1779 lines
51 KiB
C
1779 lines
51 KiB
C
/*
|
|
* srtp.c
|
|
*
|
|
* the secure real-time transport protocol
|
|
*
|
|
* David A. McGrew
|
|
* Cisco Systems, Inc.
|
|
*/
|
|
/*
|
|
*
|
|
* Copyright (c) 2001-2005, Cisco Systems, Inc.
|
|
* All rights reserved.
|
|
*
|
|
* Redistribution and use in source and binary forms, with or without
|
|
* modification, are permitted provided that the following conditions
|
|
* are met:
|
|
*
|
|
* Redistributions of source code must retain the above copyright
|
|
* notice, this list of conditions and the following disclaimer.
|
|
*
|
|
* Redistributions in binary form must reproduce the above
|
|
* copyright notice, this list of conditions and the following
|
|
* disclaimer in the documentation and/or other materials provided
|
|
* with the distribution.
|
|
*
|
|
* Neither the name of the Cisco Systems, Inc. nor the names of its
|
|
* contributors may be used to endorse or promote products derived
|
|
* from this software without specific prior written permission.
|
|
*
|
|
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
|
|
* "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
|
|
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
|
|
* FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
|
|
* COPYRIGHT HOLDERS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
|
|
* INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
|
|
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
|
|
* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
|
|
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
|
|
* STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
|
|
* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED
|
|
* OF THE POSSIBILITY OF SUCH DAMAGE.
|
|
*
|
|
*/
|
|
|
|
|
|
#include "srtp.h"
|
|
#include "aes_icm.h" /* aes_icm is used in the KDF */
|
|
#include "alloc.h" /* for crypto_alloc() */
|
|
|
|
#ifndef SRTP_KERNEL
|
|
# include <limits.h>
|
|
# ifdef HAVE_NETINET_IN_H
|
|
# include <netinet/in.h>
|
|
# elif defined(HAVE_WINSOCK2_H)
|
|
# include <winsock2.h>
|
|
# endif
|
|
#endif /* ! SRTP_KERNEL */
|
|
|
|
|
|
extern cipher_type_t aes_icm;
|
|
extern auth_type_t tmmhv2;
|
|
|
|
/* the debug module for srtp */
|
|
|
|
debug_module_t mod_srtp = {
|
|
0, /* debugging is off by default */
|
|
"srtp" /* printable name for module */
|
|
};
|
|
|
|
#define octets_in_rtp_header 12
|
|
#define uint32s_in_rtp_header 3
|
|
#define octets_in_rtcp_header 8
|
|
#define uint32s_in_rtcp_header 2
|
|
|
|
|
|
err_status_t
|
|
srtp_stream_alloc(srtp_stream_ctx_t **str_ptr,
|
|
const srtp_policy_t *p) {
|
|
srtp_stream_ctx_t *str;
|
|
err_status_t stat;
|
|
|
|
/*
|
|
* This function allocates the stream context, rtp and rtcp ciphers
|
|
* and auth functions, and key limit structure. If there is a
|
|
* failure during allocation, we free all previously allocated
|
|
* memory and return a failure code. The code could probably
|
|
* be improved, but it works and should be clear.
|
|
*/
|
|
|
|
/* allocate srtp stream and set str_ptr */
|
|
str = (srtp_stream_ctx_t *) crypto_alloc(sizeof(srtp_stream_ctx_t));
|
|
if (str == NULL)
|
|
return err_status_alloc_fail;
|
|
*str_ptr = str;
|
|
|
|
/* allocate cipher */
|
|
stat = crypto_kernel_alloc_cipher(p->rtp.cipher_type,
|
|
&str->rtp_cipher,
|
|
p->rtp.cipher_key_len);
|
|
if (stat) {
|
|
crypto_free(str);
|
|
return stat;
|
|
}
|
|
|
|
/* allocate auth function */
|
|
stat = crypto_kernel_alloc_auth(p->rtp.auth_type,
|
|
&str->rtp_auth,
|
|
p->rtp.auth_key_len,
|
|
p->rtp.auth_tag_len);
|
|
if (stat) {
|
|
cipher_dealloc(str->rtp_cipher);
|
|
crypto_free(str);
|
|
return stat;
|
|
}
|
|
|
|
/* allocate key limit structure */
|
|
str->limit = crypto_alloc(sizeof(key_limit_ctx_t));
|
|
if (str->limit == NULL) {
|
|
auth_dealloc(str->rtp_auth);
|
|
cipher_dealloc(str->rtp_cipher);
|
|
crypto_free(str);
|
|
return err_status_alloc_fail;
|
|
}
|
|
|
|
/*
|
|
* ...and now the RTCP-specific initialization - first, allocate
|
|
* the cipher
|
|
*/
|
|
stat = crypto_kernel_alloc_cipher(p->rtcp.cipher_type,
|
|
&str->rtcp_cipher,
|
|
p->rtcp.cipher_key_len);
|
|
if (stat) {
|
|
auth_dealloc(str->rtp_auth);
|
|
cipher_dealloc(str->rtp_cipher);
|
|
crypto_free(str->limit);
|
|
crypto_free(str);
|
|
return stat;
|
|
}
|
|
|
|
/* allocate auth function */
|
|
stat = crypto_kernel_alloc_auth(p->rtcp.auth_type,
|
|
&str->rtcp_auth,
|
|
p->rtcp.auth_key_len,
|
|
p->rtcp.auth_tag_len);
|
|
if (stat) {
|
|
cipher_dealloc(str->rtcp_cipher);
|
|
auth_dealloc(str->rtp_auth);
|
|
cipher_dealloc(str->rtp_cipher);
|
|
crypto_free(str->limit);
|
|
crypto_free(str);
|
|
return stat;
|
|
}
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
err_status_t
|
|
srtp_stream_dealloc(srtp_t session, srtp_stream_ctx_t *stream) {
|
|
err_status_t status;
|
|
|
|
/*
|
|
* we use a conservative deallocation strategy - if any deallocation
|
|
* fails, then we report that fact without trying to deallocate
|
|
* anything else
|
|
*/
|
|
|
|
/* deallocate cipher, if it is not the same as that in template */
|
|
if (session->stream_template
|
|
&& stream->rtp_cipher == session->stream_template->rtp_cipher) {
|
|
/* do nothing */
|
|
} else {
|
|
status = cipher_dealloc(stream->rtp_cipher);
|
|
if (status)
|
|
return status;
|
|
}
|
|
|
|
/* deallocate auth function, if it is not the same as that in template */
|
|
if (session->stream_template
|
|
&& stream->rtp_auth == session->stream_template->rtp_auth) {
|
|
/* do nothing */
|
|
} else {
|
|
status = auth_dealloc(stream->rtp_auth);
|
|
if (status)
|
|
return status;
|
|
}
|
|
|
|
/* deallocate key usage limit, if it is not the same as that in template */
|
|
if (session->stream_template
|
|
&& stream->limit == session->stream_template->limit) {
|
|
/* do nothing */
|
|
} else {
|
|
crypto_free(stream->limit);
|
|
}
|
|
|
|
/*
|
|
* deallocate rtcp cipher, if it is not the same as that in
|
|
* template
|
|
*/
|
|
if (session->stream_template
|
|
&& stream->rtcp_cipher == session->stream_template->rtcp_cipher) {
|
|
/* do nothing */
|
|
} else {
|
|
status = cipher_dealloc(stream->rtcp_cipher);
|
|
if (status)
|
|
return status;
|
|
}
|
|
|
|
/*
|
|
* deallocate rtcp auth function, if it is not the same as that in
|
|
* template
|
|
*/
|
|
if (session->stream_template
|
|
&& stream->rtcp_auth == session->stream_template->rtcp_auth) {
|
|
/* do nothing */
|
|
} else {
|
|
status = auth_dealloc(stream->rtcp_auth);
|
|
if (status)
|
|
return status;
|
|
}
|
|
|
|
/* deallocate srtp stream context */
|
|
crypto_free(stream);
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
|
|
/*
|
|
* srtp_stream_clone(stream_template, new) allocates a new stream and
|
|
* initializes it using the cipher and auth of the stream_template
|
|
*
|
|
* the only unique data in a cloned stream is the replay database and
|
|
* the SSRC
|
|
*/
|
|
|
|
err_status_t
|
|
srtp_stream_clone(const srtp_stream_ctx_t *stream_template,
|
|
uint32_t ssrc,
|
|
srtp_stream_ctx_t **str_ptr) {
|
|
err_status_t status;
|
|
srtp_stream_ctx_t *str;
|
|
|
|
debug_print(mod_srtp, "cloning stream (SSRC: 0x%08x)", ssrc);
|
|
|
|
/* allocate srtp stream and set str_ptr */
|
|
str = (srtp_stream_ctx_t *) crypto_alloc(sizeof(srtp_stream_ctx_t));
|
|
if (str == NULL)
|
|
return err_status_alloc_fail;
|
|
*str_ptr = str;
|
|
|
|
/* set cipher and auth pointers to those of the template */
|
|
str->rtp_cipher = stream_template->rtp_cipher;
|
|
str->rtp_auth = stream_template->rtp_auth;
|
|
str->rtcp_cipher = stream_template->rtcp_cipher;
|
|
str->rtcp_auth = stream_template->rtcp_auth;
|
|
|
|
/* set key limit to point to that of the template */
|
|
status = key_limit_clone(stream_template->limit, &str->limit);
|
|
if (status)
|
|
return status;
|
|
|
|
/* initialize replay databases */
|
|
rdbx_init(&str->rtp_rdbx);
|
|
rdb_init(&str->rtcp_rdb);
|
|
|
|
/* set ssrc to that provided */
|
|
str->ssrc = ssrc;
|
|
|
|
/* set direction and security services */
|
|
str->direction = stream_template->direction;
|
|
str->rtp_services = stream_template->rtp_services;
|
|
str->rtcp_services = stream_template->rtcp_services;
|
|
|
|
/* defensive coding */
|
|
str->next = NULL;
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
|
|
/*
|
|
* key derivation functions, internal to libSRTP
|
|
*
|
|
* srtp_kdf_t is a key derivation context
|
|
*
|
|
* srtp_kdf_init(&kdf, k) initializes kdf with the key k
|
|
*
|
|
* 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 the kdf state
|
|
*/
|
|
|
|
typedef enum {
|
|
label_rtp_encryption = 0x00,
|
|
label_rtp_msg_auth = 0x01,
|
|
label_rtp_salt = 0x02,
|
|
label_rtcp_encryption = 0x03,
|
|
label_rtcp_msg_auth = 0x04,
|
|
label_rtcp_salt = 0x05
|
|
} srtp_prf_label;
|
|
|
|
|
|
/*
|
|
* srtp_kdf_t represents a key derivation function. The SRTP
|
|
* default KDF is the only one implemented at present.
|
|
*/
|
|
|
|
typedef struct {
|
|
aes_icm_ctx_t c; /* cipher used for key derivation */
|
|
} srtp_kdf_t;
|
|
|
|
err_status_t
|
|
srtp_kdf_init(srtp_kdf_t *kdf, const uint8_t key[30]) {
|
|
|
|
aes_icm_context_init(&kdf->c, key);
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
err_status_t
|
|
srtp_kdf_generate(srtp_kdf_t *kdf, srtp_prf_label label,
|
|
uint8_t *key, int length) {
|
|
|
|
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;
|
|
|
|
aes_icm_set_iv(&kdf->c, &nonce);
|
|
|
|
/* generate keystream output */
|
|
aes_icm_output(&kdf->c, key, length);
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
err_status_t
|
|
srtp_kdf_clear(srtp_kdf_t *kdf) {
|
|
|
|
/* zeroize aes context */
|
|
octet_string_set_to_zero((uint8_t *)kdf, sizeof(srtp_kdf_t));
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
/*
|
|
* end of key derivation functions
|
|
*/
|
|
|
|
#define MAX_SRTP_KEY_LEN 256
|
|
|
|
|
|
|
|
err_status_t
|
|
srtp_stream_init(srtp_stream_ctx_t *srtp,
|
|
const srtp_policy_t *p) {
|
|
err_status_t stat;
|
|
srtp_kdf_t kdf;
|
|
uint8_t tmp_key[MAX_SRTP_KEY_LEN];
|
|
uint8_t *key = p->key;
|
|
|
|
debug_print(mod_srtp, "initializing stream (SSRC: 0x%08x)",
|
|
p->ssrc.value);
|
|
|
|
/* initialize replay database */
|
|
rdbx_init(&srtp->rtp_rdbx);
|
|
|
|
/* initialize key limit to maximum value */
|
|
#ifdef NO_64BIT_MATH
|
|
{
|
|
uint64_t temp;
|
|
temp = make64(UINT_MAX,UINT_MAX);
|
|
key_limit_set(srtp->limit, temp);
|
|
}
|
|
#else
|
|
key_limit_set(srtp->limit, 0xffffffffffffLL);
|
|
#endif
|
|
|
|
/* set the SSRC value */
|
|
srtp->ssrc = htonl(p->ssrc.value);
|
|
|
|
/* set the security service flags */
|
|
srtp->rtp_services = p->rtp.sec_serv;
|
|
srtp->rtcp_services = p->rtcp.sec_serv;
|
|
|
|
/*
|
|
* set direction to unknown - this flag gets checked in srtp_protect(),
|
|
* srtp_unprotect(), srtp_protect_rtcp(), and srtp_unprotect_rtcp(), and
|
|
* gets set appropriately if it is set to unknown.
|
|
*/
|
|
srtp->direction = dir_unknown;
|
|
|
|
/* initialize KDF state */
|
|
srtp_kdf_init(&kdf, key);
|
|
|
|
/* generate encryption key */
|
|
srtp_kdf_generate(&kdf, label_rtp_encryption,
|
|
tmp_key, cipher_get_key_length(srtp->rtp_cipher));
|
|
/*
|
|
* if the cipher in the srtp context is aes_icm, then we need
|
|
* to generate the salt value
|
|
*/
|
|
if (srtp->rtp_cipher->type == &aes_icm) {
|
|
/* FIX!!! this is really the cipher key length; rest is salt */
|
|
int base_key_len = 16;
|
|
int salt_len = cipher_get_key_length(srtp->rtp_cipher) - base_key_len;
|
|
|
|
debug_print(mod_srtp, "found aes_icm, generating salt", NULL);
|
|
|
|
/* generate encryption salt, put after encryption key */
|
|
srtp_kdf_generate(&kdf, label_rtp_salt,
|
|
tmp_key + base_key_len, salt_len);
|
|
}
|
|
debug_print(mod_srtp, "cipher key: %s",
|
|
octet_string_hex_string(tmp_key,
|
|
cipher_get_key_length(srtp->rtp_cipher)));
|
|
|
|
/* initialize cipher */
|
|
stat = cipher_init(srtp->rtp_cipher, tmp_key, direction_any);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return err_status_init_fail;
|
|
}
|
|
|
|
/* generate authentication key */
|
|
srtp_kdf_generate(&kdf, label_rtp_msg_auth,
|
|
tmp_key, auth_get_key_length(srtp->rtp_auth));
|
|
debug_print(mod_srtp, "auth key: %s",
|
|
octet_string_hex_string(tmp_key,
|
|
auth_get_key_length(srtp->rtp_auth)));
|
|
|
|
/* initialize auth function */
|
|
stat = auth_init(srtp->rtp_auth, tmp_key);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return err_status_init_fail;
|
|
}
|
|
|
|
/*
|
|
* ...now initialize RTCP-specific structures
|
|
*/
|
|
|
|
/* initialize replay database */
|
|
rdb_init(&srtp->rtcp_rdb);
|
|
|
|
/* DAM - no RTCP key limit at present */
|
|
|
|
/* generate encryption key */
|
|
srtp_kdf_generate(&kdf, label_rtcp_encryption,
|
|
tmp_key, cipher_get_key_length(srtp->rtcp_cipher));
|
|
/*
|
|
* if the cipher in the srtp context is aes_icm, then we need
|
|
* to generate the salt value
|
|
*/
|
|
if (srtp->rtcp_cipher->type == &aes_icm) {
|
|
/* FIX!!! this is really the cipher key length; rest is salt */
|
|
int base_key_len = 16;
|
|
int salt_len = cipher_get_key_length(srtp->rtcp_cipher) - base_key_len;
|
|
|
|
debug_print(mod_srtp, "found aes_icm, generating rtcp salt", NULL);
|
|
|
|
/* generate encryption salt, put after encryption key */
|
|
srtp_kdf_generate(&kdf, label_rtcp_salt,
|
|
tmp_key + base_key_len, salt_len);
|
|
}
|
|
debug_print(mod_srtp, "rtcp cipher key: %s",
|
|
octet_string_hex_string(tmp_key,
|
|
cipher_get_key_length(srtp->rtcp_cipher)));
|
|
|
|
/* initialize cipher */
|
|
stat = cipher_init(srtp->rtcp_cipher, tmp_key, direction_any);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return err_status_init_fail;
|
|
}
|
|
|
|
/* generate authentication key */
|
|
srtp_kdf_generate(&kdf, label_rtcp_msg_auth,
|
|
tmp_key, auth_get_key_length(srtp->rtcp_auth));
|
|
debug_print(mod_srtp, "rtcp auth key: %s",
|
|
octet_string_hex_string(tmp_key,
|
|
auth_get_key_length(srtp->rtcp_auth)));
|
|
|
|
/* initialize auth function */
|
|
stat = auth_init(srtp->rtcp_auth, tmp_key);
|
|
if (stat) {
|
|
/* zeroize temp buffer */
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return err_status_init_fail;
|
|
}
|
|
|
|
/* clear memory then return */
|
|
srtp_kdf_clear(&kdf);
|
|
octet_string_set_to_zero(tmp_key, MAX_SRTP_KEY_LEN);
|
|
return 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) {
|
|
|
|
err_report(err_level_warning, "srtp: in stream 0x%x: ",
|
|
data->stream->ssrc);
|
|
|
|
switch(data->event) {
|
|
case event_ssrc_collision:
|
|
err_report(err_level_warning, "\tSSRC collision\n");
|
|
break;
|
|
case event_key_soft_limit:
|
|
err_report(err_level_warning, "\tkey usage soft limit reached\n");
|
|
break;
|
|
case event_key_hard_limit:
|
|
err_report(err_level_warning, "\tkey usage hard limit reached\n");
|
|
break;
|
|
case event_packet_index_limit:
|
|
err_report(err_level_warning, "\tpacket index limit reached\n");
|
|
break;
|
|
default:
|
|
err_report(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;
|
|
|
|
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 err_status_ok;
|
|
}
|
|
|
|
err_status_t
|
|
srtp_protect(srtp_ctx_t *ctx, void *rtp_hdr, int *pkt_octet_len) {
|
|
srtp_hdr_t *hdr = rtp_hdr;
|
|
uint32_t *enc_start; /* pointer to start of encrypted portion */
|
|
uint32_t *auth_start; /* pointer to start of auth. portion */
|
|
unsigned enc_octet_len = 0; /* number of octets in encrypted portion */
|
|
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 */
|
|
err_status_t status;
|
|
int tag_len;
|
|
srtp_stream_ctx_t *stream;
|
|
int prefix_len;
|
|
|
|
debug_print(mod_srtp, "function srtp_protect", NULL);
|
|
|
|
/* we assume the hdr is 32-bit aligned to start */
|
|
|
|
/* check the packet length - it must at least contain a full header */
|
|
if (*pkt_octet_len < octets_in_rtp_header)
|
|
return 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 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);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* 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(key_limit_update(stream->limit)) {
|
|
case key_event_normal:
|
|
break;
|
|
case key_event_soft_limit:
|
|
srtp_handle_event(ctx, stream, event_key_soft_limit);
|
|
break;
|
|
case key_event_hard_limit:
|
|
srtp_handle_event(ctx, stream, event_key_hard_limit);
|
|
return err_status_key_expired;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* get tag length from stream */
|
|
tag_len = auth_get_tag_length(stream->rtp_auth);
|
|
|
|
/*
|
|
* find starting point for encryption and length of data to be
|
|
* encrypted - the encrypted portion starts after the rtp header
|
|
* extension, if present; otherwise, it starts after the last csrc,
|
|
* if any are present
|
|
*
|
|
* if we're not providing confidentiality, set enc_start to NULL
|
|
*/
|
|
if (stream->rtp_services & sec_serv_conf) {
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtp_header + hdr->cc;
|
|
if (hdr->x == 1) {
|
|
srtp_hdr_xtnd_t *xtn_hdr = (srtp_hdr_xtnd_t *)enc_start;
|
|
enc_start += (ntohs(xtn_hdr->length) + 1);
|
|
}
|
|
enc_octet_len = (unsigned int)(*pkt_octet_len - ((enc_start - (uint32_t *)hdr) << 2));
|
|
} else {
|
|
enc_start = NULL;
|
|
}
|
|
|
|
/*
|
|
* if we're providing authentication, set the auth_start and auth_tag
|
|
* pointers to the proper locations; otherwise, set auth_start to NULL
|
|
* to indicate that no authentication is needed
|
|
*/
|
|
if (stream->rtp_services & sec_serv_auth) {
|
|
auth_start = (uint32_t *)hdr;
|
|
auth_tag = (uint8_t *)hdr + *pkt_octet_len;
|
|
} else {
|
|
auth_start = NULL;
|
|
auth_tag = NULL;
|
|
}
|
|
|
|
/*
|
|
* estimate the packet index using the start of the replay window
|
|
* and the sequence number from the header
|
|
*/
|
|
delta = rdbx_estimate_index(&stream->rtp_rdbx, &est, ntohs(hdr->seq));
|
|
status = rdbx_check(&stream->rtp_rdbx, delta);
|
|
if (status)
|
|
return status; /* we've been asked to reuse an index */
|
|
rdbx_add_index(&stream->rtp_rdbx, delta);
|
|
|
|
#ifdef NO_64BIT_MATH
|
|
debug_print2(mod_srtp, "estimated packet index: %08x%08x",
|
|
high32(est),low32(est));
|
|
#else
|
|
debug_print(mod_srtp, "estimated packet index: %016llx", est);
|
|
#endif
|
|
|
|
/*
|
|
* if we're using rindael counter mode, set nonce and seq
|
|
*/
|
|
if (stream->rtp_cipher->type == &aes_icm) {
|
|
v128_t iv;
|
|
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = hdr->ssrc;
|
|
#ifdef NO_64BIT_MATH
|
|
iv.v64[1] = be64_to_cpu(make64((high32(est) << 16) | (low32(est) >> 16),
|
|
low32(est) << 16));
|
|
#else
|
|
iv.v64[1] = be64_to_cpu(est << 16);
|
|
#endif
|
|
status = cipher_set_iv(stream->rtp_cipher, &iv);
|
|
|
|
} else {
|
|
v128_t iv;
|
|
|
|
/* otherwise, set the index to est */
|
|
#ifdef NO_64BIT_MATH
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = 0;
|
|
#else
|
|
iv.v64[0] = 0;
|
|
#endif
|
|
iv.v64[1] = be64_to_cpu(est);
|
|
status = cipher_set_iv(stream->rtp_cipher, &iv);
|
|
}
|
|
if (status)
|
|
return 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 = auth_get_prefix_length(stream->rtp_auth);
|
|
if (prefix_len) {
|
|
status = cipher_output(stream->rtp_cipher, auth_tag, prefix_len);
|
|
if (status)
|
|
return err_status_cipher_fail;
|
|
debug_print(mod_srtp, "keystream prefix: %s",
|
|
octet_string_hex_string(auth_tag, prefix_len));
|
|
}
|
|
}
|
|
|
|
/* if we're encrypting, exor keystream into the message */
|
|
if (enc_start) {
|
|
status = cipher_encrypt(stream->rtp_cipher,
|
|
(uint8_t *)enc_start, &enc_octet_len);
|
|
if (status)
|
|
return err_status_cipher_fail;
|
|
}
|
|
|
|
/*
|
|
* if we're authenticating, run authentication function and put result
|
|
* into the auth_tag
|
|
*/
|
|
if (auth_start) {
|
|
|
|
/* initialize auth func context */
|
|
status = auth_start(stream->rtp_auth);
|
|
if (status) return status;
|
|
|
|
/* run auth func over packet */
|
|
status = auth_update(stream->rtp_auth,
|
|
(uint8_t *)auth_start, *pkt_octet_len);
|
|
if (status) return status;
|
|
|
|
/* run auth func over ROC, put result into auth_tag */
|
|
debug_print(mod_srtp, "estimated packet index: %016llx", est);
|
|
status = auth_compute(stream->rtp_auth, (uint8_t *)&est, 4, auth_tag);
|
|
debug_print(mod_srtp, "srtp auth tag: %s",
|
|
octet_string_hex_string(auth_tag, tag_len));
|
|
if (status)
|
|
return err_status_auth_fail;
|
|
|
|
}
|
|
|
|
if (auth_tag) {
|
|
|
|
/* increase the packet length by the length of the auth tag */
|
|
*pkt_octet_len += tag_len;
|
|
}
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
|
|
err_status_t
|
|
srtp_unprotect(srtp_ctx_t *ctx, void *srtp_hdr, int *pkt_octet_len) {
|
|
srtp_hdr_t *hdr = srtp_hdr;
|
|
uint32_t *enc_start; /* pointer to start of encrypted portion */
|
|
uint32_t *auth_start; /* pointer to start of auth. portion */
|
|
uint32_t enc_octet_len = 0;/* number of octets in encrypted portion */
|
|
uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
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;
|
|
err_status_t status;
|
|
srtp_stream_ctx_t *stream;
|
|
uint8_t tmp_tag[SRTP_MAX_TAG_LEN];
|
|
int tag_len, prefix_len;
|
|
|
|
debug_print(mod_srtp, "function srtp_unprotect", NULL);
|
|
|
|
/* we assume the hdr is 32-bit aligned to start */
|
|
|
|
/* check the packet length - it must at least contain a full header */
|
|
if (*pkt_octet_len < octets_in_rtp_header)
|
|
return err_status_bad_param;
|
|
|
|
/*
|
|
* look up ssrc in srtp_stream list, and process the packet with
|
|
* the appropriate stream. if we haven't seen this stream before,
|
|
* there's only one key for this srtp_session, and the cipher
|
|
* supports key-sharing, then we assume that a new stream using
|
|
* that key has just started up
|
|
*/
|
|
stream = srtp_get_stream(ctx, hdr->ssrc);
|
|
if (stream == NULL) {
|
|
if (ctx->stream_template != NULL) {
|
|
stream = ctx->stream_template;
|
|
debug_print(mod_srtp, "using provisional stream (SSRC: 0x%08x)",
|
|
hdr->ssrc);
|
|
|
|
/*
|
|
* set estimated packet index to sequence number from header,
|
|
* and set delta equal to the same value
|
|
*/
|
|
#ifdef NO_64BIT_MATH
|
|
est = (xtd_seq_num_t) make64(0,ntohs(hdr->seq));
|
|
delta = low32(est);
|
|
#else
|
|
est = (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 err_status_no_ctx;
|
|
}
|
|
} else {
|
|
|
|
/* estimate packet index from seq. num. in header */
|
|
delta = rdbx_estimate_index(&stream->rtp_rdbx, &est, ntohs(hdr->seq));
|
|
|
|
/* check replay database */
|
|
status = 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
|
|
|
|
/* get tag length from stream */
|
|
tag_len = auth_get_tag_length(stream->rtp_auth);
|
|
|
|
/*
|
|
* set the cipher's IV properly, depending on whatever cipher we
|
|
* happen to be using
|
|
*/
|
|
if (stream->rtp_cipher->type == &aes_icm) {
|
|
|
|
/* aes counter mode */
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = hdr->ssrc; /* still in network order */
|
|
#ifdef NO_64BIT_MATH
|
|
iv.v64[1] = be64_to_cpu(make64((high32(est) << 16) | (low32(est) >> 16),
|
|
low32(est) << 16));
|
|
#else
|
|
iv.v64[1] = be64_to_cpu(est << 16);
|
|
#endif
|
|
status = aes_icm_set_iv(stream->rtp_cipher->state, &iv);
|
|
} else {
|
|
|
|
/* no particular format - set the iv to the pakcet index */
|
|
#ifdef NO_64BIT_MATH
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = 0;
|
|
#else
|
|
iv.v64[0] = 0;
|
|
#endif
|
|
iv.v64[1] = be64_to_cpu(est);
|
|
status = cipher_set_iv(stream->rtp_cipher, &iv);
|
|
}
|
|
if (status)
|
|
return err_status_cipher_fail;
|
|
|
|
/* shift est, put into network byte order */
|
|
#ifdef NO_64BIT_MATH
|
|
est = be64_to_cpu(make64((high32(est) << 16) |
|
|
(low32(est) >> 16),
|
|
low32(est) << 16));
|
|
#else
|
|
est = be64_to_cpu(est << 16);
|
|
#endif
|
|
|
|
/*
|
|
* find starting point for decryption and length of data to be
|
|
* decrypted - the encrypted portion starts after the rtp header
|
|
* extension, if present; otherwise, it starts after the last csrc,
|
|
* if any are present
|
|
*
|
|
* if we're not providing confidentiality, set enc_start to NULL
|
|
*/
|
|
if (stream->rtp_services & sec_serv_conf) {
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtp_header + hdr->cc;
|
|
if (hdr->x == 1) {
|
|
srtp_hdr_xtnd_t *xtn_hdr = (srtp_hdr_xtnd_t *)enc_start;
|
|
enc_start += (ntohs(xtn_hdr->length) + 1);
|
|
}
|
|
enc_octet_len = (uint32_t)(*pkt_octet_len - tag_len - ((enc_start - (uint32_t *)hdr) << 2));
|
|
} else {
|
|
enc_start = NULL;
|
|
}
|
|
|
|
/*
|
|
* if we're providing authentication, set the auth_start and auth_tag
|
|
* pointers to the proper locations; otherwise, set auth_start to NULL
|
|
* to indicate that no authentication is needed
|
|
*/
|
|
if (stream->rtp_services & sec_serv_auth) {
|
|
auth_start = (uint32_t *)hdr;
|
|
auth_tag = (uint8_t *)hdr + *pkt_octet_len - tag_len;
|
|
} else {
|
|
auth_start = NULL;
|
|
auth_tag = NULL;
|
|
}
|
|
|
|
/*
|
|
* if we expect message authentication, run the authentication
|
|
* function and compare the result with the value of the auth_tag
|
|
*/
|
|
if (auth_start) {
|
|
|
|
/*
|
|
* if we're using a universal hash, then we need to compute the
|
|
* keystream prefix for encrypting the universal hash output
|
|
*
|
|
* if the keystream prefix length is zero, then we know that
|
|
* the authenticator isn't using a universal hash function
|
|
*/
|
|
if (stream->rtp_auth->prefix_len != 0) {
|
|
|
|
prefix_len = auth_get_prefix_length(stream->rtp_auth);
|
|
status = cipher_output(stream->rtp_cipher, tmp_tag, prefix_len);
|
|
debug_print(mod_srtp, "keystream prefix: %s",
|
|
octet_string_hex_string(tmp_tag, prefix_len));
|
|
if (status)
|
|
return err_status_cipher_fail;
|
|
}
|
|
|
|
/* initialize auth func context */
|
|
status = auth_start(stream->rtp_auth);
|
|
if (status) return status;
|
|
|
|
/* now compute auth function over packet */
|
|
status = auth_update(stream->rtp_auth, (uint8_t *)auth_start,
|
|
*pkt_octet_len - tag_len);
|
|
|
|
/* run auth func over ROC, then write tmp tag */
|
|
status = auth_compute(stream->rtp_auth, (uint8_t *)&est, 4, tmp_tag);
|
|
|
|
debug_print(mod_srtp, "computed auth tag: %s",
|
|
octet_string_hex_string(tmp_tag, tag_len));
|
|
debug_print(mod_srtp, "packet auth tag: %s",
|
|
octet_string_hex_string(auth_tag, tag_len));
|
|
if (status)
|
|
return err_status_auth_fail;
|
|
|
|
if (octet_string_is_eq(tmp_tag, auth_tag, tag_len))
|
|
return 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(key_limit_update(stream->limit)) {
|
|
case key_event_normal:
|
|
break;
|
|
case key_event_soft_limit:
|
|
srtp_handle_event(ctx, stream, event_key_soft_limit);
|
|
break;
|
|
case key_event_hard_limit:
|
|
srtp_handle_event(ctx, stream, event_key_hard_limit);
|
|
return err_status_key_expired;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* if we're encrypting, add keystream into ciphertext */
|
|
if (enc_start) {
|
|
status = cipher_encrypt(stream->rtp_cipher,
|
|
(uint8_t *)enc_start, &enc_octet_len);
|
|
if (status)
|
|
return 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
|
|
*/
|
|
rdbx_add_index(&stream->rtp_rdbx, delta);
|
|
|
|
/* decrease the packet length by the length of the auth tag */
|
|
*pkt_octet_len -= tag_len;
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
err_status_t
|
|
srtp_init() {
|
|
err_status_t status;
|
|
|
|
/* initialize crypto kernel */
|
|
status = crypto_kernel_init();
|
|
if (status)
|
|
return status;
|
|
|
|
/* load srtp debug module into the kernel */
|
|
status = crypto_kernel_load_debug_module(&mod_srtp);
|
|
if (status)
|
|
return status;
|
|
|
|
return 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 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;
|
|
}
|
|
|
|
err_status_t
|
|
srtp_dealloc(srtp_t session) {
|
|
srtp_stream_ctx_t *stream;
|
|
err_status_t status;
|
|
|
|
/*
|
|
* we take a conservative deallocation strategy - if we encounter an
|
|
* error deallocating a stream, then we stop trying to deallocate
|
|
* memory and just return an error
|
|
*/
|
|
|
|
/* walk list of streams, deallocating as we go */
|
|
stream = session->stream_list;
|
|
while (stream != NULL) {
|
|
srtp_stream_t next = stream->next;
|
|
status = srtp_stream_dealloc(session, stream);
|
|
if (status)
|
|
return status;
|
|
stream = next;
|
|
}
|
|
|
|
/* deallocate stream template, if there is one */
|
|
if (session->stream_template != NULL) {
|
|
status = auth_dealloc(session->stream_template->rtcp_auth);
|
|
if (status)
|
|
return status;
|
|
status = cipher_dealloc(session->stream_template->rtcp_cipher);
|
|
if (status)
|
|
return status;
|
|
crypto_free(session->stream_template->limit);
|
|
status = cipher_dealloc(session->stream_template->rtp_cipher);
|
|
if (status)
|
|
return status;
|
|
status = auth_dealloc(session->stream_template->rtp_auth);
|
|
if (status)
|
|
return status;
|
|
crypto_free(session->stream_template);
|
|
}
|
|
|
|
/* deallocate session context */
|
|
crypto_free(session);
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
|
|
err_status_t
|
|
srtp_add_stream(srtp_t session,
|
|
const srtp_policy_t *policy) {
|
|
err_status_t status;
|
|
srtp_stream_t tmp;
|
|
|
|
/* sanity check arguments */
|
|
if ((session == NULL) || (policy == NULL) || (policy->key == NULL))
|
|
return 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) {
|
|
crypto_free(tmp);
|
|
return status;
|
|
}
|
|
|
|
/*
|
|
* set the head of the stream list or the template to point to the
|
|
* stream that we've just alloced and init'ed, depending on whether
|
|
* or not it has a wildcard SSRC value or not
|
|
*
|
|
* if the template stream has already been set, then the policy is
|
|
* inconsistent, so we return a bad_param error code
|
|
*/
|
|
switch (policy->ssrc.type) {
|
|
case (ssrc_any_outbound):
|
|
if (session->stream_template) {
|
|
return err_status_bad_param;
|
|
}
|
|
session->stream_template = tmp;
|
|
session->stream_template->direction = dir_srtp_sender;
|
|
break;
|
|
case (ssrc_any_inbound):
|
|
if (session->stream_template) {
|
|
return 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:
|
|
crypto_free(tmp);
|
|
return err_status_bad_param;
|
|
}
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
|
|
err_status_t
|
|
srtp_create(srtp_t *session, /* handle for session */
|
|
const srtp_policy_t *policy) { /* SRTP policy (list) */
|
|
err_status_t stat;
|
|
srtp_ctx_t *ctx;
|
|
|
|
/* sanity check arguments */
|
|
if (session == NULL)
|
|
return err_status_bad_param;
|
|
|
|
/* allocate srtp context and set ctx_ptr */
|
|
ctx = (srtp_ctx_t *) crypto_alloc(sizeof(srtp_ctx_t));
|
|
if (ctx == NULL)
|
|
return 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;
|
|
while (policy != NULL) {
|
|
|
|
stat = srtp_add_stream(ctx, policy);
|
|
if (stat) {
|
|
/* clean up everything */
|
|
srtp_dealloc(*session);
|
|
return stat;
|
|
}
|
|
|
|
/* set policy to next item in list */
|
|
policy = policy->next;
|
|
}
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
|
|
err_status_t
|
|
srtp_remove_stream(srtp_t session, uint32_t ssrc) {
|
|
srtp_stream_ctx_t *stream, *last_stream;
|
|
err_status_t status;
|
|
|
|
/* sanity check arguments */
|
|
if (session == NULL)
|
|
return 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 err_status_no_ctx;
|
|
|
|
/* remove stream from the list */
|
|
last_stream->next = stream->next;
|
|
|
|
/* deallocate the stream */
|
|
status = srtp_stream_dealloc(session, stream);
|
|
if (status)
|
|
return status;
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
|
|
/*
|
|
* the default policy - provides a convenient way for callers to use
|
|
* the default security policy
|
|
*
|
|
* this policy is that defined in the current SRTP internet draft.
|
|
*
|
|
*/
|
|
|
|
/*
|
|
* NOTE: cipher_key_len is really key len (128 bits) plus salt len
|
|
* (112 bits)
|
|
*/
|
|
/* There are hard-coded 16's for base_key_len in the key generation code */
|
|
|
|
void
|
|
crypto_policy_set_rtp_default(crypto_policy_t *p) {
|
|
|
|
p->cipher_type = AES_128_ICM;
|
|
p->cipher_key_len = 30; /* default 128 bits per RFC 3711 */
|
|
p->auth_type = 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
|
|
crypto_policy_set_rtcp_default(crypto_policy_t *p) {
|
|
|
|
p->cipher_type = AES_128_ICM;
|
|
p->cipher_key_len = 30; /* default 128 bits per RFC 3711 */
|
|
p->auth_type = 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
|
|
crypto_policy_set_aes_cm_128_hmac_sha1_32(crypto_policy_t *p) {
|
|
|
|
/*
|
|
* corresponds to draft-ietf-mmusic-sdescriptions-12.txt
|
|
*
|
|
* note that this crypto policy is intended for SRTP, but not SRTCP
|
|
*/
|
|
|
|
p->cipher_type = AES_128_ICM;
|
|
p->cipher_key_len = 30; /* 128 bit key, 112 bit salt */
|
|
p->auth_type = 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
|
|
crypto_policy_set_aes_cm_128_null_auth(crypto_policy_t *p) {
|
|
|
|
/*
|
|
* corresponds to draft-ietf-mmusic-sdescriptions-12.txt
|
|
*
|
|
* note that this crypto policy is intended for SRTP, but not SRTCP
|
|
*/
|
|
|
|
p->cipher_type = AES_128_ICM;
|
|
p->cipher_key_len = 30; /* 128 bit key, 112 bit salt */
|
|
p->auth_type = NULL_AUTH;
|
|
p->auth_key_len = 0;
|
|
p->auth_tag_len = 0;
|
|
p->sec_serv = sec_serv_conf;
|
|
|
|
}
|
|
|
|
|
|
void
|
|
crypto_policy_set_null_cipher_hmac_sha1_80(crypto_policy_t *p) {
|
|
|
|
/*
|
|
* corresponds to draft-ietf-mmusic-sdescriptions-12.txt
|
|
*/
|
|
|
|
p->cipher_type = NULL_CIPHER;
|
|
p->cipher_key_len = 0;
|
|
p->auth_type = HMAC_SHA1;
|
|
p->auth_key_len = 20;
|
|
p->auth_tag_len = 10;
|
|
p->sec_serv = sec_serv_auth;
|
|
|
|
}
|
|
|
|
|
|
/*
|
|
* secure rtcp functions
|
|
*/
|
|
|
|
err_status_t
|
|
srtp_protect_rtcp(srtp_t ctx, void *rtcp_hdr, int *pkt_octet_len) {
|
|
srtcp_hdr_t *hdr = 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 */
|
|
uint32_t enc_octet_len = 0;/* number of octets in encrypted portion */
|
|
uint8_t *auth_tag = NULL; /* location of auth_tag within packet */
|
|
err_status_t status;
|
|
int tag_len;
|
|
srtp_stream_ctx_t *stream;
|
|
int prefix_len;
|
|
uint32_t seq_num;
|
|
|
|
/* we assume the hdr is 32-bit aligned to start */
|
|
/*
|
|
* 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 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);
|
|
}
|
|
}
|
|
|
|
/* get tag length from stream context */
|
|
tag_len = auth_get_tag_length(stream->rtcp_auth);
|
|
|
|
/*
|
|
* set encryption start and encryption length - if we're not
|
|
* providing confidentiality, set enc_start to NULL
|
|
*/
|
|
enc_start = (uint32_t *)hdr + uint32s_in_rtcp_header;
|
|
enc_octet_len = *pkt_octet_len - octets_in_rtcp_header;
|
|
|
|
/* all of the packet, except the header, gets encrypted */
|
|
/* NOTE: hdr->length is not usable - it refers to only the first
|
|
RTCP report in the compound packet! */
|
|
/* NOTE: trailer is 32-bit aligned because RTCP 'packets' are always
|
|
multiples of 32-bits (RFC 3550 6.1) */
|
|
trailer = (uint32_t *) ((char *)enc_start + enc_octet_len);
|
|
|
|
if (stream->rtcp_services & sec_serv_conf) {
|
|
*trailer = htonl(SRTCP_E_BIT); /* set encrypt bit */
|
|
} else {
|
|
enc_start = NULL;
|
|
enc_octet_len = 0;
|
|
/* 0 is network-order independant */
|
|
*trailer = 0x00000000; /* set encrypt bit */
|
|
}
|
|
|
|
/*
|
|
* set the auth_start and auth_tag pointers to the proper locations
|
|
* (note that srtpc *always* provides authentication, unlike srtp)
|
|
*/
|
|
/* Note: This would need to change for optional mikey data */
|
|
auth_start = (uint32_t *)hdr;
|
|
auth_tag = (uint8_t *)hdr + *pkt_octet_len + sizeof(srtcp_trailer_t);
|
|
|
|
/*
|
|
* check sequence number for overruns, and copy it into the packet
|
|
* if its value isn't too big
|
|
*/
|
|
status = rdb_increment(&stream->rtcp_rdb);
|
|
if (status)
|
|
return status;
|
|
seq_num = rdb_get_value(&stream->rtcp_rdb);
|
|
*trailer |= htonl(seq_num);
|
|
debug_print(mod_srtp, "srtcp index: %x", seq_num);
|
|
|
|
/*
|
|
* if we're using rindael counter mode, set nonce and seq
|
|
*/
|
|
if (stream->rtcp_cipher->type == &aes_icm) {
|
|
v128_t iv;
|
|
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = hdr->ssrc; /* still in network order! */
|
|
iv.v32[2] = htonl(seq_num >> 16);
|
|
iv.v32[3] = htonl(seq_num << 16);
|
|
status = aes_icm_set_iv(stream->rtcp_cipher->state, &iv);
|
|
|
|
} else {
|
|
v128_t iv;
|
|
|
|
/* otherwise, just set the index to seq_num */
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = 0;
|
|
iv.v32[2] = 0;
|
|
iv.v32[3] = htonl(seq_num);
|
|
status = cipher_set_iv(stream->rtcp_cipher, &iv);
|
|
}
|
|
if (status)
|
|
return 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 = auth_get_prefix_length(stream->rtcp_auth);
|
|
status = cipher_output(stream->rtcp_cipher, auth_tag, prefix_len);
|
|
|
|
debug_print(mod_srtp, "keystream prefix: %s",
|
|
octet_string_hex_string(auth_tag, prefix_len));
|
|
|
|
if (status)
|
|
return err_status_cipher_fail;
|
|
}
|
|
|
|
/* if we're encrypting, exor keystream into the message */
|
|
if (enc_start) {
|
|
status = cipher_encrypt(stream->rtcp_cipher,
|
|
(uint8_t *)enc_start, &enc_octet_len);
|
|
if (status)
|
|
return err_status_cipher_fail;
|
|
}
|
|
|
|
/* initialize auth func context */
|
|
auth_start(stream->rtcp_auth);
|
|
|
|
/* per spec, do auth after encryption */
|
|
/* run auth func over packet, put result into auth_tag */
|
|
status = auth_compute(stream->rtcp_auth,
|
|
(uint8_t *)auth_start, *pkt_octet_len, auth_tag);
|
|
debug_print(mod_srtp, "srtcp auth tag: %s",
|
|
octet_string_hex_string(auth_tag, tag_len));
|
|
if (status)
|
|
return err_status_auth_fail;
|
|
|
|
/* increase the packet length by the length of the auth tag and seq_num*/
|
|
*pkt_octet_len += (tag_len + sizeof(srtcp_trailer_t));
|
|
|
|
return err_status_ok;
|
|
}
|
|
|
|
|
|
err_status_t
|
|
srtp_unprotect_rtcp(srtp_t ctx, void *srtcp_hdr, int *pkt_octet_len) {
|
|
srtcp_hdr_t *hdr = 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 */
|
|
uint32_t 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];
|
|
err_status_t status;
|
|
int tag_len;
|
|
srtp_stream_ctx_t *stream;
|
|
int prefix_len;
|
|
uint32_t seq_num;
|
|
|
|
/* we assume the hdr is 32-bit aligned to start */
|
|
/*
|
|
* look up ssrc in srtp_stream list, and process the packet with
|
|
* the appropriate stream. if we haven't seen this stream before,
|
|
* there's only one key for this srtp_session, and the cipher
|
|
* supports key-sharing, then we assume that a new stream using
|
|
* that key has just started up
|
|
*/
|
|
stream = srtp_get_stream(ctx, hdr->ssrc);
|
|
if (stream == NULL) {
|
|
if (ctx->stream_template != NULL) {
|
|
stream = ctx->stream_template;
|
|
debug_print(mod_srtp, "srtcp using provisional stream (SSRC: 0x%08x)",
|
|
hdr->ssrc);
|
|
} else {
|
|
/* no template stream, so we return an error */
|
|
return err_status_no_ctx;
|
|
}
|
|
}
|
|
|
|
/* get tag length from stream context */
|
|
tag_len = auth_get_tag_length(stream->rtcp_auth);
|
|
|
|
/*
|
|
* set encryption start, encryption length, and trailer
|
|
*/
|
|
enc_octet_len = *pkt_octet_len -
|
|
(octets_in_rtcp_header + tag_len + sizeof(srtcp_trailer_t));
|
|
/* index & E (encryption) bit follow normal data. hdr->len
|
|
is the number of words (32-bit) in the normal packet minus 1 */
|
|
/* This should point trailer to the word past the end of the
|
|
normal data. */
|
|
/* This would need to be modified for optional mikey data */
|
|
/*
|
|
* NOTE: trailer is 32-bit aligned because RTCP 'packets' are always
|
|
* multiples of 32-bits (RFC 3550 6.1)
|
|
*/
|
|
trailer = (uint32_t *) ((char *) hdr +
|
|
*pkt_octet_len -(tag_len + sizeof(srtcp_trailer_t)));
|
|
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 */
|
|
}
|
|
|
|
/*
|
|
* set the auth_start and auth_tag pointers to the proper locations
|
|
* (note that srtcp *always* uses authentication, unlike srtp)
|
|
*/
|
|
auth_start = (uint32_t *)hdr;
|
|
auth_tag = (uint8_t *)hdr + *pkt_octet_len - tag_len;
|
|
|
|
/*
|
|
* check the sequence number for replays
|
|
*/
|
|
/* this is easier than dealing with bitfield access */
|
|
seq_num = ntohl(*trailer) & SRTCP_INDEX_MASK;
|
|
status = rdb_check(&stream->rtcp_rdb, seq_num);
|
|
if (status)
|
|
return status;
|
|
debug_print(mod_srtp, "srtcp index: %x", seq_num);
|
|
|
|
/*
|
|
* if we're using aes counter mode, set nonce and seq
|
|
*/
|
|
if (stream->rtcp_cipher->type == &aes_icm) {
|
|
v128_t iv;
|
|
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = hdr->ssrc; /* still in network order! */
|
|
iv.v32[2] = htonl(seq_num >> 16);
|
|
iv.v32[3] = htonl(seq_num << 16);
|
|
status = aes_icm_set_iv(stream->rtcp_cipher->state, &iv);
|
|
|
|
} else {
|
|
v128_t iv;
|
|
|
|
/* otherwise, just set the index to seq_num */
|
|
iv.v32[0] = 0;
|
|
iv.v32[1] = 0;
|
|
iv.v32[2] = 0;
|
|
iv.v32[3] = htonl(seq_num);
|
|
status = cipher_set_iv(stream->rtcp_cipher, &iv);
|
|
|
|
}
|
|
if (status)
|
|
return err_status_cipher_fail;
|
|
|
|
/* initialize auth func context */
|
|
auth_start(stream->rtcp_auth);
|
|
|
|
/* run auth func over packet, put result into tmp_tag */
|
|
status = auth_compute(stream->rtcp_auth, (uint8_t *)auth_start,
|
|
*pkt_octet_len - (tag_len + sizeof(srtcp_trailer_t)),
|
|
tmp_tag);
|
|
debug_print(mod_srtp, "srtcp computed tag: %s",
|
|
octet_string_hex_string(tmp_tag, tag_len));
|
|
if (status)
|
|
return err_status_auth_fail;
|
|
|
|
/* compare the tag just computed with the one in the packet */
|
|
debug_print(mod_srtp, "srtcp tag from packet: %s",
|
|
octet_string_hex_string(auth_tag, tag_len));
|
|
if (octet_string_is_eq(tmp_tag, auth_tag, tag_len))
|
|
return err_status_auth_fail;
|
|
|
|
/*
|
|
* if we're authenticating using a universal hash, put the keystream
|
|
* prefix into the authentication tag
|
|
*/
|
|
prefix_len = auth_get_prefix_length(stream->rtcp_auth);
|
|
if (prefix_len) {
|
|
status = cipher_output(stream->rtcp_cipher, auth_tag, prefix_len);
|
|
debug_print(mod_srtp, "keystream prefix: %s",
|
|
octet_string_hex_string(auth_tag, prefix_len));
|
|
if (status)
|
|
return err_status_cipher_fail;
|
|
}
|
|
|
|
/* if we're decrypting, exor keystream into the message */
|
|
if (enc_start) {
|
|
status = cipher_encrypt(stream->rtcp_cipher,
|
|
(uint8_t *)enc_start, &enc_octet_len);
|
|
if (status)
|
|
return 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));
|
|
|
|
/*
|
|
* 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 */
|
|
rdb_add_index(&stream->rtcp_rdb, seq_num);
|
|
|
|
|
|
return err_status_ok;
|
|
}
|