (!status && session_keys->rtp_xtn_hdr_cipher) {
|
|
status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
|
|
(uint8_t *)&iv, srtp_direction_encrypt);
|
|
}
|
|
} else {
|
|
v128_t iv;
|
|
|
|
/* otherwise, set the index to est */
|
|
#ifdef NO_64BIT_MATH
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = 0;
|
|
#else
|
|
iv.v64[0] = 0;
|
|
#endif
|
|
iv.v64[1] = be64_to_cpu(est);
|
|
status = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_encrypt);
|
|
if (!status && session_keys->rtp_xtn_hdr_cipher) {
|
|
status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
|
|
(uint8_t *)&iv, srtp_direction_encrypt);
|
|
}
|
|
}
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
|
|
/* shift est, put into network byte order */
|
|
#ifdef NO_64BIT_MATH
|
|
est = be64_to_cpu(
|
|
make64((high32(est) << 16) | (low32(est) >> 16), low32(est) << 16));
|
|
#else
|
|
est = be64_to_cpu(est << 16);
|
|
#endif
|
|
|
|
/*
|
|
* if we're authenticating using a universal hash, put the keystream
|
|
* prefix into the authentication tag
|
|
*/
|
|
if (auth_start) {
|
|
prefix_len = srtp_auth_get_prefix_length(session_keys->rtp_auth);
|
|
if (prefix_len) {
|
|
status = srtp_cipher_output(session_keys->rtp_cipher, auth_tag,
|
|
&prefix_len);
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
debug_print(mod_srtp, "keystream prefix: %s",
|
|
srtp_octet_string_hex_string(auth_tag, prefix_len));
|
|
}
|
|
}
|
|
|
|
if (xtn_hdr && session_keys->rtp_xtn_hdr_cipher) {
|
|
/*
|
|
* extensions header encryption RFC 6904
|
|
*/
|
|
status = srtp_process_header_encryption(stream, xtn_hdr, session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
|
|
/* if we're encrypting, exor keystream into the message */
|
|
if (enc_start) {
|
|
status =
|
|
srtp_cipher_encrypt(session_keys->rtp_cipher, (uint8_t *)enc_start,
|
|
(unsigned int *)&enc_octet_len);
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/*
|
|
* if we're authenticating, run authentication function and put result
|
|
* into the auth_tag
|
|
*/
|
|
if (auth_start) {
|
|
/* initialize auth func context */
|
|
status = srtp_auth_start(session_keys->rtp_auth);
|
|
if (status)
|
|
return status;
|
|
|
|
/* run auth func over packet */
|
|
status = srtp_auth_update(session_keys->rtp_auth, (uint8_t *)auth_start,
|
|
*pkt_octet_len);
|
|
if (status)
|
|
return status;
|
|
|
|
/* run auth func over ROC, put result into auth_tag */
|
|
debug_print(mod_srtp, "estimated packet index: %016llx", est);
|
|
status = srtp_auth_compute(session_keys->rtp_auth, (uint8_t *)&est, 4,
|
|
auth_tag);
|
|
debug_print(mod_srtp, "srtp auth tag: %s",
|
|
srtp_octet_string_hex_string(auth_tag, tag_len));
|
|
if (status)
|
|
return srtp_err_status_auth_fail;
|
|
}
|
|
|
|
if (auth_tag) {
|
|
/* increase the packet length by the length of the auth tag */
|
|
*pkt_octet_len += tag_len;
|
|
}
|
|
|
|
if (use_mki) {
|
|
/* increate the packet length by the mki size */
|
|
*pkt_octet_len += mki_size;
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_unprotect(srtp_ctx_t *ctx,
|
|
void *srtp_hdr,
|
|
int *pkt_octet_len)
|
|
{
|
|
return srtp_unprotect_mki(ctx, srtp_hdr, pkt_octet_len, 0);
|
|
}
|
|
|
|
srtp_err_status_t srtp_unprotect_mki(srtp_ctx_t *ctx,
|
|
void *srtp_hdr,
|
|
int *pkt_octet_len,
|
|
unsigned int use_mki)
|
|
{
|
|
srtp_hdr_t *hdr = (srtp_hdr_t *)srtp_hdr;
|
|
uint32_t *enc_start; /* pointer to start of encrypted portion */
|
|
uint32_t *auth_start; /* pointer to start of auth. portion */
|
|
unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
srtp_xtd_seq_num_t est; /* estimated xtd_seq_num_t of *hdr */
|
|
int delta; /* delta of local pkt idx and that in hdr */
|
|
v128_t iv;
|
|
srtp_err_status_t status;
|
|
srtp_stream_ctx_t *stream;
|
|
uint8_t tmp_tag[SRTP_MAX_TAG_LEN];
|
|
uint32_t tag_len, prefix_len;
|
|
srtp_hdr_xtnd_t *xtn_hdr = NULL;
|
|
unsigned int mki_size = 0;
|
|
srtp_session_keys_t *session_keys = NULL;
|
|
int advance_packet_index = 0;
|
|
uint32_t roc_to_set = 0;
|
|
uint16_t seq_to_set = 0;
|
|
|
|
debug_print(mod_srtp, "function srtp_unprotect", NULL);
|
|
|
|
/* we assume the hdr is 32-bit aligned to start */
|
|
|
|
/* Verify RTP header */
|
|
status = srtp_validate_rtp_header(srtp_hdr, pkt_octet_len);
|
|
if (status)
|
|
return status;
|
|
|
|
/* check the packet length - it must at least contain a full header */
|
|
if (*pkt_octet_len < octets_in_rtp_header)
|
|
return srtp_err_status_bad_param;
|
|
|
|
/*
|
|
* look up ssrc in srtp_stream list, and process the packet with
|
|
* the appropriate stream. if we haven't seen this stream before,
|
|
* there's only one key for this srtp_session, and the cipher
|
|
* supports key-sharing, then we assume that a new stream using
|
|
* that key has just started up
|
|
*/
|
|
stream = srtp_get_stream(ctx, hdr->ssrc);
|
|
if (stream == NULL) {
|
|
if (ctx->stream_template != NULL) {
|
|
stream = ctx->stream_template;
|
|
debug_print(mod_srtp, "using provisional stream (SSRC: 0x%08x)",
|
|
ntohl(hdr->ssrc));
|
|
|
|
/*
|
|
* set estimated packet index to sequence number from header,
|
|
* and set delta equal to the same value
|
|
*/
|
|
#ifdef NO_64BIT_MATH
|
|
est = (srtp_xtd_seq_num_t)make64(0, ntohs(hdr->seq));
|
|
delta = low32(est);
|
|
#else
|
|
est = (srtp_xtd_seq_num_t)ntohs(hdr->seq);
|
|
delta = (int)est;
|
|
#endif
|
|
} else {
|
|
/*
|
|
* no stream corresponding to SSRC found, and we don't do
|
|
* key-sharing, so return an error
|
|
*/
|
|
return srtp_err_status_no_ctx;
|
|
}
|
|
} else {
|
|
status = srtp_get_est_pkt_index(hdr, stream, &est, &delta);
|
|
|
|
if (status && (status != srtp_err_status_pkt_idx_adv))
|
|
return status;
|
|
|
|
if (status == srtp_err_status_pkt_idx_adv) {
|
|
advance_packet_index = 1;
|
|
roc_to_set = (uint32_t)(est >> 16);
|
|
seq_to_set = (uint16_t)(est & 0xFFFF);
|
|
}
|
|
|
|
/* check replay database */
|
|
if (!advance_packet_index) {
|
|
status = srtp_rdbx_check(&stream->rtp_rdbx, delta);
|
|
if (status)
|
|
return status;
|
|
}
|
|
}
|
|
|
|
#ifdef NO_64BIT_MATH
|
|
debug_print2(mod_srtp, "estimated u_packet index: %08x%08x", high32(est),
|
|
low32(est));
|
|
#else
|
|
debug_print(mod_srtp, "estimated u_packet index: %016llx", est);
|
|
#endif
|
|
|
|
/* Determine if MKI is being used and what session keys should be used */
|
|
if (use_mki) {
|
|
session_keys = srtp_get_session_keys(
|
|
stream, (uint8_t *)hdr, (const unsigned int *)pkt_octet_len,
|
|
&mki_size);
|
|
|
|
if (session_keys == NULL)
|
|
return srtp_err_status_bad_mki;
|
|
} else {
|
|
session_keys = &stream->session_keys[0];
|
|
}
|
|
|
|
/*
|
|
* Check if this is an AEAD stream (GCM mode). If so, then dispatch
|
|
* the request to our AEAD handler.
|
|
*/
|
|
if (session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_128 ||
|
|
session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_256) {
|
|
return srtp_unprotect_aead(ctx, stream, delta, est, srtp_hdr,
|
|
(unsigned int *)pkt_octet_len, session_keys,
|
|
mki_size);
|
|
}
|
|
|
|
/* get tag length from stream */
|
|
tag_len = srtp_auth_get_tag_length(session_keys->rtp_auth);
|
|
|
|
/*
|
|
* set the cipher's IV properly, depending on whatever cipher we
|
|
* happen to be using
|
|
*/
|
|
if (session_keys->rtp_cipher->type->id == SRTP_AES_ICM_128 ||
|
|
session_keys->rtp_cipher->type->id == SRTP_AES_ICM_192 ||
|
|
session_keys->rtp_cipher->type->id == SRTP_AES_ICM_256) {
|
|
/* aes counter mode */
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = hdr->ssrc; /* still in network order */
|
|
#ifdef NO_64BIT_MATH
|
|
iv.v64[1] = be64_to_cpu(
|
|
make64((high32(est) << 16) | (low32(est) >> 16), low32(est) << 16));
|
|
#else
|
|
iv.v64[1] = be64_to_cpu(est << 16);
|
|
#endif
|
|
status = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_decrypt);
|
|
if (!status && session_keys->rtp_xtn_hdr_cipher) {
|
|
status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
|
|
(uint8_t *)&iv, srtp_direction_decrypt);
|
|
}
|
|
} else {
|
|
/* no particular format - set the iv to the pakcet index */
|
|
#ifdef NO_64BIT_MATH
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = 0;
|
|
#else
|
|
iv.v64[0] = 0;
|
|
#endif
|
|
iv.v64[1] = be64_to_cpu(est);
|
|
status = srtp_cipher_set_iv(session_keys->rtp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_decrypt);
|
|
if (!status && session_keys->rtp_xtn_hdr_cipher) {
|
|
status = srtp_cipher_set_iv(session_keys->rtp_xtn_hdr_cipher,
|
|
(uint8_t *)&iv, srtp_direction_decrypt);
|
|
}
|
|
}
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
|
|
/* shift est, put into network byte order */
|
|
#ifdef NO_64BIT_MATH
|
|
est = be64_to_cpu(
|
|
make64((high32(est) << 16) | (low32(est) >> 16), low32(est) << 16));
|
|
#else
|
|
est = be64_to_cpu(est << 16);
|
|
#endif
|
|
|
|
/*
|
|
* find starting point for decryption and length of data to be
|
|
* decrypted - the encrypted portion starts after the rtp header
|
|
* extension, if present; otherwise, it starts after the last csrc,
|
|
* if any are present
|
|
*
|
|
* if we're not providing confidentiality, set enc_start to NULL
|
|
*/
|
|
if (stream->rtp_services & sec_serv_conf) {
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtp_header + hdr->cc;
|
|
if (hdr->x == 1) {
|
|
xtn_hdr = (srtp_hdr_xtnd_t *)enc_start;
|
|
enc_start += (ntohs(xtn_hdr->length) + 1);
|
|
}
|
|
if (!((uint8_t *)enc_start <=
|
|
(uint8_t *)hdr + (*pkt_octet_len - tag_len - mki_size)))
|
|
return srtp_err_status_parse_err;
|
|
enc_octet_len = (uint32_t)(*pkt_octet_len - tag_len - mki_size -
|
|
((uint8_t *)enc_start - (uint8_t *)hdr));
|
|
} else {
|
|
enc_start = NULL;
|
|
}
|
|
|
|
/*
|
|
* if we're providing authentication, set the auth_start and auth_tag
|
|
* pointers to the proper locations; otherwise, set auth_start to NULL
|
|
* to indicate that no authentication is needed
|
|
*/
|
|
if (stream->rtp_services & sec_serv_auth) {
|
|
auth_start = (uint32_t *)hdr;
|
|
auth_tag = (uint8_t *)hdr + *pkt_octet_len - tag_len;
|
|
} else {
|
|
auth_start = NULL;
|
|
auth_tag = NULL;
|
|
}
|
|
|
|
/*
|
|
* if we expect message authentication, run the authentication
|
|
* function and compare the result with the value of the auth_tag
|
|
*/
|
|
if (auth_start) {
|
|
/*
|
|
* if we're using a universal hash, then we need to compute the
|
|
* keystream prefix for encrypting the universal hash output
|
|
*
|
|
* if the keystream prefix length is zero, then we know that
|
|
* the authenticator isn't using a universal hash function
|
|
*/
|
|
if (session_keys->rtp_auth->prefix_len != 0) {
|
|
prefix_len = srtp_auth_get_prefix_length(session_keys->rtp_auth);
|
|
status = srtp_cipher_output(session_keys->rtp_cipher, tmp_tag,
|
|
&prefix_len);
|
|
debug_print(mod_srtp, "keystream prefix: %s",
|
|
srtp_octet_string_hex_string(tmp_tag, prefix_len));
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/* initialize auth func context */
|
|
status = srtp_auth_start(session_keys->rtp_auth);
|
|
if (status)
|
|
return status;
|
|
|
|
/* now compute auth function over packet */
|
|
status = srtp_auth_update(session_keys->rtp_auth, (uint8_t *)auth_start,
|
|
*pkt_octet_len - tag_len - mki_size);
|
|
|
|
/* 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 (octet_string_is_eq(tmp_tag, auth_tag, tag_len))
|
|
return srtp_err_status_auth_fail;
|
|
}
|
|
|
|
/*
|
|
* update the key usage limit, and check it to make sure that we
|
|
* didn't just hit either the soft limit or the hard limit, and call
|
|
* the event handler if we hit either.
|
|
*/
|
|
switch (srtp_key_limit_update(session_keys->limit)) {
|
|
case srtp_key_event_normal:
|
|
break;
|
|
case srtp_key_event_soft_limit:
|
|
srtp_handle_event(ctx, stream, event_key_soft_limit);
|
|
break;
|
|
case srtp_key_event_hard_limit:
|
|
srtp_handle_event(ctx, stream, event_key_hard_limit);
|
|
return srtp_err_status_key_expired;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (xtn_hdr && session_keys->rtp_xtn_hdr_cipher) {
|
|
/* extensions header encryption RFC 6904 */
|
|
status = srtp_process_header_encryption(stream, xtn_hdr, session_keys);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
|
|
/* if we're decrypting, add keystream into ciphertext */
|
|
if (enc_start) {
|
|
status = srtp_cipher_decrypt(session_keys->rtp_cipher,
|
|
(uint8_t *)enc_start, &enc_octet_len);
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/*
|
|
* verify that stream is for received traffic - this check will
|
|
* detect SSRC collisions, since a stream that appears in both
|
|
* srtp_protect() and srtp_unprotect() will fail this test in one of
|
|
* those functions.
|
|
*
|
|
* we do this check *after* the authentication check, so that the
|
|
* latter check will catch any attempts to fool us into thinking
|
|
* that we've got a collision
|
|
*/
|
|
if (stream->direction != dir_srtp_receiver) {
|
|
if (stream->direction == dir_unknown) {
|
|
stream->direction = dir_srtp_receiver;
|
|
} else {
|
|
srtp_handle_event(ctx, stream, event_ssrc_collision);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* if the stream is a 'provisional' one, in which the template context
|
|
* is used, then we need to allocate a new stream at this point, since
|
|
* the authentication passed
|
|
*/
|
|
if (stream == ctx->stream_template) {
|
|
srtp_stream_ctx_t *new_stream;
|
|
|
|
/*
|
|
* allocate and initialize a new stream
|
|
*
|
|
* note that we indicate failure if we can't allocate the new
|
|
* stream, and some implementations will want to not return
|
|
* failure here
|
|
*/
|
|
status =
|
|
srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream);
|
|
if (status)
|
|
return status;
|
|
|
|
/* add new stream to the head of the stream_list */
|
|
new_stream->next = ctx->stream_list;
|
|
ctx->stream_list = new_stream;
|
|
|
|
/* set stream (the pointer used in this function) */
|
|
stream = new_stream;
|
|
}
|
|
|
|
/*
|
|
* the message authentication function passed, so add the packet
|
|
* index into the replay database
|
|
*/
|
|
if (advance_packet_index) {
|
|
srtp_rdbx_set_roc_seq(&stream->rtp_rdbx, roc_to_set, seq_to_set);
|
|
stream->pending_roc = 0;
|
|
srtp_rdbx_add_index(&stream->rtp_rdbx, 0);
|
|
} else {
|
|
srtp_rdbx_add_index(&stream->rtp_rdbx, delta);
|
|
}
|
|
|
|
/* decrease the packet length by the length of the auth tag */
|
|
*pkt_octet_len -= tag_len;
|
|
|
|
/* decrease the packet length by the mki size */
|
|
*pkt_octet_len -= mki_size;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_init()
|
|
{
|
|
srtp_err_status_t status;
|
|
|
|
/* initialize crypto kernel */
|
|
status = srtp_crypto_kernel_init();
|
|
if (status)
|
|
return status;
|
|
|
|
/* load srtp debug module into the kernel */
|
|
status = srtp_crypto_kernel_load_debug_module(&mod_srtp);
|
|
if (status)
|
|
return status;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_shutdown()
|
|
{
|
|
srtp_err_status_t status;
|
|
|
|
/* shut down crypto kernel */
|
|
status = srtp_crypto_kernel_shutdown();
|
|
if (status)
|
|
return status;
|
|
|
|
/* shutting down crypto kernel frees the srtp debug module as well */
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* The following code is under consideration for removal. See
|
|
* SRTP_MAX_TRAILER_LEN
|
|
*/
|
|
#if 0
|
|
|
|
/*
|
|
* srtp_get_trailer_length(&a) returns the number of octets that will
|
|
* be added to an RTP packet by the SRTP processing. This value
|
|
* is constant for a given srtp_stream_t (i.e. between initializations).
|
|
*/
|
|
|
|
int
|
|
srtp_get_trailer_length(const srtp_stream_t s) {
|
|
return srtp_auth_get_tag_length(s->rtp_auth);
|
|
}
|
|
|
|
#endif
|
|
|
|
/*
|
|
* srtp_get_stream(ssrc) returns a pointer to the stream corresponding
|
|
* to ssrc, or NULL if no stream exists for that ssrc
|
|
*
|
|
* this is an internal function
|
|
*/
|
|
|
|
srtp_stream_ctx_t *srtp_get_stream(srtp_t srtp, uint32_t ssrc)
|
|
{
|
|
srtp_stream_ctx_t *stream;
|
|
|
|
/* walk down list until ssrc is found */
|
|
stream = srtp->stream_list;
|
|
while (stream != NULL) {
|
|
if (stream->ssrc == ssrc)
|
|
return stream;
|
|
stream = stream->next;
|
|
}
|
|
|
|
/* we haven't found our ssrc, so return a null */
|
|
return NULL;
|
|
}
|
|
|
|
srtp_err_status_t srtp_dealloc(srtp_t session)
|
|
{
|
|
srtp_stream_ctx_t *stream;
|
|
srtp_err_status_t status;
|
|
|
|
/*
|
|
* we take a conservative deallocation strategy - if we encounter an
|
|
* error deallocating a stream, then we stop trying to deallocate
|
|
* memory and just return an error
|
|
*/
|
|
|
|
/* walk list of streams, deallocating as we go */
|
|
stream = session->stream_list;
|
|
while (stream != NULL) {
|
|
srtp_stream_t next = stream->next;
|
|
status = srtp_stream_dealloc(stream, session->stream_template);
|
|
if (status)
|
|
return status;
|
|
stream = next;
|
|
}
|
|
|
|
/* deallocate stream template, if there is one */
|
|
if (session->stream_template != NULL) {
|
|
status = srtp_stream_dealloc(session->stream_template, NULL);
|
|
if (status)
|
|
return status;
|
|
}
|
|
|
|
/* deallocate session context */
|
|
srtp_crypto_free(session);
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_add_stream(srtp_t session, const srtp_policy_t *policy)
|
|
{
|
|
srtp_err_status_t status;
|
|
srtp_stream_t tmp;
|
|
|
|
/* sanity check arguments */
|
|
if ((session == NULL) || (policy == NULL) ||
|
|
(!srtp_validate_policy_master_keys(policy)))
|
|
return srtp_err_status_bad_param;
|
|
|
|
/* allocate stream */
|
|
status = srtp_stream_alloc(&tmp, policy);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* initialize stream */
|
|
status = srtp_stream_init(tmp, policy);
|
|
if (status) {
|
|
srtp_stream_dealloc(tmp, NULL);
|
|
return status;
|
|
}
|
|
|
|
/*
|
|
* set the head of the stream list or the template to point to the
|
|
* stream that we've just alloced and init'ed, depending on whether
|
|
* or not it has a wildcard SSRC value or not
|
|
*
|
|
* if the template stream has already been set, then the policy is
|
|
* inconsistent, so we return a bad_param error code
|
|
*/
|
|
switch (policy->ssrc.type) {
|
|
case (ssrc_any_outbound):
|
|
if (session->stream_template) {
|
|
srtp_stream_dealloc(tmp, NULL);
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
session->stream_template = tmp;
|
|
session->stream_template->direction = dir_srtp_sender;
|
|
break;
|
|
case (ssrc_any_inbound):
|
|
if (session->stream_template) {
|
|
srtp_stream_dealloc(tmp, NULL);
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
session->stream_template = tmp;
|
|
session->stream_template->direction = dir_srtp_receiver;
|
|
break;
|
|
case (ssrc_specific):
|
|
tmp->next = session->stream_list;
|
|
session->stream_list = tmp;
|
|
break;
|
|
case (ssrc_undefined):
|
|
default:
|
|
srtp_stream_dealloc(tmp, NULL);
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_create(srtp_t *session, /* handle for session */
|
|
const srtp_policy_t *policy)
|
|
{ /* SRTP policy (list) */
|
|
srtp_err_status_t stat;
|
|
srtp_ctx_t *ctx;
|
|
|
|
/* sanity check arguments */
|
|
if (session == NULL)
|
|
return srtp_err_status_bad_param;
|
|
|
|
/* allocate srtp context and set ctx_ptr */
|
|
ctx = (srtp_ctx_t *)srtp_crypto_alloc(sizeof(srtp_ctx_t));
|
|
if (ctx == NULL)
|
|
return srtp_err_status_alloc_fail;
|
|
*session = ctx;
|
|
|
|
/*
|
|
* loop over elements in the policy list, allocating and
|
|
* initializing a stream for each element
|
|
*/
|
|
ctx->stream_template = NULL;
|
|
ctx->stream_list = NULL;
|
|
ctx->user_data = NULL;
|
|
while (policy != NULL) {
|
|
stat = srtp_add_stream(ctx, policy);
|
|
if (stat) {
|
|
/* clean up everything */
|
|
srtp_dealloc(*session);
|
|
*session = NULL;
|
|
return stat;
|
|
}
|
|
|
|
/* set policy to next item in list */
|
|
policy = policy->next;
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_remove_stream(srtp_t session, uint32_t ssrc)
|
|
{
|
|
srtp_stream_ctx_t *stream, *last_stream;
|
|
srtp_err_status_t status;
|
|
|
|
/* sanity check arguments */
|
|
if (session == NULL)
|
|
return srtp_err_status_bad_param;
|
|
|
|
/* find stream in list; complain if not found */
|
|
last_stream = stream = session->stream_list;
|
|
while ((stream != NULL) && (ssrc != stream->ssrc)) {
|
|
last_stream = stream;
|
|
stream = stream->next;
|
|
}
|
|
if (stream == NULL)
|
|
return srtp_err_status_no_ctx;
|
|
|
|
/* remove stream from the list */
|
|
if (last_stream == stream)
|
|
/* stream was first in list */
|
|
session->stream_list = stream->next;
|
|
else
|
|
last_stream->next = stream->next;
|
|
|
|
/* deallocate the stream */
|
|
status = srtp_stream_dealloc(stream, session->stream_template);
|
|
if (status)
|
|
return status;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_update(srtp_t session, const srtp_policy_t *policy)
|
|
{
|
|
srtp_err_status_t stat;
|
|
|
|
/* sanity check arguments */
|
|
if ((session == NULL) || (policy == NULL) ||
|
|
(!srtp_validate_policy_master_keys(policy))) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
while (policy != NULL) {
|
|
stat = srtp_update_stream(session, policy);
|
|
if (stat) {
|
|
return stat;
|
|
}
|
|
|
|
/* set policy to next item in list */
|
|
policy = policy->next;
|
|
}
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
static srtp_err_status_t update_template_streams(srtp_t session,
|
|
const srtp_policy_t *policy)
|
|
{
|
|
srtp_err_status_t status;
|
|
srtp_stream_t new_stream_template;
|
|
srtp_stream_t new_stream_list = NULL;
|
|
|
|
if (session->stream_template == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
/* allocate new template stream */
|
|
status = srtp_stream_alloc(&new_stream_template, policy);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* initialize new template stream */
|
|
status = srtp_stream_init(new_stream_template, policy);
|
|
if (status) {
|
|
srtp_crypto_free(new_stream_template);
|
|
return status;
|
|
}
|
|
|
|
/* for all old templated streams */
|
|
for (;;) {
|
|
srtp_stream_t stream;
|
|
uint32_t ssrc;
|
|
srtp_xtd_seq_num_t old_index;
|
|
srtp_rdb_t old_rtcp_rdb;
|
|
|
|
stream = session->stream_list;
|
|
while ((stream != NULL) &&
|
|
(stream->session_keys[0].rtp_auth !=
|
|
session->stream_template->session_keys[0].rtp_auth)) {
|
|
stream = stream->next;
|
|
}
|
|
if (stream == NULL) {
|
|
/* no more templated streams */
|
|
break;
|
|
}
|
|
|
|
/* save old extendard seq */
|
|
ssrc = stream->ssrc;
|
|
old_index = stream->rtp_rdbx.index;
|
|
old_rtcp_rdb = stream->rtcp_rdb;
|
|
|
|
/* remove stream */
|
|
status = srtp_remove_stream(session, ssrc);
|
|
if (status) {
|
|
/* free new allocations */
|
|
while (new_stream_list != NULL) {
|
|
srtp_stream_t next = new_stream_list->next;
|
|
srtp_stream_dealloc(new_stream_list, new_stream_template);
|
|
new_stream_list = next;
|
|
}
|
|
srtp_stream_dealloc(new_stream_template, NULL);
|
|
return status;
|
|
}
|
|
|
|
/* allocate and initialize a new stream */
|
|
status = srtp_stream_clone(new_stream_template, ssrc, &stream);
|
|
if (status) {
|
|
/* free new allocations */
|
|
while (new_stream_list != NULL) {
|
|
srtp_stream_t next = new_stream_list->next;
|
|
srtp_stream_dealloc(new_stream_list, new_stream_template);
|
|
new_stream_list = next;
|
|
}
|
|
srtp_stream_dealloc(new_stream_template, NULL);
|
|
return status;
|
|
}
|
|
|
|
/* add new stream to the head of the new_stream_list */
|
|
stream->next = new_stream_list;
|
|
new_stream_list = stream;
|
|
|
|
/* restore old extended seq */
|
|
stream->rtp_rdbx.index = old_index;
|
|
stream->rtcp_rdb = old_rtcp_rdb;
|
|
}
|
|
/* dealloc old template */
|
|
srtp_stream_dealloc(session->stream_template, NULL);
|
|
/* set new template */
|
|
session->stream_template = new_stream_template;
|
|
/* add new list */
|
|
if (new_stream_list) {
|
|
srtp_stream_t tail = new_stream_list;
|
|
while (tail->next) {
|
|
tail = tail->next;
|
|
}
|
|
tail->next = session->stream_list;
|
|
session->stream_list = new_stream_list;
|
|
}
|
|
return status;
|
|
}
|
|
|
|
static srtp_err_status_t update_stream(srtp_t session,
|
|
const srtp_policy_t *policy)
|
|
{
|
|
srtp_err_status_t status;
|
|
srtp_xtd_seq_num_t old_index;
|
|
srtp_rdb_t old_rtcp_rdb;
|
|
srtp_stream_t stream;
|
|
|
|
stream = srtp_get_stream(session, htonl(policy->ssrc.value));
|
|
if (stream == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
/* save old extendard seq */
|
|
old_index = stream->rtp_rdbx.index;
|
|
old_rtcp_rdb = stream->rtcp_rdb;
|
|
|
|
status = srtp_remove_stream(session, htonl(policy->ssrc.value));
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
status = srtp_add_stream(session, policy);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
stream = srtp_get_stream(session, htonl(policy->ssrc.value));
|
|
if (stream == NULL) {
|
|
return srtp_err_status_fail;
|
|
}
|
|
|
|
/* restore old extended seq */
|
|
stream->rtp_rdbx.index = old_index;
|
|
stream->rtcp_rdb = old_rtcp_rdb;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_update_stream(srtp_t session,
|
|
const srtp_policy_t *policy)
|
|
{
|
|
srtp_err_status_t status;
|
|
|
|
/* sanity check arguments */
|
|
if ((session == NULL) || (policy == NULL) ||
|
|
(!srtp_validate_policy_master_keys(policy)))
|
|
return srtp_err_status_bad_param;
|
|
|
|
switch (policy->ssrc.type) {
|
|
case (ssrc_any_outbound):
|
|
case (ssrc_any_inbound):
|
|
status = update_template_streams(session, policy);
|
|
break;
|
|
case (ssrc_specific):
|
|
status = update_stream(session, policy);
|
|
break;
|
|
case (ssrc_undefined):
|
|
default:
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
return status;
|
|
}
|
|
|
|
/*
|
|
* The default policy - provides a convenient way for callers to use
|
|
* the default security policy
|
|
*
|
|
* The default policy is defined in RFC 3711
|
|
* (Section 5. Default and mandatory-to-implement Transforms)
|
|
*
|
|
*/
|
|
|
|
/*
|
|
* NOTE: cipher_key_len is really key len (128 bits) plus salt len
|
|
* (112 bits)
|
|
*/
|
|
/* There are hard-coded 16's for base_key_len in the key generation code */
|
|
|
|
void srtp_crypto_policy_set_rtp_default(srtp_crypto_policy_t *p)
|
|
{
|
|
p->cipher_type = SRTP_AES_ICM_128;
|
|
p->cipher_key_len =
|
|
SRTP_AES_ICM_128_KEY_LEN_WSALT; /* default 128 bits per RFC 3711 */
|
|
p->auth_type = SRTP_HMAC_SHA1;
|
|
p->auth_key_len = 20; /* default 160 bits per RFC 3711 */
|
|
p->auth_tag_len = 10; /* default 80 bits per RFC 3711 */
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
}
|
|
|
|
void srtp_crypto_policy_set_rtcp_default(srtp_crypto_policy_t *p)
|
|
{
|
|
p->cipher_type = SRTP_AES_ICM_128;
|
|
p->cipher_key_len =
|
|
SRTP_AES_ICM_128_KEY_LEN_WSALT; /* default 128 bits per RFC 3711 */
|
|
p->auth_type = SRTP_HMAC_SHA1;
|
|
p->auth_key_len = 20; /* default 160 bits per RFC 3711 */
|
|
p->auth_tag_len = 10; /* default 80 bits per RFC 3711 */
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
}
|
|
|
|
void srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(srtp_crypto_policy_t *p)
|
|
{
|
|
/*
|
|
* corresponds to RFC 4568
|
|
*
|
|
* note that this crypto policy is intended for SRTP, but not SRTCP
|
|
*/
|
|
|
|
p->cipher_type = SRTP_AES_ICM_128;
|
|
p->cipher_key_len =
|
|
SRTP_AES_ICM_128_KEY_LEN_WSALT; /* 128 bit key, 112 bit salt */
|
|
p->auth_type = SRTP_HMAC_SHA1;
|
|
p->auth_key_len = 20; /* 160 bit key */
|
|
p->auth_tag_len = 4; /* 32 bit tag */
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
}
|
|
|
|
void srtp_crypto_policy_set_aes_cm_128_null_auth(srtp_crypto_policy_t *p)
|
|
{
|
|
/*
|
|
* corresponds to RFC 4568
|
|
*
|
|
* note that this crypto policy is intended for SRTP, but not SRTCP
|
|
*/
|
|
|
|
p->cipher_type = SRTP_AES_ICM_128;
|
|
p->cipher_key_len =
|
|
SRTP_AES_ICM_128_KEY_LEN_WSALT; /* 128 bit key, 112 bit salt */
|
|
p->auth_type = SRTP_NULL_AUTH;
|
|
p->auth_key_len = 0;
|
|
p->auth_tag_len = 0;
|
|
p->sec_serv = sec_serv_conf;
|
|
}
|
|
|
|
void srtp_crypto_policy_set_null_cipher_hmac_sha1_80(srtp_crypto_policy_t *p)
|
|
{
|
|
/*
|
|
* corresponds to RFC 4568
|
|
*/
|
|
|
|
p->cipher_type = SRTP_NULL_CIPHER;
|
|
p->cipher_key_len = 0;
|
|
p->auth_type = SRTP_HMAC_SHA1;
|
|
p->auth_key_len = 20;
|
|
p->auth_tag_len = 10;
|
|
p->sec_serv = sec_serv_auth;
|
|
}
|
|
|
|
void srtp_crypto_policy_set_null_cipher_hmac_null(srtp_crypto_policy_t *p)
|
|
{
|
|
/*
|
|
* Should only be used for testing
|
|
*/
|
|
|
|
p->cipher_type = SRTP_NULL_CIPHER;
|
|
p->cipher_key_len = 0;
|
|
p->auth_type = SRTP_NULL_AUTH;
|
|
p->auth_key_len = 0;
|
|
p->auth_tag_len = 0;
|
|
p->sec_serv = sec_serv_none;
|
|
}
|
|
|
|
void srtp_crypto_policy_set_aes_cm_256_hmac_sha1_80(srtp_crypto_policy_t *p)
|
|
{
|
|
/*
|
|
* corresponds to 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;
|
|
}
|
|
|
|
#ifdef OPENSSL
|
|
void srtp_crypto_policy_set_aes_cm_192_hmac_sha1_80(srtp_crypto_policy_t *p)
|
|
{
|
|
/*
|
|
* corresponds to RFC 6188
|
|
*/
|
|
|
|
p->cipher_type = SRTP_AES_ICM_192;
|
|
p->cipher_key_len = SRTP_AES_ICM_192_KEY_LEN_WSALT;
|
|
p->auth_type = SRTP_HMAC_SHA1;
|
|
p->auth_key_len = 20; /* default 160 bits per RFC 3711 */
|
|
p->auth_tag_len = 10; /* default 80 bits per RFC 3711 */
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
}
|
|
|
|
void srtp_crypto_policy_set_aes_cm_192_hmac_sha1_32(srtp_crypto_policy_t *p)
|
|
{
|
|
/*
|
|
* corresponds to RFC 6188
|
|
*
|
|
* note that this crypto policy is intended for SRTP, but not SRTCP
|
|
*/
|
|
|
|
p->cipher_type = SRTP_AES_ICM_192;
|
|
p->cipher_key_len = SRTP_AES_ICM_192_KEY_LEN_WSALT;
|
|
p->auth_type = SRTP_HMAC_SHA1;
|
|
p->auth_key_len = 20; /* default 160 bits per RFC 3711 */
|
|
p->auth_tag_len = 4; /* default 80 bits per RFC 3711 */
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
}
|
|
|
|
/*
|
|
* AES-192 with no authentication.
|
|
*/
|
|
void srtp_crypto_policy_set_aes_cm_192_null_auth(srtp_crypto_policy_t *p)
|
|
{
|
|
p->cipher_type = SRTP_AES_ICM_192;
|
|
p->cipher_key_len = SRTP_AES_ICM_192_KEY_LEN_WSALT;
|
|
p->auth_type = SRTP_NULL_AUTH;
|
|
p->auth_key_len = 0;
|
|
p->auth_tag_len = 0;
|
|
p->sec_serv = sec_serv_conf;
|
|
}
|
|
|
|
/*
|
|
* AES-128 GCM mode with 8 octet auth tag.
|
|
*/
|
|
void srtp_crypto_policy_set_aes_gcm_128_8_auth(srtp_crypto_policy_t *p)
|
|
{
|
|
p->cipher_type = SRTP_AES_GCM_128;
|
|
p->cipher_key_len = SRTP_AES_GCM_128_KEY_LEN_WSALT;
|
|
p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */
|
|
p->auth_key_len = 0;
|
|
p->auth_tag_len = 8; /* 8 octet tag length */
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
}
|
|
|
|
/*
|
|
* AES-256 GCM mode with 8 octet auth tag.
|
|
*/
|
|
void srtp_crypto_policy_set_aes_gcm_256_8_auth(srtp_crypto_policy_t *p)
|
|
{
|
|
p->cipher_type = SRTP_AES_GCM_256;
|
|
p->cipher_key_len = SRTP_AES_GCM_256_KEY_LEN_WSALT;
|
|
p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */
|
|
p->auth_key_len = 0;
|
|
p->auth_tag_len = 8; /* 8 octet tag length */
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
}
|
|
|
|
/*
|
|
* AES-128 GCM mode with 8 octet auth tag, no RTCP encryption.
|
|
*/
|
|
void srtp_crypto_policy_set_aes_gcm_128_8_only_auth(srtp_crypto_policy_t *p)
|
|
{
|
|
p->cipher_type = SRTP_AES_GCM_128;
|
|
p->cipher_key_len = SRTP_AES_GCM_128_KEY_LEN_WSALT;
|
|
p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */
|
|
p->auth_key_len = 0;
|
|
p->auth_tag_len = 8; /* 8 octet tag length */
|
|
p->sec_serv = sec_serv_auth; /* This only applies to RTCP */
|
|
}
|
|
|
|
/*
|
|
* AES-256 GCM mode with 8 octet auth tag, no RTCP encryption.
|
|
*/
|
|
void srtp_crypto_policy_set_aes_gcm_256_8_only_auth(srtp_crypto_policy_t *p)
|
|
{
|
|
p->cipher_type = SRTP_AES_GCM_256;
|
|
p->cipher_key_len = SRTP_AES_GCM_256_KEY_LEN_WSALT;
|
|
p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */
|
|
p->auth_key_len = 0;
|
|
p->auth_tag_len = 8; /* 8 octet tag length */
|
|
p->sec_serv = sec_serv_auth; /* This only applies to RTCP */
|
|
}
|
|
|
|
/*
|
|
* AES-128 GCM mode with 16 octet auth tag.
|
|
*/
|
|
void srtp_crypto_policy_set_aes_gcm_128_16_auth(srtp_crypto_policy_t *p)
|
|
{
|
|
p->cipher_type = SRTP_AES_GCM_128;
|
|
p->cipher_key_len = SRTP_AES_GCM_128_KEY_LEN_WSALT;
|
|
p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */
|
|
p->auth_key_len = 0;
|
|
p->auth_tag_len = 16; /* 16 octet tag length */
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
}
|
|
|
|
/*
|
|
* AES-256 GCM mode with 16 octet auth tag.
|
|
*/
|
|
void srtp_crypto_policy_set_aes_gcm_256_16_auth(srtp_crypto_policy_t *p)
|
|
{
|
|
p->cipher_type = SRTP_AES_GCM_256;
|
|
p->cipher_key_len = SRTP_AES_GCM_256_KEY_LEN_WSALT;
|
|
p->auth_type = SRTP_NULL_AUTH; /* GCM handles the auth for us */
|
|
p->auth_key_len = 0;
|
|
p->auth_tag_len = 16; /* 16 octet tag length */
|
|
p->sec_serv = sec_serv_conf_and_auth;
|
|
}
|
|
|
|
#endif
|
|
|
|
/*
|
|
* secure rtcp functions
|
|
*/
|
|
|
|
/*
|
|
* AEAD uses a new IV formation method. This function implements
|
|
* section 9.1 (SRTCP IV Formation for AES-GCM) from RFC7714.
|
|
* The calculation is defined as, where (+) is the xor operation:
|
|
*
|
|
* 0 1 2 3 4 5 6 7 8 9 10 11
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+
|
|
* |00|00| SSRC |00|00|0+SRTCP Idx|---+
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+ |
|
|
* |
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+ |
|
|
* | Encryption Salt |->(+)
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+ |
|
|
* |
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+ |
|
|
* | Initialization Vector |<--+
|
|
* +--+--+--+--+--+--+--+--+--+--+--+--+*
|
|
*
|
|
* Input: *session_keys - pointer to SRTP stream context session keys,
|
|
* used to retrieve the SALT
|
|
* *iv - Pointer to recieve the calculated IV
|
|
* seq_num - The SEQ value to use for the IV calculation.
|
|
* *hdr - The RTP header, used to get the SSRC value
|
|
*
|
|
* Returns: srtp_err_status_ok if no error or srtp_err_status_bad_param
|
|
* if seq_num is invalid
|
|
*
|
|
*/
|
|
static srtp_err_status_t srtp_calc_aead_iv_srtcp(
|
|
srtp_session_keys_t *session_keys,
|
|
v128_t *iv,
|
|
uint32_t seq_num,
|
|
srtcp_hdr_t *hdr)
|
|
{
|
|
v128_t in;
|
|
v128_t salt;
|
|
|
|
memset(&in, 0, sizeof(v128_t));
|
|
memset(&salt, 0, sizeof(v128_t));
|
|
|
|
in.v16[0] = 0;
|
|
memcpy(&in.v16[1], &hdr->ssrc, 4); /* still in network order! */
|
|
in.v16[3] = 0;
|
|
|
|
/*
|
|
* The SRTCP index (seq_num) spans bits 0 through 30 inclusive.
|
|
* The most significant bit should be zero.
|
|
*/
|
|
if (seq_num & 0x80000000UL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
in.v32[2] = htonl(seq_num);
|
|
|
|
debug_print(mod_srtp, "Pre-salted RTCP IV = %s\n", v128_hex_string(&in));
|
|
|
|
/*
|
|
* Get the SALT value from the context
|
|
*/
|
|
memcpy(salt.v8, session_keys->c_salt, 12);
|
|
debug_print(mod_srtp, "RTCP SALT = %s\n", v128_hex_string(&salt));
|
|
|
|
/*
|
|
* Finally, apply the SALT to the input
|
|
*/
|
|
v128_xor(iv, &in, &salt);
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* This code handles AEAD ciphers for outgoing RTCP. We currently support
|
|
* AES-GCM mode with 128 or 256 bit keys.
|
|
*/
|
|
static srtp_err_status_t srtp_protect_rtcp_aead(
|
|
srtp_t ctx,
|
|
srtp_stream_ctx_t *stream,
|
|
void *rtcp_hdr,
|
|
unsigned int *pkt_octet_len,
|
|
srtp_session_keys_t *session_keys,
|
|
unsigned int use_mki)
|
|
{
|
|
srtcp_hdr_t *hdr = (srtcp_hdr_t *)rtcp_hdr;
|
|
uint32_t *enc_start; /* pointer to start of encrypted portion */
|
|
uint32_t *trailer; /* pointer to start of trailer */
|
|
unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
srtp_err_status_t status;
|
|
uint32_t tag_len;
|
|
uint32_t seq_num;
|
|
v128_t iv;
|
|
uint32_t tseq;
|
|
unsigned int mki_size = 0;
|
|
|
|
/* get tag length from stream context */
|
|
tag_len = srtp_auth_get_tag_length(session_keys->rtcp_auth);
|
|
|
|
/*
|
|
* set encryption start and encryption length - if we're not
|
|
* providing confidentiality, set enc_start to NULL
|
|
*/
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtcp_header;
|
|
enc_octet_len = *pkt_octet_len - octets_in_rtcp_header;
|
|
|
|
/* NOTE: hdr->length is not usable - it refers to only the first
|
|
* RTCP report in the compound packet!
|
|
*/
|
|
/* NOTE: trailer is 32-bit aligned because RTCP 'packets' are always
|
|
* multiples of 32-bits (RFC 3550 6.1)
|
|
*/
|
|
trailer = (uint32_t *)((char *)enc_start + enc_octet_len + tag_len);
|
|
|
|
if (stream->rtcp_services & sec_serv_conf) {
|
|
*trailer = htonl(SRTCP_E_BIT); /* set encrypt bit */
|
|
} else {
|
|
enc_start = NULL;
|
|
enc_octet_len = 0;
|
|
/* 0 is network-order independant */
|
|
*trailer = 0x00000000; /* set encrypt bit */
|
|
}
|
|
|
|
mki_size = srtp_inject_mki((uint8_t *)hdr + *pkt_octet_len + tag_len +
|
|
sizeof(srtcp_trailer_t),
|
|
session_keys, use_mki);
|
|
|
|
/*
|
|
* set the auth_tag pointer to the proper location, which is after
|
|
* the payload, but before the trailer
|
|
* (note that srtpc *always* provides authentication, unlike srtp)
|
|
*/
|
|
/* Note: This would need to change for optional mikey data */
|
|
auth_tag = (uint8_t *)hdr + *pkt_octet_len;
|
|
|
|
/*
|
|
* check sequence number for overruns, and copy it into the packet
|
|
* if its value isn't too big
|
|
*/
|
|
status = srtp_rdb_increment(&stream->rtcp_rdb);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
seq_num = srtp_rdb_get_value(&stream->rtcp_rdb);
|
|
*trailer |= htonl(seq_num);
|
|
debug_print(mod_srtp, "srtcp index: %x", seq_num);
|
|
|
|
/*
|
|
* Calculate and set the IV
|
|
*/
|
|
status = srtp_calc_aead_iv_srtcp(session_keys, &iv, seq_num, hdr);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
status = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_encrypt);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/*
|
|
* Set the AAD for GCM mode
|
|
*/
|
|
if (enc_start) {
|
|
/*
|
|
* If payload encryption is enabled, then the AAD consist of
|
|
* the RTCP header and the seq# at the end of the packet
|
|
*/
|
|
status = srtp_cipher_set_aad(session_keys->rtcp_cipher, (uint8_t *)hdr,
|
|
octets_in_rtcp_header);
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
} else {
|
|
/*
|
|
* Since payload encryption is not enabled, we must authenticate
|
|
* the entire packet as described in RFC 7714 (Section 9.3. Data
|
|
* Types in Unencrypted SRTCP Compound Packets)
|
|
*/
|
|
status = srtp_cipher_set_aad(session_keys->rtcp_cipher, (uint8_t *)hdr,
|
|
*pkt_octet_len);
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
}
|
|
/*
|
|
* Process the sequence# as AAD
|
|
*/
|
|
tseq = *trailer;
|
|
status = srtp_cipher_set_aad(session_keys->rtcp_cipher, (uint8_t *)&tseq,
|
|
sizeof(srtcp_trailer_t));
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
|
|
/* if we're encrypting, exor keystream into the message */
|
|
if (enc_start) {
|
|
status = srtp_cipher_encrypt(session_keys->rtcp_cipher,
|
|
(uint8_t *)enc_start, &enc_octet_len);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
/*
|
|
* Get the tag and append that to the output
|
|
*/
|
|
status = srtp_cipher_get_tag(session_keys->rtcp_cipher,
|
|
(uint8_t *)auth_tag, &tag_len);
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
enc_octet_len += tag_len;
|
|
} else {
|
|
/*
|
|
* Even though we're not encrypting the payload, we need
|
|
* to run the cipher to get the auth tag.
|
|
*/
|
|
unsigned int nolen = 0;
|
|
status = srtp_cipher_encrypt(session_keys->rtcp_cipher, NULL, &nolen);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
/*
|
|
* Get the tag and append that to the output
|
|
*/
|
|
status = srtp_cipher_get_tag(session_keys->rtcp_cipher,
|
|
(uint8_t *)auth_tag, &tag_len);
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
enc_octet_len += tag_len;
|
|
}
|
|
|
|
/* increase the packet length by the length of the auth tag and seq_num*/
|
|
*pkt_octet_len += (tag_len + sizeof(srtcp_trailer_t));
|
|
|
|
/* increase the packet by the mki_size */
|
|
*pkt_octet_len += mki_size;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* This function handles incoming SRTCP packets while in AEAD mode,
|
|
* which currently supports AES-GCM encryption. Note, the auth tag is
|
|
* at the end of the packet stream and is automatically checked by GCM
|
|
* when decrypting the payload.
|
|
*/
|
|
static srtp_err_status_t srtp_unprotect_rtcp_aead(
|
|
srtp_t ctx,
|
|
srtp_stream_ctx_t *stream,
|
|
void *srtcp_hdr,
|
|
unsigned int *pkt_octet_len,
|
|
srtp_session_keys_t *session_keys,
|
|
unsigned int use_mki)
|
|
{
|
|
srtcp_hdr_t *hdr = (srtcp_hdr_t *)srtcp_hdr;
|
|
uint32_t *enc_start; /* pointer to start of encrypted portion */
|
|
uint32_t *trailer; /* pointer to start of trailer */
|
|
unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
srtp_err_status_t status;
|
|
int tag_len;
|
|
unsigned int tmp_len;
|
|
uint32_t seq_num;
|
|
v128_t iv;
|
|
uint32_t tseq;
|
|
unsigned int mki_size = 0;
|
|
|
|
/* get tag length from stream context */
|
|
tag_len = srtp_auth_get_tag_length(session_keys->rtcp_auth);
|
|
|
|
if (use_mki) {
|
|
mki_size = session_keys->mki_size;
|
|
}
|
|
|
|
/*
|
|
* set encryption start, encryption length, and trailer
|
|
*/
|
|
/* index & E (encryption) bit follow normal data. hdr->len is the number of
|
|
* words (32-bit) in the normal packet minus 1
|
|
*/
|
|
/* This should point trailer to the word past the end of the normal data. */
|
|
/* This would need to be modified for optional mikey data */
|
|
/*
|
|
* NOTE: trailer is 32-bit aligned because RTCP 'packets' are always
|
|
* multiples of 32-bits (RFC 3550 6.1)
|
|
*/
|
|
trailer = (uint32_t *)((char *)hdr + *pkt_octet_len -
|
|
sizeof(srtcp_trailer_t) - mki_size);
|
|
/*
|
|
* We pass the tag down to the cipher when doing GCM mode
|
|
*/
|
|
enc_octet_len = *pkt_octet_len - (octets_in_rtcp_header +
|
|
sizeof(srtcp_trailer_t) + mki_size);
|
|
auth_tag = (uint8_t *)hdr + *pkt_octet_len - tag_len - mki_size -
|
|
sizeof(srtcp_trailer_t);
|
|
|
|
if (*((unsigned char *)trailer) & SRTCP_E_BYTE_BIT) {
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtcp_header;
|
|
} else {
|
|
enc_octet_len = 0;
|
|
enc_start = NULL; /* this indicates that there's no encryption */
|
|
}
|
|
|
|
/*
|
|
* check the sequence number for replays
|
|
*/
|
|
/* this is easier than dealing with bitfield access */
|
|
seq_num = ntohl(*trailer) & SRTCP_INDEX_MASK;
|
|
debug_print(mod_srtp, "srtcp index: %x", seq_num);
|
|
status = srtp_rdb_check(&stream->rtcp_rdb, seq_num);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/*
|
|
* Calculate and set the IV
|
|
*/
|
|
status = srtp_calc_aead_iv_srtcp(session_keys, &iv, seq_num, hdr);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
status = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_decrypt);
|
|
if (status) {
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/*
|
|
* Set the AAD for GCM mode
|
|
*/
|
|
if (enc_start) {
|
|
/*
|
|
* If payload encryption is enabled, then the AAD consist of
|
|
* the RTCP header and the seq# at the end of the packet
|
|
*/
|
|
status = srtp_cipher_set_aad(session_keys->rtcp_cipher, (uint8_t *)hdr,
|
|
octets_in_rtcp_header);
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
} else {
|
|
/*
|
|
* Since payload encryption is not enabled, we must authenticate
|
|
* the entire packet as described in RFC 7714 (Section 9.3. Data
|
|
* Types in Unencrypted SRTCP Compound Packets)
|
|
*/
|
|
status = srtp_cipher_set_aad(
|
|
session_keys->rtcp_cipher, (uint8_t *)hdr,
|
|
(*pkt_octet_len - tag_len - sizeof(srtcp_trailer_t) - mki_size));
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Process the sequence# as AAD
|
|
*/
|
|
tseq = *trailer;
|
|
status = srtp_cipher_set_aad(session_keys->rtcp_cipher, (uint8_t *)&tseq,
|
|
sizeof(srtcp_trailer_t));
|
|
if (status) {
|
|
return (srtp_err_status_cipher_fail);
|
|
}
|
|
|
|
/* if we're decrypting, exor keystream into the message */
|
|
if (enc_start) {
|
|
status = srtp_cipher_decrypt(session_keys->rtcp_cipher,
|
|
(uint8_t *)enc_start, &enc_octet_len);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
} else {
|
|
/*
|
|
* Still need to run the cipher to check the tag
|
|
*/
|
|
tmp_len = tag_len;
|
|
status = srtp_cipher_decrypt(session_keys->rtcp_cipher,
|
|
(uint8_t *)auth_tag, &tmp_len);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
}
|
|
|
|
/* decrease the packet length by the length of the auth tag and seq_num*/
|
|
*pkt_octet_len -= (tag_len + sizeof(srtcp_trailer_t) + mki_size);
|
|
|
|
/*
|
|
* verify that stream is for received traffic - this check will
|
|
* detect SSRC collisions, since a stream that appears in both
|
|
* srtp_protect() and srtp_unprotect() will fail this test in one of
|
|
* those functions.
|
|
*
|
|
* we do this check *after* the authentication check, so that the
|
|
* latter check will catch any attempts to fool us into thinking
|
|
* that we've got a collision
|
|
*/
|
|
if (stream->direction != dir_srtp_receiver) {
|
|
if (stream->direction == dir_unknown) {
|
|
stream->direction = dir_srtp_receiver;
|
|
} else {
|
|
srtp_handle_event(ctx, stream, event_ssrc_collision);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* if the stream is a 'provisional' one, in which the template context
|
|
* is used, then we need to allocate a new stream at this point, since
|
|
* the authentication passed
|
|
*/
|
|
if (stream == ctx->stream_template) {
|
|
srtp_stream_ctx_t *new_stream;
|
|
|
|
/*
|
|
* allocate and initialize a new stream
|
|
*
|
|
* note that we indicate failure if we can't allocate the new
|
|
* stream, and some implementations will want to not return
|
|
* failure here
|
|
*/
|
|
status =
|
|
srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream);
|
|
if (status) {
|
|
return status;
|
|
}
|
|
|
|
/* add new stream to the head of the stream_list */
|
|
new_stream->next = ctx->stream_list;
|
|
ctx->stream_list = new_stream;
|
|
|
|
/* set stream (the pointer used in this function) */
|
|
stream = new_stream;
|
|
}
|
|
|
|
/* we've passed the authentication check, so add seq_num to the rdb */
|
|
srtp_rdb_add_index(&stream->rtcp_rdb, seq_num);
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_protect_rtcp(srtp_t ctx,
|
|
void *rtcp_hdr,
|
|
int *pkt_octet_len)
|
|
{
|
|
return srtp_protect_rtcp_mki(ctx, rtcp_hdr, pkt_octet_len, 0, 0);
|
|
}
|
|
|
|
srtp_err_status_t srtp_protect_rtcp_mki(srtp_t ctx,
|
|
void *rtcp_hdr,
|
|
int *pkt_octet_len,
|
|
unsigned int use_mki,
|
|
unsigned int mki_index)
|
|
{
|
|
srtcp_hdr_t *hdr = (srtcp_hdr_t *)rtcp_hdr;
|
|
uint32_t *enc_start; /* pointer to start of encrypted portion */
|
|
uint32_t *auth_start; /* pointer to start of auth. portion */
|
|
uint32_t *trailer; /* pointer to start of trailer */
|
|
unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
srtp_err_status_t status;
|
|
int tag_len;
|
|
srtp_stream_ctx_t *stream;
|
|
uint32_t prefix_len;
|
|
uint32_t seq_num;
|
|
unsigned int mki_size = 0;
|
|
srtp_session_keys_t *session_keys = NULL;
|
|
|
|
/* we assume the hdr is 32-bit aligned to start */
|
|
|
|
/* check the packet length - it must at least contain a full header */
|
|
if (*pkt_octet_len < octets_in_rtcp_header)
|
|
return srtp_err_status_bad_param;
|
|
|
|
/*
|
|
* look up ssrc in srtp_stream list, and process the packet with
|
|
* the appropriate stream. if we haven't seen this stream before,
|
|
* there's only one key for this srtp_session, and the cipher
|
|
* supports key-sharing, then we assume that a new stream using
|
|
* that key has just started up
|
|
*/
|
|
stream = srtp_get_stream(ctx, hdr->ssrc);
|
|
if (stream == NULL) {
|
|
if (ctx->stream_template != NULL) {
|
|
srtp_stream_ctx_t *new_stream;
|
|
|
|
/* allocate and initialize a new stream */
|
|
status =
|
|
srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream);
|
|
if (status)
|
|
return status;
|
|
|
|
/* add new stream to the head of the stream_list */
|
|
new_stream->next = ctx->stream_list;
|
|
ctx->stream_list = new_stream;
|
|
|
|
/* set stream (the pointer used in this function) */
|
|
stream = new_stream;
|
|
} else {
|
|
/* no template stream, so we return an error */
|
|
return srtp_err_status_no_ctx;
|
|
}
|
|
}
|
|
|
|
/*
|
|
* verify that stream is for sending traffic - this check will
|
|
* detect SSRC collisions, since a stream that appears in both
|
|
* srtp_protect() and srtp_unprotect() will fail this test in one of
|
|
* those functions.
|
|
*/
|
|
if (stream->direction != dir_srtp_sender) {
|
|
if (stream->direction == dir_unknown) {
|
|
stream->direction = dir_srtp_sender;
|
|
} else {
|
|
srtp_handle_event(ctx, stream, event_ssrc_collision);
|
|
}
|
|
}
|
|
|
|
session_keys =
|
|
srtp_get_session_keys_with_mki_index(stream, use_mki, mki_index);
|
|
|
|
if (session_keys == NULL)
|
|
return srtp_err_status_bad_mki;
|
|
|
|
/*
|
|
* Check if this is an AEAD stream (GCM mode). If so, then dispatch
|
|
* the request to our AEAD handler.
|
|
*/
|
|
if (session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_128 ||
|
|
session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_256) {
|
|
return srtp_protect_rtcp_aead(ctx, stream, rtcp_hdr,
|
|
(unsigned int *)pkt_octet_len,
|
|
session_keys, use_mki);
|
|
}
|
|
|
|
/* get tag length from stream context */
|
|
tag_len = srtp_auth_get_tag_length(session_keys->rtcp_auth);
|
|
|
|
/*
|
|
* set encryption start and encryption length - if we're not
|
|
* providing confidentiality, set enc_start to NULL
|
|
*/
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtcp_header;
|
|
enc_octet_len = *pkt_octet_len - octets_in_rtcp_header;
|
|
|
|
/* all of the packet, except the header, gets encrypted */
|
|
/*
|
|
* NOTE: hdr->length is not usable - it refers to only the first RTCP report
|
|
* in the compound packet!
|
|
*/
|
|
/*
|
|
* NOTE: trailer is 32-bit aligned because RTCP 'packets' are always
|
|
* multiples of 32-bits (RFC 3550 6.1)
|
|
*/
|
|
trailer = (uint32_t *)((char *)enc_start + enc_octet_len);
|
|
|
|
if (stream->rtcp_services & sec_serv_conf) {
|
|
*trailer = htonl(SRTCP_E_BIT); /* set encrypt bit */
|
|
} else {
|
|
enc_start = NULL;
|
|
enc_octet_len = 0;
|
|
/* 0 is network-order independant */
|
|
*trailer = 0x00000000; /* set encrypt bit */
|
|
}
|
|
|
|
mki_size = srtp_inject_mki((uint8_t *)hdr + *pkt_octet_len +
|
|
sizeof(srtcp_trailer_t),
|
|
session_keys, use_mki);
|
|
|
|
/*
|
|
* set the auth_start and auth_tag pointers to the proper locations
|
|
* (note that srtpc *always* provides authentication, unlike srtp)
|
|
*/
|
|
/* Note: This would need to change for optional mikey data */
|
|
auth_start = (uint32_t *)hdr;
|
|
auth_tag =
|
|
(uint8_t *)hdr + *pkt_octet_len + sizeof(srtcp_trailer_t) + mki_size;
|
|
|
|
/* perform EKT processing if needed */
|
|
srtp_ekt_write_data(stream->ekt, auth_tag, tag_len, pkt_octet_len,
|
|
srtp_rdbx_get_packet_index(&stream->rtp_rdbx));
|
|
|
|
/*
|
|
* check sequence number for overruns, and copy it into the packet
|
|
* if its value isn't too big
|
|
*/
|
|
status = srtp_rdb_increment(&stream->rtcp_rdb);
|
|
if (status)
|
|
return status;
|
|
seq_num = srtp_rdb_get_value(&stream->rtcp_rdb);
|
|
*trailer |= htonl(seq_num);
|
|
debug_print(mod_srtp, "srtcp index: %x", seq_num);
|
|
|
|
/*
|
|
* if we're using rindael counter mode, set nonce and seq
|
|
*/
|
|
if (session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_128 ||
|
|
session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_192 ||
|
|
session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_256) {
|
|
v128_t iv;
|
|
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = hdr->ssrc; /* still in network order! */
|
|
iv.v32[2] = htonl(seq_num >> 16);
|
|
iv.v32[3] = htonl(seq_num << 16);
|
|
status = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_encrypt);
|
|
|
|
} else {
|
|
v128_t iv;
|
|
|
|
/* otherwise, just set the index to seq_num */
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = 0;
|
|
iv.v32[2] = 0;
|
|
iv.v32[3] = htonl(seq_num);
|
|
status = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_encrypt);
|
|
}
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
|
|
/*
|
|
* if we're authenticating using a universal hash, put the keystream
|
|
* prefix into the authentication tag
|
|
*/
|
|
|
|
/* if auth_start is non-null, then put keystream into tag */
|
|
if (auth_start) {
|
|
/* put keystream prefix into auth_tag */
|
|
prefix_len = srtp_auth_get_prefix_length(session_keys->rtcp_auth);
|
|
status = srtp_cipher_output(session_keys->rtcp_cipher, auth_tag,
|
|
&prefix_len);
|
|
|
|
debug_print(mod_srtp, "keystream prefix: %s",
|
|
srtp_octet_string_hex_string(auth_tag, prefix_len));
|
|
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/* if we're encrypting, exor keystream into the message */
|
|
if (enc_start) {
|
|
status = srtp_cipher_encrypt(session_keys->rtcp_cipher,
|
|
(uint8_t *)enc_start, &enc_octet_len);
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/* initialize auth func context */
|
|
srtp_auth_start(session_keys->rtcp_auth);
|
|
|
|
/*
|
|
* run auth func over packet (including trailer), and write the
|
|
* result at auth_tag
|
|
*/
|
|
status =
|
|
srtp_auth_compute(session_keys->rtcp_auth, (uint8_t *)auth_start,
|
|
(*pkt_octet_len) + sizeof(srtcp_trailer_t), auth_tag);
|
|
debug_print(mod_srtp, "srtcp auth tag: %s",
|
|
srtp_octet_string_hex_string(auth_tag, tag_len));
|
|
if (status)
|
|
return srtp_err_status_auth_fail;
|
|
|
|
/* increase the packet length by the length of the auth tag and seq_num*/
|
|
*pkt_octet_len += (tag_len + sizeof(srtcp_trailer_t));
|
|
|
|
/* increase the packet by the mki_size */
|
|
*pkt_octet_len += mki_size;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_unprotect_rtcp(srtp_t ctx,
|
|
void *srtcp_hdr,
|
|
int *pkt_octet_len)
|
|
{
|
|
return srtp_unprotect_rtcp_mki(ctx, srtcp_hdr, pkt_octet_len, 0);
|
|
}
|
|
|
|
srtp_err_status_t srtp_unprotect_rtcp_mki(srtp_t ctx,
|
|
void *srtcp_hdr,
|
|
int *pkt_octet_len,
|
|
unsigned int use_mki)
|
|
{
|
|
srtcp_hdr_t *hdr = (srtcp_hdr_t *)srtcp_hdr;
|
|
uint32_t *enc_start; /* pointer to start of encrypted portion */
|
|
uint32_t *auth_start; /* pointer to start of auth. portion */
|
|
uint32_t *trailer; /* pointer to start of trailer */
|
|
unsigned int enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
uint8_t tmp_tag[SRTP_MAX_TAG_LEN];
|
|
uint8_t tag_copy[SRTP_MAX_TAG_LEN];
|
|
srtp_err_status_t status;
|
|
unsigned int auth_len;
|
|
int tag_len;
|
|
srtp_stream_ctx_t *stream;
|
|
uint32_t prefix_len;
|
|
uint32_t seq_num;
|
|
int e_bit_in_packet; /* whether the E-bit was found in the packet */
|
|
int sec_serv_confidentiality; /* whether confidentiality was requested */
|
|
unsigned int mki_size = 0;
|
|
srtp_session_keys_t *session_keys = NULL;
|
|
|
|
/* we assume the hdr is 32-bit aligned to start */
|
|
|
|
if (*pkt_octet_len < 0)
|
|
return srtp_err_status_bad_param;
|
|
|
|
/*
|
|
* check that the length value is sane; we'll check again once we
|
|
* know the tag length, but we at least want to know that it is
|
|
* a positive value
|
|
*/
|
|
if ((unsigned int)(*pkt_octet_len) <
|
|
octets_in_rtcp_header + sizeof(srtcp_trailer_t))
|
|
return srtp_err_status_bad_param;
|
|
|
|
/*
|
|
* look up ssrc in srtp_stream list, and process the packet with
|
|
* the appropriate stream. if we haven't seen this stream before,
|
|
* there's only one key for this srtp_session, and the cipher
|
|
* supports key-sharing, then we assume that a new stream using
|
|
* that key has just started up
|
|
*/
|
|
stream = srtp_get_stream(ctx, hdr->ssrc);
|
|
if (stream == NULL) {
|
|
if (ctx->stream_template != NULL) {
|
|
stream = ctx->stream_template;
|
|
|
|
/*
|
|
* check to see if stream_template has an EKT data structure, in
|
|
* which case we initialize the template using the EKT policy
|
|
* referenced by that data (which consists of decrypting the
|
|
* master key from the EKT field)
|
|
*
|
|
* this function initializes a *provisional* stream, and this
|
|
* stream should not be accepted until and unless the packet
|
|
* passes its authentication check
|
|
*/
|
|
if (stream->ekt != NULL) {
|
|
status = srtp_stream_init_from_ekt(stream, srtcp_hdr,
|
|
*pkt_octet_len);
|
|
if (status)
|
|
return status;
|
|
}
|
|
|
|
debug_print(mod_srtp,
|
|
"srtcp using provisional stream (SSRC: 0x%08x)",
|
|
ntohl(hdr->ssrc));
|
|
} else {
|
|
/* no template stream, so we return an error */
|
|
return srtp_err_status_no_ctx;
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Determine if MKI is being used and what session keys should be used
|
|
*/
|
|
if (use_mki) {
|
|
session_keys = srtp_get_session_keys(
|
|
stream, (uint8_t *)hdr, (const unsigned int *)pkt_octet_len,
|
|
&mki_size);
|
|
|
|
if (session_keys == NULL)
|
|
return srtp_err_status_bad_mki;
|
|
} else {
|
|
session_keys = &stream->session_keys[0];
|
|
}
|
|
|
|
/* get tag length from stream context */
|
|
tag_len = srtp_auth_get_tag_length(session_keys->rtcp_auth);
|
|
|
|
/* check the packet length - it must contain at least a full RTCP
|
|
header, an auth tag (if applicable), and the SRTCP encrypted flag
|
|
and 31-bit index value */
|
|
if (*pkt_octet_len < (int)(octets_in_rtcp_header + tag_len + mki_size +
|
|
sizeof(srtcp_trailer_t))) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
/*
|
|
* Check if this is an AEAD stream (GCM mode). If so, then dispatch
|
|
* the request to our AEAD handler.
|
|
*/
|
|
if (session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_128 ||
|
|
session_keys->rtp_cipher->algorithm == SRTP_AES_GCM_256) {
|
|
return srtp_unprotect_rtcp_aead(ctx, stream, srtcp_hdr,
|
|
(unsigned int *)pkt_octet_len,
|
|
session_keys, mki_size);
|
|
}
|
|
|
|
sec_serv_confidentiality = stream->rtcp_services == sec_serv_conf ||
|
|
stream->rtcp_services == sec_serv_conf_and_auth;
|
|
|
|
/*
|
|
* set encryption start, encryption length, and trailer
|
|
*/
|
|
enc_octet_len = *pkt_octet_len - (octets_in_rtcp_header + tag_len +
|
|
mki_size + sizeof(srtcp_trailer_t));
|
|
/*
|
|
*index & E (encryption) bit follow normal data. hdr->len is the number of
|
|
* words (32-bit) in the normal packet minus 1
|
|
*/
|
|
/* This should point trailer to the word past the end of the normal data. */
|
|
/* This would need to be modified for optional mikey data */
|
|
/*
|
|
* NOTE: trailer is 32-bit aligned because RTCP 'packets' are always
|
|
* multiples of 32-bits (RFC 3550 6.1)
|
|
*/
|
|
trailer = (uint32_t *)((char *)hdr + *pkt_octet_len -
|
|
(tag_len + mki_size + sizeof(srtcp_trailer_t)));
|
|
e_bit_in_packet =
|
|
(*((unsigned char *)trailer) & SRTCP_E_BYTE_BIT) == SRTCP_E_BYTE_BIT;
|
|
if (e_bit_in_packet != sec_serv_confidentiality) {
|
|
return srtp_err_status_cant_check;
|
|
}
|
|
if (sec_serv_confidentiality) {
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtcp_header;
|
|
} else {
|
|
enc_octet_len = 0;
|
|
enc_start = NULL; /* this indicates that there's no encryption */
|
|
}
|
|
|
|
/*
|
|
* set the auth_start and auth_tag pointers to the proper locations
|
|
* (note that srtcp *always* uses authentication, unlike srtp)
|
|
*/
|
|
auth_start = (uint32_t *)hdr;
|
|
|
|
/*
|
|
* The location of the auth tag in the packet needs to know MKI
|
|
* could be present. The data needed to calculate the Auth tag
|
|
* must not include the MKI
|
|
*/
|
|
auth_len = *pkt_octet_len - tag_len - mki_size;
|
|
auth_tag = (uint8_t *)hdr + auth_len + mki_size;
|
|
|
|
/*
|
|
* if EKT is in use, then we make a copy of the tag from the packet,
|
|
* and then zeroize the location of the base tag
|
|
*
|
|
* we first re-position the auth_tag pointer so that it points to
|
|
* the base tag
|
|
*/
|
|
if (stream->ekt) {
|
|
auth_tag -= srtp_ekt_octets_after_base_tag(stream->ekt);
|
|
memcpy(tag_copy, auth_tag, tag_len);
|
|
octet_string_set_to_zero(auth_tag, tag_len);
|
|
auth_tag = tag_copy;
|
|
auth_len += tag_len;
|
|
}
|
|
|
|
/*
|
|
* check the sequence number for replays
|
|
*/
|
|
/* this is easier than dealing with bitfield access */
|
|
seq_num = ntohl(*trailer) & SRTCP_INDEX_MASK;
|
|
debug_print(mod_srtp, "srtcp index: %x", seq_num);
|
|
status = srtp_rdb_check(&stream->rtcp_rdb, seq_num);
|
|
if (status)
|
|
return status;
|
|
|
|
/*
|
|
* if we're using aes counter mode, set nonce and seq
|
|
*/
|
|
if (session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_128 ||
|
|
session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_192 ||
|
|
session_keys->rtcp_cipher->type->id == SRTP_AES_ICM_256) {
|
|
v128_t iv;
|
|
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = hdr->ssrc; /* still in network order! */
|
|
iv.v32[2] = htonl(seq_num >> 16);
|
|
iv.v32[3] = htonl(seq_num << 16);
|
|
status = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_decrypt);
|
|
|
|
} else {
|
|
v128_t iv;
|
|
|
|
/* otherwise, just set the index to seq_num */
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = 0;
|
|
iv.v32[2] = 0;
|
|
iv.v32[3] = htonl(seq_num);
|
|
status = srtp_cipher_set_iv(session_keys->rtcp_cipher, (uint8_t *)&iv,
|
|
srtp_direction_decrypt);
|
|
}
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
|
|
/* initialize auth func context */
|
|
srtp_auth_start(session_keys->rtcp_auth);
|
|
|
|
/* run auth func over packet, put result into tmp_tag */
|
|
status = srtp_auth_compute(session_keys->rtcp_auth, (uint8_t *)auth_start,
|
|
auth_len, tmp_tag);
|
|
debug_print(mod_srtp, "srtcp computed tag: %s",
|
|
srtp_octet_string_hex_string(tmp_tag, tag_len));
|
|
if (status)
|
|
return srtp_err_status_auth_fail;
|
|
|
|
/* compare the tag just computed with the one in the packet */
|
|
debug_print(mod_srtp, "srtcp tag from packet: %s",
|
|
srtp_octet_string_hex_string(auth_tag, tag_len));
|
|
if (octet_string_is_eq(tmp_tag, auth_tag, tag_len))
|
|
return srtp_err_status_auth_fail;
|
|
|
|
/*
|
|
* if we're authenticating using a universal hash, put the keystream
|
|
* prefix into the authentication tag
|
|
*/
|
|
prefix_len = srtp_auth_get_prefix_length(session_keys->rtcp_auth);
|
|
if (prefix_len) {
|
|
status = srtp_cipher_output(session_keys->rtcp_cipher, auth_tag,
|
|
&prefix_len);
|
|
debug_print(mod_srtp, "keystream prefix: %s",
|
|
srtp_octet_string_hex_string(auth_tag, prefix_len));
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/* if we're decrypting, exor keystream into the message */
|
|
if (enc_start) {
|
|
status = srtp_cipher_decrypt(session_keys->rtcp_cipher,
|
|
(uint8_t *)enc_start, &enc_octet_len);
|
|
if (status)
|
|
return srtp_err_status_cipher_fail;
|
|
}
|
|
|
|
/* decrease the packet length by the length of the auth tag and seq_num */
|
|
*pkt_octet_len -= (tag_len + sizeof(srtcp_trailer_t));
|
|
|
|
/* decrease the packet length by the length of the mki_size */
|
|
*pkt_octet_len -= mki_size;
|
|
|
|
/*
|
|
* if EKT is in effect, subtract the EKT data out of the packet
|
|
* length
|
|
*/
|
|
*pkt_octet_len -= srtp_ekt_octets_after_base_tag(stream->ekt);
|
|
|
|
/*
|
|
* verify that stream is for received traffic - this check will
|
|
* detect SSRC collisions, since a stream that appears in both
|
|
* srtp_protect() and srtp_unprotect() will fail this test in one of
|
|
* those functions.
|
|
*
|
|
* we do this check *after* the authentication check, so that the
|
|
* latter check will catch any attempts to fool us into thinking
|
|
* that we've got a collision
|
|
*/
|
|
if (stream->direction != dir_srtp_receiver) {
|
|
if (stream->direction == dir_unknown) {
|
|
stream->direction = dir_srtp_receiver;
|
|
} else {
|
|
srtp_handle_event(ctx, stream, event_ssrc_collision);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* if the stream is a 'provisional' one, in which the template context
|
|
* is used, then we need to allocate a new stream at this point, since
|
|
* the authentication passed
|
|
*/
|
|
if (stream == ctx->stream_template) {
|
|
srtp_stream_ctx_t *new_stream;
|
|
|
|
/*
|
|
* allocate and initialize a new stream
|
|
*
|
|
* note that we indicate failure if we can't allocate the new
|
|
* stream, and some implementations will want to not return
|
|
* failure here
|
|
*/
|
|
status =
|
|
srtp_stream_clone(ctx->stream_template, hdr->ssrc, &new_stream);
|
|
if (status)
|
|
return status;
|
|
|
|
/* add new stream to the head of the stream_list */
|
|
new_stream->next = ctx->stream_list;
|
|
ctx->stream_list = new_stream;
|
|
|
|
/* set stream (the pointer used in this function) */
|
|
stream = new_stream;
|
|
}
|
|
|
|
/* we've passed the authentication check, so add seq_num to the rdb */
|
|
srtp_rdb_add_index(&stream->rtcp_rdb, seq_num);
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* user data within srtp_t context
|
|
*/
|
|
|
|
void srtp_set_user_data(srtp_t ctx, void *data)
|
|
{
|
|
ctx->user_data = data;
|
|
}
|
|
|
|
void *srtp_get_user_data(srtp_t ctx)
|
|
{
|
|
return ctx->user_data;
|
|
}
|
|
|
|
/*
|
|
* dtls keying for srtp
|
|
*/
|
|
|
|
srtp_err_status_t srtp_crypto_policy_set_from_profile_for_rtp(
|
|
srtp_crypto_policy_t *policy,
|
|
srtp_profile_t profile)
|
|
{
|
|
/* set SRTP policy from the SRTP profile in the key set */
|
|
switch (profile) {
|
|
case srtp_profile_aes128_cm_sha1_80:
|
|
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(policy);
|
|
break;
|
|
case srtp_profile_aes128_cm_sha1_32:
|
|
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(policy);
|
|
break;
|
|
case srtp_profile_null_sha1_80:
|
|
srtp_crypto_policy_set_null_cipher_hmac_sha1_80(policy);
|
|
break;
|
|
#if defined(OPENSSL)
|
|
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;
|
|
#if defined(OPENSSL)
|
|
case srtp_profile_aead_aes_128_gcm:
|
|
srtp_crypto_policy_set_aes_gcm_128_16_auth(policy);
|
|
break;
|
|
case srtp_profile_aead_aes_256_gcm:
|
|
srtp_crypto_policy_set_aes_gcm_256_16_auth(policy);
|
|
break;
|
|
#endif
|
|
/* the following profiles are not (yet) supported */
|
|
case srtp_profile_null_sha1_32:
|
|
default:
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
void srtp_append_salt_to_key(uint8_t *key,
|
|
unsigned int bytes_in_key,
|
|
uint8_t *salt,
|
|
unsigned int bytes_in_salt)
|
|
{
|
|
memcpy(key + bytes_in_key, salt, bytes_in_salt);
|
|
}
|
|
|
|
unsigned int srtp_profile_get_master_key_length(srtp_profile_t profile)
|
|
{
|
|
switch (profile) {
|
|
case srtp_profile_aes128_cm_sha1_80:
|
|
return SRTP_AES_128_KEY_LEN;
|
|
break;
|
|
case srtp_profile_aes128_cm_sha1_32:
|
|
return SRTP_AES_128_KEY_LEN;
|
|
break;
|
|
case srtp_profile_null_sha1_80:
|
|
return SRTP_AES_128_KEY_LEN;
|
|
break;
|
|
case srtp_profile_aead_aes_128_gcm:
|
|
return SRTP_AES_128_KEY_LEN;
|
|
break;
|
|
case srtp_profile_aead_aes_256_gcm:
|
|
return SRTP_AES_256_KEY_LEN;
|
|
break;
|
|
/* the following profiles are not (yet) supported */
|
|
case srtp_profile_null_sha1_32:
|
|
default:
|
|
return 0; /* indicate error by returning a zero */
|
|
}
|
|
}
|
|
|
|
unsigned int srtp_profile_get_master_salt_length(srtp_profile_t profile)
|
|
{
|
|
switch (profile) {
|
|
case srtp_profile_aes128_cm_sha1_80:
|
|
return SRTP_SALT_LEN;
|
|
break;
|
|
case srtp_profile_aes128_cm_sha1_32:
|
|
return SRTP_SALT_LEN;
|
|
break;
|
|
case srtp_profile_null_sha1_80:
|
|
return SRTP_SALT_LEN;
|
|
break;
|
|
case srtp_profile_aead_aes_128_gcm:
|
|
return SRTP_AEAD_SALT_LEN;
|
|
break;
|
|
case srtp_profile_aead_aes_256_gcm:
|
|
return SRTP_AEAD_SALT_LEN;
|
|
break;
|
|
/* the following profiles are not (yet) supported */
|
|
case srtp_profile_null_sha1_32:
|
|
default:
|
|
return 0; /* indicate error by returning a zero */
|
|
}
|
|
}
|
|
|
|
srtp_err_status_t stream_get_protect_trailer_length(srtp_stream_ctx_t *stream,
|
|
uint32_t is_rtp,
|
|
uint32_t use_mki,
|
|
uint32_t mki_index,
|
|
uint32_t *length)
|
|
{
|
|
*length = 0;
|
|
|
|
srtp_session_keys_t *session_key;
|
|
|
|
if (use_mki) {
|
|
if (mki_index >= stream->num_master_keys) {
|
|
return srtp_err_status_bad_mki;
|
|
}
|
|
session_key = &stream->session_keys[mki_index];
|
|
|
|
*length += session_key->mki_size;
|
|
|
|
} else {
|
|
session_key = &stream->session_keys[0];
|
|
}
|
|
if (is_rtp) {
|
|
*length += srtp_auth_get_tag_length(session_key->rtp_auth);
|
|
} else {
|
|
*length += srtp_auth_get_tag_length(session_key->rtcp_auth);
|
|
*length += sizeof(srtcp_trailer_t);
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t get_protect_trailer_length(srtp_t session,
|
|
uint32_t is_rtp,
|
|
uint32_t use_mki,
|
|
uint32_t mki_index,
|
|
uint32_t *length)
|
|
{
|
|
srtp_stream_ctx_t *stream;
|
|
|
|
if (session == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
if (session->stream_template == NULL && session->stream_list == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
*length = 0;
|
|
|
|
stream = session->stream_template;
|
|
|
|
if (stream != NULL) {
|
|
stream_get_protect_trailer_length(stream, is_rtp, use_mki, mki_index,
|
|
length);
|
|
}
|
|
|
|
stream = session->stream_list;
|
|
|
|
while (stream != NULL) {
|
|
uint32_t temp_length;
|
|
if (stream_get_protect_trailer_length(stream, is_rtp, use_mki,
|
|
mki_index, &temp_length) ==
|
|
srtp_err_status_ok) {
|
|
if (temp_length > *length) {
|
|
*length = temp_length;
|
|
}
|
|
}
|
|
stream = stream->next;
|
|
}
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_get_protect_trailer_length(srtp_t session,
|
|
uint32_t use_mki,
|
|
uint32_t mki_index,
|
|
uint32_t *length)
|
|
{
|
|
return get_protect_trailer_length(session, 1, use_mki, mki_index, length);
|
|
}
|
|
|
|
srtp_err_status_t srtp_get_protect_rtcp_trailer_length(srtp_t session,
|
|
uint32_t use_mki,
|
|
uint32_t mki_index,
|
|
uint32_t *length)
|
|
{
|
|
return get_protect_trailer_length(session, 0, use_mki, mki_index, length);
|
|
}
|
|
|
|
/*
|
|
* SRTP debug interface
|
|
*/
|
|
srtp_err_status_t srtp_set_debug_module(const char *mod_name, int v)
|
|
{
|
|
return srtp_crypto_kernel_set_debug_module(mod_name, v);
|
|
}
|
|
|
|
srtp_err_status_t srtp_list_debug_modules(void)
|
|
{
|
|
return srtp_crypto_kernel_list_debug_modules();
|
|
}
|
|
|
|
/*
|
|
* srtp_log_handler is a global variable holding a pointer to the
|
|
* log handler function; this function is called for any log
|
|
* output.
|
|
*/
|
|
|
|
static srtp_log_handler_func_t *srtp_log_handler = NULL;
|
|
static void *srtp_log_handler_data = NULL;
|
|
|
|
void srtp_err_handler(srtp_err_reporting_level_t level, const char *msg)
|
|
{
|
|
if (srtp_log_handler) {
|
|
srtp_log_level_t log_level = srtp_log_level_error;
|
|
switch (level) {
|
|
case srtp_err_level_error:
|
|
log_level = srtp_log_level_error;
|
|
break;
|
|
case srtp_err_level_warning:
|
|
log_level = srtp_log_level_warning;
|
|
break;
|
|
case srtp_err_level_info:
|
|
log_level = srtp_log_level_info;
|
|
break;
|
|
case srtp_err_level_debug:
|
|
log_level = srtp_log_level_debug;
|
|
break;
|
|
}
|
|
|
|
srtp_log_handler(log_level, msg, srtp_log_handler_data);
|
|
}
|
|
}
|
|
|
|
srtp_err_status_t srtp_install_log_handler(srtp_log_handler_func_t func,
|
|
void *data)
|
|
{
|
|
/*
|
|
* note that we accept NULL arguments intentionally - calling this
|
|
* function with a NULL arguments removes a log handler that's
|
|
* been previously installed
|
|
*/
|
|
|
|
if (srtp_log_handler) {
|
|
srtp_install_err_report_handler(NULL);
|
|
}
|
|
srtp_log_handler = func;
|
|
srtp_log_handler_data = data;
|
|
if (srtp_log_handler) {
|
|
srtp_install_err_report_handler(srtp_err_handler);
|
|
}
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_set_stream_roc(srtp_t session,
|
|
uint32_t ssrc,
|
|
uint32_t roc)
|
|
{
|
|
srtp_stream_t stream;
|
|
|
|
stream = srtp_get_stream(session, htonl(ssrc));
|
|
if (stream == NULL)
|
|
return srtp_err_status_bad_param;
|
|
|
|
stream->pending_roc = roc;
|
|
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t srtp_get_stream_roc(srtp_t session,
|
|
uint32_t ssrc,
|
|
uint32_t *roc)
|
|
{
|
|
srtp_stream_t stream;
|
|
|
|
stream = srtp_get_stream(session, htonl(ssrc));
|
|
if (stream == NULL)
|
|
return srtp_err_status_bad_param;
|
|
|
|
*roc = srtp_rdbx_get_roc(&stream->rtp_rdbx);
|
|
|
|
return srtp_err_status_ok;
|
|
}
|